webrtc_m130/pc/media_session_unittest.cc

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/*
* Copyright 2004 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "pc/media_session.h"
#include <algorithm>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
#include "absl/strings/match.h"
#include "media/base/codec.h"
#include "media/base/test_utils.h"
#include "media/sctp/sctp_transport_internal.h"
#include "p2p/base/p2p_constants.h"
#include "p2p/base/transport_description.h"
#include "p2p/base/transport_info.h"
#include "pc/rtp_media_utils.h"
#include "pc/srtp_filter.h"
#include "rtc_base/checks.h"
#include "rtc_base/fake_ssl_identity.h"
#include "rtc_base/gunit.h"
#include "rtc_base/message_digest.h"
#include "rtc_base/ssl_adapter.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/unique_id_generator.h"
#include "test/gmock.h"
#define ASSERT_CRYPTO(cd, s, cs) \
ASSERT_EQ(s, cd->cryptos().size()); \
ASSERT_EQ(cs, cd->cryptos()[0].cipher_suite)
typedef std::vector<cricket::Candidate> Candidates;
using cricket::AudioCodec;
using cricket::AudioContentDescription;
using cricket::ContentInfo;
using cricket::CryptoParamsVec;
using cricket::GetFirstAudioContent;
using cricket::GetFirstAudioContentDescription;
using cricket::GetFirstDataContent;
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
using cricket::GetFirstRtpDataContentDescription;
using cricket::GetFirstVideoContent;
using cricket::GetFirstVideoContentDescription;
using cricket::kAutoBandwidth;
using cricket::MEDIA_TYPE_AUDIO;
using cricket::MEDIA_TYPE_DATA;
using cricket::MEDIA_TYPE_VIDEO;
using cricket::MediaContentDescription;
using cricket::MediaDescriptionOptions;
using cricket::MediaProtocolType;
using cricket::MediaSessionDescriptionFactory;
using cricket::MediaSessionOptions;
using cricket::MediaType;
using cricket::RidDescription;
using cricket::RidDirection;
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
using cricket::RtpDataCodec;
using cricket::RtpDataContentDescription;
using cricket::SctpDataContentDescription;
using cricket::SEC_DISABLED;
using cricket::SEC_ENABLED;
using cricket::SEC_REQUIRED;
using cricket::SessionDescription;
using cricket::SimulcastDescription;
using cricket::SimulcastLayer;
using cricket::SimulcastLayerList;
using cricket::SsrcGroup;
using cricket::StreamParams;
using cricket::StreamParamsVec;
using cricket::TransportDescription;
using cricket::TransportDescriptionFactory;
using cricket::TransportInfo;
using cricket::VideoCodec;
using cricket::VideoContentDescription;
using rtc::CS_AEAD_AES_128_GCM;
using rtc::CS_AEAD_AES_256_GCM;
using rtc::CS_AES_CM_128_HMAC_SHA1_32;
using rtc::CS_AES_CM_128_HMAC_SHA1_80;
using rtc::UniqueRandomIdGenerator;
using ::testing::Contains;
using ::testing::Each;
using ::testing::ElementsAre;
using ::testing::ElementsAreArray;
using ::testing::Eq;
using ::testing::Field;
using ::testing::IsEmpty;
using ::testing::IsFalse;
using ::testing::Ne;
using ::testing::Not;
using ::testing::Pointwise;
using ::testing::SizeIs;
using webrtc::RtpExtension;
using webrtc::RtpTransceiverDirection;
static const AudioCodec kAudioCodecs1[] = {
AudioCodec(103, "ISAC", 16000, -1, 1),
AudioCodec(102, "iLBC", 8000, 13300, 1),
AudioCodec(0, "PCMU", 8000, 64000, 1),
AudioCodec(8, "PCMA", 8000, 64000, 1),
AudioCodec(117, "red", 8000, 0, 1),
AudioCodec(107, "CN", 48000, 0, 1)};
static const AudioCodec kAudioCodecs2[] = {
AudioCodec(126, "foo", 16000, 22000, 1),
AudioCodec(0, "PCMU", 8000, 64000, 1),
AudioCodec(127, "iLBC", 8000, 13300, 1),
};
static const AudioCodec kAudioCodecsAnswer[] = {
AudioCodec(102, "iLBC", 8000, 13300, 1),
AudioCodec(0, "PCMU", 8000, 64000, 1),
};
static const VideoCodec kVideoCodecs1[] = {VideoCodec(96, "H264-SVC"),
VideoCodec(97, "H264")};
static const VideoCodec kVideoCodecs1Reverse[] = {VideoCodec(97, "H264"),
VideoCodec(96, "H264-SVC")};
static const VideoCodec kVideoCodecs2[] = {VideoCodec(126, "H264"),
VideoCodec(127, "H263")};
static const VideoCodec kVideoCodecsAnswer[] = {VideoCodec(97, "H264")};
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
static const RtpDataCodec kDataCodecs1[] = {RtpDataCodec(98, "binary-data"),
RtpDataCodec(99, "utf8-text")};
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
static const RtpDataCodec kDataCodecs2[] = {RtpDataCodec(126, "binary-data"),
RtpDataCodec(127, "utf8-text")};
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
static const RtpDataCodec kDataCodecsAnswer[] = {
RtpDataCodec(98, "binary-data"), RtpDataCodec(99, "utf8-text")};
static const RtpExtension kAudioRtpExtension1[] = {
RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 8),
RtpExtension("http://google.com/testing/audio_something", 10),
};
static const RtpExtension kAudioRtpExtensionEncrypted1[] = {
RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 8),
RtpExtension("http://google.com/testing/audio_something", 10),
RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 12, true),
};
static const RtpExtension kAudioRtpExtension2[] = {
RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 2),
RtpExtension("http://google.com/testing/audio_something_else", 8),
RtpExtension("http://google.com/testing/both_audio_and_video", 7),
};
static const RtpExtension kAudioRtpExtension3[] = {
RtpExtension("http://google.com/testing/audio_something", 2),
RtpExtension("http://google.com/testing/both_audio_and_video", 3),
};
static const RtpExtension kAudioRtpExtension3ForEncryption[] = {
RtpExtension("http://google.com/testing/audio_something", 2),
// Use RTP extension that supports encryption.
RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 3),
};
static const RtpExtension kAudioRtpExtension3ForEncryptionOffer[] = {
RtpExtension("http://google.com/testing/audio_something", 2),
RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 3),
RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 14, true),
};
static const RtpExtension kAudioRtpExtensionAnswer[] = {
RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 8),
};
static const RtpExtension kAudioRtpExtensionEncryptedAnswer[] = {
RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 12, true),
};
static const RtpExtension kVideoRtpExtension1[] = {
RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 14),
RtpExtension("http://google.com/testing/video_something", 13),
};
static const RtpExtension kVideoRtpExtensionEncrypted1[] = {
RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 14),
RtpExtension("http://google.com/testing/video_something", 13),
RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 11, true),
};
static const RtpExtension kVideoRtpExtension2[] = {
RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 2),
RtpExtension("http://google.com/testing/video_something_else", 14),
RtpExtension("http://google.com/testing/both_audio_and_video", 7),
};
static const RtpExtension kVideoRtpExtension3[] = {
RtpExtension("http://google.com/testing/video_something", 4),
RtpExtension("http://google.com/testing/both_audio_and_video", 5),
};
static const RtpExtension kVideoRtpExtension3ForEncryption[] = {
RtpExtension("http://google.com/testing/video_something", 4),
// Use RTP extension that supports encryption.
RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 5),
};
static const RtpExtension kVideoRtpExtensionAnswer[] = {
RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 14),
};
static const RtpExtension kVideoRtpExtensionEncryptedAnswer[] = {
RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 11, true),
};
static const RtpExtension kRtpExtensionTransportSequenceNumber01[] = {
RtpExtension("http://www.ietf.org/id/"
"draft-holmer-rmcat-transport-wide-cc-extensions-01",
1),
};
static const RtpExtension kRtpExtensionTransportSequenceNumber01And02[] = {
RtpExtension("http://www.ietf.org/id/"
"draft-holmer-rmcat-transport-wide-cc-extensions-01",
1),
RtpExtension(
"http://www.webrtc.org/experiments/rtp-hdrext/transport-wide-cc-02",
2),
};
static const RtpExtension kRtpExtensionTransportSequenceNumber02[] = {
RtpExtension(
"http://www.webrtc.org/experiments/rtp-hdrext/transport-wide-cc-02",
2),
};
static const RtpExtension kRtpExtensionGenericFrameDescriptorUri00[] = {
RtpExtension("http://www.webrtc.org/experiments/rtp-hdrext/"
"generic-frame-descriptor-00",
3),
};
static const uint32_t kSimulcastParamsSsrc[] = {10, 11, 20, 21, 30, 31};
static const uint32_t kSimSsrc[] = {10, 20, 30};
static const uint32_t kFec1Ssrc[] = {10, 11};
static const uint32_t kFec2Ssrc[] = {20, 21};
static const uint32_t kFec3Ssrc[] = {30, 31};
static const char kMediaStream1[] = "stream_1";
static const char kMediaStream2[] = "stream_2";
static const char kVideoTrack1[] = "video_1";
static const char kVideoTrack2[] = "video_2";
static const char kAudioTrack1[] = "audio_1";
static const char kAudioTrack2[] = "audio_2";
static const char kAudioTrack3[] = "audio_3";
static const char kDataTrack1[] = "data_1";
static const char kDataTrack2[] = "data_2";
static const char kDataTrack3[] = "data_3";
static const char* kMediaProtocols[] = {"RTP/AVP", "RTP/SAVP", "RTP/AVPF",
"RTP/SAVPF"};
static const char* kMediaProtocolsDtls[] = {
"TCP/TLS/RTP/SAVPF", "TCP/TLS/RTP/SAVP", "UDP/TLS/RTP/SAVPF",
"UDP/TLS/RTP/SAVP"};
// SRTP cipher name negotiated by the tests. This must be updated if the
// default changes.
static const char* kDefaultSrtpCryptoSuite = CS_AES_CM_128_HMAC_SHA1_80;
static const char* kDefaultSrtpCryptoSuiteGcm = CS_AEAD_AES_256_GCM;
// These constants are used to make the code using "AddMediaDescriptionOptions"
// more readable.
static constexpr bool kStopped = true;
static constexpr bool kActive = false;
static bool IsMediaContentOfType(const ContentInfo* content,
MediaType media_type) {
RTC_DCHECK(content);
return content->media_description()->type() == media_type;
}
static RtpTransceiverDirection GetMediaDirection(const ContentInfo* content) {
RTC_DCHECK(content);
return content->media_description()->direction();
}
static void AddRtxCodec(const VideoCodec& rtx_codec,
std::vector<VideoCodec>* codecs) {
ASSERT_FALSE(cricket::FindCodecById(*codecs, rtx_codec.id));
codecs->push_back(rtx_codec);
}
template <class T>
static std::vector<std::string> GetCodecNames(const std::vector<T>& codecs) {
std::vector<std::string> codec_names;
codec_names.reserve(codecs.size());
for (const auto& codec : codecs) {
codec_names.push_back(codec.name);
}
return codec_names;
}
// This is used for test only. MIDs are not the identification of the
// MediaDescriptionOptions since some end points may not support MID and the SDP
// may not contain 'mid'.
std::vector<MediaDescriptionOptions>::iterator FindFirstMediaDescriptionByMid(
const std::string& mid,
MediaSessionOptions* opts) {
return absl::c_find_if(
opts->media_description_options,
[&mid](const MediaDescriptionOptions& t) { return t.mid == mid; });
}
std::vector<MediaDescriptionOptions>::const_iterator
FindFirstMediaDescriptionByMid(const std::string& mid,
const MediaSessionOptions& opts) {
return absl::c_find_if(
opts.media_description_options,
[&mid](const MediaDescriptionOptions& t) { return t.mid == mid; });
}
// Add a media section to the |session_options|.
static void AddMediaDescriptionOptions(MediaType type,
const std::string& mid,
RtpTransceiverDirection direction,
bool stopped,
MediaSessionOptions* opts) {
opts->media_description_options.push_back(
MediaDescriptionOptions(type, mid, direction, stopped));
}
static void AddAudioVideoSections(RtpTransceiverDirection direction,
MediaSessionOptions* opts) {
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio", direction, kActive,
opts);
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video", direction, kActive,
opts);
}
static void AddDataSection(cricket::DataChannelType dct,
RtpTransceiverDirection direction,
MediaSessionOptions* opts) {
opts->data_channel_type = dct;
AddMediaDescriptionOptions(MEDIA_TYPE_DATA, "data", direction, kActive, opts);
}
static void AttachSenderToMediaDescriptionOptions(
const std::string& mid,
MediaType type,
const std::string& track_id,
const std::vector<std::string>& stream_ids,
const std::vector<RidDescription>& rids,
const SimulcastLayerList& simulcast_layers,
int num_sim_layer,
MediaSessionOptions* session_options) {
auto it = FindFirstMediaDescriptionByMid(mid, session_options);
switch (type) {
case MEDIA_TYPE_AUDIO:
it->AddAudioSender(track_id, stream_ids);
break;
case MEDIA_TYPE_VIDEO:
it->AddVideoSender(track_id, stream_ids, rids, simulcast_layers,
num_sim_layer);
break;
case MEDIA_TYPE_DATA:
RTC_CHECK(stream_ids.size() == 1U);
it->AddRtpDataChannel(track_id, stream_ids[0]);
break;
default:
RTC_NOTREACHED();
}
}
static void AttachSenderToMediaDescriptionOptions(
const std::string& mid,
MediaType type,
const std::string& track_id,
const std::vector<std::string>& stream_ids,
int num_sim_layer,
MediaSessionOptions* session_options) {
AttachSenderToMediaDescriptionOptions(mid, type, track_id, stream_ids, {},
SimulcastLayerList(), num_sim_layer,
session_options);
}
static void DetachSenderFromMediaSection(const std::string& mid,
const std::string& track_id,
MediaSessionOptions* session_options) {
std::vector<cricket::SenderOptions>& sender_options_list =
FindFirstMediaDescriptionByMid(mid, session_options)->sender_options;
auto sender_it =
absl::c_find_if(sender_options_list,
[track_id](const cricket::SenderOptions& sender_options) {
return sender_options.track_id == track_id;
});
RTC_DCHECK(sender_it != sender_options_list.end());
sender_options_list.erase(sender_it);
}
// Helper function used to create a default MediaSessionOptions for Plan B SDP.
// (https://tools.ietf.org/html/draft-uberti-rtcweb-plan-00).
static MediaSessionOptions CreatePlanBMediaSessionOptions() {
MediaSessionOptions session_options;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&session_options);
return session_options;
}
// prefers GCM SDES crypto suites by removing non-GCM defaults.
void PreferGcmCryptoParameters(CryptoParamsVec* cryptos) {
cryptos->erase(
std::remove_if(cryptos->begin(), cryptos->end(),
[](const cricket::CryptoParams& crypto) {
return crypto.cipher_suite != CS_AEAD_AES_256_GCM &&
crypto.cipher_suite != CS_AEAD_AES_128_GCM;
}),
cryptos->end());
}
// TODO(zhihuang): Most of these tests were written while MediaSessionOptions
// was designed for Plan B SDP, where only one audio "m=" section and one video
// "m=" section could be generated, and ordering couldn't be controlled. Many of
// these tests may be obsolete as a result, and should be refactored or removed.
class MediaSessionDescriptionFactoryTest : public ::testing::Test {
public:
MediaSessionDescriptionFactoryTest()
: f1_(&tdf1_, &ssrc_generator1), f2_(&tdf2_, &ssrc_generator2) {
f1_.set_audio_codecs(MAKE_VECTOR(kAudioCodecs1),
MAKE_VECTOR(kAudioCodecs1));
f1_.set_video_codecs(MAKE_VECTOR(kVideoCodecs1),
MAKE_VECTOR(kVideoCodecs1));
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
f1_.set_rtp_data_codecs(MAKE_VECTOR(kDataCodecs1));
f2_.set_audio_codecs(MAKE_VECTOR(kAudioCodecs2),
MAKE_VECTOR(kAudioCodecs2));
f2_.set_video_codecs(MAKE_VECTOR(kVideoCodecs2),
MAKE_VECTOR(kVideoCodecs2));
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
f2_.set_rtp_data_codecs(MAKE_VECTOR(kDataCodecs2));
tdf1_.set_certificate(rtc::RTCCertificate::Create(
std::unique_ptr<rtc::SSLIdentity>(new rtc::FakeSSLIdentity("id1"))));
tdf2_.set_certificate(rtc::RTCCertificate::Create(
std::unique_ptr<rtc::SSLIdentity>(new rtc::FakeSSLIdentity("id2"))));
}
// Create a video StreamParamsVec object with:
// - one video stream with 3 simulcast streams and FEC,
StreamParamsVec CreateComplexVideoStreamParamsVec() {
SsrcGroup sim_group("SIM", MAKE_VECTOR(kSimSsrc));
SsrcGroup fec_group1("FEC", MAKE_VECTOR(kFec1Ssrc));
SsrcGroup fec_group2("FEC", MAKE_VECTOR(kFec2Ssrc));
SsrcGroup fec_group3("FEC", MAKE_VECTOR(kFec3Ssrc));
std::vector<SsrcGroup> ssrc_groups;
ssrc_groups.push_back(sim_group);
ssrc_groups.push_back(fec_group1);
ssrc_groups.push_back(fec_group2);
ssrc_groups.push_back(fec_group3);
StreamParams simulcast_params;
simulcast_params.id = kVideoTrack1;
simulcast_params.ssrcs = MAKE_VECTOR(kSimulcastParamsSsrc);
simulcast_params.ssrc_groups = ssrc_groups;
simulcast_params.cname = "Video_SIM_FEC";
simulcast_params.set_stream_ids({kMediaStream1});
StreamParamsVec video_streams;
video_streams.push_back(simulcast_params);
return video_streams;
}
bool CompareCryptoParams(const CryptoParamsVec& c1,
const CryptoParamsVec& c2) {
if (c1.size() != c2.size())
return false;
for (size_t i = 0; i < c1.size(); ++i)
if (c1[i].tag != c2[i].tag || c1[i].cipher_suite != c2[i].cipher_suite ||
c1[i].key_params != c2[i].key_params ||
c1[i].session_params != c2[i].session_params)
return false;
return true;
}
// Returns true if the transport info contains "renomination" as an
// ICE option.
bool GetIceRenomination(const TransportInfo* transport_info) {
return absl::c_linear_search(transport_info->description.transport_options,
"renomination");
}
void TestTransportInfo(bool offer,
const MediaSessionOptions& options,
bool has_current_desc) {
const std::string current_audio_ufrag = "current_audio_ufrag";
const std::string current_audio_pwd = "current_audio_pwd";
const std::string current_video_ufrag = "current_video_ufrag";
const std::string current_video_pwd = "current_video_pwd";
const std::string current_data_ufrag = "current_data_ufrag";
const std::string current_data_pwd = "current_data_pwd";
std::unique_ptr<SessionDescription> current_desc;
std::unique_ptr<SessionDescription> desc;
if (has_current_desc) {
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
current_desc = std::make_unique<SessionDescription>();
current_desc->AddTransportInfo(TransportInfo(
"audio",
TransportDescription(current_audio_ufrag, current_audio_pwd)));
current_desc->AddTransportInfo(TransportInfo(
"video",
TransportDescription(current_video_ufrag, current_video_pwd)));
current_desc->AddTransportInfo(TransportInfo(
"data", TransportDescription(current_data_ufrag, current_data_pwd)));
}
if (offer) {
desc = f1_.CreateOffer(options, current_desc.get());
} else {
std::unique_ptr<SessionDescription> offer;
offer = f1_.CreateOffer(options, NULL);
desc = f1_.CreateAnswer(offer.get(), options, current_desc.get());
}
ASSERT_TRUE(desc.get() != NULL);
const TransportInfo* ti_audio = desc->GetTransportInfoByName("audio");
if (options.has_audio()) {
EXPECT_TRUE(ti_audio != NULL);
if (has_current_desc) {
EXPECT_EQ(current_audio_ufrag, ti_audio->description.ice_ufrag);
EXPECT_EQ(current_audio_pwd, ti_audio->description.ice_pwd);
} else {
EXPECT_EQ(static_cast<size_t>(cricket::ICE_UFRAG_LENGTH),
ti_audio->description.ice_ufrag.size());
EXPECT_EQ(static_cast<size_t>(cricket::ICE_PWD_LENGTH),
ti_audio->description.ice_pwd.size());
}
auto media_desc_options_it =
FindFirstMediaDescriptionByMid("audio", options);
EXPECT_EQ(
media_desc_options_it->transport_options.enable_ice_renomination,
GetIceRenomination(ti_audio));
Reland: Implement true negotiation for DatagramTransport with fallback to RTP. In short, the caller places a x-opaque line in SDP for each m= section that uses datagram transport. If the answerer supports datagram transport, it will parse this line and create a datagram transport. It will then echo the x-opaque line into the answer (to indicate that it accepted use of datagram transport). If the offer and answer contain exactly the same x-opaque line, both peers will use datagram transport. If the x-opaque line is omitted from the answer (or is different in the answer) they will fall back to RTP. Note that a different x-opaque line in the answer means the answerer did not understand something in the negotiation proto. Since WebRTC cannot know what was misunderstood, or whether it's still possible to use the datagram transport, it must fall back to RTP. This may change in the future, possibly by passing the answer to the datagram transport, but it's good enough for now. Negotiation consists of four parts: 1. DatagramTransport exposes transport parameters for both client and server perspectives. The client just echoes what it received from the server (modulo any fields it might not have understood). 2. SDP adds a x-opaque line for opaque transport parameters. Identical to x-mt, but this is specific to datagram transport and goes in each m= section, and appears in the answer as well as the offer. - This is propagated to Jsep as part of the TransportDescription. - SDP files: transport_description.h,cc, transport_description_factory.h,cc, media_session.cc, webrtc_sdp.cc 3. JsepTransport/Controller: - Exposes opaque parameters for each mid (m= section). On offerer, this means pre-allocating a datagram transport and getting its parameters. On the answerer, this means echoing the offerer's parameters. - Uses a composite RTP transport to receive from either default RTP or datagram transport until both offer and answer arrive. - If a provisional answer arrives, sets the composite to send on the provisionally selected transport. - Once both offer and answer are set, deletes the unneeded transports and keeps whichever transport is selected. 4. PeerConnection pulls transport parameters out of Jsep and adds them to SDP. Bug: webrtc:9719 Change-Id: Ifcc428c8d76fb77dcc8abaa79507c620bcfb31b9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140920 Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28198}
2019-06-07 10:28:06 -07:00
EXPECT_EQ(media_desc_options_it->transport_options.opaque_parameters,
ti_audio->description.opaque_parameters);
} else {
EXPECT_TRUE(ti_audio == NULL);
}
const TransportInfo* ti_video = desc->GetTransportInfoByName("video");
if (options.has_video()) {
EXPECT_TRUE(ti_video != NULL);
Reland: Implement true negotiation for DatagramTransport with fallback to RTP. In short, the caller places a x-opaque line in SDP for each m= section that uses datagram transport. If the answerer supports datagram transport, it will parse this line and create a datagram transport. It will then echo the x-opaque line into the answer (to indicate that it accepted use of datagram transport). If the offer and answer contain exactly the same x-opaque line, both peers will use datagram transport. If the x-opaque line is omitted from the answer (or is different in the answer) they will fall back to RTP. Note that a different x-opaque line in the answer means the answerer did not understand something in the negotiation proto. Since WebRTC cannot know what was misunderstood, or whether it's still possible to use the datagram transport, it must fall back to RTP. This may change in the future, possibly by passing the answer to the datagram transport, but it's good enough for now. Negotiation consists of four parts: 1. DatagramTransport exposes transport parameters for both client and server perspectives. The client just echoes what it received from the server (modulo any fields it might not have understood). 2. SDP adds a x-opaque line for opaque transport parameters. Identical to x-mt, but this is specific to datagram transport and goes in each m= section, and appears in the answer as well as the offer. - This is propagated to Jsep as part of the TransportDescription. - SDP files: transport_description.h,cc, transport_description_factory.h,cc, media_session.cc, webrtc_sdp.cc 3. JsepTransport/Controller: - Exposes opaque parameters for each mid (m= section). On offerer, this means pre-allocating a datagram transport and getting its parameters. On the answerer, this means echoing the offerer's parameters. - Uses a composite RTP transport to receive from either default RTP or datagram transport until both offer and answer arrive. - If a provisional answer arrives, sets the composite to send on the provisionally selected transport. - Once both offer and answer are set, deletes the unneeded transports and keeps whichever transport is selected. 4. PeerConnection pulls transport parameters out of Jsep and adds them to SDP. Bug: webrtc:9719 Change-Id: Ifcc428c8d76fb77dcc8abaa79507c620bcfb31b9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140920 Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28198}
2019-06-07 10:28:06 -07:00
auto media_desc_options_it =
FindFirstMediaDescriptionByMid("video", options);
if (options.bundle_enabled) {
EXPECT_EQ(ti_audio->description.ice_ufrag,
ti_video->description.ice_ufrag);
EXPECT_EQ(ti_audio->description.ice_pwd, ti_video->description.ice_pwd);
Reland: Implement true negotiation for DatagramTransport with fallback to RTP. In short, the caller places a x-opaque line in SDP for each m= section that uses datagram transport. If the answerer supports datagram transport, it will parse this line and create a datagram transport. It will then echo the x-opaque line into the answer (to indicate that it accepted use of datagram transport). If the offer and answer contain exactly the same x-opaque line, both peers will use datagram transport. If the x-opaque line is omitted from the answer (or is different in the answer) they will fall back to RTP. Note that a different x-opaque line in the answer means the answerer did not understand something in the negotiation proto. Since WebRTC cannot know what was misunderstood, or whether it's still possible to use the datagram transport, it must fall back to RTP. This may change in the future, possibly by passing the answer to the datagram transport, but it's good enough for now. Negotiation consists of four parts: 1. DatagramTransport exposes transport parameters for both client and server perspectives. The client just echoes what it received from the server (modulo any fields it might not have understood). 2. SDP adds a x-opaque line for opaque transport parameters. Identical to x-mt, but this is specific to datagram transport and goes in each m= section, and appears in the answer as well as the offer. - This is propagated to Jsep as part of the TransportDescription. - SDP files: transport_description.h,cc, transport_description_factory.h,cc, media_session.cc, webrtc_sdp.cc 3. JsepTransport/Controller: - Exposes opaque parameters for each mid (m= section). On offerer, this means pre-allocating a datagram transport and getting its parameters. On the answerer, this means echoing the offerer's parameters. - Uses a composite RTP transport to receive from either default RTP or datagram transport until both offer and answer arrive. - If a provisional answer arrives, sets the composite to send on the provisionally selected transport. - Once both offer and answer are set, deletes the unneeded transports and keeps whichever transport is selected. 4. PeerConnection pulls transport parameters out of Jsep and adds them to SDP. Bug: webrtc:9719 Change-Id: Ifcc428c8d76fb77dcc8abaa79507c620bcfb31b9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140920 Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28198}
2019-06-07 10:28:06 -07:00
EXPECT_EQ(ti_audio->description.opaque_parameters,
ti_video->description.opaque_parameters);
} else {
if (has_current_desc) {
EXPECT_EQ(current_video_ufrag, ti_video->description.ice_ufrag);
EXPECT_EQ(current_video_pwd, ti_video->description.ice_pwd);
} else {
EXPECT_EQ(static_cast<size_t>(cricket::ICE_UFRAG_LENGTH),
ti_video->description.ice_ufrag.size());
EXPECT_EQ(static_cast<size_t>(cricket::ICE_PWD_LENGTH),
ti_video->description.ice_pwd.size());
}
Reland: Implement true negotiation for DatagramTransport with fallback to RTP. In short, the caller places a x-opaque line in SDP for each m= section that uses datagram transport. If the answerer supports datagram transport, it will parse this line and create a datagram transport. It will then echo the x-opaque line into the answer (to indicate that it accepted use of datagram transport). If the offer and answer contain exactly the same x-opaque line, both peers will use datagram transport. If the x-opaque line is omitted from the answer (or is different in the answer) they will fall back to RTP. Note that a different x-opaque line in the answer means the answerer did not understand something in the negotiation proto. Since WebRTC cannot know what was misunderstood, or whether it's still possible to use the datagram transport, it must fall back to RTP. This may change in the future, possibly by passing the answer to the datagram transport, but it's good enough for now. Negotiation consists of four parts: 1. DatagramTransport exposes transport parameters for both client and server perspectives. The client just echoes what it received from the server (modulo any fields it might not have understood). 2. SDP adds a x-opaque line for opaque transport parameters. Identical to x-mt, but this is specific to datagram transport and goes in each m= section, and appears in the answer as well as the offer. - This is propagated to Jsep as part of the TransportDescription. - SDP files: transport_description.h,cc, transport_description_factory.h,cc, media_session.cc, webrtc_sdp.cc 3. JsepTransport/Controller: - Exposes opaque parameters for each mid (m= section). On offerer, this means pre-allocating a datagram transport and getting its parameters. On the answerer, this means echoing the offerer's parameters. - Uses a composite RTP transport to receive from either default RTP or datagram transport until both offer and answer arrive. - If a provisional answer arrives, sets the composite to send on the provisionally selected transport. - Once both offer and answer are set, deletes the unneeded transports and keeps whichever transport is selected. 4. PeerConnection pulls transport parameters out of Jsep and adds them to SDP. Bug: webrtc:9719 Change-Id: Ifcc428c8d76fb77dcc8abaa79507c620bcfb31b9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140920 Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28198}
2019-06-07 10:28:06 -07:00
EXPECT_EQ(media_desc_options_it->transport_options.opaque_parameters,
ti_video->description.opaque_parameters);
}
EXPECT_EQ(
media_desc_options_it->transport_options.enable_ice_renomination,
GetIceRenomination(ti_video));
} else {
EXPECT_TRUE(ti_video == NULL);
}
const TransportInfo* ti_data = desc->GetTransportInfoByName("data");
if (options.has_data()) {
EXPECT_TRUE(ti_data != NULL);
if (options.bundle_enabled) {
EXPECT_EQ(ti_audio->description.ice_ufrag,
ti_data->description.ice_ufrag);
EXPECT_EQ(ti_audio->description.ice_pwd, ti_data->description.ice_pwd);
} else {
if (has_current_desc) {
EXPECT_EQ(current_data_ufrag, ti_data->description.ice_ufrag);
EXPECT_EQ(current_data_pwd, ti_data->description.ice_pwd);
} else {
EXPECT_EQ(static_cast<size_t>(cricket::ICE_UFRAG_LENGTH),
ti_data->description.ice_ufrag.size());
EXPECT_EQ(static_cast<size_t>(cricket::ICE_PWD_LENGTH),
ti_data->description.ice_pwd.size());
}
}
auto media_desc_options_it =
FindFirstMediaDescriptionByMid("data", options);
EXPECT_EQ(
media_desc_options_it->transport_options.enable_ice_renomination,
GetIceRenomination(ti_data));
} else {
Reland: Implement true negotiation for DatagramTransport with fallback to RTP. In short, the caller places a x-opaque line in SDP for each m= section that uses datagram transport. If the answerer supports datagram transport, it will parse this line and create a datagram transport. It will then echo the x-opaque line into the answer (to indicate that it accepted use of datagram transport). If the offer and answer contain exactly the same x-opaque line, both peers will use datagram transport. If the x-opaque line is omitted from the answer (or is different in the answer) they will fall back to RTP. Note that a different x-opaque line in the answer means the answerer did not understand something in the negotiation proto. Since WebRTC cannot know what was misunderstood, or whether it's still possible to use the datagram transport, it must fall back to RTP. This may change in the future, possibly by passing the answer to the datagram transport, but it's good enough for now. Negotiation consists of four parts: 1. DatagramTransport exposes transport parameters for both client and server perspectives. The client just echoes what it received from the server (modulo any fields it might not have understood). 2. SDP adds a x-opaque line for opaque transport parameters. Identical to x-mt, but this is specific to datagram transport and goes in each m= section, and appears in the answer as well as the offer. - This is propagated to Jsep as part of the TransportDescription. - SDP files: transport_description.h,cc, transport_description_factory.h,cc, media_session.cc, webrtc_sdp.cc 3. JsepTransport/Controller: - Exposes opaque parameters for each mid (m= section). On offerer, this means pre-allocating a datagram transport and getting its parameters. On the answerer, this means echoing the offerer's parameters. - Uses a composite RTP transport to receive from either default RTP or datagram transport until both offer and answer arrive. - If a provisional answer arrives, sets the composite to send on the provisionally selected transport. - Once both offer and answer are set, deletes the unneeded transports and keeps whichever transport is selected. 4. PeerConnection pulls transport parameters out of Jsep and adds them to SDP. Bug: webrtc:9719 Change-Id: Ifcc428c8d76fb77dcc8abaa79507c620bcfb31b9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140920 Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28198}
2019-06-07 10:28:06 -07:00
EXPECT_TRUE(ti_data == NULL);
}
}
void TestCryptoWithBundle(bool offer) {
f1_.set_secure(SEC_ENABLED);
MediaSessionOptions options;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly,
&options);
std::unique_ptr<SessionDescription> ref_desc;
std::unique_ptr<SessionDescription> desc;
if (offer) {
options.bundle_enabled = false;
ref_desc = f1_.CreateOffer(options, NULL);
options.bundle_enabled = true;
desc = f1_.CreateOffer(options, ref_desc.get());
} else {
options.bundle_enabled = true;
ref_desc = f1_.CreateOffer(options, NULL);
desc = f1_.CreateAnswer(ref_desc.get(), options, NULL);
}
ASSERT_TRUE(desc);
const cricket::MediaContentDescription* audio_media_desc =
desc->GetContentDescriptionByName("audio");
ASSERT_TRUE(audio_media_desc);
const cricket::MediaContentDescription* video_media_desc =
desc->GetContentDescriptionByName("video");
ASSERT_TRUE(video_media_desc);
EXPECT_TRUE(CompareCryptoParams(audio_media_desc->cryptos(),
video_media_desc->cryptos()));
EXPECT_EQ(1u, audio_media_desc->cryptos().size());
EXPECT_EQ(kDefaultSrtpCryptoSuite,
audio_media_desc->cryptos()[0].cipher_suite);
// Verify the selected crypto is one from the reference audio
// media content.
const cricket::MediaContentDescription* ref_audio_media_desc =
ref_desc->GetContentDescriptionByName("audio");
bool found = false;
for (size_t i = 0; i < ref_audio_media_desc->cryptos().size(); ++i) {
if (ref_audio_media_desc->cryptos()[i].Matches(
audio_media_desc->cryptos()[0])) {
found = true;
break;
}
}
EXPECT_TRUE(found);
}
// This test that the audio and video media direction is set to
// |expected_direction_in_answer| in an answer if the offer direction is set
// to |direction_in_offer| and the answer is willing to both send and receive.
void TestMediaDirectionInAnswer(
RtpTransceiverDirection direction_in_offer,
RtpTransceiverDirection expected_direction_in_answer) {
MediaSessionOptions offer_opts;
AddAudioVideoSections(direction_in_offer, &offer_opts);
std::unique_ptr<SessionDescription> offer =
f1_.CreateOffer(offer_opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
ContentInfo* ac_offer = offer->GetContentByName("audio");
ASSERT_TRUE(ac_offer != NULL);
ContentInfo* vc_offer = offer->GetContentByName("video");
ASSERT_TRUE(vc_offer != NULL);
MediaSessionOptions answer_opts;
AddAudioVideoSections(RtpTransceiverDirection::kSendRecv, &answer_opts);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), answer_opts, NULL);
const AudioContentDescription* acd_answer =
GetFirstAudioContentDescription(answer.get());
EXPECT_EQ(expected_direction_in_answer, acd_answer->direction());
const VideoContentDescription* vcd_answer =
GetFirstVideoContentDescription(answer.get());
EXPECT_EQ(expected_direction_in_answer, vcd_answer->direction());
}
bool VerifyNoCNCodecs(const cricket::ContentInfo* content) {
RTC_DCHECK(content);
RTC_CHECK(content->media_description());
const cricket::AudioContentDescription* audio_desc =
content->media_description()->as_audio();
RTC_CHECK(audio_desc);
for (const cricket::AudioCodec& codec : audio_desc->codecs()) {
if (codec.name == "CN") {
return false;
}
}
return true;
}
void TestVideoGcmCipher(bool gcm_offer, bool gcm_answer) {
MediaSessionOptions offer_opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &offer_opts);
Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class that only handles SRTP configuration to a more generic structure that can be used and extended for all per peer connection CryptoOptions that can be on a given PeerConnection. Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be accessed as crypto_options.srtp.whatever_option_name. This is more inline with other structures we have in WebRTC such as VideoConfig. As additional features are added over time this will allow the structure to remain compartmentalized and concerned components can only request a subset of the overall configuration structure e.g: void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config); In addition to this it made little sense for sslstreamadapter.h to hold all Srtp related configuration options. The header has become loo large and takes on too many responsibilities and spilting this up will lead to more maintainable code going forward. This will be used in a future CL to enable configuration options for the newly supported Frame Crypto. Reland Fix: - cryptooptions.h - now has enable_aes128_sha1_32_crypto_cipher as an optional root level configuration. - peerconnectionfactory - If this optional is set will now overwrite the underyling value. This along with the other field will be deprecated once dependent projects are updated. TBR=sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org Bug: webrtc:9681 Change-Id: Iaa6b741baafb85d352e42f54226119f19d97151d Reviewed-on: https://webrtc-review.googlesource.com/c/105560 Reviewed-by: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Emad Omara <emadomara@webrtc.org> Commit-Queue: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25135}
2018-10-11 15:33:17 -07:00
offer_opts.crypto_options.srtp.enable_gcm_crypto_suites = gcm_offer;
MediaSessionOptions answer_opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &answer_opts);
Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class that only handles SRTP configuration to a more generic structure that can be used and extended for all per peer connection CryptoOptions that can be on a given PeerConnection. Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be accessed as crypto_options.srtp.whatever_option_name. This is more inline with other structures we have in WebRTC such as VideoConfig. As additional features are added over time this will allow the structure to remain compartmentalized and concerned components can only request a subset of the overall configuration structure e.g: void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config); In addition to this it made little sense for sslstreamadapter.h to hold all Srtp related configuration options. The header has become loo large and takes on too many responsibilities and spilting this up will lead to more maintainable code going forward. This will be used in a future CL to enable configuration options for the newly supported Frame Crypto. Reland Fix: - cryptooptions.h - now has enable_aes128_sha1_32_crypto_cipher as an optional root level configuration. - peerconnectionfactory - If this optional is set will now overwrite the underyling value. This along with the other field will be deprecated once dependent projects are updated. TBR=sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org Bug: webrtc:9681 Change-Id: Iaa6b741baafb85d352e42f54226119f19d97151d Reviewed-on: https://webrtc-review.googlesource.com/c/105560 Reviewed-by: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Emad Omara <emadomara@webrtc.org> Commit-Queue: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25135}
2018-10-11 15:33:17 -07:00
answer_opts.crypto_options.srtp.enable_gcm_crypto_suites = gcm_answer;
f1_.set_secure(SEC_ENABLED);
f2_.set_secure(SEC_ENABLED);
std::unique_ptr<SessionDescription> offer =
f1_.CreateOffer(offer_opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
if (gcm_offer && gcm_answer) {
for (cricket::ContentInfo& content : offer->contents()) {
auto cryptos = content.media_description()->cryptos();
PreferGcmCryptoParameters(&cryptos);
content.media_description()->set_cryptos(cryptos);
}
}
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), answer_opts, NULL);
const ContentInfo* ac = answer->GetContentByName("audio");
const ContentInfo* vc = answer->GetContentByName("video");
ASSERT_TRUE(ac != NULL);
ASSERT_TRUE(vc != NULL);
EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
EXPECT_EQ(MediaProtocolType::kRtp, vc->type);
const AudioContentDescription* acd = ac->media_description()->as_audio();
const VideoContentDescription* vcd = vc->media_description()->as_video();
EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
EXPECT_EQ(kAutoBandwidth, acd->bandwidth()); // negotiated auto bw
EXPECT_EQ(0U, acd->first_ssrc()); // no sender is attached
EXPECT_TRUE(acd->rtcp_mux()); // negotiated rtcp-mux
if (gcm_offer && gcm_answer) {
ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuiteGcm);
} else {
ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
}
EXPECT_EQ(MEDIA_TYPE_VIDEO, vcd->type());
EXPECT_THAT(vcd->codecs(), ElementsAreArray(kVideoCodecsAnswer));
EXPECT_EQ(0U, vcd->first_ssrc()); // no sender is attached
EXPECT_TRUE(vcd->rtcp_mux()); // negotiated rtcp-mux
if (gcm_offer && gcm_answer) {
ASSERT_CRYPTO(vcd, 1U, kDefaultSrtpCryptoSuiteGcm);
} else {
ASSERT_CRYPTO(vcd, 1U, kDefaultSrtpCryptoSuite);
}
EXPECT_EQ(cricket::kMediaProtocolSavpf, vcd->protocol());
}
void TestTransportSequenceNumberNegotiation(
const cricket::RtpHeaderExtensions& local,
const cricket::RtpHeaderExtensions& offered,
const cricket::RtpHeaderExtensions& expectedAnswer) {
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
f1_.set_audio_rtp_header_extensions(offered);
f1_.set_video_rtp_header_extensions(offered);
f2_.set_audio_rtp_header_extensions(local);
f2_.set_video_rtp_header_extensions(local);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, NULL);
EXPECT_EQ(
expectedAnswer,
GetFirstAudioContentDescription(answer.get())->rtp_header_extensions());
EXPECT_EQ(
expectedAnswer,
GetFirstVideoContentDescription(answer.get())->rtp_header_extensions());
}
protected:
UniqueRandomIdGenerator ssrc_generator1;
UniqueRandomIdGenerator ssrc_generator2;
MediaSessionDescriptionFactory f1_;
MediaSessionDescriptionFactory f2_;
TransportDescriptionFactory tdf1_;
TransportDescriptionFactory tdf2_;
};
// Create a typical audio offer, and ensure it matches what we expect.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateAudioOffer) {
f1_.set_secure(SEC_ENABLED);
std::unique_ptr<SessionDescription> offer =
f1_.CreateOffer(CreatePlanBMediaSessionOptions(), NULL);
ASSERT_TRUE(offer.get() != NULL);
const ContentInfo* ac = offer->GetContentByName("audio");
const ContentInfo* vc = offer->GetContentByName("video");
ASSERT_TRUE(ac != NULL);
ASSERT_TRUE(vc == NULL);
EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
const AudioContentDescription* acd = ac->media_description()->as_audio();
EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
EXPECT_EQ(f1_.audio_sendrecv_codecs(), acd->codecs());
EXPECT_EQ(0U, acd->first_ssrc()); // no sender is attached.
EXPECT_EQ(kAutoBandwidth, acd->bandwidth()); // default bandwidth (auto)
EXPECT_TRUE(acd->rtcp_mux()); // rtcp-mux defaults on
ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
EXPECT_EQ(cricket::kMediaProtocolSavpf, acd->protocol());
}
// Create a typical video offer, and ensure it matches what we expect.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateVideoOffer) {
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
f1_.set_secure(SEC_ENABLED);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
const ContentInfo* ac = offer->GetContentByName("audio");
const ContentInfo* vc = offer->GetContentByName("video");
ASSERT_TRUE(ac != NULL);
ASSERT_TRUE(vc != NULL);
EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
EXPECT_EQ(MediaProtocolType::kRtp, vc->type);
const AudioContentDescription* acd = ac->media_description()->as_audio();
const VideoContentDescription* vcd = vc->media_description()->as_video();
EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
EXPECT_EQ(f1_.audio_sendrecv_codecs(), acd->codecs());
EXPECT_EQ(0U, acd->first_ssrc()); // no sender is attached
EXPECT_EQ(kAutoBandwidth, acd->bandwidth()); // default bandwidth (auto)
EXPECT_TRUE(acd->rtcp_mux()); // rtcp-mux defaults on
ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
EXPECT_EQ(cricket::kMediaProtocolSavpf, acd->protocol());
EXPECT_EQ(MEDIA_TYPE_VIDEO, vcd->type());
EXPECT_EQ(f1_.video_sendrecv_codecs(), vcd->codecs());
EXPECT_EQ(0U, vcd->first_ssrc()); // no sender is attached
EXPECT_EQ(kAutoBandwidth, vcd->bandwidth()); // default bandwidth (auto)
EXPECT_TRUE(vcd->rtcp_mux()); // rtcp-mux defaults on
ASSERT_CRYPTO(vcd, 1U, kDefaultSrtpCryptoSuite);
EXPECT_EQ(cricket::kMediaProtocolSavpf, vcd->protocol());
}
// Test creating an offer with bundle where the Codecs have the same dynamic
// RTP playlod type. The test verifies that the offer don't contain the
// duplicate RTP payload types.
TEST_F(MediaSessionDescriptionFactoryTest, TestBundleOfferWithSameCodecPlType) {
const VideoCodec& offered_video_codec = f2_.video_sendrecv_codecs()[0];
const AudioCodec& offered_audio_codec = f2_.audio_sendrecv_codecs()[0];
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
const RtpDataCodec& offered_data_codec = f2_.rtp_data_codecs()[0];
ASSERT_EQ(offered_video_codec.id, offered_audio_codec.id);
ASSERT_EQ(offered_video_codec.id, offered_data_codec.id);
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly, &opts);
opts.bundle_enabled = true;
std::unique_ptr<SessionDescription> offer = f2_.CreateOffer(opts, NULL);
const VideoContentDescription* vcd =
GetFirstVideoContentDescription(offer.get());
const AudioContentDescription* acd =
GetFirstAudioContentDescription(offer.get());
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
const RtpDataContentDescription* dcd =
GetFirstRtpDataContentDescription(offer.get());
ASSERT_TRUE(NULL != vcd);
ASSERT_TRUE(NULL != acd);
ASSERT_TRUE(NULL != dcd);
EXPECT_NE(vcd->codecs()[0].id, acd->codecs()[0].id);
EXPECT_NE(vcd->codecs()[0].id, dcd->codecs()[0].id);
EXPECT_NE(acd->codecs()[0].id, dcd->codecs()[0].id);
EXPECT_EQ(vcd->codecs()[0].name, offered_video_codec.name);
EXPECT_EQ(acd->codecs()[0].name, offered_audio_codec.name);
EXPECT_EQ(dcd->codecs()[0].name, offered_data_codec.name);
}
// Test creating an updated offer with bundle, audio, video and data
// after an audio only session has been negotiated.
TEST_F(MediaSessionDescriptionFactoryTest,
TestCreateUpdatedVideoOfferWithBundle) {
f1_.set_secure(SEC_ENABLED);
f2_.set_secure(SEC_ENABLED);
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
RtpTransceiverDirection::kInactive, kStopped,
&opts);
opts.data_channel_type = cricket::DCT_NONE;
opts.bundle_enabled = true;
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, NULL);
MediaSessionOptions updated_opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &updated_opts);
AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly,
&updated_opts);
updated_opts.bundle_enabled = true;
std::unique_ptr<SessionDescription> updated_offer(
f1_.CreateOffer(updated_opts, answer.get()));
const AudioContentDescription* acd =
GetFirstAudioContentDescription(updated_offer.get());
const VideoContentDescription* vcd =
GetFirstVideoContentDescription(updated_offer.get());
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
const RtpDataContentDescription* dcd =
GetFirstRtpDataContentDescription(updated_offer.get());
EXPECT_TRUE(NULL != vcd);
EXPECT_TRUE(NULL != acd);
EXPECT_TRUE(NULL != dcd);
ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
EXPECT_EQ(cricket::kMediaProtocolSavpf, acd->protocol());
ASSERT_CRYPTO(vcd, 1U, kDefaultSrtpCryptoSuite);
EXPECT_EQ(cricket::kMediaProtocolSavpf, vcd->protocol());
ASSERT_CRYPTO(dcd, 1U, kDefaultSrtpCryptoSuite);
EXPECT_EQ(cricket::kMediaProtocolSavpf, dcd->protocol());
}
// Create a RTP data offer, and ensure it matches what we expect.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateRtpDataOffer) {
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly, &opts);
f1_.set_secure(SEC_ENABLED);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
const ContentInfo* ac = offer->GetContentByName("audio");
const ContentInfo* dc = offer->GetContentByName("data");
ASSERT_TRUE(ac != NULL);
ASSERT_TRUE(dc != NULL);
EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
EXPECT_EQ(MediaProtocolType::kRtp, dc->type);
const AudioContentDescription* acd = ac->media_description()->as_audio();
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
const RtpDataContentDescription* dcd = dc->media_description()->as_rtp_data();
EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
EXPECT_EQ(f1_.audio_sendrecv_codecs(), acd->codecs());
EXPECT_EQ(0U, acd->first_ssrc()); // no sender is attched.
EXPECT_EQ(kAutoBandwidth, acd->bandwidth()); // default bandwidth (auto)
EXPECT_TRUE(acd->rtcp_mux()); // rtcp-mux defaults on
ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
EXPECT_EQ(cricket::kMediaProtocolSavpf, acd->protocol());
EXPECT_EQ(MEDIA_TYPE_DATA, dcd->type());
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
EXPECT_EQ(f1_.rtp_data_codecs(), dcd->codecs());
EXPECT_EQ(0U, dcd->first_ssrc()); // no sender is attached.
EXPECT_EQ(cricket::kDataMaxBandwidth,
dcd->bandwidth()); // default bandwidth (auto)
EXPECT_TRUE(dcd->rtcp_mux()); // rtcp-mux defaults on
ASSERT_CRYPTO(dcd, 1U, kDefaultSrtpCryptoSuite);
EXPECT_EQ(cricket::kMediaProtocolSavpf, dcd->protocol());
}
// Create an SCTP data offer with bundle without error.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateSctpDataOffer) {
MediaSessionOptions opts;
opts.bundle_enabled = true;
AddDataSection(cricket::DCT_SCTP, RtpTransceiverDirection::kSendRecv, &opts);
f1_.set_secure(SEC_ENABLED);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
EXPECT_TRUE(offer.get() != NULL);
EXPECT_TRUE(offer->GetContentByName("data") != NULL);
Reland "Change default secure SCTP protocol to UDP/DTLS/SCTP" This reverts commit 4436887ed2d3324279e0f2e091c9e9355392721a. Reason for revert: The original revert was intended to be temporary. Original change's description: > Revert "Change default secure SCTP protocol to UDP/DTLS/SCTP" > > This reverts commit c3f4820e129d44471b366b8885a67b5392918d5a. > > Reason for revert: Will temporarily revert to fix an issue and reland afterwards. > > Original change's description: > > Change default secure SCTP protocol to UDP/DTLS/SCTP > > > > The old value - DTLS/SCTP - is not standards conformant, > > and the new value should be parsable since Chrome M61. > > > > Bug: webrtc:7706 > > Change-Id: I7468cc9597dec4ef4b102fccddc4e981fed7e8d8 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136804 > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#27940} > > TBR=steveanton@webrtc.org,mbonadei@webrtc.org,hbos@webrtc.org,hta@webrtc.org > > # Not skipping CQ checks because original CL landed > 1 day ago. > > Bug: webrtc:7706 > Change-Id: Ida8ae20767485c75edc44dff8a3fa1af2006f207 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139244 > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Commit-Queue: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28121} TBR=steveanton@webrtc.org,mbonadei@webrtc.org,hbos@webrtc.org,hta@webrtc.org,guidou@webrtc.org Change-Id: I381fa18b644874c20ddaa4cd13fec79a5fd9555a No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7706 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139246 Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28122}
2019-05-31 10:17:38 +00:00
auto dcd = GetFirstSctpDataContentDescription(offer.get());
ASSERT_TRUE(dcd);
// Since this transport is insecure, the protocol should be "SCTP".
EXPECT_EQ(cricket::kMediaProtocolSctp, dcd->protocol());
}
// Create an SCTP data offer with bundle without error.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateSecureSctpDataOffer) {
MediaSessionOptions opts;
opts.bundle_enabled = true;
AddDataSection(cricket::DCT_SCTP, RtpTransceiverDirection::kSendRecv, &opts);
f1_.set_secure(SEC_ENABLED);
tdf1_.set_secure(SEC_ENABLED);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
EXPECT_TRUE(offer.get() != NULL);
EXPECT_TRUE(offer->GetContentByName("data") != NULL);
auto dcd = GetFirstSctpDataContentDescription(offer.get());
ASSERT_TRUE(dcd);
// The protocol should now be "UDP/DTLS/SCTP"
EXPECT_EQ(cricket::kMediaProtocolUdpDtlsSctp, dcd->protocol());
}
// Test creating an sctp data channel from an already generated offer.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateImplicitSctpDataOffer) {
MediaSessionOptions opts;
opts.bundle_enabled = true;
AddDataSection(cricket::DCT_SCTP, RtpTransceiverDirection::kSendRecv, &opts);
f1_.set_secure(SEC_ENABLED);
std::unique_ptr<SessionDescription> offer1(f1_.CreateOffer(opts, NULL));
ASSERT_TRUE(offer1.get() != NULL);
const ContentInfo* data = offer1->GetContentByName("data");
ASSERT_TRUE(data != NULL);
ASSERT_EQ(cricket::kMediaProtocolSctp, data->media_description()->protocol());
// Now set data_channel_type to 'none' (default) and make sure that the
// datachannel type that gets generated from the previous offer, is of the
// same type.
opts.data_channel_type = cricket::DCT_NONE;
std::unique_ptr<SessionDescription> offer2(
f1_.CreateOffer(opts, offer1.get()));
data = offer2->GetContentByName("data");
ASSERT_TRUE(data != NULL);
EXPECT_EQ(cricket::kMediaProtocolSctp, data->media_description()->protocol());
}
// Test that if BUNDLE is enabled and all media sections are rejected then the
// BUNDLE group is not present in the re-offer.
TEST_F(MediaSessionDescriptionFactoryTest, ReOfferNoBundleGroupIfAllRejected) {
MediaSessionOptions opts;
opts.bundle_enabled = true;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
opts.media_description_options[0].stopped = true;
std::unique_ptr<SessionDescription> reoffer =
f1_.CreateOffer(opts, offer.get());
EXPECT_FALSE(reoffer->GetGroupByName(cricket::GROUP_TYPE_BUNDLE));
}
// Test that if BUNDLE is enabled and the remote re-offer does not include a
// BUNDLE group since all media sections are rejected, then the re-answer also
// does not include a BUNDLE group.
TEST_F(MediaSessionDescriptionFactoryTest, ReAnswerNoBundleGroupIfAllRejected) {
MediaSessionOptions opts;
opts.bundle_enabled = true;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, nullptr);
opts.media_description_options[0].stopped = true;
std::unique_ptr<SessionDescription> reoffer =
f1_.CreateOffer(opts, offer.get());
std::unique_ptr<SessionDescription> reanswer =
f2_.CreateAnswer(reoffer.get(), opts, answer.get());
EXPECT_FALSE(reanswer->GetGroupByName(cricket::GROUP_TYPE_BUNDLE));
}
// Test that if BUNDLE is enabled and the previous offerer-tagged media section
// was rejected then the new offerer-tagged media section is the non-rejected
// media section.
TEST_F(MediaSessionDescriptionFactoryTest, ReOfferChangeBundleOffererTagged) {
MediaSessionOptions opts;
opts.bundle_enabled = true;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
// Reject the audio m= section and add a video m= section.
opts.media_description_options[0].stopped = true;
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::unique_ptr<SessionDescription> reoffer =
f1_.CreateOffer(opts, offer.get());
const cricket::ContentGroup* bundle_group =
reoffer->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
ASSERT_TRUE(bundle_group);
EXPECT_FALSE(bundle_group->HasContentName("audio"));
EXPECT_TRUE(bundle_group->HasContentName("video"));
}
// Test that if BUNDLE is enabled and the previous offerer-tagged media section
// was rejected and a new media section is added, then the re-answer BUNDLE
// group will contain only the non-rejected media section.
TEST_F(MediaSessionDescriptionFactoryTest, ReAnswerChangedBundleOffererTagged) {
MediaSessionOptions opts;
opts.bundle_enabled = true;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, nullptr);
// Reject the audio m= section and add a video m= section.
opts.media_description_options[0].stopped = true;
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::unique_ptr<SessionDescription> reoffer =
f1_.CreateOffer(opts, offer.get());
std::unique_ptr<SessionDescription> reanswer =
f2_.CreateAnswer(reoffer.get(), opts, answer.get());
const cricket::ContentGroup* bundle_group =
reanswer->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
ASSERT_TRUE(bundle_group);
EXPECT_FALSE(bundle_group->HasContentName("audio"));
EXPECT_TRUE(bundle_group->HasContentName("video"));
}
// Test that if the BUNDLE offerer-tagged media section is changed in a reoffer
// and there is still a non-rejected media section that was in the initial
// offer, then the ICE credentials do not change in the reoffer offerer-tagged
// media section.
TEST_F(MediaSessionDescriptionFactoryTest,
ReOfferChangeBundleOffererTaggedKeepsIceCredentials) {
MediaSessionOptions opts;
opts.bundle_enabled = true;
AddAudioVideoSections(RtpTransceiverDirection::kSendRecv, &opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, nullptr);
// Reject the audio m= section.
opts.media_description_options[0].stopped = true;
std::unique_ptr<SessionDescription> reoffer =
f1_.CreateOffer(opts, offer.get());
const TransportDescription* offer_tagged =
offer->GetTransportDescriptionByName("audio");
ASSERT_TRUE(offer_tagged);
const TransportDescription* reoffer_tagged =
reoffer->GetTransportDescriptionByName("video");
ASSERT_TRUE(reoffer_tagged);
EXPECT_EQ(offer_tagged->ice_ufrag, reoffer_tagged->ice_ufrag);
EXPECT_EQ(offer_tagged->ice_pwd, reoffer_tagged->ice_pwd);
}
// Test that if the BUNDLE offerer-tagged media section is changed in a reoffer
// and there is still a non-rejected media section that was in the initial
// offer, then the ICE credentials do not change in the reanswer answerer-tagged
// media section.
TEST_F(MediaSessionDescriptionFactoryTest,
ReAnswerChangeBundleOffererTaggedKeepsIceCredentials) {
MediaSessionOptions opts;
opts.bundle_enabled = true;
AddAudioVideoSections(RtpTransceiverDirection::kSendRecv, &opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, nullptr);
// Reject the audio m= section.
opts.media_description_options[0].stopped = true;
std::unique_ptr<SessionDescription> reoffer =
f1_.CreateOffer(opts, offer.get());
std::unique_ptr<SessionDescription> reanswer =
f2_.CreateAnswer(reoffer.get(), opts, answer.get());
const TransportDescription* answer_tagged =
answer->GetTransportDescriptionByName("audio");
ASSERT_TRUE(answer_tagged);
const TransportDescription* reanswer_tagged =
reanswer->GetTransportDescriptionByName("video");
ASSERT_TRUE(reanswer_tagged);
EXPECT_EQ(answer_tagged->ice_ufrag, reanswer_tagged->ice_ufrag);
EXPECT_EQ(answer_tagged->ice_pwd, reanswer_tagged->ice_pwd);
}
// Create an audio, video offer without legacy StreamParams.
TEST_F(MediaSessionDescriptionFactoryTest,
TestCreateOfferWithoutLegacyStreams) {
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
const ContentInfo* ac = offer->GetContentByName("audio");
const ContentInfo* vc = offer->GetContentByName("video");
ASSERT_TRUE(ac != NULL);
ASSERT_TRUE(vc != NULL);
const AudioContentDescription* acd = ac->media_description()->as_audio();
const VideoContentDescription* vcd = vc->media_description()->as_video();
EXPECT_FALSE(vcd->has_ssrcs()); // No StreamParams.
EXPECT_FALSE(acd->has_ssrcs()); // No StreamParams.
}
// Creates an audio+video sendonly offer.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateSendOnlyOffer) {
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kSendOnly, &opts);
AttachSenderToMediaDescriptionOptions("video", MEDIA_TYPE_VIDEO, kVideoTrack1,
{kMediaStream1}, 1, &opts);
AttachSenderToMediaDescriptionOptions("audio", MEDIA_TYPE_AUDIO, kAudioTrack1,
{kMediaStream1}, 1, &opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
EXPECT_EQ(2u, offer->contents().size());
EXPECT_TRUE(IsMediaContentOfType(&offer->contents()[0], MEDIA_TYPE_AUDIO));
EXPECT_TRUE(IsMediaContentOfType(&offer->contents()[1], MEDIA_TYPE_VIDEO));
EXPECT_EQ(RtpTransceiverDirection::kSendOnly,
GetMediaDirection(&offer->contents()[0]));
EXPECT_EQ(RtpTransceiverDirection::kSendOnly,
GetMediaDirection(&offer->contents()[1]));
}
// Verifies that the order of the media contents in the current
// SessionDescription is preserved in the new SessionDescription.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateOfferContentOrder) {
MediaSessionOptions opts;
AddDataSection(cricket::DCT_SCTP, RtpTransceiverDirection::kSendRecv, &opts);
std::unique_ptr<SessionDescription> offer1(f1_.CreateOffer(opts, NULL));
ASSERT_TRUE(offer1.get() != NULL);
EXPECT_EQ(1u, offer1->contents().size());
EXPECT_TRUE(IsMediaContentOfType(&offer1->contents()[0], MEDIA_TYPE_DATA));
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
std::unique_ptr<SessionDescription> offer2(
f1_.CreateOffer(opts, offer1.get()));
ASSERT_TRUE(offer2.get() != NULL);
EXPECT_EQ(2u, offer2->contents().size());
EXPECT_TRUE(IsMediaContentOfType(&offer2->contents()[0], MEDIA_TYPE_DATA));
EXPECT_TRUE(IsMediaContentOfType(&offer2->contents()[1], MEDIA_TYPE_VIDEO));
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
std::unique_ptr<SessionDescription> offer3(
f1_.CreateOffer(opts, offer2.get()));
ASSERT_TRUE(offer3.get() != NULL);
EXPECT_EQ(3u, offer3->contents().size());
EXPECT_TRUE(IsMediaContentOfType(&offer3->contents()[0], MEDIA_TYPE_DATA));
EXPECT_TRUE(IsMediaContentOfType(&offer3->contents()[1], MEDIA_TYPE_VIDEO));
EXPECT_TRUE(IsMediaContentOfType(&offer3->contents()[2], MEDIA_TYPE_AUDIO));
}
// Create a typical audio answer, and ensure it matches what we expect.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateAudioAnswer) {
f1_.set_secure(SEC_ENABLED);
f2_.set_secure(SEC_ENABLED);
std::unique_ptr<SessionDescription> offer =
f1_.CreateOffer(CreatePlanBMediaSessionOptions(), NULL);
ASSERT_TRUE(offer.get() != NULL);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), CreatePlanBMediaSessionOptions(), NULL);
const ContentInfo* ac = answer->GetContentByName("audio");
const ContentInfo* vc = answer->GetContentByName("video");
ASSERT_TRUE(ac != NULL);
ASSERT_TRUE(vc == NULL);
EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
const AudioContentDescription* acd = ac->media_description()->as_audio();
EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
EXPECT_EQ(0U, acd->first_ssrc()); // no sender is attached
EXPECT_EQ(kAutoBandwidth, acd->bandwidth()); // negotiated auto bw
EXPECT_TRUE(acd->rtcp_mux()); // negotiated rtcp-mux
ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
EXPECT_EQ(cricket::kMediaProtocolSavpf, acd->protocol());
}
// Create a typical audio answer with GCM ciphers enabled, and ensure it
// matches what we expect.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateAudioAnswerGcm) {
f1_.set_secure(SEC_ENABLED);
f2_.set_secure(SEC_ENABLED);
MediaSessionOptions opts = CreatePlanBMediaSessionOptions();
Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class that only handles SRTP configuration to a more generic structure that can be used and extended for all per peer connection CryptoOptions that can be on a given PeerConnection. Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be accessed as crypto_options.srtp.whatever_option_name. This is more inline with other structures we have in WebRTC such as VideoConfig. As additional features are added over time this will allow the structure to remain compartmentalized and concerned components can only request a subset of the overall configuration structure e.g: void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config); In addition to this it made little sense for sslstreamadapter.h to hold all Srtp related configuration options. The header has become loo large and takes on too many responsibilities and spilting this up will lead to more maintainable code going forward. This will be used in a future CL to enable configuration options for the newly supported Frame Crypto. Reland Fix: - cryptooptions.h - now has enable_aes128_sha1_32_crypto_cipher as an optional root level configuration. - peerconnectionfactory - If this optional is set will now overwrite the underyling value. This along with the other field will be deprecated once dependent projects are updated. TBR=sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org Bug: webrtc:9681 Change-Id: Iaa6b741baafb85d352e42f54226119f19d97151d Reviewed-on: https://webrtc-review.googlesource.com/c/105560 Reviewed-by: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Emad Omara <emadomara@webrtc.org> Commit-Queue: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25135}
2018-10-11 15:33:17 -07:00
opts.crypto_options.srtp.enable_gcm_crypto_suites = true;
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
for (cricket::ContentInfo& content : offer->contents()) {
auto cryptos = content.media_description()->cryptos();
PreferGcmCryptoParameters(&cryptos);
content.media_description()->set_cryptos(cryptos);
}
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, NULL);
const ContentInfo* ac = answer->GetContentByName("audio");
const ContentInfo* vc = answer->GetContentByName("video");
ASSERT_TRUE(ac != NULL);
ASSERT_TRUE(vc == NULL);
EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
const AudioContentDescription* acd = ac->media_description()->as_audio();
EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
EXPECT_EQ(0U, acd->first_ssrc()); // no sender is attached
EXPECT_EQ(kAutoBandwidth, acd->bandwidth()); // negotiated auto bw
EXPECT_TRUE(acd->rtcp_mux()); // negotiated rtcp-mux
ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuiteGcm);
EXPECT_EQ(cricket::kMediaProtocolSavpf, acd->protocol());
}
// Create a typical video answer, and ensure it matches what we expect.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateVideoAnswer) {
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
f1_.set_secure(SEC_ENABLED);
f2_.set_secure(SEC_ENABLED);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, NULL);
const ContentInfo* ac = answer->GetContentByName("audio");
const ContentInfo* vc = answer->GetContentByName("video");
ASSERT_TRUE(ac != NULL);
ASSERT_TRUE(vc != NULL);
EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
EXPECT_EQ(MediaProtocolType::kRtp, vc->type);
const AudioContentDescription* acd = ac->media_description()->as_audio();
const VideoContentDescription* vcd = vc->media_description()->as_video();
EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
EXPECT_EQ(kAutoBandwidth, acd->bandwidth()); // negotiated auto bw
EXPECT_EQ(0U, acd->first_ssrc()); // no sender is attached
EXPECT_TRUE(acd->rtcp_mux()); // negotiated rtcp-mux
ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
EXPECT_EQ(MEDIA_TYPE_VIDEO, vcd->type());
EXPECT_THAT(vcd->codecs(), ElementsAreArray(kVideoCodecsAnswer));
EXPECT_EQ(0U, vcd->first_ssrc()); // no sender is attached
EXPECT_TRUE(vcd->rtcp_mux()); // negotiated rtcp-mux
ASSERT_CRYPTO(vcd, 1U, kDefaultSrtpCryptoSuite);
EXPECT_EQ(cricket::kMediaProtocolSavpf, vcd->protocol());
}
// Create a typical video answer with GCM ciphers enabled, and ensure it
// matches what we expect.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateVideoAnswerGcm) {
TestVideoGcmCipher(true, true);
}
// Create a typical video answer with GCM ciphers enabled for the offer only,
// and ensure it matches what we expect.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateVideoAnswerGcmOffer) {
TestVideoGcmCipher(true, false);
}
// Create a typical video answer with GCM ciphers enabled for the answer only,
// and ensure it matches what we expect.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateVideoAnswerGcmAnswer) {
TestVideoGcmCipher(false, true);
}
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateDataAnswer) {
MediaSessionOptions opts = CreatePlanBMediaSessionOptions();
AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly, &opts);
f1_.set_secure(SEC_ENABLED);
f2_.set_secure(SEC_ENABLED);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, NULL);
const ContentInfo* ac = answer->GetContentByName("audio");
const ContentInfo* dc = answer->GetContentByName("data");
ASSERT_TRUE(ac != NULL);
ASSERT_TRUE(dc != NULL);
EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
EXPECT_EQ(MediaProtocolType::kRtp, dc->type);
const AudioContentDescription* acd = ac->media_description()->as_audio();
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
const RtpDataContentDescription* dcd = dc->media_description()->as_rtp_data();
EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
EXPECT_EQ(kAutoBandwidth, acd->bandwidth()); // negotiated auto bw
EXPECT_EQ(0U, acd->first_ssrc()); // no sender is attached
EXPECT_TRUE(acd->rtcp_mux()); // negotiated rtcp-mux
ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
EXPECT_EQ(MEDIA_TYPE_DATA, dcd->type());
EXPECT_THAT(dcd->codecs(), ElementsAreArray(kDataCodecsAnswer));
EXPECT_EQ(0U, dcd->first_ssrc()); // no sender is attached
EXPECT_TRUE(dcd->rtcp_mux()); // negotiated rtcp-mux
ASSERT_CRYPTO(dcd, 1U, kDefaultSrtpCryptoSuite);
EXPECT_EQ(cricket::kMediaProtocolSavpf, dcd->protocol());
}
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateDataAnswerGcm) {
MediaSessionOptions opts = CreatePlanBMediaSessionOptions();
AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly, &opts);
Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class that only handles SRTP configuration to a more generic structure that can be used and extended for all per peer connection CryptoOptions that can be on a given PeerConnection. Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be accessed as crypto_options.srtp.whatever_option_name. This is more inline with other structures we have in WebRTC such as VideoConfig. As additional features are added over time this will allow the structure to remain compartmentalized and concerned components can only request a subset of the overall configuration structure e.g: void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config); In addition to this it made little sense for sslstreamadapter.h to hold all Srtp related configuration options. The header has become loo large and takes on too many responsibilities and spilting this up will lead to more maintainable code going forward. This will be used in a future CL to enable configuration options for the newly supported Frame Crypto. Reland Fix: - cryptooptions.h - now has enable_aes128_sha1_32_crypto_cipher as an optional root level configuration. - peerconnectionfactory - If this optional is set will now overwrite the underyling value. This along with the other field will be deprecated once dependent projects are updated. TBR=sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org Bug: webrtc:9681 Change-Id: Iaa6b741baafb85d352e42f54226119f19d97151d Reviewed-on: https://webrtc-review.googlesource.com/c/105560 Reviewed-by: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Emad Omara <emadomara@webrtc.org> Commit-Queue: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25135}
2018-10-11 15:33:17 -07:00
opts.crypto_options.srtp.enable_gcm_crypto_suites = true;
f1_.set_secure(SEC_ENABLED);
f2_.set_secure(SEC_ENABLED);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
for (cricket::ContentInfo& content : offer->contents()) {
auto cryptos = content.media_description()->cryptos();
PreferGcmCryptoParameters(&cryptos);
content.media_description()->set_cryptos(cryptos);
}
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, NULL);
const ContentInfo* ac = answer->GetContentByName("audio");
const ContentInfo* dc = answer->GetContentByName("data");
ASSERT_TRUE(ac != NULL);
ASSERT_TRUE(dc != NULL);
EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
EXPECT_EQ(MediaProtocolType::kRtp, dc->type);
const AudioContentDescription* acd = ac->media_description()->as_audio();
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
const RtpDataContentDescription* dcd = dc->media_description()->as_rtp_data();
EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
EXPECT_EQ(kAutoBandwidth, acd->bandwidth()); // negotiated auto bw
EXPECT_EQ(0U, acd->first_ssrc()); // no sender is attached
EXPECT_TRUE(acd->rtcp_mux()); // negotiated rtcp-mux
ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuiteGcm);
EXPECT_EQ(MEDIA_TYPE_DATA, dcd->type());
EXPECT_THAT(dcd->codecs(), ElementsAreArray(kDataCodecsAnswer));
EXPECT_EQ(0U, dcd->first_ssrc()); // no sender is attached
EXPECT_TRUE(dcd->rtcp_mux()); // negotiated rtcp-mux
ASSERT_CRYPTO(dcd, 1U, kDefaultSrtpCryptoSuiteGcm);
EXPECT_EQ(cricket::kMediaProtocolSavpf, dcd->protocol());
}
// The use_sctpmap flag should be set in an Sctp DataContentDescription by
// default. The answer's use_sctpmap flag should match the offer's.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateDataAnswerUsesSctpmap) {
MediaSessionOptions opts;
AddDataSection(cricket::DCT_SCTP, RtpTransceiverDirection::kSendRecv, &opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
ContentInfo* dc_offer = offer->GetContentByName("data");
ASSERT_TRUE(dc_offer != NULL);
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
SctpDataContentDescription* dcd_offer =
dc_offer->media_description()->as_sctp();
EXPECT_TRUE(dcd_offer->use_sctpmap());
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, NULL);
const ContentInfo* dc_answer = answer->GetContentByName("data");
ASSERT_TRUE(dc_answer != NULL);
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
const SctpDataContentDescription* dcd_answer =
dc_answer->media_description()->as_sctp();
EXPECT_TRUE(dcd_answer->use_sctpmap());
}
// The answer's use_sctpmap flag should match the offer's.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateDataAnswerWithoutSctpmap) {
MediaSessionOptions opts;
AddDataSection(cricket::DCT_SCTP, RtpTransceiverDirection::kSendRecv, &opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
ContentInfo* dc_offer = offer->GetContentByName("data");
ASSERT_TRUE(dc_offer != NULL);
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
SctpDataContentDescription* dcd_offer =
dc_offer->media_description()->as_sctp();
dcd_offer->set_use_sctpmap(false);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, NULL);
const ContentInfo* dc_answer = answer->GetContentByName("data");
ASSERT_TRUE(dc_answer != NULL);
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
const SctpDataContentDescription* dcd_answer =
dc_answer->media_description()->as_sctp();
EXPECT_FALSE(dcd_answer->use_sctpmap());
}
// Test that a valid answer will be created for "DTLS/SCTP", "UDP/DTLS/SCTP"
// and "TCP/DTLS/SCTP" offers.
TEST_F(MediaSessionDescriptionFactoryTest,
TestCreateDataAnswerToDifferentOfferedProtos) {
// Need to enable DTLS offer/answer generation (disabled by default in this
// test).
f1_.set_secure(SEC_ENABLED);
f2_.set_secure(SEC_ENABLED);
tdf1_.set_secure(SEC_ENABLED);
tdf2_.set_secure(SEC_ENABLED);
MediaSessionOptions opts;
AddDataSection(cricket::DCT_SCTP, RtpTransceiverDirection::kSendRecv, &opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
ASSERT_TRUE(offer.get() != nullptr);
ContentInfo* dc_offer = offer->GetContentByName("data");
ASSERT_TRUE(dc_offer != nullptr);
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
SctpDataContentDescription* dcd_offer =
dc_offer->media_description()->as_sctp();
ASSERT_TRUE(dcd_offer);
std::vector<std::string> protos = {"DTLS/SCTP", "UDP/DTLS/SCTP",
"TCP/DTLS/SCTP"};
for (const std::string& proto : protos) {
dcd_offer->set_protocol(proto);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, nullptr);
const ContentInfo* dc_answer = answer->GetContentByName("data");
ASSERT_TRUE(dc_answer != nullptr);
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
const SctpDataContentDescription* dcd_answer =
dc_answer->media_description()->as_sctp();
EXPECT_FALSE(dc_answer->rejected);
EXPECT_EQ(proto, dcd_answer->protocol());
}
}
TEST_F(MediaSessionDescriptionFactoryTest,
TestCreateDataAnswerToOfferWithDefinedMessageSize) {
// Need to enable DTLS offer/answer generation (disabled by default in this
// test).
f1_.set_secure(SEC_ENABLED);
f2_.set_secure(SEC_ENABLED);
tdf1_.set_secure(SEC_ENABLED);
tdf2_.set_secure(SEC_ENABLED);
MediaSessionOptions opts;
AddDataSection(cricket::DCT_SCTP, RtpTransceiverDirection::kSendRecv, &opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
ASSERT_TRUE(offer.get() != nullptr);
ContentInfo* dc_offer = offer->GetContentByName("data");
ASSERT_TRUE(dc_offer != nullptr);
SctpDataContentDescription* dcd_offer =
dc_offer->media_description()->as_sctp();
ASSERT_TRUE(dcd_offer);
dcd_offer->set_max_message_size(1234);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, nullptr);
const ContentInfo* dc_answer = answer->GetContentByName("data");
ASSERT_TRUE(dc_answer != nullptr);
const SctpDataContentDescription* dcd_answer =
dc_answer->media_description()->as_sctp();
EXPECT_FALSE(dc_answer->rejected);
EXPECT_EQ(1234, dcd_answer->max_message_size());
}
TEST_F(MediaSessionDescriptionFactoryTest,
TestCreateDataAnswerToOfferWithZeroMessageSize) {
// Need to enable DTLS offer/answer generation (disabled by default in this
// test).
f1_.set_secure(SEC_ENABLED);
f2_.set_secure(SEC_ENABLED);
tdf1_.set_secure(SEC_ENABLED);
tdf2_.set_secure(SEC_ENABLED);
MediaSessionOptions opts;
AddDataSection(cricket::DCT_SCTP, RtpTransceiverDirection::kSendRecv, &opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
ASSERT_TRUE(offer.get() != nullptr);
ContentInfo* dc_offer = offer->GetContentByName("data");
ASSERT_TRUE(dc_offer != nullptr);
SctpDataContentDescription* dcd_offer =
dc_offer->media_description()->as_sctp();
ASSERT_TRUE(dcd_offer);
dcd_offer->set_max_message_size(0);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, nullptr);
const ContentInfo* dc_answer = answer->GetContentByName("data");
ASSERT_TRUE(dc_answer != nullptr);
const SctpDataContentDescription* dcd_answer =
dc_answer->media_description()->as_sctp();
EXPECT_FALSE(dc_answer->rejected);
EXPECT_EQ(cricket::kSctpSendBufferSize, dcd_answer->max_message_size());
}
// Verifies that the order of the media contents in the offer is preserved in
// the answer.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateAnswerContentOrder) {
MediaSessionOptions opts;
// Creates a data only offer.
AddDataSection(cricket::DCT_SCTP, RtpTransceiverDirection::kSendRecv, &opts);
std::unique_ptr<SessionDescription> offer1(f1_.CreateOffer(opts, NULL));
ASSERT_TRUE(offer1.get() != NULL);
// Appends audio to the offer.
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
std::unique_ptr<SessionDescription> offer2(
f1_.CreateOffer(opts, offer1.get()));
ASSERT_TRUE(offer2.get() != NULL);
// Appends video to the offer.
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
std::unique_ptr<SessionDescription> offer3(
f1_.CreateOffer(opts, offer2.get()));
ASSERT_TRUE(offer3.get() != NULL);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer3.get(), opts, NULL);
ASSERT_TRUE(answer.get() != NULL);
EXPECT_EQ(3u, answer->contents().size());
EXPECT_TRUE(IsMediaContentOfType(&answer->contents()[0], MEDIA_TYPE_DATA));
EXPECT_TRUE(IsMediaContentOfType(&answer->contents()[1], MEDIA_TYPE_AUDIO));
EXPECT_TRUE(IsMediaContentOfType(&answer->contents()[2], MEDIA_TYPE_VIDEO));
}
// TODO(deadbeef): Extend these tests to ensure the correct direction with other
// answerer settings.
// This test that the media direction is set to send/receive in an answer if
// the offer is send receive.
TEST_F(MediaSessionDescriptionFactoryTest, CreateAnswerToSendReceiveOffer) {
TestMediaDirectionInAnswer(RtpTransceiverDirection::kSendRecv,
RtpTransceiverDirection::kSendRecv);
}
// This test that the media direction is set to receive only in an answer if
// the offer is send only.
TEST_F(MediaSessionDescriptionFactoryTest, CreateAnswerToSendOnlyOffer) {
TestMediaDirectionInAnswer(RtpTransceiverDirection::kSendOnly,
RtpTransceiverDirection::kRecvOnly);
}
// This test that the media direction is set to send only in an answer if
// the offer is recv only.
TEST_F(MediaSessionDescriptionFactoryTest, CreateAnswerToRecvOnlyOffer) {
TestMediaDirectionInAnswer(RtpTransceiverDirection::kRecvOnly,
RtpTransceiverDirection::kSendOnly);
}
// This test that the media direction is set to inactive in an answer if
// the offer is inactive.
TEST_F(MediaSessionDescriptionFactoryTest, CreateAnswerToInactiveOffer) {
TestMediaDirectionInAnswer(RtpTransceiverDirection::kInactive,
RtpTransceiverDirection::kInactive);
}
// Test that a data content with an unknown protocol is rejected in an answer.
TEST_F(MediaSessionDescriptionFactoryTest,
CreateDataAnswerToOfferWithUnknownProtocol) {
MediaSessionOptions opts;
AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly, &opts);
f1_.set_secure(SEC_ENABLED);
f2_.set_secure(SEC_ENABLED);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ContentInfo* dc_offer = offer->GetContentByName("data");
ASSERT_TRUE(dc_offer != NULL);
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
RtpDataContentDescription* dcd_offer =
dc_offer->media_description()->as_rtp_data();
ASSERT_TRUE(dcd_offer != NULL);
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
// Offer must be acceptable as an RTP protocol in order to be set.
std::string protocol = "RTP/a weird unknown protocol";
dcd_offer->set_protocol(protocol);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, NULL);
const ContentInfo* dc_answer = answer->GetContentByName("data");
ASSERT_TRUE(dc_answer != NULL);
EXPECT_TRUE(dc_answer->rejected);
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
const RtpDataContentDescription* dcd_answer =
dc_answer->media_description()->as_rtp_data();
ASSERT_TRUE(dcd_answer != NULL);
EXPECT_EQ(protocol, dcd_answer->protocol());
}
// Test that the media protocol is RTP/AVPF if DTLS and SDES are disabled.
TEST_F(MediaSessionDescriptionFactoryTest, AudioOfferAnswerWithCryptoDisabled) {
MediaSessionOptions opts = CreatePlanBMediaSessionOptions();
f1_.set_secure(SEC_DISABLED);
f2_.set_secure(SEC_DISABLED);
tdf1_.set_secure(SEC_DISABLED);
tdf2_.set_secure(SEC_DISABLED);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
const AudioContentDescription* offer_acd =
GetFirstAudioContentDescription(offer.get());
ASSERT_TRUE(offer_acd != NULL);
EXPECT_EQ(cricket::kMediaProtocolAvpf, offer_acd->protocol());
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, NULL);
const ContentInfo* ac_answer = answer->GetContentByName("audio");
ASSERT_TRUE(ac_answer != NULL);
EXPECT_FALSE(ac_answer->rejected);
const AudioContentDescription* answer_acd =
GetFirstAudioContentDescription(answer.get());
ASSERT_TRUE(answer_acd != NULL);
EXPECT_EQ(cricket::kMediaProtocolAvpf, answer_acd->protocol());
}
// Create a video offer and answer and ensure the RTP header extensions
// matches what we expect.
TEST_F(MediaSessionDescriptionFactoryTest, TestOfferAnswerWithRtpExtensions) {
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
f1_.set_audio_rtp_header_extensions(MAKE_VECTOR(kAudioRtpExtension1));
f1_.set_video_rtp_header_extensions(MAKE_VECTOR(kVideoRtpExtension1));
f2_.set_audio_rtp_header_extensions(MAKE_VECTOR(kAudioRtpExtension2));
f2_.set_video_rtp_header_extensions(MAKE_VECTOR(kVideoRtpExtension2));
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, NULL);
EXPECT_EQ(
MAKE_VECTOR(kAudioRtpExtension1),
GetFirstAudioContentDescription(offer.get())->rtp_header_extensions());
EXPECT_EQ(
MAKE_VECTOR(kVideoRtpExtension1),
GetFirstVideoContentDescription(offer.get())->rtp_header_extensions());
EXPECT_EQ(
MAKE_VECTOR(kAudioRtpExtensionAnswer),
GetFirstAudioContentDescription(answer.get())->rtp_header_extensions());
EXPECT_EQ(
MAKE_VECTOR(kVideoRtpExtensionAnswer),
GetFirstVideoContentDescription(answer.get())->rtp_header_extensions());
}
// Create a audio/video offer and answer and ensure that the
// TransportSequenceNumber RTP header extensions are handled correctly. 02 is
// supported and should take precedence even though not listed among locally
// supported extensions.
TEST_F(MediaSessionDescriptionFactoryTest,
TestOfferAnswerWithTransportSequenceNumberInOffer) {
TestTransportSequenceNumberNegotiation(
MAKE_VECTOR(kRtpExtensionTransportSequenceNumber01), // Local.
MAKE_VECTOR(kRtpExtensionTransportSequenceNumber01), // Offer.
MAKE_VECTOR(kRtpExtensionTransportSequenceNumber01)); // Expected answer.
}
TEST_F(MediaSessionDescriptionFactoryTest,
TestOfferAnswerWithTransportSequenceNumber01And02InOffer) {
TestTransportSequenceNumberNegotiation(
MAKE_VECTOR(kRtpExtensionTransportSequenceNumber01), // Local.
MAKE_VECTOR(kRtpExtensionTransportSequenceNumber01And02), // Offer.
MAKE_VECTOR(kRtpExtensionTransportSequenceNumber02)); // Expected answer.
}
TEST_F(MediaSessionDescriptionFactoryTest,
TestOfferAnswerWithTransportSequenceNumber02InOffer) {
TestTransportSequenceNumberNegotiation(
MAKE_VECTOR(kRtpExtensionTransportSequenceNumber01), // Local.
MAKE_VECTOR(kRtpExtensionTransportSequenceNumber02), // Offer.
MAKE_VECTOR(kRtpExtensionTransportSequenceNumber02)); // Expected answer.
}
TEST_F(MediaSessionDescriptionFactoryTest,
TestNegotiateFrameDescriptorWhenUnexposedLocally) {
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
const auto offered = MAKE_VECTOR(kRtpExtensionGenericFrameDescriptorUri00);
f1_.set_audio_rtp_header_extensions(offered);
f1_.set_video_rtp_header_extensions(offered);
const auto local = MAKE_VECTOR(kRtpExtensionTransportSequenceNumber01);
f2_.set_audio_rtp_header_extensions(local);
f2_.set_video_rtp_header_extensions(local);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, nullptr);
EXPECT_THAT(
GetFirstAudioContentDescription(answer.get())->rtp_header_extensions(),
ElementsAreArray(offered));
EXPECT_THAT(
GetFirstVideoContentDescription(answer.get())->rtp_header_extensions(),
ElementsAreArray(offered));
}
TEST_F(MediaSessionDescriptionFactoryTest,
TestNegotiateFrameDescriptorWhenExposedLocally) {
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
const auto offered = MAKE_VECTOR(kRtpExtensionGenericFrameDescriptorUri00);
f1_.set_audio_rtp_header_extensions(offered);
f1_.set_video_rtp_header_extensions(offered);
const auto local = MAKE_VECTOR(kRtpExtensionGenericFrameDescriptorUri00);
f2_.set_audio_rtp_header_extensions(local);
f2_.set_video_rtp_header_extensions(local);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, nullptr);
EXPECT_THAT(
GetFirstAudioContentDescription(answer.get())->rtp_header_extensions(),
ElementsAreArray(offered));
EXPECT_THAT(
GetFirstVideoContentDescription(answer.get())->rtp_header_extensions(),
ElementsAreArray(offered));
}
TEST_F(MediaSessionDescriptionFactoryTest,
NegotiateDependencyDescriptorWhenUnexposedLocally) {
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
RtpExtension offer_dd(RtpExtension::kDependencyDescriptorUri, 7);
RtpExtension local_tsn(RtpExtension::kTransportSequenceNumberUri, 5);
f1_.set_video_rtp_header_extensions({offer_dd});
f2_.set_video_rtp_header_extensions({local_tsn});
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, nullptr);
EXPECT_THAT(
GetFirstVideoContentDescription(answer.get())->rtp_header_extensions(),
ElementsAre(offer_dd));
}
TEST_F(MediaSessionDescriptionFactoryTest,
NegotiateDependencyDescriptorWhenExposedLocally) {
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
RtpExtension offer_dd(RtpExtension::kDependencyDescriptorUri, 7);
RtpExtension local_dd(RtpExtension::kDependencyDescriptorUri, 5);
f1_.set_video_rtp_header_extensions({offer_dd});
f2_.set_video_rtp_header_extensions({local_dd});
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, nullptr);
EXPECT_THAT(
GetFirstVideoContentDescription(answer.get())->rtp_header_extensions(),
ElementsAre(offer_dd));
}
TEST_F(MediaSessionDescriptionFactoryTest,
NegotiateAbsoluteCaptureTimeWhenUnexposedLocally) {
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
const cricket::RtpHeaderExtensions offered_extensions = {
RtpExtension(RtpExtension::kAbsoluteCaptureTimeUri, 7)};
const cricket::RtpHeaderExtensions local_extensions = {
RtpExtension(RtpExtension::kTransportSequenceNumberUri, 5)};
f1_.set_video_rtp_header_extensions(offered_extensions);
f1_.set_audio_rtp_header_extensions(offered_extensions);
f2_.set_video_rtp_header_extensions(local_extensions);
f2_.set_audio_rtp_header_extensions(local_extensions);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, nullptr);
EXPECT_THAT(
GetFirstVideoContentDescription(answer.get())->rtp_header_extensions(),
ElementsAreArray(offered_extensions));
EXPECT_THAT(
GetFirstAudioContentDescription(answer.get())->rtp_header_extensions(),
ElementsAreArray(offered_extensions));
}
TEST_F(MediaSessionDescriptionFactoryTest,
NegotiateAbsoluteCaptureTimeWhenExposedLocally) {
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
const cricket::RtpHeaderExtensions offered_extensions = {
RtpExtension(RtpExtension::kAbsoluteCaptureTimeUri, 7)};
const cricket::RtpHeaderExtensions local_extensions = {
RtpExtension(RtpExtension::kAbsoluteCaptureTimeUri, 5)};
f1_.set_video_rtp_header_extensions(offered_extensions);
f1_.set_audio_rtp_header_extensions(offered_extensions);
f2_.set_video_rtp_header_extensions(local_extensions);
f2_.set_audio_rtp_header_extensions(local_extensions);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, nullptr);
EXPECT_THAT(
GetFirstVideoContentDescription(answer.get())->rtp_header_extensions(),
ElementsAreArray(offered_extensions));
EXPECT_THAT(
GetFirstAudioContentDescription(answer.get())->rtp_header_extensions(),
ElementsAreArray(offered_extensions));
}
TEST_F(MediaSessionDescriptionFactoryTest,
DoNotNegotiateAbsoluteCaptureTimeWhenNotOffered) {
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
const cricket::RtpHeaderExtensions offered_extensions = {
RtpExtension(RtpExtension::kTransportSequenceNumberUri, 7)};
const cricket::RtpHeaderExtensions local_extensions = {
RtpExtension(RtpExtension::kAbsoluteCaptureTimeUri, 5)};
f1_.set_video_rtp_header_extensions(offered_extensions);
f1_.set_audio_rtp_header_extensions(offered_extensions);
f2_.set_video_rtp_header_extensions(local_extensions);
f2_.set_audio_rtp_header_extensions(local_extensions);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, nullptr);
EXPECT_THAT(
GetFirstVideoContentDescription(answer.get())->rtp_header_extensions(),
IsEmpty());
EXPECT_THAT(
GetFirstAudioContentDescription(answer.get())->rtp_header_extensions(),
IsEmpty());
}
TEST_F(MediaSessionDescriptionFactoryTest,
TestOfferAnswerWithEncryptedRtpExtensionsBoth) {
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
f1_.set_enable_encrypted_rtp_header_extensions(true);
f2_.set_enable_encrypted_rtp_header_extensions(true);
f1_.set_audio_rtp_header_extensions(MAKE_VECTOR(kAudioRtpExtension1));
f1_.set_video_rtp_header_extensions(MAKE_VECTOR(kVideoRtpExtension1));
f2_.set_audio_rtp_header_extensions(MAKE_VECTOR(kAudioRtpExtension2));
f2_.set_video_rtp_header_extensions(MAKE_VECTOR(kVideoRtpExtension2));
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, NULL);
EXPECT_EQ(
MAKE_VECTOR(kAudioRtpExtensionEncrypted1),
GetFirstAudioContentDescription(offer.get())->rtp_header_extensions());
EXPECT_EQ(
MAKE_VECTOR(kVideoRtpExtensionEncrypted1),
GetFirstVideoContentDescription(offer.get())->rtp_header_extensions());
EXPECT_EQ(
MAKE_VECTOR(kAudioRtpExtensionEncryptedAnswer),
GetFirstAudioContentDescription(answer.get())->rtp_header_extensions());
EXPECT_EQ(
MAKE_VECTOR(kVideoRtpExtensionEncryptedAnswer),
GetFirstVideoContentDescription(answer.get())->rtp_header_extensions());
}
TEST_F(MediaSessionDescriptionFactoryTest,
TestOfferAnswerWithEncryptedRtpExtensionsOffer) {
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
f1_.set_enable_encrypted_rtp_header_extensions(true);
f1_.set_audio_rtp_header_extensions(MAKE_VECTOR(kAudioRtpExtension1));
f1_.set_video_rtp_header_extensions(MAKE_VECTOR(kVideoRtpExtension1));
f2_.set_audio_rtp_header_extensions(MAKE_VECTOR(kAudioRtpExtension2));
f2_.set_video_rtp_header_extensions(MAKE_VECTOR(kVideoRtpExtension2));
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, NULL);
EXPECT_EQ(
MAKE_VECTOR(kAudioRtpExtensionEncrypted1),
GetFirstAudioContentDescription(offer.get())->rtp_header_extensions());
EXPECT_EQ(
MAKE_VECTOR(kVideoRtpExtensionEncrypted1),
GetFirstVideoContentDescription(offer.get())->rtp_header_extensions());
EXPECT_EQ(
MAKE_VECTOR(kAudioRtpExtensionAnswer),
GetFirstAudioContentDescription(answer.get())->rtp_header_extensions());
EXPECT_EQ(
MAKE_VECTOR(kVideoRtpExtensionAnswer),
GetFirstVideoContentDescription(answer.get())->rtp_header_extensions());
}
TEST_F(MediaSessionDescriptionFactoryTest,
TestOfferAnswerWithEncryptedRtpExtensionsAnswer) {
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
f2_.set_enable_encrypted_rtp_header_extensions(true);
f1_.set_audio_rtp_header_extensions(MAKE_VECTOR(kAudioRtpExtension1));
f1_.set_video_rtp_header_extensions(MAKE_VECTOR(kVideoRtpExtension1));
f2_.set_audio_rtp_header_extensions(MAKE_VECTOR(kAudioRtpExtension2));
f2_.set_video_rtp_header_extensions(MAKE_VECTOR(kVideoRtpExtension2));
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, NULL);
EXPECT_EQ(
MAKE_VECTOR(kAudioRtpExtension1),
GetFirstAudioContentDescription(offer.get())->rtp_header_extensions());
EXPECT_EQ(
MAKE_VECTOR(kVideoRtpExtension1),
GetFirstVideoContentDescription(offer.get())->rtp_header_extensions());
EXPECT_EQ(
MAKE_VECTOR(kAudioRtpExtensionAnswer),
GetFirstAudioContentDescription(answer.get())->rtp_header_extensions());
EXPECT_EQ(
MAKE_VECTOR(kVideoRtpExtensionAnswer),
GetFirstVideoContentDescription(answer.get())->rtp_header_extensions());
}
// Create an audio, video, data answer without legacy StreamParams.
TEST_F(MediaSessionDescriptionFactoryTest,
TestCreateAnswerWithoutLegacyStreams) {
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly, &opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, NULL);
const ContentInfo* ac = answer->GetContentByName("audio");
const ContentInfo* vc = answer->GetContentByName("video");
const ContentInfo* dc = answer->GetContentByName("data");
ASSERT_TRUE(ac != NULL);
ASSERT_TRUE(vc != NULL);
const AudioContentDescription* acd = ac->media_description()->as_audio();
const VideoContentDescription* vcd = vc->media_description()->as_video();
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
const RtpDataContentDescription* dcd = dc->media_description()->as_rtp_data();
EXPECT_FALSE(acd->has_ssrcs()); // No StreamParams.
EXPECT_FALSE(vcd->has_ssrcs()); // No StreamParams.
EXPECT_FALSE(dcd->has_ssrcs()); // No StreamParams.
}
// Create a typical video answer, and ensure it matches what we expect.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateVideoAnswerRtcpMux) {
MediaSessionOptions offer_opts;
AddAudioVideoSections(RtpTransceiverDirection::kSendRecv, &offer_opts);
AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kSendRecv,
&offer_opts);
MediaSessionOptions answer_opts;
AddAudioVideoSections(RtpTransceiverDirection::kSendRecv, &answer_opts);
AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kSendRecv,
&answer_opts);
std::unique_ptr<SessionDescription> offer;
std::unique_ptr<SessionDescription> answer;
offer_opts.rtcp_mux_enabled = true;
answer_opts.rtcp_mux_enabled = true;
offer = f1_.CreateOffer(offer_opts, NULL);
answer = f2_.CreateAnswer(offer.get(), answer_opts, NULL);
ASSERT_TRUE(NULL != GetFirstAudioContentDescription(offer.get()));
ASSERT_TRUE(NULL != GetFirstVideoContentDescription(offer.get()));
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
ASSERT_TRUE(NULL != GetFirstRtpDataContentDescription(offer.get()));
ASSERT_TRUE(NULL != GetFirstAudioContentDescription(answer.get()));
ASSERT_TRUE(NULL != GetFirstVideoContentDescription(answer.get()));
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
ASSERT_TRUE(NULL != GetFirstRtpDataContentDescription(answer.get()));
EXPECT_TRUE(GetFirstAudioContentDescription(offer.get())->rtcp_mux());
EXPECT_TRUE(GetFirstVideoContentDescription(offer.get())->rtcp_mux());
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
EXPECT_TRUE(GetFirstRtpDataContentDescription(offer.get())->rtcp_mux());
EXPECT_TRUE(GetFirstAudioContentDescription(answer.get())->rtcp_mux());
EXPECT_TRUE(GetFirstVideoContentDescription(answer.get())->rtcp_mux());
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
EXPECT_TRUE(GetFirstRtpDataContentDescription(answer.get())->rtcp_mux());
offer_opts.rtcp_mux_enabled = true;
answer_opts.rtcp_mux_enabled = false;
offer = f1_.CreateOffer(offer_opts, NULL);
answer = f2_.CreateAnswer(offer.get(), answer_opts, NULL);
ASSERT_TRUE(NULL != GetFirstAudioContentDescription(offer.get()));
ASSERT_TRUE(NULL != GetFirstVideoContentDescription(offer.get()));
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
ASSERT_TRUE(NULL != GetFirstRtpDataContentDescription(offer.get()));
ASSERT_TRUE(NULL != GetFirstAudioContentDescription(answer.get()));
ASSERT_TRUE(NULL != GetFirstVideoContentDescription(answer.get()));
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
ASSERT_TRUE(NULL != GetFirstRtpDataContentDescription(answer.get()));
EXPECT_TRUE(GetFirstAudioContentDescription(offer.get())->rtcp_mux());
EXPECT_TRUE(GetFirstVideoContentDescription(offer.get())->rtcp_mux());
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
EXPECT_TRUE(GetFirstRtpDataContentDescription(offer.get())->rtcp_mux());
EXPECT_FALSE(GetFirstAudioContentDescription(answer.get())->rtcp_mux());
EXPECT_FALSE(GetFirstVideoContentDescription(answer.get())->rtcp_mux());
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
EXPECT_FALSE(GetFirstRtpDataContentDescription(answer.get())->rtcp_mux());
offer_opts.rtcp_mux_enabled = false;
answer_opts.rtcp_mux_enabled = true;
offer = f1_.CreateOffer(offer_opts, NULL);
answer = f2_.CreateAnswer(offer.get(), answer_opts, NULL);
ASSERT_TRUE(NULL != GetFirstAudioContentDescription(offer.get()));
ASSERT_TRUE(NULL != GetFirstVideoContentDescription(offer.get()));
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
ASSERT_TRUE(NULL != GetFirstRtpDataContentDescription(offer.get()));
ASSERT_TRUE(NULL != GetFirstAudioContentDescription(answer.get()));
ASSERT_TRUE(NULL != GetFirstVideoContentDescription(answer.get()));
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
ASSERT_TRUE(NULL != GetFirstRtpDataContentDescription(answer.get()));
EXPECT_FALSE(GetFirstAudioContentDescription(offer.get())->rtcp_mux());
EXPECT_FALSE(GetFirstVideoContentDescription(offer.get())->rtcp_mux());
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
EXPECT_FALSE(GetFirstRtpDataContentDescription(offer.get())->rtcp_mux());
EXPECT_FALSE(GetFirstAudioContentDescription(answer.get())->rtcp_mux());
EXPECT_FALSE(GetFirstVideoContentDescription(answer.get())->rtcp_mux());
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
EXPECT_FALSE(GetFirstRtpDataContentDescription(answer.get())->rtcp_mux());
offer_opts.rtcp_mux_enabled = false;
answer_opts.rtcp_mux_enabled = false;
offer = f1_.CreateOffer(offer_opts, NULL);
answer = f2_.CreateAnswer(offer.get(), answer_opts, NULL);
ASSERT_TRUE(NULL != GetFirstAudioContentDescription(offer.get()));
ASSERT_TRUE(NULL != GetFirstVideoContentDescription(offer.get()));
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
ASSERT_TRUE(NULL != GetFirstRtpDataContentDescription(offer.get()));
ASSERT_TRUE(NULL != GetFirstAudioContentDescription(answer.get()));
ASSERT_TRUE(NULL != GetFirstVideoContentDescription(answer.get()));
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
ASSERT_TRUE(NULL != GetFirstRtpDataContentDescription(answer.get()));
EXPECT_FALSE(GetFirstAudioContentDescription(offer.get())->rtcp_mux());
EXPECT_FALSE(GetFirstVideoContentDescription(offer.get())->rtcp_mux());
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
EXPECT_FALSE(GetFirstRtpDataContentDescription(offer.get())->rtcp_mux());
EXPECT_FALSE(GetFirstAudioContentDescription(answer.get())->rtcp_mux());
EXPECT_FALSE(GetFirstVideoContentDescription(answer.get())->rtcp_mux());
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
EXPECT_FALSE(GetFirstRtpDataContentDescription(answer.get())->rtcp_mux());
}
// Create an audio-only answer to a video offer.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateAudioAnswerToVideo) {
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
opts.media_description_options[1].stopped = true;
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, NULL);
const ContentInfo* ac = answer->GetContentByName("audio");
const ContentInfo* vc = answer->GetContentByName("video");
ASSERT_TRUE(ac != NULL);
ASSERT_TRUE(vc != NULL);
ASSERT_TRUE(vc->media_description() != NULL);
EXPECT_TRUE(vc->rejected);
}
// Create an audio-only answer to an offer with data.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateNoDataAnswerToDataOffer) {
MediaSessionOptions opts = CreatePlanBMediaSessionOptions();
opts.data_channel_type = cricket::DCT_RTP;
AddMediaDescriptionOptions(MEDIA_TYPE_DATA, "data",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
opts.media_description_options[1].stopped = true;
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, NULL);
const ContentInfo* ac = answer->GetContentByName("audio");
const ContentInfo* dc = answer->GetContentByName("data");
ASSERT_TRUE(ac != NULL);
ASSERT_TRUE(dc != NULL);
ASSERT_TRUE(dc->media_description() != NULL);
EXPECT_TRUE(dc->rejected);
}
// Create an answer that rejects the contents which are rejected in the offer.
TEST_F(MediaSessionDescriptionFactoryTest,
CreateAnswerToOfferWithRejectedMedia) {
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly, &opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
ContentInfo* ac = offer->GetContentByName("audio");
ContentInfo* vc = offer->GetContentByName("video");
ContentInfo* dc = offer->GetContentByName("data");
ASSERT_TRUE(ac != NULL);
ASSERT_TRUE(vc != NULL);
ASSERT_TRUE(dc != NULL);
ac->rejected = true;
vc->rejected = true;
dc->rejected = true;
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, NULL);
ac = answer->GetContentByName("audio");
vc = answer->GetContentByName("video");
dc = answer->GetContentByName("data");
ASSERT_TRUE(ac != NULL);
ASSERT_TRUE(vc != NULL);
ASSERT_TRUE(dc != NULL);
EXPECT_TRUE(ac->rejected);
EXPECT_TRUE(vc->rejected);
EXPECT_TRUE(dc->rejected);
}
TEST_F(MediaSessionDescriptionFactoryTest,
CreateAnswerSupportsMixedOneAndTwoByteHeaderExtensions) {
MediaSessionOptions opts;
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
// Offer without request of mixed one- and two-byte header extensions.
offer->set_extmap_allow_mixed(false);
ASSERT_TRUE(offer.get() != NULL);
std::unique_ptr<SessionDescription> answer_no_support(
f2_.CreateAnswer(offer.get(), opts, NULL));
EXPECT_FALSE(answer_no_support->extmap_allow_mixed());
// Offer with request of mixed one- and two-byte header extensions.
offer->set_extmap_allow_mixed(true);
ASSERT_TRUE(offer.get() != NULL);
std::unique_ptr<SessionDescription> answer_support(
f2_.CreateAnswer(offer.get(), opts, NULL));
EXPECT_TRUE(answer_support->extmap_allow_mixed());
}
TEST_F(MediaSessionDescriptionFactoryTest,
CreateAnswerSupportsMixedOneAndTwoByteHeaderExtensionsOnMediaLevel) {
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kSendRecv, &opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
MediaContentDescription* video_offer =
offer->GetContentDescriptionByName("video");
ASSERT_TRUE(video_offer);
MediaContentDescription* audio_offer =
offer->GetContentDescriptionByName("audio");
ASSERT_TRUE(audio_offer);
// Explicit disable of mixed one-two byte header support in offer.
video_offer->set_extmap_allow_mixed_enum(MediaContentDescription::kNo);
audio_offer->set_extmap_allow_mixed_enum(MediaContentDescription::kNo);
ASSERT_TRUE(offer.get() != NULL);
std::unique_ptr<SessionDescription> answer_no_support(
f2_.CreateAnswer(offer.get(), opts, NULL));
MediaContentDescription* video_answer =
answer_no_support->GetContentDescriptionByName("video");
MediaContentDescription* audio_answer =
answer_no_support->GetContentDescriptionByName("audio");
EXPECT_EQ(MediaContentDescription::kNo,
video_answer->extmap_allow_mixed_enum());
EXPECT_EQ(MediaContentDescription::kNo,
audio_answer->extmap_allow_mixed_enum());
// Enable mixed one-two byte header support in offer.
video_offer->set_extmap_allow_mixed_enum(MediaContentDescription::kMedia);
audio_offer->set_extmap_allow_mixed_enum(MediaContentDescription::kMedia);
ASSERT_TRUE(offer.get() != NULL);
std::unique_ptr<SessionDescription> answer_support(
f2_.CreateAnswer(offer.get(), opts, NULL));
video_answer = answer_support->GetContentDescriptionByName("video");
audio_answer = answer_support->GetContentDescriptionByName("audio");
EXPECT_EQ(MediaContentDescription::kMedia,
video_answer->extmap_allow_mixed_enum());
EXPECT_EQ(MediaContentDescription::kMedia,
audio_answer->extmap_allow_mixed_enum());
}
// Create an audio and video offer with:
// - one video track
// - two audio tracks
// - two data tracks
// and ensure it matches what we expect. Also updates the initial offer by
// adding a new video track and replaces one of the audio tracks.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoOffer) {
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kSendRecv, &opts);
AttachSenderToMediaDescriptionOptions("video", MEDIA_TYPE_VIDEO, kVideoTrack1,
{kMediaStream1}, 1, &opts);
AttachSenderToMediaDescriptionOptions("audio", MEDIA_TYPE_AUDIO, kAudioTrack1,
{kMediaStream1}, 1, &opts);
AttachSenderToMediaDescriptionOptions("audio", MEDIA_TYPE_AUDIO, kAudioTrack2,
{kMediaStream1}, 1, &opts);
AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kSendRecv, &opts);
AttachSenderToMediaDescriptionOptions("data", MEDIA_TYPE_DATA, kDataTrack1,
{kMediaStream1}, 1, &opts);
AttachSenderToMediaDescriptionOptions("data", MEDIA_TYPE_DATA, kDataTrack2,
{kMediaStream1}, 1, &opts);
f1_.set_secure(SEC_ENABLED);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
const ContentInfo* ac = offer->GetContentByName("audio");
const ContentInfo* vc = offer->GetContentByName("video");
const ContentInfo* dc = offer->GetContentByName("data");
ASSERT_TRUE(ac != NULL);
ASSERT_TRUE(vc != NULL);
ASSERT_TRUE(dc != NULL);
const AudioContentDescription* acd = ac->media_description()->as_audio();
const VideoContentDescription* vcd = vc->media_description()->as_video();
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
const RtpDataContentDescription* dcd = dc->media_description()->as_rtp_data();
EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
EXPECT_EQ(f1_.audio_sendrecv_codecs(), acd->codecs());
const StreamParamsVec& audio_streams = acd->streams();
ASSERT_EQ(2U, audio_streams.size());
EXPECT_EQ(audio_streams[0].cname, audio_streams[1].cname);
EXPECT_EQ(kAudioTrack1, audio_streams[0].id);
ASSERT_EQ(1U, audio_streams[0].ssrcs.size());
EXPECT_NE(0U, audio_streams[0].ssrcs[0]);
EXPECT_EQ(kAudioTrack2, audio_streams[1].id);
ASSERT_EQ(1U, audio_streams[1].ssrcs.size());
EXPECT_NE(0U, audio_streams[1].ssrcs[0]);
EXPECT_EQ(kAutoBandwidth, acd->bandwidth()); // default bandwidth (auto)
EXPECT_TRUE(acd->rtcp_mux()); // rtcp-mux defaults on
ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
EXPECT_EQ(MEDIA_TYPE_VIDEO, vcd->type());
EXPECT_EQ(f1_.video_sendrecv_codecs(), vcd->codecs());
ASSERT_CRYPTO(vcd, 1U, kDefaultSrtpCryptoSuite);
const StreamParamsVec& video_streams = vcd->streams();
ASSERT_EQ(1U, video_streams.size());
EXPECT_EQ(video_streams[0].cname, audio_streams[0].cname);
EXPECT_EQ(kVideoTrack1, video_streams[0].id);
EXPECT_EQ(kAutoBandwidth, vcd->bandwidth()); // default bandwidth (auto)
EXPECT_TRUE(vcd->rtcp_mux()); // rtcp-mux defaults on
EXPECT_EQ(MEDIA_TYPE_DATA, dcd->type());
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
EXPECT_EQ(f1_.rtp_data_codecs(), dcd->codecs());
ASSERT_CRYPTO(dcd, 1U, kDefaultSrtpCryptoSuite);
const StreamParamsVec& data_streams = dcd->streams();
ASSERT_EQ(2U, data_streams.size());
EXPECT_EQ(data_streams[0].cname, data_streams[1].cname);
EXPECT_EQ(kDataTrack1, data_streams[0].id);
ASSERT_EQ(1U, data_streams[0].ssrcs.size());
EXPECT_NE(0U, data_streams[0].ssrcs[0]);
EXPECT_EQ(kDataTrack2, data_streams[1].id);
ASSERT_EQ(1U, data_streams[1].ssrcs.size());
EXPECT_NE(0U, data_streams[1].ssrcs[0]);
EXPECT_EQ(cricket::kDataMaxBandwidth,
dcd->bandwidth()); // default bandwidth (auto)
EXPECT_TRUE(dcd->rtcp_mux()); // rtcp-mux defaults on
ASSERT_CRYPTO(dcd, 1U, kDefaultSrtpCryptoSuite);
// Update the offer. Add a new video track that is not synched to the
// other tracks and replace audio track 2 with audio track 3.
AttachSenderToMediaDescriptionOptions("video", MEDIA_TYPE_VIDEO, kVideoTrack2,
{kMediaStream2}, 1, &opts);
DetachSenderFromMediaSection("audio", kAudioTrack2, &opts);
AttachSenderToMediaDescriptionOptions("audio", MEDIA_TYPE_AUDIO, kAudioTrack3,
{kMediaStream1}, 1, &opts);
DetachSenderFromMediaSection("data", kDataTrack2, &opts);
AttachSenderToMediaDescriptionOptions("data", MEDIA_TYPE_DATA, kDataTrack3,
{kMediaStream1}, 1, &opts);
std::unique_ptr<SessionDescription> updated_offer(
f1_.CreateOffer(opts, offer.get()));
ASSERT_TRUE(updated_offer.get() != NULL);
ac = updated_offer->GetContentByName("audio");
vc = updated_offer->GetContentByName("video");
dc = updated_offer->GetContentByName("data");
ASSERT_TRUE(ac != NULL);
ASSERT_TRUE(vc != NULL);
ASSERT_TRUE(dc != NULL);
const AudioContentDescription* updated_acd =
ac->media_description()->as_audio();
const VideoContentDescription* updated_vcd =
vc->media_description()->as_video();
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
const RtpDataContentDescription* updated_dcd =
dc->media_description()->as_rtp_data();
EXPECT_EQ(acd->type(), updated_acd->type());
EXPECT_EQ(acd->codecs(), updated_acd->codecs());
EXPECT_EQ(vcd->type(), updated_vcd->type());
EXPECT_EQ(vcd->codecs(), updated_vcd->codecs());
EXPECT_EQ(dcd->type(), updated_dcd->type());
EXPECT_EQ(dcd->codecs(), updated_dcd->codecs());
ASSERT_CRYPTO(updated_acd, 1U, kDefaultSrtpCryptoSuite);
EXPECT_TRUE(CompareCryptoParams(acd->cryptos(), updated_acd->cryptos()));
ASSERT_CRYPTO(updated_vcd, 1U, kDefaultSrtpCryptoSuite);
EXPECT_TRUE(CompareCryptoParams(vcd->cryptos(), updated_vcd->cryptos()));
ASSERT_CRYPTO(updated_dcd, 1U, kDefaultSrtpCryptoSuite);
EXPECT_TRUE(CompareCryptoParams(dcd->cryptos(), updated_dcd->cryptos()));
const StreamParamsVec& updated_audio_streams = updated_acd->streams();
ASSERT_EQ(2U, updated_audio_streams.size());
EXPECT_EQ(audio_streams[0], updated_audio_streams[0]);
EXPECT_EQ(kAudioTrack3, updated_audio_streams[1].id); // New audio track.
ASSERT_EQ(1U, updated_audio_streams[1].ssrcs.size());
EXPECT_NE(0U, updated_audio_streams[1].ssrcs[0]);
EXPECT_EQ(updated_audio_streams[0].cname, updated_audio_streams[1].cname);
const StreamParamsVec& updated_video_streams = updated_vcd->streams();
ASSERT_EQ(2U, updated_video_streams.size());
EXPECT_EQ(video_streams[0], updated_video_streams[0]);
EXPECT_EQ(kVideoTrack2, updated_video_streams[1].id);
// All the media streams in one PeerConnection share one RTCP CNAME.
EXPECT_EQ(updated_video_streams[1].cname, updated_video_streams[0].cname);
const StreamParamsVec& updated_data_streams = updated_dcd->streams();
ASSERT_EQ(2U, updated_data_streams.size());
EXPECT_EQ(data_streams[0], updated_data_streams[0]);
EXPECT_EQ(kDataTrack3, updated_data_streams[1].id); // New data track.
ASSERT_EQ(1U, updated_data_streams[1].ssrcs.size());
EXPECT_NE(0U, updated_data_streams[1].ssrcs[0]);
EXPECT_EQ(updated_data_streams[0].cname, updated_data_streams[1].cname);
// The stream correctly got the CNAME from the MediaSessionOptions.
// The Expected RTCP CNAME is the default one as we are using the default
// MediaSessionOptions.
EXPECT_EQ(updated_data_streams[0].cname, cricket::kDefaultRtcpCname);
}
// Create an offer with simulcast video stream.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateSimulcastVideoOffer) {
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
const int num_sim_layers = 3;
AttachSenderToMediaDescriptionOptions("video", MEDIA_TYPE_VIDEO, kVideoTrack1,
{kMediaStream1}, num_sim_layers, &opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
const ContentInfo* vc = offer->GetContentByName("video");
ASSERT_TRUE(vc != NULL);
const VideoContentDescription* vcd = vc->media_description()->as_video();
const StreamParamsVec& video_streams = vcd->streams();
ASSERT_EQ(1U, video_streams.size());
EXPECT_EQ(kVideoTrack1, video_streams[0].id);
const SsrcGroup* sim_ssrc_group =
video_streams[0].get_ssrc_group(cricket::kSimSsrcGroupSemantics);
ASSERT_TRUE(sim_ssrc_group != NULL);
EXPECT_EQ(static_cast<size_t>(num_sim_layers), sim_ssrc_group->ssrcs.size());
}
MATCHER(RidDescriptionEquals, "Verifies that two RidDescriptions are equal.") {
const RidDescription& rid1 = ::testing::get<0>(arg);
const RidDescription& rid2 = ::testing::get<1>(arg);
return rid1.rid == rid2.rid && rid1.direction == rid2.direction;
}
static void CheckSimulcastInSessionDescription(
const SessionDescription* description,
const std::string& content_name,
const std::vector<RidDescription>& send_rids,
const SimulcastLayerList& send_layers) {
ASSERT_NE(description, nullptr);
const ContentInfo* content = description->GetContentByName(content_name);
ASSERT_NE(content, nullptr);
const MediaContentDescription* cd = content->media_description();
ASSERT_NE(cd, nullptr);
const StreamParamsVec& streams = cd->streams();
ASSERT_THAT(streams, SizeIs(1));
const StreamParams& stream = streams[0];
ASSERT_THAT(stream.ssrcs, IsEmpty());
EXPECT_TRUE(stream.has_rids());
const std::vector<RidDescription> rids = stream.rids();
EXPECT_THAT(rids, Pointwise(RidDescriptionEquals(), send_rids));
EXPECT_TRUE(cd->HasSimulcast());
const SimulcastDescription& simulcast = cd->simulcast_description();
EXPECT_THAT(simulcast.send_layers(), SizeIs(send_layers.size()));
EXPECT_THAT(simulcast.send_layers(), Pointwise(Eq(), send_layers));
ASSERT_THAT(simulcast.receive_layers().GetAllLayers(), SizeIs(0));
}
// Create an offer with spec-compliant simulcast video stream.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateCompliantSimulcastOffer) {
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::vector<RidDescription> send_rids;
send_rids.push_back(RidDescription("f", RidDirection::kSend));
send_rids.push_back(RidDescription("h", RidDirection::kSend));
send_rids.push_back(RidDescription("q", RidDirection::kSend));
SimulcastLayerList simulcast_layers;
simulcast_layers.AddLayer(SimulcastLayer(send_rids[0].rid, false));
simulcast_layers.AddLayer(SimulcastLayer(send_rids[1].rid, true));
simulcast_layers.AddLayer(SimulcastLayer(send_rids[2].rid, false));
AttachSenderToMediaDescriptionOptions("video", MEDIA_TYPE_VIDEO, kVideoTrack1,
{kMediaStream1}, send_rids,
simulcast_layers, 0, &opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
CheckSimulcastInSessionDescription(offer.get(), "video", send_rids,
simulcast_layers);
}
// Create an offer that signals RIDs (not SSRCs) without Simulcast.
// In this scenario, RIDs do not need to be negotiated (there is only one).
TEST_F(MediaSessionDescriptionFactoryTest, TestOfferWithRidsNoSimulcast) {
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
RidDescription rid("f", RidDirection::kSend);
AttachSenderToMediaDescriptionOptions("video", MEDIA_TYPE_VIDEO, kVideoTrack1,
{kMediaStream1}, {rid},
SimulcastLayerList(), 0, &opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
ASSERT_NE(offer.get(), nullptr);
const ContentInfo* content = offer->GetContentByName("video");
ASSERT_NE(content, nullptr);
const MediaContentDescription* cd = content->media_description();
ASSERT_NE(cd, nullptr);
const StreamParamsVec& streams = cd->streams();
ASSERT_THAT(streams, SizeIs(1));
const StreamParams& stream = streams[0];
ASSERT_THAT(stream.ssrcs, IsEmpty());
EXPECT_FALSE(stream.has_rids());
EXPECT_FALSE(cd->HasSimulcast());
}
// Create an answer with spec-compliant simulcast video stream.
// In this scenario, the SFU is the caller requesting that we send Simulcast.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateCompliantSimulcastAnswer) {
MediaSessionOptions offer_opts;
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&offer_opts);
AttachSenderToMediaDescriptionOptions("video", MEDIA_TYPE_VIDEO, kVideoTrack1,
{kMediaStream1}, 1, &offer_opts);
std::unique_ptr<SessionDescription> offer =
f1_.CreateOffer(offer_opts, nullptr);
MediaSessionOptions answer_opts;
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&answer_opts);
std::vector<RidDescription> rid_descriptions{
RidDescription("f", RidDirection::kSend),
RidDescription("h", RidDirection::kSend),
RidDescription("q", RidDirection::kSend),
};
SimulcastLayerList simulcast_layers;
simulcast_layers.AddLayer(SimulcastLayer(rid_descriptions[0].rid, false));
simulcast_layers.AddLayer(SimulcastLayer(rid_descriptions[1].rid, true));
simulcast_layers.AddLayer(SimulcastLayer(rid_descriptions[2].rid, false));
AttachSenderToMediaDescriptionOptions("video", MEDIA_TYPE_VIDEO, kVideoTrack1,
{kMediaStream1}, rid_descriptions,
simulcast_layers, 0, &answer_opts);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), answer_opts, nullptr);
CheckSimulcastInSessionDescription(answer.get(), "video", rid_descriptions,
simulcast_layers);
}
// Create an answer that signals RIDs (not SSRCs) without Simulcast.
// In this scenario, RIDs do not need to be negotiated (there is only one).
// Note that RID Direction is not the same as the transceiver direction.
TEST_F(MediaSessionDescriptionFactoryTest, TestAnswerWithRidsNoSimulcast) {
MediaSessionOptions offer_opts;
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&offer_opts);
RidDescription rid_offer("f", RidDirection::kSend);
AttachSenderToMediaDescriptionOptions("video", MEDIA_TYPE_VIDEO, kVideoTrack1,
{kMediaStream1}, {rid_offer},
SimulcastLayerList(), 0, &offer_opts);
std::unique_ptr<SessionDescription> offer =
f1_.CreateOffer(offer_opts, nullptr);
MediaSessionOptions answer_opts;
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&answer_opts);
RidDescription rid_answer("f", RidDirection::kReceive);
AttachSenderToMediaDescriptionOptions("video", MEDIA_TYPE_VIDEO, kVideoTrack1,
{kMediaStream1}, {rid_answer},
SimulcastLayerList(), 0, &answer_opts);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), answer_opts, nullptr);
ASSERT_NE(answer.get(), nullptr);
const ContentInfo* content = offer->GetContentByName("video");
ASSERT_NE(content, nullptr);
const MediaContentDescription* cd = content->media_description();
ASSERT_NE(cd, nullptr);
const StreamParamsVec& streams = cd->streams();
ASSERT_THAT(streams, SizeIs(1));
const StreamParams& stream = streams[0];
ASSERT_THAT(stream.ssrcs, IsEmpty());
EXPECT_FALSE(stream.has_rids());
EXPECT_FALSE(cd->HasSimulcast());
}
// Create an audio and video answer to a standard video offer with:
// - one video track
// - two audio tracks
// - two data tracks
// and ensure it matches what we expect. Also updates the initial answer by
// adding a new video track and removes one of the audio tracks.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoAnswer) {
MediaSessionOptions offer_opts;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&offer_opts);
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&offer_opts);
offer_opts.data_channel_type = cricket::DCT_RTP;
AddMediaDescriptionOptions(MEDIA_TYPE_DATA, "data",
RtpTransceiverDirection::kRecvOnly, kActive,
&offer_opts);
f1_.set_secure(SEC_ENABLED);
f2_.set_secure(SEC_ENABLED);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(offer_opts, NULL);
MediaSessionOptions answer_opts;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio",
RtpTransceiverDirection::kSendRecv, kActive,
&answer_opts);
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&answer_opts);
AttachSenderToMediaDescriptionOptions("video", MEDIA_TYPE_VIDEO, kVideoTrack1,
{kMediaStream1}, 1, &answer_opts);
AttachSenderToMediaDescriptionOptions("audio", MEDIA_TYPE_AUDIO, kAudioTrack1,
{kMediaStream1}, 1, &answer_opts);
AttachSenderToMediaDescriptionOptions("audio", MEDIA_TYPE_AUDIO, kAudioTrack2,
{kMediaStream1}, 1, &answer_opts);
AddMediaDescriptionOptions(MEDIA_TYPE_DATA, "data",
RtpTransceiverDirection::kSendRecv, kActive,
&answer_opts);
AttachSenderToMediaDescriptionOptions("data", MEDIA_TYPE_DATA, kDataTrack1,
{kMediaStream1}, 1, &answer_opts);
AttachSenderToMediaDescriptionOptions("data", MEDIA_TYPE_DATA, kDataTrack2,
{kMediaStream1}, 1, &answer_opts);
answer_opts.data_channel_type = cricket::DCT_RTP;
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), answer_opts, NULL);
ASSERT_TRUE(answer.get() != NULL);
const ContentInfo* ac = answer->GetContentByName("audio");
const ContentInfo* vc = answer->GetContentByName("video");
const ContentInfo* dc = answer->GetContentByName("data");
ASSERT_TRUE(ac != NULL);
ASSERT_TRUE(vc != NULL);
ASSERT_TRUE(dc != NULL);
const AudioContentDescription* acd = ac->media_description()->as_audio();
const VideoContentDescription* vcd = vc->media_description()->as_video();
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
const RtpDataContentDescription* dcd = dc->media_description()->as_rtp_data();
ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
ASSERT_CRYPTO(vcd, 1U, kDefaultSrtpCryptoSuite);
ASSERT_CRYPTO(dcd, 1U, kDefaultSrtpCryptoSuite);
EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
const StreamParamsVec& audio_streams = acd->streams();
ASSERT_EQ(2U, audio_streams.size());
EXPECT_TRUE(audio_streams[0].cname == audio_streams[1].cname);
EXPECT_EQ(kAudioTrack1, audio_streams[0].id);
ASSERT_EQ(1U, audio_streams[0].ssrcs.size());
EXPECT_NE(0U, audio_streams[0].ssrcs[0]);
EXPECT_EQ(kAudioTrack2, audio_streams[1].id);
ASSERT_EQ(1U, audio_streams[1].ssrcs.size());
EXPECT_NE(0U, audio_streams[1].ssrcs[0]);
EXPECT_EQ(kAutoBandwidth, acd->bandwidth()); // default bandwidth (auto)
EXPECT_TRUE(acd->rtcp_mux()); // rtcp-mux defaults on
EXPECT_EQ(MEDIA_TYPE_VIDEO, vcd->type());
EXPECT_THAT(vcd->codecs(), ElementsAreArray(kVideoCodecsAnswer));
const StreamParamsVec& video_streams = vcd->streams();
ASSERT_EQ(1U, video_streams.size());
EXPECT_EQ(video_streams[0].cname, audio_streams[0].cname);
EXPECT_EQ(kVideoTrack1, video_streams[0].id);
EXPECT_EQ(kAutoBandwidth, vcd->bandwidth()); // default bandwidth (auto)
EXPECT_TRUE(vcd->rtcp_mux()); // rtcp-mux defaults on
EXPECT_EQ(MEDIA_TYPE_DATA, dcd->type());
EXPECT_THAT(dcd->codecs(), ElementsAreArray(kDataCodecsAnswer));
const StreamParamsVec& data_streams = dcd->streams();
ASSERT_EQ(2U, data_streams.size());
EXPECT_TRUE(data_streams[0].cname == data_streams[1].cname);
EXPECT_EQ(kDataTrack1, data_streams[0].id);
ASSERT_EQ(1U, data_streams[0].ssrcs.size());
EXPECT_NE(0U, data_streams[0].ssrcs[0]);
EXPECT_EQ(kDataTrack2, data_streams[1].id);
ASSERT_EQ(1U, data_streams[1].ssrcs.size());
EXPECT_NE(0U, data_streams[1].ssrcs[0]);
EXPECT_EQ(cricket::kDataMaxBandwidth,
dcd->bandwidth()); // default bandwidth (auto)
EXPECT_TRUE(dcd->rtcp_mux()); // rtcp-mux defaults on
// Update the answer. Add a new video track that is not synched to the
// other tracks and remove 1 audio track.
AttachSenderToMediaDescriptionOptions("video", MEDIA_TYPE_VIDEO, kVideoTrack2,
{kMediaStream2}, 1, &answer_opts);
DetachSenderFromMediaSection("audio", kAudioTrack2, &answer_opts);
DetachSenderFromMediaSection("data", kDataTrack2, &answer_opts);
std::unique_ptr<SessionDescription> updated_answer(
f2_.CreateAnswer(offer.get(), answer_opts, answer.get()));
ASSERT_TRUE(updated_answer.get() != NULL);
ac = updated_answer->GetContentByName("audio");
vc = updated_answer->GetContentByName("video");
dc = updated_answer->GetContentByName("data");
ASSERT_TRUE(ac != NULL);
ASSERT_TRUE(vc != NULL);
ASSERT_TRUE(dc != NULL);
const AudioContentDescription* updated_acd =
ac->media_description()->as_audio();
const VideoContentDescription* updated_vcd =
vc->media_description()->as_video();
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
const RtpDataContentDescription* updated_dcd =
dc->media_description()->as_rtp_data();
ASSERT_CRYPTO(updated_acd, 1U, kDefaultSrtpCryptoSuite);
EXPECT_TRUE(CompareCryptoParams(acd->cryptos(), updated_acd->cryptos()));
ASSERT_CRYPTO(updated_vcd, 1U, kDefaultSrtpCryptoSuite);
EXPECT_TRUE(CompareCryptoParams(vcd->cryptos(), updated_vcd->cryptos()));
ASSERT_CRYPTO(updated_dcd, 1U, kDefaultSrtpCryptoSuite);
EXPECT_TRUE(CompareCryptoParams(dcd->cryptos(), updated_dcd->cryptos()));
EXPECT_EQ(acd->type(), updated_acd->type());
EXPECT_EQ(acd->codecs(), updated_acd->codecs());
EXPECT_EQ(vcd->type(), updated_vcd->type());
EXPECT_EQ(vcd->codecs(), updated_vcd->codecs());
EXPECT_EQ(dcd->type(), updated_dcd->type());
EXPECT_EQ(dcd->codecs(), updated_dcd->codecs());
const StreamParamsVec& updated_audio_streams = updated_acd->streams();
ASSERT_EQ(1U, updated_audio_streams.size());
EXPECT_TRUE(audio_streams[0] == updated_audio_streams[0]);
const StreamParamsVec& updated_video_streams = updated_vcd->streams();
ASSERT_EQ(2U, updated_video_streams.size());
EXPECT_EQ(video_streams[0], updated_video_streams[0]);
EXPECT_EQ(kVideoTrack2, updated_video_streams[1].id);
// All media streams in one PeerConnection share one CNAME.
EXPECT_EQ(updated_video_streams[1].cname, updated_video_streams[0].cname);
const StreamParamsVec& updated_data_streams = updated_dcd->streams();
ASSERT_EQ(1U, updated_data_streams.size());
EXPECT_TRUE(data_streams[0] == updated_data_streams[0]);
}
// Create an updated offer after creating an answer to the original offer and
// verify that the codecs that were part of the original answer are not changed
// in the updated offer.
TEST_F(MediaSessionDescriptionFactoryTest,
RespondentCreatesOfferAfterCreatingAnswer) {
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, NULL);
const AudioContentDescription* acd =
GetFirstAudioContentDescription(answer.get());
EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
const VideoContentDescription* vcd =
GetFirstVideoContentDescription(answer.get());
EXPECT_THAT(vcd->codecs(), ElementsAreArray(kVideoCodecsAnswer));
std::unique_ptr<SessionDescription> updated_offer(
f2_.CreateOffer(opts, answer.get()));
// The expected audio codecs are the common audio codecs from the first
// offer/answer exchange plus the audio codecs only |f2_| offer, sorted in
// preference order.
// TODO(wu): |updated_offer| should not include the codec
// (i.e. |kAudioCodecs2[0]|) the other side doesn't support.
const AudioCodec kUpdatedAudioCodecOffer[] = {
kAudioCodecsAnswer[0],
kAudioCodecsAnswer[1],
kAudioCodecs2[0],
};
// The expected video codecs are the common video codecs from the first
// offer/answer exchange plus the video codecs only |f2_| offer, sorted in
// preference order.
const VideoCodec kUpdatedVideoCodecOffer[] = {
kVideoCodecsAnswer[0],
kVideoCodecs2[1],
};
const AudioContentDescription* updated_acd =
GetFirstAudioContentDescription(updated_offer.get());
EXPECT_THAT(updated_acd->codecs(), ElementsAreArray(kUpdatedAudioCodecOffer));
const VideoContentDescription* updated_vcd =
GetFirstVideoContentDescription(updated_offer.get());
EXPECT_THAT(updated_vcd->codecs(), ElementsAreArray(kUpdatedVideoCodecOffer));
}
// Test that a reoffer does not reuse audio codecs from a previous media section
// that is being recycled.
TEST_F(MediaSessionDescriptionFactoryTest,
ReOfferDoesNotReUseRecycledAudioCodecs) {
f1_.set_video_codecs({}, {});
f2_.set_video_codecs({}, {});
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "a0",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, nullptr);
// Recycle the media section by changing its mid.
opts.media_description_options[0].mid = "a1";
std::unique_ptr<SessionDescription> reoffer =
f2_.CreateOffer(opts, answer.get());
// Expect that the results of the first negotiation are ignored. If the m=
// section was not recycled the payload types would match the initial offerer.
const AudioContentDescription* acd =
GetFirstAudioContentDescription(reoffer.get());
EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecs2));
}
// Test that a reoffer does not reuse video codecs from a previous media section
// that is being recycled.
TEST_F(MediaSessionDescriptionFactoryTest,
ReOfferDoesNotReUseRecycledVideoCodecs) {
f1_.set_audio_codecs({}, {});
f2_.set_audio_codecs({}, {});
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "v0",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
auto answer = f2_.CreateAnswer(offer.get(), opts, nullptr);
// Recycle the media section by changing its mid.
opts.media_description_options[0].mid = "v1";
std::unique_ptr<SessionDescription> reoffer =
f2_.CreateOffer(opts, answer.get());
// Expect that the results of the first negotiation are ignored. If the m=
// section was not recycled the payload types would match the initial offerer.
const VideoContentDescription* vcd =
GetFirstVideoContentDescription(reoffer.get());
EXPECT_THAT(vcd->codecs(), ElementsAreArray(kVideoCodecs2));
}
// Test that a reanswer does not reuse audio codecs from a previous media
// section that is being recycled.
TEST_F(MediaSessionDescriptionFactoryTest,
ReAnswerDoesNotReUseRecycledAudioCodecs) {
f1_.set_video_codecs({}, {});
f2_.set_video_codecs({}, {});
// Perform initial offer/answer in reverse (|f2_| as offerer) so that the
// second offer/answer is forward (|f1_| as offerer).
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "a0",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::unique_ptr<SessionDescription> offer = f2_.CreateOffer(opts, nullptr);
std::unique_ptr<SessionDescription> answer =
f1_.CreateAnswer(offer.get(), opts, nullptr);
// Recycle the media section by changing its mid.
opts.media_description_options[0].mid = "a1";
std::unique_ptr<SessionDescription> reoffer =
f1_.CreateOffer(opts, answer.get());
std::unique_ptr<SessionDescription> reanswer =
f2_.CreateAnswer(reoffer.get(), opts, offer.get());
// Expect that the results of the first negotiation are ignored. If the m=
// section was not recycled the payload types would match the initial offerer.
const AudioContentDescription* acd =
GetFirstAudioContentDescription(reanswer.get());
EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
}
// Test that a reanswer does not reuse video codecs from a previous media
// section that is being recycled.
TEST_F(MediaSessionDescriptionFactoryTest,
ReAnswerDoesNotReUseRecycledVideoCodecs) {
f1_.set_audio_codecs({}, {});
f2_.set_audio_codecs({}, {});
// Perform initial offer/answer in reverse (|f2_| as offerer) so that the
// second offer/answer is forward (|f1_| as offerer).
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "v0",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::unique_ptr<SessionDescription> offer = f2_.CreateOffer(opts, nullptr);
std::unique_ptr<SessionDescription> answer =
f1_.CreateAnswer(offer.get(), opts, nullptr);
// Recycle the media section by changing its mid.
opts.media_description_options[0].mid = "v1";
std::unique_ptr<SessionDescription> reoffer =
f1_.CreateOffer(opts, answer.get());
std::unique_ptr<SessionDescription> reanswer =
f2_.CreateAnswer(reoffer.get(), opts, offer.get());
// Expect that the results of the first negotiation are ignored. If the m=
// section was not recycled the payload types would match the initial offerer.
const VideoContentDescription* vcd =
GetFirstVideoContentDescription(reanswer.get());
EXPECT_THAT(vcd->codecs(), ElementsAreArray(kVideoCodecsAnswer));
}
// Create an updated offer after creating an answer to the original offer and
// verify that the codecs that were part of the original answer are not changed
// in the updated offer. In this test Rtx is enabled.
TEST_F(MediaSessionDescriptionFactoryTest,
RespondentCreatesOfferAfterCreatingAnswerWithRtx) {
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
std::vector<VideoCodec> f1_codecs = MAKE_VECTOR(kVideoCodecs1);
// This creates rtx for H264 with the payload type |f1_| uses.
AddRtxCodec(VideoCodec::CreateRtxCodec(126, kVideoCodecs1[1].id), &f1_codecs);
f1_.set_video_codecs(f1_codecs, f1_codecs);
std::vector<VideoCodec> f2_codecs = MAKE_VECTOR(kVideoCodecs2);
// This creates rtx for H264 with the payload type |f2_| uses.
AddRtxCodec(VideoCodec::CreateRtxCodec(125, kVideoCodecs2[0].id), &f2_codecs);
f2_.set_video_codecs(f2_codecs, f2_codecs);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, NULL);
const VideoContentDescription* vcd =
GetFirstVideoContentDescription(answer.get());
std::vector<VideoCodec> expected_codecs = MAKE_VECTOR(kVideoCodecsAnswer);
AddRtxCodec(VideoCodec::CreateRtxCodec(126, kVideoCodecs1[1].id),
&expected_codecs);
EXPECT_EQ(expected_codecs, vcd->codecs());
// Now, make sure we get same result (except for the order) if |f2_| creates
// an updated offer even though the default payload types between |f1_| and
// |f2_| are different.
std::unique_ptr<SessionDescription> updated_offer(
f2_.CreateOffer(opts, answer.get()));
ASSERT_TRUE(updated_offer);
std::unique_ptr<SessionDescription> updated_answer(
f1_.CreateAnswer(updated_offer.get(), opts, answer.get()));
const VideoContentDescription* updated_vcd =
GetFirstVideoContentDescription(updated_answer.get());
EXPECT_EQ(expected_codecs, updated_vcd->codecs());
}
// Regression test for:
// https://bugs.chromium.org/p/webrtc/issues/detail?id=8332
// Existing codecs should always appear before new codecs in re-offers. But
// under a specific set of circumstances, the existing RTX codec was ending up
// added to the end of the list.
TEST_F(MediaSessionDescriptionFactoryTest,
RespondentCreatesOfferAfterCreatingAnswerWithRemappedRtxPayloadType) {
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
// We specifically choose different preferred payload types for VP8 to
// trigger the issue.
cricket::VideoCodec vp8_offerer(100, "VP8");
cricket::VideoCodec vp8_offerer_rtx =
VideoCodec::CreateRtxCodec(101, vp8_offerer.id);
cricket::VideoCodec vp8_answerer(110, "VP8");
cricket::VideoCodec vp8_answerer_rtx =
VideoCodec::CreateRtxCodec(111, vp8_answerer.id);
cricket::VideoCodec vp9(120, "VP9");
cricket::VideoCodec vp9_rtx = VideoCodec::CreateRtxCodec(121, vp9.id);
std::vector<VideoCodec> f1_codecs = {vp8_offerer, vp8_offerer_rtx};
// We also specifically cause the answerer to prefer VP9, such that if it
// *doesn't* honor the existing preferred codec (VP8) we'll notice.
std::vector<VideoCodec> f2_codecs = {vp9, vp9_rtx, vp8_answerer,
vp8_answerer_rtx};
f1_.set_video_codecs(f1_codecs, f1_codecs);
f2_.set_video_codecs(f2_codecs, f2_codecs);
std::vector<AudioCodec> audio_codecs;
f1_.set_audio_codecs(audio_codecs, audio_codecs);
f2_.set_audio_codecs(audio_codecs, audio_codecs);
// Offer will be {VP8, RTX for VP8}. Answer will be the same.
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, NULL);
// Updated offer *should* be {VP8, RTX for VP8, VP9, RTX for VP9}.
// But if the bug is triggered, RTX for VP8 ends up last.
std::unique_ptr<SessionDescription> updated_offer(
f2_.CreateOffer(opts, answer.get()));
const VideoContentDescription* vcd =
GetFirstVideoContentDescription(updated_offer.get());
std::vector<cricket::VideoCodec> codecs = vcd->codecs();
ASSERT_EQ(4u, codecs.size());
EXPECT_EQ(vp8_offerer, codecs[0]);
EXPECT_EQ(vp8_offerer_rtx, codecs[1]);
EXPECT_EQ(vp9, codecs[2]);
EXPECT_EQ(vp9_rtx, codecs[3]);
}
// Create an updated offer that adds video after creating an audio only answer
// to the original offer. This test verifies that if a video codec and the RTX
// codec have the same default payload type as an audio codec that is already in
// use, the added codecs payload types are changed.
TEST_F(MediaSessionDescriptionFactoryTest,
RespondentCreatesOfferWithVideoAndRtxAfterCreatingAudioAnswer) {
std::vector<VideoCodec> f1_codecs = MAKE_VECTOR(kVideoCodecs1);
// This creates rtx for H264 with the payload type |f1_| uses.
AddRtxCodec(VideoCodec::CreateRtxCodec(126, kVideoCodecs1[1].id), &f1_codecs);
f1_.set_video_codecs(f1_codecs, f1_codecs);
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, NULL);
const AudioContentDescription* acd =
GetFirstAudioContentDescription(answer.get());
EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
// Now - let |f2_| add video with RTX and let the payload type the RTX codec
// reference be the same as an audio codec that was negotiated in the
// first offer/answer exchange.
opts.media_description_options.clear();
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
std::vector<VideoCodec> f2_codecs = MAKE_VECTOR(kVideoCodecs2);
int used_pl_type = acd->codecs()[0].id;
f2_codecs[0].id = used_pl_type; // Set the payload type for H264.
AddRtxCodec(VideoCodec::CreateRtxCodec(125, used_pl_type), &f2_codecs);
f2_.set_video_codecs(f2_codecs, f2_codecs);
std::unique_ptr<SessionDescription> updated_offer(
f2_.CreateOffer(opts, answer.get()));
ASSERT_TRUE(updated_offer);
std::unique_ptr<SessionDescription> updated_answer(
f1_.CreateAnswer(updated_offer.get(), opts, answer.get()));
const AudioContentDescription* updated_acd =
GetFirstAudioContentDescription(answer.get());
EXPECT_THAT(updated_acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
const VideoContentDescription* updated_vcd =
GetFirstVideoContentDescription(updated_answer.get());
ASSERT_EQ("H264", updated_vcd->codecs()[0].name);
ASSERT_EQ(cricket::kRtxCodecName, updated_vcd->codecs()[1].name);
int new_h264_pl_type = updated_vcd->codecs()[0].id;
EXPECT_NE(used_pl_type, new_h264_pl_type);
VideoCodec rtx = updated_vcd->codecs()[1];
int pt_referenced_by_rtx = rtc::FromString<int>(
rtx.params[cricket::kCodecParamAssociatedPayloadType]);
EXPECT_EQ(new_h264_pl_type, pt_referenced_by_rtx);
}
// Create an updated offer with RTX after creating an answer to an offer
// without RTX, and with different default payload types.
// Verify that the added RTX codec references the correct payload type.
TEST_F(MediaSessionDescriptionFactoryTest,
RespondentCreatesOfferWithRtxAfterCreatingAnswerWithoutRtx) {
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
std::vector<VideoCodec> f2_codecs = MAKE_VECTOR(kVideoCodecs2);
// This creates rtx for H264 with the payload type |f2_| uses.
AddRtxCodec(VideoCodec::CreateRtxCodec(125, kVideoCodecs2[0].id), &f2_codecs);
f2_.set_video_codecs(f2_codecs, f2_codecs);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
ASSERT_TRUE(offer.get() != nullptr);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, nullptr);
const VideoContentDescription* vcd =
GetFirstVideoContentDescription(answer.get());
std::vector<VideoCodec> expected_codecs = MAKE_VECTOR(kVideoCodecsAnswer);
EXPECT_EQ(expected_codecs, vcd->codecs());
// Now, ensure that the RTX codec is created correctly when |f2_| creates an
// updated offer, even though the default payload types are different from
// those of |f1_|.
std::unique_ptr<SessionDescription> updated_offer(
f2_.CreateOffer(opts, answer.get()));
ASSERT_TRUE(updated_offer);
const VideoContentDescription* updated_vcd =
GetFirstVideoContentDescription(updated_offer.get());
// New offer should attempt to add H263, and RTX for H264.
expected_codecs.push_back(kVideoCodecs2[1]);
AddRtxCodec(VideoCodec::CreateRtxCodec(125, kVideoCodecs1[1].id),
&expected_codecs);
EXPECT_EQ(expected_codecs, updated_vcd->codecs());
}
// Test that RTX is ignored when there is no associated payload type parameter.
TEST_F(MediaSessionDescriptionFactoryTest, RtxWithoutApt) {
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
std::vector<VideoCodec> f1_codecs = MAKE_VECTOR(kVideoCodecs1);
// This creates RTX without associated payload type parameter.
AddRtxCodec(VideoCodec(126, cricket::kRtxCodecName), &f1_codecs);
f1_.set_video_codecs(f1_codecs, f1_codecs);
std::vector<VideoCodec> f2_codecs = MAKE_VECTOR(kVideoCodecs2);
// This creates RTX for H264 with the payload type |f2_| uses.
AddRtxCodec(VideoCodec::CreateRtxCodec(125, kVideoCodecs2[0].id), &f2_codecs);
f2_.set_video_codecs(f2_codecs, f2_codecs);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
// kCodecParamAssociatedPayloadType will always be added to the offer when RTX
// is selected. Manually remove kCodecParamAssociatedPayloadType so that it
// is possible to test that that RTX is dropped when
// kCodecParamAssociatedPayloadType is missing in the offer.
MediaContentDescription* media_desc =
offer->GetContentDescriptionByName(cricket::CN_VIDEO);
ASSERT_TRUE(media_desc);
VideoContentDescription* desc = media_desc->as_video();
std::vector<VideoCodec> codecs = desc->codecs();
for (VideoCodec& codec : codecs) {
if (absl::StartsWith(codec.name, cricket::kRtxCodecName)) {
codec.params.clear();
}
}
desc->set_codecs(codecs);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, NULL);
EXPECT_THAT(
GetCodecNames(GetFirstVideoContentDescription(answer.get())->codecs()),
Not(Contains(cricket::kRtxCodecName)));
}
// Test that RTX will be filtered out in the answer if its associated payload
// type doesn't match the local value.
TEST_F(MediaSessionDescriptionFactoryTest, FilterOutRtxIfAptDoesntMatch) {
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
std::vector<VideoCodec> f1_codecs = MAKE_VECTOR(kVideoCodecs1);
// This creates RTX for H264 in sender.
AddRtxCodec(VideoCodec::CreateRtxCodec(126, kVideoCodecs1[1].id), &f1_codecs);
f1_.set_video_codecs(f1_codecs, f1_codecs);
std::vector<VideoCodec> f2_codecs = MAKE_VECTOR(kVideoCodecs2);
// This creates RTX for H263 in receiver.
AddRtxCodec(VideoCodec::CreateRtxCodec(125, kVideoCodecs2[1].id), &f2_codecs);
f2_.set_video_codecs(f2_codecs, f2_codecs);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
// Associated payload type doesn't match, therefore, RTX codec is removed in
// the answer.
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, NULL);
EXPECT_THAT(
GetCodecNames(GetFirstVideoContentDescription(answer.get())->codecs()),
Not(Contains(cricket::kRtxCodecName)));
}
// Test that when multiple RTX codecs are offered, only the matched RTX codec
// is added in the answer, and the unsupported RTX codec is filtered out.
TEST_F(MediaSessionDescriptionFactoryTest,
FilterOutUnsupportedRtxWhenCreatingAnswer) {
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
std::vector<VideoCodec> f1_codecs = MAKE_VECTOR(kVideoCodecs1);
// This creates RTX for H264-SVC in sender.
AddRtxCodec(VideoCodec::CreateRtxCodec(125, kVideoCodecs1[0].id), &f1_codecs);
f1_.set_video_codecs(f1_codecs, f1_codecs);
// This creates RTX for H264 in sender.
AddRtxCodec(VideoCodec::CreateRtxCodec(126, kVideoCodecs1[1].id), &f1_codecs);
f1_.set_video_codecs(f1_codecs, f1_codecs);
std::vector<VideoCodec> f2_codecs = MAKE_VECTOR(kVideoCodecs2);
// This creates RTX for H264 in receiver.
AddRtxCodec(VideoCodec::CreateRtxCodec(124, kVideoCodecs2[0].id), &f2_codecs);
f2_.set_video_codecs(f2_codecs, f1_codecs);
// H264-SVC codec is removed in the answer, therefore, associated RTX codec
// for H264-SVC should also be removed.
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, NULL);
const VideoContentDescription* vcd =
GetFirstVideoContentDescription(answer.get());
std::vector<VideoCodec> expected_codecs = MAKE_VECTOR(kVideoCodecsAnswer);
AddRtxCodec(VideoCodec::CreateRtxCodec(126, kVideoCodecs1[1].id),
&expected_codecs);
EXPECT_EQ(expected_codecs, vcd->codecs());
}
// Test that after one RTX codec has been negotiated, a new offer can attempt
// to add another.
TEST_F(MediaSessionDescriptionFactoryTest, AddSecondRtxInNewOffer) {
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
std::vector<VideoCodec> f1_codecs = MAKE_VECTOR(kVideoCodecs1);
// This creates RTX for H264 for the offerer.
AddRtxCodec(VideoCodec::CreateRtxCodec(126, kVideoCodecs1[1].id), &f1_codecs);
f1_.set_video_codecs(f1_codecs, f1_codecs);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
ASSERT_TRUE(offer);
const VideoContentDescription* vcd =
GetFirstVideoContentDescription(offer.get());
std::vector<VideoCodec> expected_codecs = MAKE_VECTOR(kVideoCodecs1);
AddRtxCodec(VideoCodec::CreateRtxCodec(126, kVideoCodecs1[1].id),
&expected_codecs);
EXPECT_EQ(expected_codecs, vcd->codecs());
// Now, attempt to add RTX for H264-SVC.
AddRtxCodec(VideoCodec::CreateRtxCodec(125, kVideoCodecs1[0].id), &f1_codecs);
f1_.set_video_codecs(f1_codecs, f1_codecs);
std::unique_ptr<SessionDescription> updated_offer(
f1_.CreateOffer(opts, offer.get()));
ASSERT_TRUE(updated_offer);
vcd = GetFirstVideoContentDescription(updated_offer.get());
AddRtxCodec(VideoCodec::CreateRtxCodec(125, kVideoCodecs1[0].id),
&expected_codecs);
EXPECT_EQ(expected_codecs, vcd->codecs());
}
// Test that when RTX is used in conjunction with simulcast, an RTX ssrc is
// generated for each simulcast ssrc and correctly grouped.
TEST_F(MediaSessionDescriptionFactoryTest, SimSsrcsGenerateMultipleRtxSsrcs) {
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
// Add simulcast streams.
AttachSenderToMediaDescriptionOptions("video", MEDIA_TYPE_VIDEO, "stream1",
{"stream1label"}, 3, &opts);
// Use a single real codec, and then add RTX for it.
std::vector<VideoCodec> f1_codecs;
f1_codecs.push_back(VideoCodec(97, "H264"));
AddRtxCodec(VideoCodec::CreateRtxCodec(125, 97), &f1_codecs);
f1_.set_video_codecs(f1_codecs, f1_codecs);
// Ensure that the offer has an RTX ssrc for each regular ssrc, and that there
// is a FID ssrc + grouping for each.
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
MediaContentDescription* media_desc =
offer->GetContentDescriptionByName(cricket::CN_VIDEO);
ASSERT_TRUE(media_desc);
VideoContentDescription* desc = media_desc->as_video();
const StreamParamsVec& streams = desc->streams();
// Single stream.
ASSERT_EQ(1u, streams.size());
// Stream should have 6 ssrcs: 3 for video, 3 for RTX.
EXPECT_EQ(6u, streams[0].ssrcs.size());
// And should have a SIM group for the simulcast.
EXPECT_TRUE(streams[0].has_ssrc_group("SIM"));
// And a FID group for RTX.
EXPECT_TRUE(streams[0].has_ssrc_group("FID"));
std::vector<uint32_t> primary_ssrcs;
streams[0].GetPrimarySsrcs(&primary_ssrcs);
EXPECT_EQ(3u, primary_ssrcs.size());
std::vector<uint32_t> fid_ssrcs;
streams[0].GetFidSsrcs(primary_ssrcs, &fid_ssrcs);
EXPECT_EQ(3u, fid_ssrcs.size());
}
// Test that, when the FlexFEC codec is added, a FlexFEC ssrc is created
// together with a FEC-FR grouping.
TEST_F(MediaSessionDescriptionFactoryTest, GenerateFlexfecSsrc) {
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
// Add single stream.
AttachSenderToMediaDescriptionOptions("video", MEDIA_TYPE_VIDEO, "stream1",
{"stream1label"}, 1, &opts);
// Use a single real codec, and then add FlexFEC for it.
std::vector<VideoCodec> f1_codecs;
f1_codecs.push_back(VideoCodec(97, "H264"));
f1_codecs.push_back(VideoCodec(118, "flexfec-03"));
f1_.set_video_codecs(f1_codecs, f1_codecs);
// Ensure that the offer has a single FlexFEC ssrc and that
// there is no FEC-FR ssrc + grouping for each.
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
ASSERT_TRUE(offer.get() != nullptr);
MediaContentDescription* media_desc =
offer->GetContentDescriptionByName(cricket::CN_VIDEO);
ASSERT_TRUE(media_desc);
VideoContentDescription* desc = media_desc->as_video();
const StreamParamsVec& streams = desc->streams();
// Single stream.
ASSERT_EQ(1u, streams.size());
// Stream should have 2 ssrcs: 1 for video, 1 for FlexFEC.
EXPECT_EQ(2u, streams[0].ssrcs.size());
// And should have a FEC-FR group for FlexFEC.
EXPECT_TRUE(streams[0].has_ssrc_group("FEC-FR"));
std::vector<uint32_t> primary_ssrcs;
streams[0].GetPrimarySsrcs(&primary_ssrcs);
ASSERT_EQ(1u, primary_ssrcs.size());
uint32_t flexfec_ssrc;
EXPECT_TRUE(streams[0].GetFecFrSsrc(primary_ssrcs[0], &flexfec_ssrc));
EXPECT_NE(flexfec_ssrc, 0u);
}
// Test that FlexFEC is disabled for simulcast.
// TODO(brandtr): Remove this test when we support simulcast, either through
// multiple FlexfecSenders, or through multistream protection.
TEST_F(MediaSessionDescriptionFactoryTest, SimSsrcsGenerateNoFlexfecSsrcs) {
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
// Add simulcast streams.
AttachSenderToMediaDescriptionOptions("video", MEDIA_TYPE_VIDEO, "stream1",
{"stream1label"}, 3, &opts);
// Use a single real codec, and then add FlexFEC for it.
std::vector<VideoCodec> f1_codecs;
f1_codecs.push_back(VideoCodec(97, "H264"));
f1_codecs.push_back(VideoCodec(118, "flexfec-03"));
f1_.set_video_codecs(f1_codecs, f1_codecs);
// Ensure that the offer has no FlexFEC ssrcs for each regular ssrc, and that
// there is no FEC-FR ssrc + grouping for each.
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
ASSERT_TRUE(offer.get() != nullptr);
MediaContentDescription* media_desc =
offer->GetContentDescriptionByName(cricket::CN_VIDEO);
ASSERT_TRUE(media_desc);
VideoContentDescription* desc = media_desc->as_video();
const StreamParamsVec& streams = desc->streams();
// Single stream.
ASSERT_EQ(1u, streams.size());
// Stream should have 3 ssrcs: 3 for video, 0 for FlexFEC.
EXPECT_EQ(3u, streams[0].ssrcs.size());
// And should have a SIM group for the simulcast.
EXPECT_TRUE(streams[0].has_ssrc_group("SIM"));
// And not a FEC-FR group for FlexFEC.
EXPECT_FALSE(streams[0].has_ssrc_group("FEC-FR"));
std::vector<uint32_t> primary_ssrcs;
streams[0].GetPrimarySsrcs(&primary_ssrcs);
EXPECT_EQ(3u, primary_ssrcs.size());
for (uint32_t primary_ssrc : primary_ssrcs) {
uint32_t flexfec_ssrc;
EXPECT_FALSE(streams[0].GetFecFrSsrc(primary_ssrc, &flexfec_ssrc));
}
}
// Create an updated offer after creating an answer to the original offer and
// verify that the RTP header extensions that were part of the original answer
// are not changed in the updated offer.
TEST_F(MediaSessionDescriptionFactoryTest,
RespondentCreatesOfferAfterCreatingAnswerWithRtpExtensions) {
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
f1_.set_audio_rtp_header_extensions(MAKE_VECTOR(kAudioRtpExtension1));
f1_.set_video_rtp_header_extensions(MAKE_VECTOR(kVideoRtpExtension1));
f2_.set_audio_rtp_header_extensions(MAKE_VECTOR(kAudioRtpExtension2));
f2_.set_video_rtp_header_extensions(MAKE_VECTOR(kVideoRtpExtension2));
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, NULL);
EXPECT_EQ(
MAKE_VECTOR(kAudioRtpExtensionAnswer),
GetFirstAudioContentDescription(answer.get())->rtp_header_extensions());
EXPECT_EQ(
MAKE_VECTOR(kVideoRtpExtensionAnswer),
GetFirstVideoContentDescription(answer.get())->rtp_header_extensions());
std::unique_ptr<SessionDescription> updated_offer(
f2_.CreateOffer(opts, answer.get()));
// The expected RTP header extensions in the new offer are the resulting
// extensions from the first offer/answer exchange plus the extensions only
// |f2_| offer.
// Since the default local extension id |f2_| uses has already been used by
// |f1_| for another extensions, it is changed to 13.
const RtpExtension kUpdatedAudioRtpExtensions[] = {
kAudioRtpExtensionAnswer[0],
RtpExtension(kAudioRtpExtension2[1].uri, 13),
kAudioRtpExtension2[2],
};
// Since the default local extension id |f2_| uses has already been used by
// |f1_| for another extensions, is is changed to 12.
const RtpExtension kUpdatedVideoRtpExtensions[] = {
kVideoRtpExtensionAnswer[0],
RtpExtension(kVideoRtpExtension2[1].uri, 12),
kVideoRtpExtension2[2],
};
const AudioContentDescription* updated_acd =
GetFirstAudioContentDescription(updated_offer.get());
EXPECT_EQ(MAKE_VECTOR(kUpdatedAudioRtpExtensions),
updated_acd->rtp_header_extensions());
const VideoContentDescription* updated_vcd =
GetFirstVideoContentDescription(updated_offer.get());
EXPECT_EQ(MAKE_VECTOR(kUpdatedVideoRtpExtensions),
updated_vcd->rtp_header_extensions());
}
// Verify that if the same RTP extension URI is used for audio and video, the
// same ID is used. Also verify that the ID isn't changed when creating an
// updated offer (this was previously a bug).
TEST_F(MediaSessionDescriptionFactoryTest, RtpExtensionIdReused) {
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
f1_.set_audio_rtp_header_extensions(MAKE_VECTOR(kAudioRtpExtension3));
f1_.set_video_rtp_header_extensions(MAKE_VECTOR(kVideoRtpExtension3));
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
// Since the audio extensions used ID 3 for "both_audio_and_video", so should
// the video extensions.
const RtpExtension kExpectedVideoRtpExtension[] = {
kVideoRtpExtension3[0],
kAudioRtpExtension3[1],
};
EXPECT_EQ(
MAKE_VECTOR(kAudioRtpExtension3),
GetFirstAudioContentDescription(offer.get())->rtp_header_extensions());
EXPECT_EQ(
MAKE_VECTOR(kExpectedVideoRtpExtension),
GetFirstVideoContentDescription(offer.get())->rtp_header_extensions());
// Nothing should change when creating a new offer
std::unique_ptr<SessionDescription> updated_offer(
f1_.CreateOffer(opts, offer.get()));
EXPECT_EQ(MAKE_VECTOR(kAudioRtpExtension3),
GetFirstAudioContentDescription(updated_offer.get())
->rtp_header_extensions());
EXPECT_EQ(MAKE_VECTOR(kExpectedVideoRtpExtension),
GetFirstVideoContentDescription(updated_offer.get())
->rtp_header_extensions());
}
// Same as "RtpExtensionIdReused" above for encrypted RTP extensions.
TEST_F(MediaSessionDescriptionFactoryTest, RtpExtensionIdReusedEncrypted) {
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
f1_.set_enable_encrypted_rtp_header_extensions(true);
f2_.set_enable_encrypted_rtp_header_extensions(true);
f1_.set_audio_rtp_header_extensions(
MAKE_VECTOR(kAudioRtpExtension3ForEncryption));
f1_.set_video_rtp_header_extensions(
MAKE_VECTOR(kVideoRtpExtension3ForEncryption));
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
// The extensions that are shared between audio and video should use the same
// id.
const RtpExtension kExpectedVideoRtpExtension[] = {
kVideoRtpExtension3ForEncryption[0],
kAudioRtpExtension3ForEncryptionOffer[1],
kAudioRtpExtension3ForEncryptionOffer[2],
};
EXPECT_EQ(
MAKE_VECTOR(kAudioRtpExtension3ForEncryptionOffer),
GetFirstAudioContentDescription(offer.get())->rtp_header_extensions());
EXPECT_EQ(
MAKE_VECTOR(kExpectedVideoRtpExtension),
GetFirstVideoContentDescription(offer.get())->rtp_header_extensions());
// Nothing should change when creating a new offer
std::unique_ptr<SessionDescription> updated_offer(
f1_.CreateOffer(opts, offer.get()));
EXPECT_EQ(MAKE_VECTOR(kAudioRtpExtension3ForEncryptionOffer),
GetFirstAudioContentDescription(updated_offer.get())
->rtp_header_extensions());
EXPECT_EQ(MAKE_VECTOR(kExpectedVideoRtpExtension),
GetFirstVideoContentDescription(updated_offer.get())
->rtp_header_extensions());
}
TEST(MediaSessionDescription, CopySessionDescription) {
SessionDescription source;
cricket::ContentGroup group(cricket::CN_AUDIO);
source.AddGroup(group);
std::unique_ptr<AudioContentDescription> acd =
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
std::make_unique<AudioContentDescription>();
acd->set_codecs(MAKE_VECTOR(kAudioCodecs1));
acd->AddLegacyStream(1);
source.AddContent(cricket::CN_AUDIO, MediaProtocolType::kRtp, acd->Clone());
std::unique_ptr<VideoContentDescription> vcd =
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
std::make_unique<VideoContentDescription>();
vcd->set_codecs(MAKE_VECTOR(kVideoCodecs1));
vcd->AddLegacyStream(2);
source.AddContent(cricket::CN_VIDEO, MediaProtocolType::kRtp, vcd->Clone());
std::unique_ptr<SessionDescription> copy = source.Clone();
ASSERT_TRUE(copy.get() != NULL);
EXPECT_TRUE(copy->HasGroup(cricket::CN_AUDIO));
const ContentInfo* ac = copy->GetContentByName("audio");
const ContentInfo* vc = copy->GetContentByName("video");
ASSERT_TRUE(ac != NULL);
ASSERT_TRUE(vc != NULL);
EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
const AudioContentDescription* acd_copy = ac->media_description()->as_audio();
EXPECT_EQ(acd->codecs(), acd_copy->codecs());
EXPECT_EQ(1u, acd->first_ssrc());
EXPECT_EQ(MediaProtocolType::kRtp, vc->type);
const VideoContentDescription* vcd_copy = vc->media_description()->as_video();
EXPECT_EQ(vcd->codecs(), vcd_copy->codecs());
EXPECT_EQ(2u, vcd->first_ssrc());
}
// The below TestTransportInfoXXX tests create different offers/answers, and
// ensure the TransportInfo in the SessionDescription matches what we expect.
TEST_F(MediaSessionDescriptionFactoryTest, TestTransportInfoOfferAudio) {
MediaSessionOptions options;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&options);
TestTransportInfo(true, options, false);
}
TEST_F(MediaSessionDescriptionFactoryTest,
TestTransportInfoOfferIceRenomination) {
MediaSessionOptions options;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&options);
options.media_description_options[0]
.transport_options.enable_ice_renomination = true;
TestTransportInfo(true, options, false);
}
TEST_F(MediaSessionDescriptionFactoryTest, TestTransportInfoOfferAudioCurrent) {
MediaSessionOptions options;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&options);
TestTransportInfo(true, options, true);
}
TEST_F(MediaSessionDescriptionFactoryTest, TestTransportInfoOfferMultimedia) {
MediaSessionOptions options;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly,
&options);
TestTransportInfo(true, options, false);
}
TEST_F(MediaSessionDescriptionFactoryTest,
TestTransportInfoOfferMultimediaCurrent) {
MediaSessionOptions options;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly,
&options);
TestTransportInfo(true, options, true);
}
TEST_F(MediaSessionDescriptionFactoryTest, TestTransportInfoOfferBundle) {
MediaSessionOptions options;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly,
&options);
options.bundle_enabled = true;
TestTransportInfo(true, options, false);
}
TEST_F(MediaSessionDescriptionFactoryTest,
TestTransportInfoOfferBundleCurrent) {
MediaSessionOptions options;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly,
&options);
options.bundle_enabled = true;
TestTransportInfo(true, options, true);
}
TEST_F(MediaSessionDescriptionFactoryTest, TestTransportInfoAnswerAudio) {
MediaSessionOptions options;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&options);
TestTransportInfo(false, options, false);
}
TEST_F(MediaSessionDescriptionFactoryTest,
TestTransportInfoAnswerIceRenomination) {
MediaSessionOptions options;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&options);
options.media_description_options[0]
.transport_options.enable_ice_renomination = true;
TestTransportInfo(false, options, false);
}
TEST_F(MediaSessionDescriptionFactoryTest,
TestTransportInfoAnswerAudioCurrent) {
MediaSessionOptions options;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&options);
TestTransportInfo(false, options, true);
}
TEST_F(MediaSessionDescriptionFactoryTest, TestTransportInfoAnswerMultimedia) {
MediaSessionOptions options;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly,
&options);
TestTransportInfo(false, options, false);
}
TEST_F(MediaSessionDescriptionFactoryTest,
TestTransportInfoAnswerMultimediaCurrent) {
MediaSessionOptions options;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly,
&options);
TestTransportInfo(false, options, true);
}
TEST_F(MediaSessionDescriptionFactoryTest, TestTransportInfoAnswerBundle) {
MediaSessionOptions options;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly,
&options);
options.bundle_enabled = true;
TestTransportInfo(false, options, false);
}
TEST_F(MediaSessionDescriptionFactoryTest,
TestTransportInfoAnswerBundleCurrent) {
MediaSessionOptions options;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly,
&options);
options.bundle_enabled = true;
TestTransportInfo(false, options, true);
}
Reland: Implement true negotiation for DatagramTransport with fallback to RTP. In short, the caller places a x-opaque line in SDP for each m= section that uses datagram transport. If the answerer supports datagram transport, it will parse this line and create a datagram transport. It will then echo the x-opaque line into the answer (to indicate that it accepted use of datagram transport). If the offer and answer contain exactly the same x-opaque line, both peers will use datagram transport. If the x-opaque line is omitted from the answer (or is different in the answer) they will fall back to RTP. Note that a different x-opaque line in the answer means the answerer did not understand something in the negotiation proto. Since WebRTC cannot know what was misunderstood, or whether it's still possible to use the datagram transport, it must fall back to RTP. This may change in the future, possibly by passing the answer to the datagram transport, but it's good enough for now. Negotiation consists of four parts: 1. DatagramTransport exposes transport parameters for both client and server perspectives. The client just echoes what it received from the server (modulo any fields it might not have understood). 2. SDP adds a x-opaque line for opaque transport parameters. Identical to x-mt, but this is specific to datagram transport and goes in each m= section, and appears in the answer as well as the offer. - This is propagated to Jsep as part of the TransportDescription. - SDP files: transport_description.h,cc, transport_description_factory.h,cc, media_session.cc, webrtc_sdp.cc 3. JsepTransport/Controller: - Exposes opaque parameters for each mid (m= section). On offerer, this means pre-allocating a datagram transport and getting its parameters. On the answerer, this means echoing the offerer's parameters. - Uses a composite RTP transport to receive from either default RTP or datagram transport until both offer and answer arrive. - If a provisional answer arrives, sets the composite to send on the provisionally selected transport. - Once both offer and answer are set, deletes the unneeded transports and keeps whichever transport is selected. 4. PeerConnection pulls transport parameters out of Jsep and adds them to SDP. Bug: webrtc:9719 Change-Id: Ifcc428c8d76fb77dcc8abaa79507c620bcfb31b9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140920 Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28198}
2019-06-07 10:28:06 -07:00
TEST_F(MediaSessionDescriptionFactoryTest,
TestTransportInfoOfferBundlesTransportOptions) {
MediaSessionOptions options;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
cricket::OpaqueTransportParameters audio_params;
audio_params.protocol = "audio-transport";
audio_params.parameters = "audio-params";
FindFirstMediaDescriptionByMid("audio", &options)
->transport_options.opaque_parameters = audio_params;
cricket::OpaqueTransportParameters video_params;
video_params.protocol = "video-transport";
video_params.parameters = "video-params";
FindFirstMediaDescriptionByMid("video", &options)
->transport_options.opaque_parameters = video_params;
TestTransportInfo(/*offer=*/true, options, /*has_current_desc=*/false);
}
TEST_F(MediaSessionDescriptionFactoryTest,
TestTransportInfoAnswerBundlesTransportOptions) {
MediaSessionOptions options;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
cricket::OpaqueTransportParameters audio_params;
audio_params.protocol = "audio-transport";
audio_params.parameters = "audio-params";
FindFirstMediaDescriptionByMid("audio", &options)
->transport_options.opaque_parameters = audio_params;
cricket::OpaqueTransportParameters video_params;
video_params.protocol = "video-transport";
video_params.parameters = "video-params";
FindFirstMediaDescriptionByMid("video", &options)
->transport_options.opaque_parameters = video_params;
TestTransportInfo(/*offer=*/false, options, /*has_current_desc=*/false);
}
Add an alt-protocol to SDP to indicate which m= sections use a plugin transport. The plugin transport parameters (a=x-opaque: lines) relate to how to create and set up a plugin transport. When SDP bundle is used, the x-opaque line needs to be copied into the bundled m= section. This means x-opaque can appear on a section even if the offerer does not intend to use the transport for the media described by that section. Consequently, the answerer cannot currently tell whether the caller is offering an alternate transport for media, data, or both. This change adds an a=x-alt-protocol: line to SDP. The value following this line matches the <protocol> part of the x-opaque:<protocol>:<params> line. However, alt-protocol is not bundled--it only ever applies to the m= section that contains the line. This allows the offerer to express which m= sections should actually use an alternate transport, even in the case of bundle. Note that this is still limited by the available configuration options: datagram transport can be used for media (audio + video) and/or data. It is still not possible to use it for audio but not video, or vice versa. PeerConnection places an alt-protocol line in each media (audio/video) m= section if it is configured to use a datagram transport for media. It places an alt-protocol line in each data m= section if it is configured to use a datagram transport for data channels. PeerConnection leaves alt-protocol in media (audio/video) m= sections of the answer if it is configured to use a datagram transport for media, and in data m= sections of the answer if it is configured to use a datagram transport for data channels. JsepTransport now negotiates use of the datagram transport independently for media and data channels. It only uses it for media if the m= sections for bundled audio/video have an alt-protocol line matching the x-opaque protocol, and only uses it for data channels if a bundled m= section for data has an alt-protocol line matching the x-opaque protocol. Bug: webrtc:9719 Change-Id: I773e4fc10c57d815afcd76a2a74da38dd0c52b3b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154763 Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29351}
2019-09-30 15:12:47 -07:00
TEST_F(MediaSessionDescriptionFactoryTest, AltProtocolAddedToOffer) {
MediaSessionOptions options;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly,
&options);
FindFirstMediaDescriptionByMid("audio", &options)->alt_protocol = "foo";
FindFirstMediaDescriptionByMid("video", &options)->alt_protocol = "bar";
FindFirstMediaDescriptionByMid("data", &options)->alt_protocol = "baz";
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(options, nullptr);
EXPECT_EQ(offer->GetContentDescriptionByName("audio")->alt_protocol(), "foo");
EXPECT_EQ(offer->GetContentDescriptionByName("video")->alt_protocol(), "bar");
EXPECT_EQ(offer->GetContentDescriptionByName("data")->alt_protocol(), "baz");
}
TEST_F(MediaSessionDescriptionFactoryTest, AltProtocolAddedToAnswer) {
MediaSessionOptions options;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
AddDataSection(cricket::DCT_SCTP, RtpTransceiverDirection::kRecvOnly,
&options);
FindFirstMediaDescriptionByMid("audio", &options)->alt_protocol = "foo";
FindFirstMediaDescriptionByMid("video", &options)->alt_protocol = "bar";
FindFirstMediaDescriptionByMid("data", &options)->alt_protocol = "baz";
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(options, nullptr);
std::unique_ptr<SessionDescription> answer =
f1_.CreateAnswer(offer.get(), options, nullptr);
EXPECT_EQ(answer->GetContentDescriptionByName("audio")->alt_protocol(),
"foo");
EXPECT_EQ(answer->GetContentDescriptionByName("video")->alt_protocol(),
"bar");
EXPECT_EQ(answer->GetContentDescriptionByName("data")->alt_protocol(), "baz");
}
TEST_F(MediaSessionDescriptionFactoryTest, AltProtocolNotInOffer) {
MediaSessionOptions options;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
AddDataSection(cricket::DCT_SCTP, RtpTransceiverDirection::kRecvOnly,
&options);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(options, nullptr);
FindFirstMediaDescriptionByMid("audio", &options)->alt_protocol = "foo";
FindFirstMediaDescriptionByMid("video", &options)->alt_protocol = "bar";
FindFirstMediaDescriptionByMid("data", &options)->alt_protocol = "baz";
std::unique_ptr<SessionDescription> answer =
f1_.CreateAnswer(offer.get(), options, nullptr);
EXPECT_EQ(answer->GetContentDescriptionByName("audio")->alt_protocol(),
absl::nullopt);
EXPECT_EQ(answer->GetContentDescriptionByName("video")->alt_protocol(),
absl::nullopt);
EXPECT_EQ(answer->GetContentDescriptionByName("data")->alt_protocol(),
absl::nullopt);
}
TEST_F(MediaSessionDescriptionFactoryTest, AltProtocolDifferentInOffer) {
MediaSessionOptions options;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
AddDataSection(cricket::DCT_SCTP, RtpTransceiverDirection::kRecvOnly,
&options);
FindFirstMediaDescriptionByMid("audio", &options)->alt_protocol = "not-foo";
FindFirstMediaDescriptionByMid("video", &options)->alt_protocol = "not-bar";
FindFirstMediaDescriptionByMid("data", &options)->alt_protocol = "not-baz";
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(options, nullptr);
FindFirstMediaDescriptionByMid("audio", &options)->alt_protocol = "foo";
FindFirstMediaDescriptionByMid("video", &options)->alt_protocol = "bar";
FindFirstMediaDescriptionByMid("data", &options)->alt_protocol = "baz";
std::unique_ptr<SessionDescription> answer =
f1_.CreateAnswer(offer.get(), options, nullptr);
EXPECT_EQ(answer->GetContentDescriptionByName("audio")->alt_protocol(),
absl::nullopt);
EXPECT_EQ(answer->GetContentDescriptionByName("video")->alt_protocol(),
absl::nullopt);
EXPECT_EQ(answer->GetContentDescriptionByName("data")->alt_protocol(),
absl::nullopt);
}
TEST_F(MediaSessionDescriptionFactoryTest, AltProtocolNotInAnswer) {
MediaSessionOptions options;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
AddDataSection(cricket::DCT_SCTP, RtpTransceiverDirection::kRecvOnly,
&options);
FindFirstMediaDescriptionByMid("audio", &options)->alt_protocol = "foo";
FindFirstMediaDescriptionByMid("video", &options)->alt_protocol = "bar";
FindFirstMediaDescriptionByMid("data", &options)->alt_protocol = "baz";
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(options, nullptr);
FindFirstMediaDescriptionByMid("audio", &options)->alt_protocol =
absl::nullopt;
FindFirstMediaDescriptionByMid("video", &options)->alt_protocol =
absl::nullopt;
FindFirstMediaDescriptionByMid("data", &options)->alt_protocol =
absl::nullopt;
std::unique_ptr<SessionDescription> answer =
f1_.CreateAnswer(offer.get(), options, nullptr);
EXPECT_EQ(answer->GetContentDescriptionByName("audio")->alt_protocol(),
absl::nullopt);
EXPECT_EQ(answer->GetContentDescriptionByName("video")->alt_protocol(),
absl::nullopt);
EXPECT_EQ(answer->GetContentDescriptionByName("data")->alt_protocol(),
absl::nullopt);
}
// Create an offer with bundle enabled and verify the crypto parameters are
// the common set of the available cryptos.
TEST_F(MediaSessionDescriptionFactoryTest, TestCryptoWithOfferBundle) {
TestCryptoWithBundle(true);
}
// Create an answer with bundle enabled and verify the crypto parameters are
// the common set of the available cryptos.
TEST_F(MediaSessionDescriptionFactoryTest, TestCryptoWithAnswerBundle) {
TestCryptoWithBundle(false);
}
// Verifies that creating answer fails if the offer has UDP/TLS/RTP/SAVPF but
// DTLS is not enabled locally.
TEST_F(MediaSessionDescriptionFactoryTest,
TestOfferDtlsSavpfWithoutDtlsFailed) {
f1_.set_secure(SEC_ENABLED);
f2_.set_secure(SEC_ENABLED);
tdf1_.set_secure(SEC_DISABLED);
tdf2_.set_secure(SEC_DISABLED);
std::unique_ptr<SessionDescription> offer =
f1_.CreateOffer(CreatePlanBMediaSessionOptions(), NULL);
ASSERT_TRUE(offer.get() != NULL);
ContentInfo* offer_content = offer->GetContentByName("audio");
ASSERT_TRUE(offer_content != NULL);
AudioContentDescription* offer_audio_desc =
offer_content->media_description()->as_audio();
offer_audio_desc->set_protocol(cricket::kMediaProtocolDtlsSavpf);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), CreatePlanBMediaSessionOptions(), NULL);
ASSERT_TRUE(answer != NULL);
ContentInfo* answer_content = answer->GetContentByName("audio");
ASSERT_TRUE(answer_content != NULL);
ASSERT_TRUE(answer_content->rejected);
}
// Offers UDP/TLS/RTP/SAVPF and verifies the answer can be created and contains
// UDP/TLS/RTP/SAVPF.
TEST_F(MediaSessionDescriptionFactoryTest, TestOfferDtlsSavpfCreateAnswer) {
f1_.set_secure(SEC_ENABLED);
f2_.set_secure(SEC_ENABLED);
tdf1_.set_secure(SEC_ENABLED);
tdf2_.set_secure(SEC_ENABLED);
std::unique_ptr<SessionDescription> offer =
f1_.CreateOffer(CreatePlanBMediaSessionOptions(), NULL);
ASSERT_TRUE(offer.get() != NULL);
ContentInfo* offer_content = offer->GetContentByName("audio");
ASSERT_TRUE(offer_content != NULL);
AudioContentDescription* offer_audio_desc =
offer_content->media_description()->as_audio();
offer_audio_desc->set_protocol(cricket::kMediaProtocolDtlsSavpf);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), CreatePlanBMediaSessionOptions(), NULL);
ASSERT_TRUE(answer != NULL);
const ContentInfo* answer_content = answer->GetContentByName("audio");
ASSERT_TRUE(answer_content != NULL);
ASSERT_FALSE(answer_content->rejected);
const AudioContentDescription* answer_audio_desc =
answer_content->media_description()->as_audio();
EXPECT_EQ(cricket::kMediaProtocolDtlsSavpf, answer_audio_desc->protocol());
}
// Test that we include both SDES and DTLS in the offer, but only include SDES
// in the answer if DTLS isn't negotiated.
TEST_F(MediaSessionDescriptionFactoryTest, TestCryptoDtls) {
f1_.set_secure(SEC_ENABLED);
f2_.set_secure(SEC_ENABLED);
tdf1_.set_secure(SEC_ENABLED);
tdf2_.set_secure(SEC_DISABLED);
MediaSessionOptions options;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
std::unique_ptr<SessionDescription> offer, answer;
const cricket::MediaContentDescription* audio_media_desc;
const cricket::MediaContentDescription* video_media_desc;
const cricket::TransportDescription* audio_trans_desc;
const cricket::TransportDescription* video_trans_desc;
// Generate an offer with SDES and DTLS support.
offer = f1_.CreateOffer(options, NULL);
ASSERT_TRUE(offer.get() != NULL);
audio_media_desc = offer->GetContentDescriptionByName("audio");
ASSERT_TRUE(audio_media_desc != NULL);
video_media_desc = offer->GetContentDescriptionByName("video");
ASSERT_TRUE(video_media_desc != NULL);
EXPECT_EQ(1u, audio_media_desc->cryptos().size());
EXPECT_EQ(1u, video_media_desc->cryptos().size());
audio_trans_desc = offer->GetTransportDescriptionByName("audio");
ASSERT_TRUE(audio_trans_desc != NULL);
video_trans_desc = offer->GetTransportDescriptionByName("video");
ASSERT_TRUE(video_trans_desc != NULL);
ASSERT_TRUE(audio_trans_desc->identity_fingerprint.get() != NULL);
ASSERT_TRUE(video_trans_desc->identity_fingerprint.get() != NULL);
// Generate an answer with only SDES support, since tdf2 has crypto disabled.
answer = f2_.CreateAnswer(offer.get(), options, NULL);
ASSERT_TRUE(answer.get() != NULL);
audio_media_desc = answer->GetContentDescriptionByName("audio");
ASSERT_TRUE(audio_media_desc != NULL);
video_media_desc = answer->GetContentDescriptionByName("video");
ASSERT_TRUE(video_media_desc != NULL);
EXPECT_EQ(1u, audio_media_desc->cryptos().size());
EXPECT_EQ(1u, video_media_desc->cryptos().size());
audio_trans_desc = answer->GetTransportDescriptionByName("audio");
ASSERT_TRUE(audio_trans_desc != NULL);
video_trans_desc = answer->GetTransportDescriptionByName("video");
ASSERT_TRUE(video_trans_desc != NULL);
ASSERT_TRUE(audio_trans_desc->identity_fingerprint.get() == NULL);
ASSERT_TRUE(video_trans_desc->identity_fingerprint.get() == NULL);
// Enable DTLS; the answer should now only have DTLS support.
tdf2_.set_secure(SEC_ENABLED);
answer = f2_.CreateAnswer(offer.get(), options, NULL);
ASSERT_TRUE(answer.get() != NULL);
audio_media_desc = answer->GetContentDescriptionByName("audio");
ASSERT_TRUE(audio_media_desc != NULL);
video_media_desc = answer->GetContentDescriptionByName("video");
ASSERT_TRUE(video_media_desc != NULL);
EXPECT_TRUE(audio_media_desc->cryptos().empty());
EXPECT_TRUE(video_media_desc->cryptos().empty());
EXPECT_EQ(cricket::kMediaProtocolSavpf, audio_media_desc->protocol());
EXPECT_EQ(cricket::kMediaProtocolSavpf, video_media_desc->protocol());
audio_trans_desc = answer->GetTransportDescriptionByName("audio");
ASSERT_TRUE(audio_trans_desc != NULL);
video_trans_desc = answer->GetTransportDescriptionByName("video");
ASSERT_TRUE(video_trans_desc != NULL);
ASSERT_TRUE(audio_trans_desc->identity_fingerprint.get() != NULL);
ASSERT_TRUE(video_trans_desc->identity_fingerprint.get() != NULL);
// Try creating offer again. DTLS enabled now, crypto's should be empty
// in new offer.
offer = f1_.CreateOffer(options, offer.get());
ASSERT_TRUE(offer.get() != NULL);
audio_media_desc = offer->GetContentDescriptionByName("audio");
ASSERT_TRUE(audio_media_desc != NULL);
video_media_desc = offer->GetContentDescriptionByName("video");
ASSERT_TRUE(video_media_desc != NULL);
EXPECT_TRUE(audio_media_desc->cryptos().empty());
EXPECT_TRUE(video_media_desc->cryptos().empty());
audio_trans_desc = offer->GetTransportDescriptionByName("audio");
ASSERT_TRUE(audio_trans_desc != NULL);
video_trans_desc = offer->GetTransportDescriptionByName("video");
ASSERT_TRUE(video_trans_desc != NULL);
ASSERT_TRUE(audio_trans_desc->identity_fingerprint.get() != NULL);
ASSERT_TRUE(video_trans_desc->identity_fingerprint.get() != NULL);
}
// Test that an answer can't be created if cryptos are required but the offer is
// unsecure.
TEST_F(MediaSessionDescriptionFactoryTest, TestSecureAnswerToUnsecureOffer) {
MediaSessionOptions options = CreatePlanBMediaSessionOptions();
f1_.set_secure(SEC_DISABLED);
tdf1_.set_secure(SEC_DISABLED);
f2_.set_secure(SEC_REQUIRED);
tdf1_.set_secure(SEC_ENABLED);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(options, NULL);
ASSERT_TRUE(offer.get() != NULL);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), options, NULL);
EXPECT_TRUE(answer.get() == NULL);
}
// Test that we accept a DTLS offer without SDES and create an appropriate
// answer.
TEST_F(MediaSessionDescriptionFactoryTest, TestCryptoOfferDtlsButNotSdes) {
f1_.set_secure(SEC_DISABLED);
f2_.set_secure(SEC_ENABLED);
tdf1_.set_secure(SEC_ENABLED);
tdf2_.set_secure(SEC_ENABLED);
MediaSessionOptions options;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly,
&options);
// Generate an offer with DTLS but without SDES.
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(options, NULL);
ASSERT_TRUE(offer.get() != NULL);
const AudioContentDescription* audio_offer =
GetFirstAudioContentDescription(offer.get());
ASSERT_TRUE(audio_offer->cryptos().empty());
const VideoContentDescription* video_offer =
GetFirstVideoContentDescription(offer.get());
ASSERT_TRUE(video_offer->cryptos().empty());
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
const RtpDataContentDescription* data_offer =
GetFirstRtpDataContentDescription(offer.get());
ASSERT_TRUE(data_offer->cryptos().empty());
const cricket::TransportDescription* audio_offer_trans_desc =
offer->GetTransportDescriptionByName("audio");
ASSERT_TRUE(audio_offer_trans_desc->identity_fingerprint.get() != NULL);
const cricket::TransportDescription* video_offer_trans_desc =
offer->GetTransportDescriptionByName("video");
ASSERT_TRUE(video_offer_trans_desc->identity_fingerprint.get() != NULL);
const cricket::TransportDescription* data_offer_trans_desc =
offer->GetTransportDescriptionByName("data");
ASSERT_TRUE(data_offer_trans_desc->identity_fingerprint.get() != NULL);
// Generate an answer with DTLS.
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), options, NULL);
ASSERT_TRUE(answer.get() != NULL);
const cricket::TransportDescription* audio_answer_trans_desc =
answer->GetTransportDescriptionByName("audio");
EXPECT_TRUE(audio_answer_trans_desc->identity_fingerprint.get() != NULL);
const cricket::TransportDescription* video_answer_trans_desc =
answer->GetTransportDescriptionByName("video");
EXPECT_TRUE(video_answer_trans_desc->identity_fingerprint.get() != NULL);
const cricket::TransportDescription* data_answer_trans_desc =
answer->GetTransportDescriptionByName("data");
EXPECT_TRUE(data_answer_trans_desc->identity_fingerprint.get() != NULL);
}
// Verifies if vad_enabled option is set to false, CN codecs are not present in
// offer or answer.
TEST_F(MediaSessionDescriptionFactoryTest, TestVADEnableOption) {
MediaSessionOptions options;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(options, NULL);
ASSERT_TRUE(offer.get() != NULL);
const ContentInfo* audio_content = offer->GetContentByName("audio");
EXPECT_FALSE(VerifyNoCNCodecs(audio_content));
options.vad_enabled = false;
offer = f1_.CreateOffer(options, NULL);
ASSERT_TRUE(offer.get() != NULL);
audio_content = offer->GetContentByName("audio");
EXPECT_TRUE(VerifyNoCNCodecs(audio_content));
std::unique_ptr<SessionDescription> answer =
f1_.CreateAnswer(offer.get(), options, NULL);
ASSERT_TRUE(answer.get() != NULL);
audio_content = answer->GetContentByName("audio");
EXPECT_TRUE(VerifyNoCNCodecs(audio_content));
}
// Test that the generated MIDs match the existing offer.
TEST_F(MediaSessionDescriptionFactoryTest, TestMIDsMatchesExistingOffer) {
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio_modified",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video_modified",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
opts.data_channel_type = cricket::DCT_SCTP;
AddMediaDescriptionOptions(MEDIA_TYPE_DATA, "data_modified",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
// Create offer.
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
std::unique_ptr<SessionDescription> updated_offer(
f1_.CreateOffer(opts, offer.get()));
const ContentInfo* audio_content = GetFirstAudioContent(updated_offer.get());
const ContentInfo* video_content = GetFirstVideoContent(updated_offer.get());
const ContentInfo* data_content = GetFirstDataContent(updated_offer.get());
ASSERT_TRUE(audio_content != nullptr);
ASSERT_TRUE(video_content != nullptr);
ASSERT_TRUE(data_content != nullptr);
EXPECT_EQ("audio_modified", audio_content->name);
EXPECT_EQ("video_modified", video_content->name);
EXPECT_EQ("data_modified", data_content->name);
}
// The following tests verify that the unified plan SDP is supported.
// Test that we can create an offer with multiple media sections of same media
// type.
TEST_F(MediaSessionDescriptionFactoryTest,
CreateOfferWithMultipleAVMediaSections) {
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio_1",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
AttachSenderToMediaDescriptionOptions(
"audio_1", MEDIA_TYPE_AUDIO, kAudioTrack1, {kMediaStream1}, 1, &opts);
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video_1",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
AttachSenderToMediaDescriptionOptions(
"video_1", MEDIA_TYPE_VIDEO, kVideoTrack1, {kMediaStream1}, 1, &opts);
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio_2",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
AttachSenderToMediaDescriptionOptions(
"audio_2", MEDIA_TYPE_AUDIO, kAudioTrack2, {kMediaStream2}, 1, &opts);
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video_2",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
AttachSenderToMediaDescriptionOptions(
"video_2", MEDIA_TYPE_VIDEO, kVideoTrack2, {kMediaStream2}, 1, &opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
ASSERT_TRUE(offer);
ASSERT_EQ(4u, offer->contents().size());
EXPECT_FALSE(offer->contents()[0].rejected);
const AudioContentDescription* acd =
offer->contents()[0].media_description()->as_audio();
ASSERT_EQ(1u, acd->streams().size());
EXPECT_EQ(kAudioTrack1, acd->streams()[0].id);
EXPECT_EQ(RtpTransceiverDirection::kSendRecv, acd->direction());
EXPECT_FALSE(offer->contents()[1].rejected);
const VideoContentDescription* vcd =
offer->contents()[1].media_description()->as_video();
ASSERT_EQ(1u, vcd->streams().size());
EXPECT_EQ(kVideoTrack1, vcd->streams()[0].id);
EXPECT_EQ(RtpTransceiverDirection::kSendRecv, vcd->direction());
EXPECT_FALSE(offer->contents()[2].rejected);
acd = offer->contents()[2].media_description()->as_audio();
ASSERT_EQ(1u, acd->streams().size());
EXPECT_EQ(kAudioTrack2, acd->streams()[0].id);
EXPECT_EQ(RtpTransceiverDirection::kSendRecv, acd->direction());
EXPECT_FALSE(offer->contents()[3].rejected);
vcd = offer->contents()[3].media_description()->as_video();
ASSERT_EQ(1u, vcd->streams().size());
EXPECT_EQ(kVideoTrack2, vcd->streams()[0].id);
EXPECT_EQ(RtpTransceiverDirection::kSendRecv, vcd->direction());
}
// Test that we can create an answer with multiple media sections of same media
// type.
TEST_F(MediaSessionDescriptionFactoryTest,
CreateAnswerWithMultipleAVMediaSections) {
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio_1",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
AttachSenderToMediaDescriptionOptions(
"audio_1", MEDIA_TYPE_AUDIO, kAudioTrack1, {kMediaStream1}, 1, &opts);
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video_1",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
AttachSenderToMediaDescriptionOptions(
"video_1", MEDIA_TYPE_VIDEO, kVideoTrack1, {kMediaStream1}, 1, &opts);
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio_2",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
AttachSenderToMediaDescriptionOptions(
"audio_2", MEDIA_TYPE_AUDIO, kAudioTrack2, {kMediaStream2}, 1, &opts);
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video_2",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
AttachSenderToMediaDescriptionOptions(
"video_2", MEDIA_TYPE_VIDEO, kVideoTrack2, {kMediaStream2}, 1, &opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
ASSERT_TRUE(offer);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, nullptr);
ASSERT_EQ(4u, answer->contents().size());
EXPECT_FALSE(answer->contents()[0].rejected);
const AudioContentDescription* acd =
answer->contents()[0].media_description()->as_audio();
ASSERT_EQ(1u, acd->streams().size());
EXPECT_EQ(kAudioTrack1, acd->streams()[0].id);
EXPECT_EQ(RtpTransceiverDirection::kSendRecv, acd->direction());
EXPECT_FALSE(answer->contents()[1].rejected);
const VideoContentDescription* vcd =
answer->contents()[1].media_description()->as_video();
ASSERT_EQ(1u, vcd->streams().size());
EXPECT_EQ(kVideoTrack1, vcd->streams()[0].id);
EXPECT_EQ(RtpTransceiverDirection::kSendRecv, vcd->direction());
EXPECT_FALSE(answer->contents()[2].rejected);
acd = answer->contents()[2].media_description()->as_audio();
ASSERT_EQ(1u, acd->streams().size());
EXPECT_EQ(kAudioTrack2, acd->streams()[0].id);
EXPECT_EQ(RtpTransceiverDirection::kSendRecv, acd->direction());
EXPECT_FALSE(answer->contents()[3].rejected);
vcd = answer->contents()[3].media_description()->as_video();
ASSERT_EQ(1u, vcd->streams().size());
EXPECT_EQ(kVideoTrack2, vcd->streams()[0].id);
EXPECT_EQ(RtpTransceiverDirection::kSendRecv, vcd->direction());
}
// Test that the media section will be rejected in offer if the corresponding
// MediaDescriptionOptions is stopped by the offerer.
TEST_F(MediaSessionDescriptionFactoryTest,
CreateOfferWithMediaSectionStoppedByOfferer) {
// Create an offer with two audio sections and one of them is stopped.
MediaSessionOptions offer_opts;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio1",
RtpTransceiverDirection::kSendRecv, kActive,
&offer_opts);
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio2",
RtpTransceiverDirection::kInactive, kStopped,
&offer_opts);
std::unique_ptr<SessionDescription> offer =
f1_.CreateOffer(offer_opts, nullptr);
ASSERT_TRUE(offer);
ASSERT_EQ(2u, offer->contents().size());
EXPECT_FALSE(offer->contents()[0].rejected);
EXPECT_TRUE(offer->contents()[1].rejected);
}
// Test that the media section will be rejected in answer if the corresponding
// MediaDescriptionOptions is stopped by the offerer.
TEST_F(MediaSessionDescriptionFactoryTest,
CreateAnswerWithMediaSectionStoppedByOfferer) {
// Create an offer with two audio sections and one of them is stopped.
MediaSessionOptions offer_opts;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio1",
RtpTransceiverDirection::kSendRecv, kActive,
&offer_opts);
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio2",
RtpTransceiverDirection::kInactive, kStopped,
&offer_opts);
std::unique_ptr<SessionDescription> offer =
f1_.CreateOffer(offer_opts, nullptr);
ASSERT_TRUE(offer);
ASSERT_EQ(2u, offer->contents().size());
EXPECT_FALSE(offer->contents()[0].rejected);
EXPECT_TRUE(offer->contents()[1].rejected);
// Create an answer based on the offer.
MediaSessionOptions answer_opts;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio1",
RtpTransceiverDirection::kSendRecv, kActive,
&answer_opts);
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio2",
RtpTransceiverDirection::kSendRecv, kActive,
&answer_opts);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), answer_opts, nullptr);
ASSERT_EQ(2u, answer->contents().size());
EXPECT_FALSE(answer->contents()[0].rejected);
EXPECT_TRUE(answer->contents()[1].rejected);
}
// Test that the media section will be rejected in answer if the corresponding
// MediaDescriptionOptions is stopped by the answerer.
TEST_F(MediaSessionDescriptionFactoryTest,
CreateAnswerWithMediaSectionRejectedByAnswerer) {
// Create an offer with two audio sections.
MediaSessionOptions offer_opts;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio1",
RtpTransceiverDirection::kSendRecv, kActive,
&offer_opts);
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio2",
RtpTransceiverDirection::kSendRecv, kActive,
&offer_opts);
std::unique_ptr<SessionDescription> offer =
f1_.CreateOffer(offer_opts, nullptr);
ASSERT_TRUE(offer);
ASSERT_EQ(2u, offer->contents().size());
ASSERT_FALSE(offer->contents()[0].rejected);
ASSERT_FALSE(offer->contents()[1].rejected);
// The answerer rejects one of the audio sections.
MediaSessionOptions answer_opts;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio1",
RtpTransceiverDirection::kSendRecv, kActive,
&answer_opts);
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio2",
RtpTransceiverDirection::kInactive, kStopped,
&answer_opts);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), answer_opts, nullptr);
ASSERT_EQ(2u, answer->contents().size());
EXPECT_FALSE(answer->contents()[0].rejected);
EXPECT_TRUE(answer->contents()[1].rejected);
// The TransportInfo of the rejected m= section is expected to be added in the
// answer.
EXPECT_EQ(offer->transport_infos().size(), answer->transport_infos().size());
}
// Test the generated media sections has the same order of the
// corresponding MediaDescriptionOptions.
TEST_F(MediaSessionDescriptionFactoryTest,
CreateOfferRespectsMediaDescriptionOptionsOrder) {
MediaSessionOptions opts;
// This tests put video section first because normally audio comes first by
// default.
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
ASSERT_TRUE(offer);
ASSERT_EQ(2u, offer->contents().size());
EXPECT_EQ("video", offer->contents()[0].name);
EXPECT_EQ("audio", offer->contents()[1].name);
}
// Test that different media sections using the same codec have same payload
// type.
TEST_F(MediaSessionDescriptionFactoryTest,
PayloadTypesSharedByMediaSectionsOfSameType) {
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video1",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video2",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
// Create an offer with two video sections using same codecs.
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
ASSERT_TRUE(offer);
ASSERT_EQ(2u, offer->contents().size());
const VideoContentDescription* vcd1 =
offer->contents()[0].media_description()->as_video();
const VideoContentDescription* vcd2 =
offer->contents()[1].media_description()->as_video();
EXPECT_EQ(vcd1->codecs().size(), vcd2->codecs().size());
ASSERT_EQ(2u, vcd1->codecs().size());
EXPECT_EQ(vcd1->codecs()[0].name, vcd2->codecs()[0].name);
EXPECT_EQ(vcd1->codecs()[0].id, vcd2->codecs()[0].id);
EXPECT_EQ(vcd1->codecs()[1].name, vcd2->codecs()[1].name);
EXPECT_EQ(vcd1->codecs()[1].id, vcd2->codecs()[1].id);
// Create answer and negotiate the codecs.
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, nullptr);
ASSERT_TRUE(answer);
ASSERT_EQ(2u, answer->contents().size());
vcd1 = answer->contents()[0].media_description()->as_video();
vcd2 = answer->contents()[1].media_description()->as_video();
EXPECT_EQ(vcd1->codecs().size(), vcd2->codecs().size());
ASSERT_EQ(1u, vcd1->codecs().size());
EXPECT_EQ(vcd1->codecs()[0].name, vcd2->codecs()[0].name);
EXPECT_EQ(vcd1->codecs()[0].id, vcd2->codecs()[0].id);
}
// Test that the codec preference order per media section is respected in
// subsequent offer.
TEST_F(MediaSessionDescriptionFactoryTest,
CreateOfferRespectsCodecPreferenceOrder) {
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video1",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video2",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
// Create an offer with two video sections using same codecs.
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
ASSERT_TRUE(offer);
ASSERT_EQ(2u, offer->contents().size());
VideoContentDescription* vcd1 =
offer->contents()[0].media_description()->as_video();
const VideoContentDescription* vcd2 =
offer->contents()[1].media_description()->as_video();
auto video_codecs = MAKE_VECTOR(kVideoCodecs1);
EXPECT_EQ(video_codecs, vcd1->codecs());
EXPECT_EQ(video_codecs, vcd2->codecs());
// Change the codec preference of the first video section and create a
// follow-up offer.
auto video_codecs_reverse = MAKE_VECTOR(kVideoCodecs1Reverse);
vcd1->set_codecs(video_codecs_reverse);
std::unique_ptr<SessionDescription> updated_offer(
f1_.CreateOffer(opts, offer.get()));
vcd1 = updated_offer->contents()[0].media_description()->as_video();
vcd2 = updated_offer->contents()[1].media_description()->as_video();
// The video codec preference order should be respected.
EXPECT_EQ(video_codecs_reverse, vcd1->codecs());
EXPECT_EQ(video_codecs, vcd2->codecs());
}
// Test that the codec preference order per media section is respected in
// the answer.
TEST_F(MediaSessionDescriptionFactoryTest,
CreateAnswerRespectsCodecPreferenceOrder) {
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video1",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video2",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
// Create an offer with two video sections using same codecs.
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
ASSERT_TRUE(offer);
ASSERT_EQ(2u, offer->contents().size());
VideoContentDescription* vcd1 =
offer->contents()[0].media_description()->as_video();
const VideoContentDescription* vcd2 =
offer->contents()[1].media_description()->as_video();
auto video_codecs = MAKE_VECTOR(kVideoCodecs1);
EXPECT_EQ(video_codecs, vcd1->codecs());
EXPECT_EQ(video_codecs, vcd2->codecs());
// Change the codec preference of the first video section and create an
// answer.
auto video_codecs_reverse = MAKE_VECTOR(kVideoCodecs1Reverse);
vcd1->set_codecs(video_codecs_reverse);
std::unique_ptr<SessionDescription> answer =
f1_.CreateAnswer(offer.get(), opts, nullptr);
vcd1 = answer->contents()[0].media_description()->as_video();
vcd2 = answer->contents()[1].media_description()->as_video();
// The video codec preference order should be respected.
EXPECT_EQ(video_codecs_reverse, vcd1->codecs());
EXPECT_EQ(video_codecs, vcd2->codecs());
}
// Test that when creating an answer, the codecs use local parameters instead of
// the remote ones.
TEST_F(MediaSessionDescriptionFactoryTest, CreateAnswerWithLocalCodecParams) {
const std::string audio_param_name = "audio_param";
const std::string audio_value1 = "audio_v1";
const std::string audio_value2 = "audio_v2";
const std::string video_param_name = "video_param";
const std::string video_value1 = "video_v1";
const std::string video_value2 = "video_v2";
auto audio_codecs1 = MAKE_VECTOR(kAudioCodecs1);
auto audio_codecs2 = MAKE_VECTOR(kAudioCodecs1);
auto video_codecs1 = MAKE_VECTOR(kVideoCodecs1);
auto video_codecs2 = MAKE_VECTOR(kVideoCodecs1);
// Set the parameters for codecs.
audio_codecs1[0].SetParam(audio_param_name, audio_value1);
video_codecs1[0].SetParam(video_param_name, video_value1);
audio_codecs2[0].SetParam(audio_param_name, audio_value2);
video_codecs2[0].SetParam(video_param_name, video_value2);
f1_.set_audio_codecs(audio_codecs1, audio_codecs1);
f1_.set_video_codecs(video_codecs1, video_codecs1);
f2_.set_audio_codecs(audio_codecs2, audio_codecs2);
f2_.set_video_codecs(video_codecs2, video_codecs2);
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
ASSERT_TRUE(offer);
auto offer_acd = offer->contents()[0].media_description()->as_audio();
auto offer_vcd = offer->contents()[1].media_description()->as_video();
std::string value;
EXPECT_TRUE(offer_acd->codecs()[0].GetParam(audio_param_name, &value));
EXPECT_EQ(audio_value1, value);
EXPECT_TRUE(offer_vcd->codecs()[0].GetParam(video_param_name, &value));
EXPECT_EQ(video_value1, value);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, nullptr);
ASSERT_TRUE(answer);
auto answer_acd = answer->contents()[0].media_description()->as_audio();
auto answer_vcd = answer->contents()[1].media_description()->as_video();
// Use the parameters from the local codecs.
EXPECT_TRUE(answer_acd->codecs()[0].GetParam(audio_param_name, &value));
EXPECT_EQ(audio_value2, value);
EXPECT_TRUE(answer_vcd->codecs()[0].GetParam(video_param_name, &value));
EXPECT_EQ(video_value2, value);
}
// Test that matching packetization-mode is part of the criteria for matching
// H264 codecs (in addition to profile-level-id). Previously, this was not the
// case, so the first H264 codec with the same profile-level-id would match and
// the payload type in the answer would be incorrect.
// This is a regression test for bugs.webrtc.org/8808
TEST_F(MediaSessionDescriptionFactoryTest,
H264MatchCriteriaIncludesPacketizationMode) {
// Create two H264 codecs with the same profile level ID and different
// packetization modes.
VideoCodec h264_pm0(96, "H264");
h264_pm0.params[cricket::kH264FmtpProfileLevelId] = "42c01f";
h264_pm0.params[cricket::kH264FmtpPacketizationMode] = "0";
VideoCodec h264_pm1(97, "H264");
h264_pm1.params[cricket::kH264FmtpProfileLevelId] = "42c01f";
h264_pm1.params[cricket::kH264FmtpPacketizationMode] = "1";
// Offerer will send both codecs, answerer should choose the one with matching
// packetization mode (and not the first one it sees).
f1_.set_video_codecs({h264_pm0, h264_pm1}, {h264_pm0, h264_pm1});
f2_.set_video_codecs({h264_pm1}, {h264_pm1});
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
ASSERT_TRUE(offer);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, nullptr);
ASSERT_TRUE(answer);
// Answer should have one negotiated codec with packetization-mode=1 using the
// offered payload type.
ASSERT_EQ(1u, answer->contents().size());
auto answer_vcd = answer->contents()[0].media_description()->as_video();
ASSERT_EQ(1u, answer_vcd->codecs().size());
auto answer_codec = answer_vcd->codecs()[0];
EXPECT_EQ(h264_pm1.id, answer_codec.id);
}
class MediaProtocolTest : public ::testing::TestWithParam<const char*> {
public:
MediaProtocolTest()
: f1_(&tdf1_, &ssrc_generator1), f2_(&tdf2_, &ssrc_generator2) {
f1_.set_audio_codecs(MAKE_VECTOR(kAudioCodecs1),
MAKE_VECTOR(kAudioCodecs1));
f1_.set_video_codecs(MAKE_VECTOR(kVideoCodecs1),
MAKE_VECTOR(kVideoCodecs1));
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
f1_.set_rtp_data_codecs(MAKE_VECTOR(kDataCodecs1));
f2_.set_audio_codecs(MAKE_VECTOR(kAudioCodecs2),
MAKE_VECTOR(kAudioCodecs2));
f2_.set_video_codecs(MAKE_VECTOR(kVideoCodecs2),
MAKE_VECTOR(kVideoCodecs2));
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
f2_.set_rtp_data_codecs(MAKE_VECTOR(kDataCodecs2));
f1_.set_secure(SEC_ENABLED);
f2_.set_secure(SEC_ENABLED);
tdf1_.set_certificate(rtc::RTCCertificate::Create(
std::unique_ptr<rtc::SSLIdentity>(new rtc::FakeSSLIdentity("id1"))));
tdf2_.set_certificate(rtc::RTCCertificate::Create(
std::unique_ptr<rtc::SSLIdentity>(new rtc::FakeSSLIdentity("id2"))));
tdf1_.set_secure(SEC_ENABLED);
tdf2_.set_secure(SEC_ENABLED);
}
protected:
MediaSessionDescriptionFactory f1_;
MediaSessionDescriptionFactory f2_;
TransportDescriptionFactory tdf1_;
TransportDescriptionFactory tdf2_;
UniqueRandomIdGenerator ssrc_generator1;
UniqueRandomIdGenerator ssrc_generator2;
};
TEST_P(MediaProtocolTest, TestAudioVideoAcceptance) {
MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, nullptr);
ASSERT_TRUE(offer.get() != nullptr);
// Set the protocol for all the contents.
for (auto& content : offer.get()->contents()) {
content.media_description()->set_protocol(GetParam());
}
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswer(offer.get(), opts, nullptr);
const ContentInfo* ac = answer->GetContentByName("audio");
const ContentInfo* vc = answer->GetContentByName("video");
ASSERT_TRUE(ac != nullptr);
ASSERT_TRUE(vc != nullptr);
EXPECT_FALSE(ac->rejected); // the offer is accepted
EXPECT_FALSE(vc->rejected);
const AudioContentDescription* acd = ac->media_description()->as_audio();
const VideoContentDescription* vcd = vc->media_description()->as_video();
EXPECT_EQ(GetParam(), acd->protocol());
EXPECT_EQ(GetParam(), vcd->protocol());
}
INSTANTIATE_TEST_SUITE_P(MediaProtocolPatternTest,
MediaProtocolTest,
::testing::ValuesIn(kMediaProtocols));
INSTANTIATE_TEST_SUITE_P(MediaProtocolDtlsPatternTest,
MediaProtocolTest,
::testing::ValuesIn(kMediaProtocolsDtls));
TEST_F(MediaSessionDescriptionFactoryTest, TestSetAudioCodecs) {
TransportDescriptionFactory tdf;
UniqueRandomIdGenerator ssrc_generator;
MediaSessionDescriptionFactory sf(&tdf, &ssrc_generator);
std::vector<AudioCodec> send_codecs = MAKE_VECTOR(kAudioCodecs1);
std::vector<AudioCodec> recv_codecs = MAKE_VECTOR(kAudioCodecs2);
// The merged list of codecs should contain any send codecs that are also
// nominally in the recieve codecs list. Payload types should be picked from
// the send codecs and a number-of-channels of 0 and 1 should be equivalent
// (set to 1). This equals what happens when the send codecs are used in an
// offer and the receive codecs are used in the following answer.
const std::vector<AudioCodec> sendrecv_codecs =
MAKE_VECTOR(kAudioCodecsAnswer);
const std::vector<AudioCodec> no_codecs;
RTC_CHECK_EQ(send_codecs[1].name, "iLBC")
<< "Please don't change shared test data!";
RTC_CHECK_EQ(recv_codecs[2].name, "iLBC")
<< "Please don't change shared test data!";
// Alter iLBC send codec to have zero channels, to test that that is handled
// properly.
send_codecs[1].channels = 0;
// Alther iLBC receive codec to be lowercase, to test that case conversions
// are handled properly.
recv_codecs[2].name = "ilbc";
// Test proper merge
sf.set_audio_codecs(send_codecs, recv_codecs);
EXPECT_EQ(send_codecs, sf.audio_send_codecs());
EXPECT_EQ(recv_codecs, sf.audio_recv_codecs());
EXPECT_EQ(sendrecv_codecs, sf.audio_sendrecv_codecs());
// Test empty send codecs list
sf.set_audio_codecs(no_codecs, recv_codecs);
EXPECT_EQ(no_codecs, sf.audio_send_codecs());
EXPECT_EQ(recv_codecs, sf.audio_recv_codecs());
EXPECT_EQ(no_codecs, sf.audio_sendrecv_codecs());
// Test empty recv codecs list
sf.set_audio_codecs(send_codecs, no_codecs);
EXPECT_EQ(send_codecs, sf.audio_send_codecs());
EXPECT_EQ(no_codecs, sf.audio_recv_codecs());
EXPECT_EQ(no_codecs, sf.audio_sendrecv_codecs());
// Test all empty codec lists
sf.set_audio_codecs(no_codecs, no_codecs);
EXPECT_EQ(no_codecs, sf.audio_send_codecs());
EXPECT_EQ(no_codecs, sf.audio_recv_codecs());
EXPECT_EQ(no_codecs, sf.audio_sendrecv_codecs());
}
namespace {
// Compare the two vectors of codecs ignoring the payload type.
template <class Codec>
bool CodecsMatch(const std::vector<Codec>& codecs1,
const std::vector<Codec>& codecs2) {
if (codecs1.size() != codecs2.size()) {
return false;
}
for (size_t i = 0; i < codecs1.size(); ++i) {
if (!codecs1[i].Matches(codecs2[i])) {
return false;
}
}
return true;
}
void TestAudioCodecsOffer(RtpTransceiverDirection direction) {
TransportDescriptionFactory tdf;
UniqueRandomIdGenerator ssrc_generator;
MediaSessionDescriptionFactory sf(&tdf, &ssrc_generator);
const std::vector<AudioCodec> send_codecs = MAKE_VECTOR(kAudioCodecs1);
const std::vector<AudioCodec> recv_codecs = MAKE_VECTOR(kAudioCodecs2);
const std::vector<AudioCodec> sendrecv_codecs =
MAKE_VECTOR(kAudioCodecsAnswer);
sf.set_audio_codecs(send_codecs, recv_codecs);
MediaSessionOptions opts;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio", direction, kActive,
&opts);
if (direction == RtpTransceiverDirection::kSendRecv ||
direction == RtpTransceiverDirection::kSendOnly) {
AttachSenderToMediaDescriptionOptions(
"audio", MEDIA_TYPE_AUDIO, kAudioTrack1, {kMediaStream1}, 1, &opts);
}
std::unique_ptr<SessionDescription> offer = sf.CreateOffer(opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
ContentInfo* ac = offer->GetContentByName("audio");
// If the factory didn't add any audio content to the offer, we cannot check
// that the codecs put in are right. This happens when we neither want to
// send nor receive audio. The checks are still in place if at some point
// we'd instead create an inactive stream.
if (ac) {
AudioContentDescription* acd = ac->media_description()->as_audio();
// sendrecv and inactive should both present lists as if the channel was
// to be used for sending and receiving. Inactive essentially means it
// might eventually be used anything, but we don't know more at this
// moment.
if (acd->direction() == RtpTransceiverDirection::kSendOnly) {
EXPECT_TRUE(CodecsMatch<AudioCodec>(send_codecs, acd->codecs()));
} else if (acd->direction() == RtpTransceiverDirection::kRecvOnly) {
EXPECT_TRUE(CodecsMatch<AudioCodec>(recv_codecs, acd->codecs()));
} else {
EXPECT_TRUE(CodecsMatch<AudioCodec>(sendrecv_codecs, acd->codecs()));
}
}
}
static const AudioCodec kOfferAnswerCodecs[] = {
AudioCodec(0, "codec0", 16000, -1, 1),
AudioCodec(1, "codec1", 8000, 13300, 1),
AudioCodec(2, "codec2", 8000, 64000, 1),
AudioCodec(3, "codec3", 8000, 64000, 1),
AudioCodec(4, "codec4", 8000, 0, 2),
AudioCodec(5, "codec5", 32000, 0, 1),
AudioCodec(6, "codec6", 48000, 0, 1)};
/* The codecs groups below are chosen as per the matrix below. The objective
* is to have different sets of codecs in the inputs, to get unique sets of
* codecs after negotiation, depending on offer and answer communication
* directions. One-way directions in the offer should either result in the
* opposite direction in the answer, or an inactive answer. Regardless, the
* choice of codecs should be as if the answer contained the opposite
* direction. Inactive offers should be treated as sendrecv/sendrecv.
*
* | Offer | Answer | Result
* codec|send recv sr | send recv sr | s/r r/s sr/s sr/r sr/sr
* 0 | x - - | - x - | x - - - -
* 1 | x x x | - x - | x - - x -
* 2 | - x - | x - - | - x - - -
* 3 | x x x | x - - | - x x - -
* 4 | - x - | x x x | - x - - -
* 5 | x - - | x x x | x - - - -
* 6 | x x x | x x x | x x x x x
*/
// Codecs used by offerer in the AudioCodecsAnswerTest
static const int kOfferSendCodecs[] = {0, 1, 3, 5, 6};
static const int kOfferRecvCodecs[] = {1, 2, 3, 4, 6};
// Codecs used in the answerer in the AudioCodecsAnswerTest. The order is
// jumbled to catch the answer not following the order in the offer.
static const int kAnswerSendCodecs[] = {6, 5, 2, 3, 4};
static const int kAnswerRecvCodecs[] = {6, 5, 4, 1, 0};
// The resulting sets of codecs in the answer in the AudioCodecsAnswerTest
static const int kResultSend_RecvCodecs[] = {0, 1, 5, 6};
static const int kResultRecv_SendCodecs[] = {2, 3, 4, 6};
static const int kResultSendrecv_SendCodecs[] = {3, 6};
static const int kResultSendrecv_RecvCodecs[] = {1, 6};
static const int kResultSendrecv_SendrecvCodecs[] = {6};
template <typename T, int IDXS>
std::vector<T> VectorFromIndices(const T* array, const int (&indices)[IDXS]) {
std::vector<T> out;
out.reserve(IDXS);
for (int idx : indices)
out.push_back(array[idx]);
return out;
}
void TestAudioCodecsAnswer(RtpTransceiverDirection offer_direction,
RtpTransceiverDirection answer_direction,
bool add_legacy_stream) {
TransportDescriptionFactory offer_tdf;
TransportDescriptionFactory answer_tdf;
UniqueRandomIdGenerator ssrc_generator1, ssrc_generator2;
MediaSessionDescriptionFactory offer_factory(&offer_tdf, &ssrc_generator1);
MediaSessionDescriptionFactory answer_factory(&answer_tdf, &ssrc_generator2);
offer_factory.set_audio_codecs(
VectorFromIndices(kOfferAnswerCodecs, kOfferSendCodecs),
VectorFromIndices(kOfferAnswerCodecs, kOfferRecvCodecs));
answer_factory.set_audio_codecs(
VectorFromIndices(kOfferAnswerCodecs, kAnswerSendCodecs),
VectorFromIndices(kOfferAnswerCodecs, kAnswerRecvCodecs));
MediaSessionOptions offer_opts;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio", offer_direction,
kActive, &offer_opts);
if (webrtc::RtpTransceiverDirectionHasSend(offer_direction)) {
AttachSenderToMediaDescriptionOptions("audio", MEDIA_TYPE_AUDIO,
kAudioTrack1, {kMediaStream1}, 1,
&offer_opts);
}
std::unique_ptr<SessionDescription> offer =
offer_factory.CreateOffer(offer_opts, NULL);
ASSERT_TRUE(offer.get() != NULL);
MediaSessionOptions answer_opts;
AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio", answer_direction,
kActive, &answer_opts);
if (webrtc::RtpTransceiverDirectionHasSend(answer_direction)) {
AttachSenderToMediaDescriptionOptions("audio", MEDIA_TYPE_AUDIO,
kAudioTrack1, {kMediaStream1}, 1,
&answer_opts);
}
std::unique_ptr<SessionDescription> answer =
answer_factory.CreateAnswer(offer.get(), answer_opts, NULL);
const ContentInfo* ac = answer->GetContentByName("audio");
// If the factory didn't add any audio content to the answer, we cannot
// check that the codecs put in are right. This happens when we neither want
// to send nor receive audio. The checks are still in place if at some point
// we'd instead create an inactive stream.
if (ac) {
ASSERT_EQ(MEDIA_TYPE_AUDIO, ac->media_description()->type());
const AudioContentDescription* acd = ac->media_description()->as_audio();
std::vector<AudioCodec> target_codecs;
// For offers with sendrecv or inactive, we should never reply with more
// codecs than offered, with these codec sets.
switch (offer_direction) {
case RtpTransceiverDirection::kInactive:
target_codecs = VectorFromIndices(kOfferAnswerCodecs,
kResultSendrecv_SendrecvCodecs);
break;
case RtpTransceiverDirection::kSendOnly:
target_codecs =
VectorFromIndices(kOfferAnswerCodecs, kResultSend_RecvCodecs);
break;
case RtpTransceiverDirection::kRecvOnly:
target_codecs =
VectorFromIndices(kOfferAnswerCodecs, kResultRecv_SendCodecs);
break;
case RtpTransceiverDirection::kSendRecv:
if (acd->direction() == RtpTransceiverDirection::kSendOnly) {
target_codecs =
VectorFromIndices(kOfferAnswerCodecs, kResultSendrecv_SendCodecs);
} else if (acd->direction() == RtpTransceiverDirection::kRecvOnly) {
target_codecs =
VectorFromIndices(kOfferAnswerCodecs, kResultSendrecv_RecvCodecs);
} else {
target_codecs = VectorFromIndices(kOfferAnswerCodecs,
kResultSendrecv_SendrecvCodecs);
}
break;
default:
RTC_NOTREACHED();
}
auto format_codecs = [](const std::vector<AudioCodec>& codecs) {
rtc::StringBuilder os;
bool first = true;
os << "{";
for (const auto& c : codecs) {
os << (first ? " " : ", ") << c.id;
first = false;
}
os << " }";
return os.Release();
};
EXPECT_TRUE(acd->codecs() == target_codecs)
<< "Expected: " << format_codecs(target_codecs)
<< ", got: " << format_codecs(acd->codecs()) << "; Offered: "
<< webrtc::RtpTransceiverDirectionToString(offer_direction)
<< ", answerer wants: "
<< webrtc::RtpTransceiverDirectionToString(answer_direction)
<< "; got: "
<< webrtc::RtpTransceiverDirectionToString(acd->direction());
} else {
EXPECT_EQ(offer_direction, RtpTransceiverDirection::kInactive)
<< "Only inactive offers are allowed to not generate any audio "
"content";
}
}
} // namespace
class AudioCodecsOfferTest
: public ::testing::TestWithParam<RtpTransceiverDirection> {};
TEST_P(AudioCodecsOfferTest, TestCodecsInOffer) {
TestAudioCodecsOffer(GetParam());
}
INSTANTIATE_TEST_SUITE_P(MediaSessionDescriptionFactoryTest,
AudioCodecsOfferTest,
::testing::Values(RtpTransceiverDirection::kSendOnly,
RtpTransceiverDirection::kRecvOnly,
RtpTransceiverDirection::kSendRecv,
RtpTransceiverDirection::kInactive));
class AudioCodecsAnswerTest
: public ::testing::TestWithParam<::testing::tuple<RtpTransceiverDirection,
RtpTransceiverDirection,
bool>> {};
TEST_P(AudioCodecsAnswerTest, TestCodecsInAnswer) {
TestAudioCodecsAnswer(::testing::get<0>(GetParam()),
::testing::get<1>(GetParam()),
::testing::get<2>(GetParam()));
}
INSTANTIATE_TEST_SUITE_P(
MediaSessionDescriptionFactoryTest,
AudioCodecsAnswerTest,
::testing::Combine(::testing::Values(RtpTransceiverDirection::kSendOnly,
RtpTransceiverDirection::kRecvOnly,
RtpTransceiverDirection::kSendRecv,
RtpTransceiverDirection::kInactive),
::testing::Values(RtpTransceiverDirection::kSendOnly,
RtpTransceiverDirection::kRecvOnly,
RtpTransceiverDirection::kSendRecv,
RtpTransceiverDirection::kInactive),
::testing::Bool()));