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Implement codec selection api

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The implementation covers the latest specification, but does not
support mixed-codec simulcast at the moment.
Changing codec for audio and video is supported.

Bug: webrtc:15064
Change-Id: I09082f39e2a7d54dd4a663a8a57bf9df5a851690
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311663
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40616}
This commit is contained in:
Florent Castelli 2023-04-12 10:45:07 +00:00 committed by WebRTC LUCI CQ
parent f45c2ceae0
commit 43a5dd86c2
21 changed files with 1109 additions and 131 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

5
api/rtp_parameters.h
View File

@ -520,6 +520,9 @@ struct RTC_EXPORT RtpEncodingParameters {
// https://w3c.github.io/webrtc-extensions/#dom-rtcrtpencodingparameters-adaptiveptime
bool adaptive_ptime = false;
// Allow changing the used codec for this encoding.
absl::optional<RtpCodec> codec;
bool operator==(const RtpEncodingParameters& o) const {
return ssrc == o.ssrc && bitrate_priority == o.bitrate_priority &&
network_priority == o.network_priority &&
@ -530,7 +533,7 @@ struct RTC_EXPORT RtpEncodingParameters {
scale_resolution_down_by == o.scale_resolution_down_by &&
active == o.active && rid == o.rid &&
adaptive_ptime == o.adaptive_ptime &&
requested_resolution == o.requested_resolution;
requested_resolution == o.requested_resolution && codec == o.codec;
}
bool operator!=(const RtpEncodingParameters& o) const {
return !(*this == o);

3
media/base/codec.cc
View File

@ -211,8 +211,7 @@ bool Codec::Matches(const Codec& codec,
return matches_id && matches_type_specific();
}
bool Codec::MatchesCapability(
const webrtc::RtpCodecCapability& codec_capability) const {
bool Codec::MatchesRtpCodec(const webrtc::RtpCodec& codec_capability) const {
webrtc::RtpCodecParameters codec_parameters = ToCodecParameters();
return codec_parameters.name == codec_capability.name &&

2
media/base/codec.h
View File

@ -114,7 +114,7 @@ struct RTC_EXPORT Codec {
// H264 levels are not compared.
bool Matches(const Codec& codec,
const webrtc::FieldTrialsView* field_trials = nullptr) const;
bool MatchesCapability(const webrtc::RtpCodecCapability& capability) const;
bool MatchesRtpCodec(const webrtc::RtpCodec& capability) const;
// Find the parameter for `name` and write the value to `out`.
bool GetParam(const std::string& name, std::string* out) const;

66
media/base/media_engine.cc
View File

@ -67,19 +67,37 @@ std::vector<webrtc::RtpExtension> GetDefaultEnabledRtpHeaderExtensions(
webrtc::RTCError CheckScalabilityModeValues(
const webrtc::RtpParameters& rtp_parameters,
rtc::ArrayView<cricket::VideoCodec> codecs) {
rtc::ArrayView<cricket::Codec> codec_preferences,
absl::optional<cricket::Codec> send_codec) {
using webrtc::RTCErrorType;
if (codecs.empty()) {
if (codec_preferences.empty()) {
// This is an audio sender or an extra check in the stack where the codec
// list is not available and we can't check the scalability_mode values.
return webrtc::RTCError::OK();
}
for (size_t i = 0; i < rtp_parameters.encodings.size(); ++i) {
if (rtp_parameters.encodings[i].codec) {
bool codecFound = false;
for (const cricket::VideoCodec& codec : codec_preferences) {
if (codec.MatchesRtpCodec(*rtp_parameters.encodings[i].codec)) {
codecFound = true;
send_codec = codec;
break;
}
}
if (!codecFound) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_MODIFICATION,
"Attempted to use an unsupported codec for layer " +
std::to_string(i));
}
}
if (rtp_parameters.encodings[i].scalability_mode) {
if (!send_codec) {
bool scalabilityModeFound = false;
for (const cricket::VideoCodec& codec : codecs) {
for (const cricket::VideoCodec& codec : codec_preferences) {
for (const auto& scalability_mode : codec.scalability_modes) {
if (ScalabilityModeToString(scalability_mode) ==
*rtp_parameters.encodings[i].scalability_mode) {
@ -92,10 +110,27 @@ webrtc::RTCError CheckScalabilityModeValues(
}
if (!scalabilityModeFound) {
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_MODIFICATION,
"Attempted to set RtpParameters scalabilityMode "
"to an unsupported value for the current codecs.");
}
} else {
bool scalabilityModeFound = false;
for (const auto& scalability_mode : send_codec->scalability_modes) {
if (ScalabilityModeToString(scalability_mode) ==
*rtp_parameters.encodings[i].scalability_mode) {
scalabilityModeFound = true;
break;
}
}
if (!scalabilityModeFound) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_MODIFICATION,
"Attempted to set RtpParameters scalabilityMode "
"to an unsupported value for the current codecs.");
}
}
}
}
@ -104,7 +139,8 @@ webrtc::RTCError CheckScalabilityModeValues(
webrtc::RTCError CheckRtpParametersValues(
const webrtc::RtpParameters& rtp_parameters,
rtc::ArrayView<cricket::VideoCodec> codecs) {
rtc::ArrayView<cricket::Codec> codec_preferences,
absl::optional<cricket::Codec> send_codec) {
using webrtc::RTCErrorType;
for (size_t i = 0; i < rtp_parameters.encodings.size(); ++i) {
@ -151,22 +187,31 @@ webrtc::RTCError CheckRtpParametersValues(
"Attempted to set scale_resolution_down_by and "
"requested_resolution simultaniously.");
}
if (i > 0 && rtp_parameters.encodings[i - 1].codec !=
rtp_parameters.encodings[i].codec) {
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
"Attempted to use different codec values for "
"different encodings.");
}
}
return CheckScalabilityModeValues(rtp_parameters, codecs);
return CheckScalabilityModeValues(rtp_parameters, codec_preferences,
send_codec);
}
webrtc::RTCError CheckRtpParametersInvalidModificationAndValues(
const webrtc::RtpParameters& old_rtp_parameters,
const webrtc::RtpParameters& rtp_parameters) {
return CheckRtpParametersInvalidModificationAndValues(old_rtp_parameters,
rtp_parameters, {});
return CheckRtpParametersInvalidModificationAndValues(
old_rtp_parameters, rtp_parameters, {}, absl::nullopt);
}
webrtc::RTCError CheckRtpParametersInvalidModificationAndValues(
const webrtc::RtpParameters& old_rtp_parameters,
const webrtc::RtpParameters& rtp_parameters,
rtc::ArrayView<cricket::VideoCodec> codecs) {
rtc::ArrayView<cricket::Codec> codec_preferences,
absl::optional<cricket::Codec> send_codec) {
using webrtc::RTCErrorType;
if (rtp_parameters.encodings.size() != old_rtp_parameters.encodings.size()) {
LOG_AND_RETURN_ERROR(
@ -201,7 +246,8 @@ webrtc::RTCError CheckRtpParametersInvalidModificationAndValues(
"Attempted to set RtpParameters with modified SSRC");
}
return CheckRtpParametersValues(rtp_parameters, codecs);
return CheckRtpParametersValues(rtp_parameters, codec_preferences,
send_codec);
}
CompositeMediaEngine::CompositeMediaEngine(

9
media/base/media_engine.h
View File

@ -42,20 +42,23 @@ namespace cricket {
// least one video codec of the list. If the list is empty, no check is done.
webrtc::RTCError CheckScalabilityModeValues(
const webrtc::RtpParameters& new_parameters,
rtc::ArrayView<cricket::VideoCodec> codecs);
rtc::ArrayView<cricket::Codec> codec_preferences,
absl::optional<cricket::Codec> send_codec);
// Checks the parameters have valid and supported values, and checks parameters
// with CheckScalabilityModeValues().
webrtc::RTCError CheckRtpParametersValues(
const webrtc::RtpParameters& new_parameters,
rtc::ArrayView<cricket::VideoCodec> codecs);
rtc::ArrayView<cricket::Codec> codec_preferences,
absl::optional<cricket::Codec> send_codec);
// Checks that the immutable values have not changed in new_parameters and
// checks all parameters with CheckRtpParametersValues().
webrtc::RTCError CheckRtpParametersInvalidModificationAndValues(
const webrtc::RtpParameters& old_parameters,
const webrtc::RtpParameters& new_parameters,
rtc::ArrayView<cricket::VideoCodec> codecs);
rtc::ArrayView<cricket::Codec> codec_preferences,
absl::optional<cricket::Codec> send_codec);
// Checks that the immutable values have not changed in new_parameters and
// checks parameters (except SVC) with CheckRtpParametersValues(). It should

45
media/engine/webrtc_video_engine.cc
View File

@ -994,9 +994,36 @@ bool WebRtcVideoSendChannel::GetChangedSendParameters(
codec.flexfec_payload_type = -1;
}
absl::optional<VideoCodecSettings> force_codec;
if (!send_streams_.empty()) {
// Since we do not support mixed-codec simulcast yet,
// all send streams must have the same codec value.
auto rtp_parameters = send_streams_.begin()->second->GetRtpParameters();
if (rtp_parameters.encodings[0].codec) {
auto matched_codec =
absl::c_find_if(negotiated_codecs, [&](auto negotiated_codec) {
return negotiated_codec.codec.MatchesRtpCodec(
*rtp_parameters.encodings[0].codec);
});
if (matched_codec != negotiated_codecs.end()) {
force_codec = *matched_codec;
} else {
// The requested codec has been negotiated away, we clear it from the
// parameters.
for (auto& encoding : rtp_parameters.encodings) {
encoding.codec.reset();
}
send_streams_.begin()->second->SetRtpParameters(rtp_parameters,
nullptr);
}
}
}
if (negotiated_codecs_ != negotiated_codecs) {
if (negotiated_codecs.empty()) {
changed_params->send_codec = absl::nullopt;
} else if (force_codec) {
changed_params->send_codec = force_codec;
} else if (send_codec() != negotiated_codecs.front()) {
changed_params->send_codec = negotiated_codecs.front();
}
@ -1264,6 +1291,24 @@ webrtc::RTCError WebRtcVideoSendChannel::SetRtpSendParameters(
new_dscp = rtc::DSCP_AF41;
break;
}
// TODO(orphis): Support mixed-codec simulcast
if (parameters.encodings[0].codec && send_codec_ &&
!send_codec_->codec.MatchesRtpCodec(*parameters.encodings[0].codec)) {
RTC_LOG(LS_ERROR) << "Trying to change codec to "
<< parameters.encodings[0].codec->name;
auto matched_codec =
absl::c_find_if(negotiated_codecs_, [&](auto negotiated_codec) {
return negotiated_codec.codec.MatchesRtpCodec(
*parameters.encodings[0].codec);
});
RTC_CHECK(matched_codec != negotiated_codecs_.end());
ChangedSendParameters params;
params.send_codec = *matched_codec;
ApplyChangedParams(params);
}
SetPreferredDscp(new_dscp);
}

33
media/engine/webrtc_video_engine_unittest.cc
View File

@ -4836,6 +4836,39 @@ TEST_F(WebRtcVideoChannelTest, SetSendCodecsRejectsMaxLessThanMinBitrate) {
EXPECT_FALSE(send_channel_->SetSendParameters(send_parameters_));
}
TEST_F(WebRtcVideoChannelTest,
SetSendParametersRemovesSelectedCodecFromRtpParameters) {
EXPECT_TRUE(AddSendStream());
cricket::VideoSenderParameters parameters;
parameters.codecs.push_back(cricket::CreateVideoCodec(100, "VP8"));
parameters.codecs.push_back(cricket::CreateVideoCodec(100, "VP9"));
send_channel_->SetSendParameters(parameters);
webrtc::RtpParameters initial_params =
send_channel_->GetRtpSendParameters(last_ssrc_);
webrtc::RtpCodec vp9_rtp_codec;
vp9_rtp_codec.name = "VP9";
vp9_rtp_codec.kind = cricket::MEDIA_TYPE_VIDEO;
vp9_rtp_codec.clock_rate = 90000;
initial_params.encodings[0].codec = vp9_rtp_codec;
// We should be able to set the params with the VP9 codec that has been
// negotiated.
EXPECT_TRUE(
send_channel_->SetRtpSendParameters(last_ssrc_, initial_params).ok());
parameters.codecs.clear();
parameters.codecs.push_back(cricket::CreateVideoCodec(100, "VP8"));
send_channel_->SetSendParameters(parameters);
// Since VP9 is no longer negotiated, the RTP parameters should not have a
// forced codec anymore.
webrtc::RtpParameters new_params =
send_channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(new_params.encodings[0].codec, absl::nullopt);
}
// Test that when both the codec-specific bitrate params and max_bandwidth_bps
// are present in the same send parameters, the settings are combined correctly.
TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithBitratesAndMaxSendBandwidth) {

73
media/engine/webrtc_voice_engine.cc
View File

@ -1271,7 +1271,34 @@ bool WebRtcVoiceSendChannel::SetSendParameters(
// TODO(pthatcher): Refactor this to be more clean now that we have
// all the information at once.
if (!SetSendCodecs(params.codecs)) {
// Finding if the RtpParameters force a specific codec
absl::optional<Codec> force_codec;
if (send_streams_.size() == 1) {
// Since audio simulcast is not supported, currently, only PlanB
// has multiple tracks and we don't care about getting the
// functionality working there properly.
auto rtp_parameters = send_streams_.begin()->second->rtp_parameters();
if (rtp_parameters.encodings[0].codec) {
auto matched_codec =
absl::c_find_if(params.codecs, [&](auto negotiated_codec) {
return negotiated_codec.MatchesRtpCodec(
*rtp_parameters.encodings[0].codec);
});
if (matched_codec != params.codecs.end()) {
force_codec = *matched_codec;
} else {
// The requested codec has been negotiated away, we clear it from the
// parameters.
for (auto& encoding : rtp_parameters.encodings) {
encoding.codec.reset();
}
send_streams_.begin()->second->SetRtpParameters(rtp_parameters,
nullptr);
}
}
}
if (!SetSendCodecs(params.codecs, force_codec)) {
return false;
}
@ -1319,13 +1346,14 @@ absl::optional<Codec> WebRtcVoiceSendChannel::GetSendCodec() const {
// codec settings from the given list of codecs (originally from SDP). Both send
// and receive streams may be reconfigured based on the new settings.
bool WebRtcVoiceSendChannel::SetSendCodecs(
const std::vector<AudioCodec>& codecs) {
const std::vector<Codec>& codecs,
absl::optional<Codec> preferred_codec) {
RTC_DCHECK_RUN_ON(worker_thread_);
dtmf_payload_type_ = absl::nullopt;
dtmf_payload_freq_ = -1;
// Validate supplied codecs list.
for (const AudioCodec& codec : codecs) {
for (const Codec& codec : codecs) {
// TODO(solenberg): Validate more aspects of input - that payload types
// don't overlap, remove redundant/unsupported codecs etc -
// the same way it is done for RtpHeaderExtensions.
@ -1339,8 +1367,8 @@ bool WebRtcVoiceSendChannel::SetSendCodecs(
// Find PT of telephone-event codec with lowest clockrate, as a fallback, in
// case we don't have a DTMF codec with a rate matching the send codec's, or
// if this function returns early.
std::vector<AudioCodec> dtmf_codecs;
for (const AudioCodec& codec : codecs) {
std::vector<Codec> dtmf_codecs;
for (const Codec& codec : codecs) {
if (IsCodec(codec, kDtmfCodecName)) {
dtmf_codecs.push_back(codec);
if (!dtmf_payload_type_ || codec.clockrate < dtmf_payload_freq_) {
@ -1356,10 +1384,11 @@ bool WebRtcVoiceSendChannel::SetSendCodecs(
webrtc::BitrateConstraints bitrate_config;
absl::optional<webrtc::AudioCodecInfo> voice_codec_info;
size_t send_codec_position = 0;
for (const AudioCodec& voice_codec : codecs) {
for (const Codec& voice_codec : codecs) {
if (!(IsCodec(voice_codec, kCnCodecName) ||
IsCodec(voice_codec, kDtmfCodecName) ||
IsCodec(voice_codec, kRedCodecName))) {
IsCodec(voice_codec, kRedCodecName)) &&
(!preferred_codec || preferred_codec->Matches(voice_codec))) {
webrtc::SdpAudioFormat format(voice_codec.name, voice_codec.clockrate,
voice_codec.channels, voice_codec.params);
@ -1391,7 +1420,7 @@ bool WebRtcVoiceSendChannel::SetSendCodecs(
if (voice_codec_info->allow_comfort_noise) {
// Loop through the codecs list again to find the CN codec.
// TODO(solenberg): Break out into a separate function?
for (const AudioCodec& cn_codec : codecs) {
for (const Codec& cn_codec : codecs) {
if (IsCodec(cn_codec, kCnCodecName) &&
cn_codec.clockrate == send_codec_spec->format.clockrate_hz &&
cn_codec.channels == voice_codec_info->num_channels) {
@ -1410,7 +1439,7 @@ bool WebRtcVoiceSendChannel::SetSendCodecs(
}
// Find the telephone-event PT exactly matching the preferred send codec.
for (const AudioCodec& dtmf_codec : dtmf_codecs) {
for (const Codec& dtmf_codec : dtmf_codecs) {
if (dtmf_codec.clockrate == send_codec_spec->format.clockrate_hz) {
dtmf_payload_type_ = dtmf_codec.id;
dtmf_payload_freq_ = dtmf_codec.clockrate;
@ -1421,15 +1450,15 @@ bool WebRtcVoiceSendChannel::SetSendCodecs(
// Loop through the codecs to find the RED codec that matches opus
// with respect to clockrate and number of channels.
size_t red_codec_position = 0;
for (const AudioCodec& red_codec : codecs) {
if (red_codec_position < send_codec_position &&
IsCodec(red_codec, kRedCodecName) &&
// RED codec needs to be negotiated before the actual codec they
// reference.
for (size_t i = 0; i < send_codec_position; ++i) {
const Codec& red_codec = codecs[i];
if (IsCodec(red_codec, kRedCodecName) &&
CheckRedParameters(red_codec, *send_codec_spec)) {
send_codec_spec->red_payload_type = red_codec.id;
break;
}
red_codec_position++;
}
if (send_codec_spec_ != send_codec_spec) {
@ -1839,6 +1868,22 @@ webrtc::RTCError WebRtcVoiceSendChannel::SetRtpSendParameters(
break;
}
SetPreferredDscp(new_dscp);
absl::optional<cricket::Codec> send_codec = GetSendCodec();
// TODO(orphis): Support mixed-codec simulcast
if (parameters.encodings[0].codec && send_codec &&
!send_codec->MatchesRtpCodec(*parameters.encodings[0].codec)) {
RTC_LOG(LS_ERROR) << "Trying to change codec to "
<< parameters.encodings[0].codec->name;
auto matched_codec =
absl::c_find_if(send_codecs_, [&](auto negotiated_codec) {
return negotiated_codec.MatchesRtpCodec(
*parameters.encodings[0].codec);
});
RTC_DCHECK(matched_codec != send_codecs_.end());
SetSendCodecs(send_codecs_, *matched_codec);
}
}
// TODO(minyue): The following legacy actions go into

3
media/engine/webrtc_voice_engine.h
View File

@ -290,7 +290,8 @@ class WebRtcVoiceSendChannel final : public MediaChannelUtil,
private:
bool SetOptions(const AudioOptions& options);
bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
bool SetSendCodecs(const std::vector<Codec>& codecs,
absl::optional<Codec> preferred_codec);
bool SetLocalSource(uint32_t ssrc, AudioSource* source);
bool MuteStream(uint32_t ssrc, bool mute);

37
media/engine/webrtc_voice_engine_unittest.cc
View File

@ -15,8 +15,10 @@
#include "absl/memory/memory.h"
#include "absl/strings/match.h"
#include "absl/types/optional.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/media_types.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
@ -1370,6 +1372,41 @@ TEST_P(WebRtcVoiceEngineTestFake, SetAndGetRtpSendParameters) {
EXPECT_EQ(initial_params, send_channel_->GetRtpSendParameters(kSsrcX));
}
// Test that we remove the codec from RTP parameters if it's not negotiated
// anymore.
TEST_P(WebRtcVoiceEngineTestFake,
SetSendParametersRemovesSelectedCodecFromRtpParameters) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs.push_back(kPcmuCodec);
SetSendParameters(parameters);
webrtc::RtpParameters initial_params =
send_channel_->GetRtpSendParameters(kSsrcX);
webrtc::RtpCodec opus_rtp_codec;
opus_rtp_codec.name = "opus";
opus_rtp_codec.kind = cricket::MEDIA_TYPE_AUDIO;
opus_rtp_codec.num_channels = 2;
opus_rtp_codec.clock_rate = 48000;
initial_params.encodings[0].codec = opus_rtp_codec;
// We should be able to set the params with the opus codec that has been
// negotiated.
EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, initial_params).ok());
parameters.codecs.clear();
parameters.codecs.push_back(kPcmuCodec);
SetSendParameters(parameters);
// Since Opus is no longer negotiated, the RTP parameters should not have a
// forced codec anymore.
webrtc::RtpParameters new_params =
send_channel_->GetRtpSendParameters(kSsrcX);
EXPECT_EQ(new_params.encodings[0].codec, absl::nullopt);
}
// Test that max_bitrate_bps in send stream config gets updated correctly when
// SetRtpSendParameters is called.
TEST_P(WebRtcVoiceEngineTestFake, SetRtpSendParameterUpdatesMaxBitrate) {

2
pc/BUILD.gn
View File

@ -2770,6 +2770,7 @@ if (rtc_include_tests && !build_with_chromium) {
"../api:media_stream_interface",
"../api:rtc_error",
"../api:rtc_stats_api",
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/audio:audio_mixer_api",
@ -2827,6 +2828,7 @@ if (rtc_include_tests && !build_with_chromium) {
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}

12
pc/peer_connection.cc
View File

@ -1106,14 +1106,20 @@ PeerConnection::AddTransceiver(
}
std::vector<cricket::VideoCodec> codecs;
// Gather the current codec capabilities to allow checking scalabilityMode and
// codec selection against supported values.
if (media_type == cricket::MEDIA_TYPE_VIDEO) {
// Gather the current codec capabilities to allow checking scalabilityMode
// against supported values.
codecs = context_->media_engine()->video().send_codecs(false);
} else {
codecs = context_->media_engine()->voice().send_codecs();
}
auto result = cricket::CheckRtpParametersValues(parameters, codecs);
auto result =
cricket::CheckRtpParametersValues(parameters, codecs, absl::nullopt);
if (!result.ok()) {
if (result.type() == RTCErrorType::INVALID_MODIFICATION) {
result.set_type(RTCErrorType::UNSUPPORTED_OPERATION);
}
LOG_AND_RETURN_ERROR(result.type(), result.message());
}

770
pc/peer_connection_encodings_integrationtest.cc
View File

@ -11,11 +11,17 @@
#include <string>
#include <vector>
#include "absl/strings/match.h"
#include "absl/types/optional.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/audio_codecs/opus_audio_decoder_factory.h"
#include "api/audio_codecs/opus_audio_encoder_factory.h"
#include "api/media_types.h"
#include "api/rtc_error.h"
#include "api/rtp_parameters.h"
#include "api/rtp_transceiver_direction.h"
#include "api/rtp_transceiver_interface.h"
#include "api/stats/rtcstats_objects.h"
#include "api/units/data_rate.h"
#include "api/video_codecs/video_decoder_factory_template.h"
@ -115,8 +121,8 @@ class PeerConnectionEncodingsIntegrationTest : public ::testing::Test {
rtc::scoped_refptr<PeerConnectionTestWrapper> CreatePc() {
auto pc_wrapper = rtc::make_ref_counted<PeerConnectionTestWrapper>(
"pc", &pss_, background_thread_.get(), background_thread_.get());
pc_wrapper->CreatePc({}, webrtc::CreateOpusAudioEncoderFactory(),
webrtc::CreateOpusAudioDecoderFactory());
pc_wrapper->CreatePc({}, webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::CreateBuiltinAudioDecoderFactory());
return pc_wrapper;
}
@ -183,8 +189,7 @@ class PeerConnectionEncodingsIntegrationTest : public ::testing::Test {
void NegotiateWithSimulcastTweaks(
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper,
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper,
std::vector<cricket::SimulcastLayer> init_layers) {
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper) {
// Create and set offer for `local_pc_wrapper`.
std::unique_ptr<SessionDescriptionInterface> offer =
CreateOffer(local_pc_wrapper);
@ -226,6 +231,29 @@ class PeerConnectionEncodingsIntegrationTest : public ::testing::Test {
return callback->report();
}
bool IsCodecIdDifferent(
rtc::scoped_refptr<PeerConnectionTestWrapper> pc_wrapper,
size_t index,
const std::string& codec_id) {
return IsCodecIdDifferentWithScalabilityMode(pc_wrapper, index, codec_id,
absl::nullopt);
}
bool IsCodecIdDifferentWithScalabilityMode(
rtc::scoped_refptr<PeerConnectionTestWrapper> pc_wrapper,
size_t index,
const std::string& codec_id,
absl::optional<std::string> wanted_scalability_mode) {
rtc::scoped_refptr<const RTCStatsReport> report = GetStats(pc_wrapper);
std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRtpStreamStats>();
return outbound_rtps[index]->codec_id.value() != codec_id &&
(!wanted_scalability_mode ||
(outbound_rtps[index]->scalability_mode.has_value() &&
outbound_rtps[index]->scalability_mode.value() ==
wanted_scalability_mode));
}
bool HasOutboundRtpBytesSent(
rtc::scoped_refptr<PeerConnectionTestWrapper> pc_wrapper,
size_t num_layers) {
@ -387,7 +415,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
GetCapabilitiesAndRestrictToCodec(local_pc_wrapper, "VP8");
transceiver->SetCodecPreferences(codecs);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper, layers);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
@ -434,7 +462,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
ASSERT_EQ(parameters.encodings.size(), 1u);
EXPECT_THAT(parameters.encodings[0].scalability_mode, Eq(absl::nullopt));
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper, layers);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
@ -488,7 +516,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
Optional(std::string("L3T3_KEY")));
// Negotiate, this results in VP8 being picked and fallback happening.
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper, layers);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
// `scalaiblity_mode` is assigned the fallback value "L1T2" which is different
@ -533,7 +561,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
GetCapabilitiesAndRestrictToCodec(local_pc_wrapper, "VP9");
transceiver->SetCodecPreferences(codecs);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper, layers);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
@ -585,7 +613,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
parameters.encodings[0].scale_resolution_down_by = 1;
EXPECT_TRUE(sender->SetParameters(parameters).ok());
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper, layers);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
@ -640,7 +668,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
parameters.encodings[2].active = false;
EXPECT_TRUE(sender->SetParameters(parameters).ok());
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper, layers);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
@ -683,7 +711,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
// The original negotiation triggers legacy SVC because we didn't specify
// any scalability mode.
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper, layers);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
@ -745,7 +773,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
parameters.encodings[2].active = false;
sender->SetParameters(parameters);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper, layers);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
@ -784,7 +812,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
parameters.encodings[2].active = false;
sender->SetParameters(parameters);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper, layers);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
@ -822,7 +850,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, VP9_TargetBitrate_LegacyL1T3) {
parameters.encodings[2].active = true;
sender->SetParameters(parameters);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper, layers);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
@ -872,7 +900,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, VP9_TargetBitrate_StandardL1T3) {
parameters.encodings[2].active = false;
sender->SetParameters(parameters);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper, layers);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
@ -896,6 +924,714 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, VP9_TargetBitrate_StandardL1T3) {
EXPECT_LE(target_bitrate.kbps(), kVp9ExpectedMaxBitrateForL1T3.kbps());
}
TEST_F(PeerConnectionEncodingsIntegrationTest,
EncodingParameterCodecIsEmptyWhenCreatedAudio) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
auto transceiver_or_error =
local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver =
transceiver_or_error.MoveValue();
webrtc::RtpParameters parameters =
audio_transceiver->sender()->GetParameters();
EXPECT_FALSE(parameters.encodings[0].codec.has_value());
}
TEST_F(PeerConnectionEncodingsIntegrationTest,
EncodingParameterCodecIsEmptyWhenCreatedVideo) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
auto transceiver_or_error =
local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver =
transceiver_or_error.MoveValue();
webrtc::RtpParameters parameters =
video_transceiver->sender()->GetParameters();
EXPECT_FALSE(parameters.encodings[0].codec.has_value());
}
TEST_F(PeerConnectionEncodingsIntegrationTest,
EncodingParameterCodecIsSetByAddTransceiverAudio) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
local_pc_wrapper->GetUserMedia(
/*audio=*/true, {}, /*video=*/false, {});
rtc::scoped_refptr<AudioTrackInterface> track = stream->GetAudioTracks()[0];
absl::optional<webrtc::RtpCodecCapability> pcmu =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
"pcmu");
ASSERT_TRUE(pcmu);
webrtc::RtpTransceiverInit init;
init.direction = webrtc::RtpTransceiverDirection::kSendOnly;
webrtc::RtpEncodingParameters encoding_parameters;
encoding_parameters.codec = pcmu;
init.send_encodings.push_back(encoding_parameters);
auto transceiver_or_error =
local_pc_wrapper->pc()->AddTransceiver(track, init);
rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver =
transceiver_or_error.MoveValue();
webrtc::RtpParameters parameters =
audio_transceiver->sender()->GetParameters();
EXPECT_EQ(*parameters.encodings[0].codec, *pcmu);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper);
std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRtpStreamStats>();
ASSERT_EQ(outbound_rtps.size(), 1u);
std::string codec_name = GetCurrentCodecMimeType(report, *outbound_rtps[0]);
EXPECT_STRCASEEQ(("audio/" + pcmu->name).c_str(), codec_name.c_str());
}
TEST_F(PeerConnectionEncodingsIntegrationTest,
EncodingParameterCodecIsSetByAddTransceiverVideo) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
local_pc_wrapper->GetUserMedia(
/*audio=*/false, {}, /*video=*/true, {.width = 1280, .height = 720});
rtc::scoped_refptr<VideoTrackInterface> track = stream->GetVideoTracks()[0];
absl::optional<webrtc::RtpCodecCapability> vp9 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp9");
ASSERT_TRUE(vp9);
webrtc::RtpTransceiverInit init;
init.direction = webrtc::RtpTransceiverDirection::kSendOnly;
webrtc::RtpEncodingParameters encoding_parameters;
encoding_parameters.codec = vp9;
encoding_parameters.scalability_mode = "L3T3";
init.send_encodings.push_back(encoding_parameters);
auto transceiver_or_error =
local_pc_wrapper->pc()->AddTransceiver(track, init);
rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver =
transceiver_or_error.MoveValue();
webrtc::RtpParameters parameters =
audio_transceiver->sender()->GetParameters();
EXPECT_EQ(*parameters.encodings[0].codec, *vp9);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
EXPECT_TRUE_WAIT(
IsCodecIdDifferentWithScalabilityMode(local_pc_wrapper, 0, "", "L3T3"),
kDefaultTimeout.ms());
rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper);
std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRtpStreamStats>();
ASSERT_EQ(outbound_rtps.size(), 1u);
std::string codec_name = GetCurrentCodecMimeType(report, *outbound_rtps[0]);
EXPECT_STRCASEEQ(("video/" + vp9->name).c_str(), codec_name.c_str());
EXPECT_EQ(outbound_rtps[0]->scalability_mode.value(), "L3T3");
}
TEST_F(PeerConnectionEncodingsIntegrationTest,
EncodingParameterCodecIsSetBySetParametersBeforeNegotiationAudio) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
local_pc_wrapper->GetUserMedia(
/*audio=*/true, {}, /*video=*/false, {});
rtc::scoped_refptr<AudioTrackInterface> track = stream->GetAudioTracks()[0];
absl::optional<webrtc::RtpCodecCapability> pcmu =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
"pcmu");
auto transceiver_or_error = local_pc_wrapper->pc()->AddTransceiver(track);
rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver =
transceiver_or_error.MoveValue();
webrtc::RtpParameters parameters =
audio_transceiver->sender()->GetParameters();
parameters.encodings[0].codec = pcmu;
EXPECT_TRUE(audio_transceiver->sender()->SetParameters(parameters).ok());
parameters = audio_transceiver->sender()->GetParameters();
EXPECT_EQ(parameters.encodings[0].codec, pcmu);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper);
std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRtpStreamStats>();
ASSERT_EQ(outbound_rtps.size(), 1u);
std::string codec_name = GetCurrentCodecMimeType(report, *outbound_rtps[0]);
EXPECT_STRCASEEQ(("audio/" + pcmu->name).c_str(), codec_name.c_str());
}
TEST_F(PeerConnectionEncodingsIntegrationTest,
EncodingParameterCodecIsSetBySetParametersAfterNegotiationAudio) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
local_pc_wrapper->GetUserMedia(
/*audio=*/true, {}, /*video=*/false, {});
rtc::scoped_refptr<AudioTrackInterface> track = stream->GetAudioTracks()[0];
absl::optional<webrtc::RtpCodecCapability> pcmu =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
"pcmu");
auto transceiver_or_error = local_pc_wrapper->pc()->AddTransceiver(track);
rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver =
transceiver_or_error.MoveValue();
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper);
std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRtpStreamStats>();
ASSERT_EQ(outbound_rtps.size(), 1u);
std::string codec_name = GetCurrentCodecMimeType(report, *outbound_rtps[0]);
EXPECT_STRCASENE(("audio/" + pcmu->name).c_str(), codec_name.c_str());
std::string last_codec_id = outbound_rtps[0]->codec_id.value();
webrtc::RtpParameters parameters =
audio_transceiver->sender()->GetParameters();
parameters.encodings[0].codec = pcmu;
EXPECT_TRUE(audio_transceiver->sender()->SetParameters(parameters).ok());
parameters = audio_transceiver->sender()->GetParameters();
EXPECT_EQ(parameters.encodings[0].codec, pcmu);
EXPECT_TRUE_WAIT(IsCodecIdDifferent(local_pc_wrapper, 0, last_codec_id),
kDefaultTimeout.ms());
report = GetStats(local_pc_wrapper);
outbound_rtps = report->GetStatsOfType<RTCOutboundRtpStreamStats>();
ASSERT_EQ(outbound_rtps.size(), 1u);
codec_name = GetCurrentCodecMimeType(report, *outbound_rtps[0]);
EXPECT_STRCASEEQ(("audio/" + pcmu->name).c_str(), codec_name.c_str());
}
TEST_F(PeerConnectionEncodingsIntegrationTest,
EncodingParameterCodecIsSetBySetParametersBeforeNegotiationVideo) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
local_pc_wrapper->GetUserMedia(
/*audio=*/false, {}, /*video=*/true, {.width = 1280, .height = 720});
rtc::scoped_refptr<VideoTrackInterface> track = stream->GetVideoTracks()[0];
absl::optional<webrtc::RtpCodecCapability> vp9 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp9");
auto transceiver_or_error = local_pc_wrapper->pc()->AddTransceiver(track);
rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver =
transceiver_or_error.MoveValue();
webrtc::RtpParameters parameters =
video_transceiver->sender()->GetParameters();
parameters.encodings[0].codec = vp9;
parameters.encodings[0].scalability_mode = "L3T3";
EXPECT_TRUE(video_transceiver->sender()->SetParameters(parameters).ok());
parameters = video_transceiver->sender()->GetParameters();
EXPECT_EQ(parameters.encodings[0].codec, vp9);
EXPECT_EQ(parameters.encodings[0].scalability_mode, "L3T3");
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
EXPECT_TRUE_WAIT(
IsCodecIdDifferentWithScalabilityMode(local_pc_wrapper, 0, "", "L3T3"),
kDefaultTimeout.ms());
rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper);
std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRtpStreamStats>();
ASSERT_EQ(outbound_rtps.size(), 1u);
std::string codec_name = GetCurrentCodecMimeType(report, *outbound_rtps[0]);
EXPECT_STRCASEEQ(("video/" + vp9->name).c_str(), codec_name.c_str());
EXPECT_EQ(outbound_rtps[0]->scalability_mode.ValueOrDefault(""), "L3T3");
}
TEST_F(PeerConnectionEncodingsIntegrationTest,
EncodingParameterCodecIsSetBySetParametersAfterNegotiationVideo) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
local_pc_wrapper->GetUserMedia(
/*audio=*/false, {}, /*video=*/true, {.width = 1280, .height = 720});
rtc::scoped_refptr<VideoTrackInterface> track = stream->GetVideoTracks()[0];
absl::optional<webrtc::RtpCodecCapability> vp9 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp9");
auto transceiver_or_error = local_pc_wrapper->pc()->AddTransceiver(track);
rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver =
transceiver_or_error.MoveValue();
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper);
std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRtpStreamStats>();
ASSERT_EQ(outbound_rtps.size(), 1u);
std::string codec_name = GetCurrentCodecMimeType(report, *outbound_rtps[0]);
EXPECT_STRCASENE(("audio/" + vp9->name).c_str(), codec_name.c_str());
std::string last_codec_id = outbound_rtps[0]->codec_id.value();
webrtc::RtpParameters parameters =
video_transceiver->sender()->GetParameters();
parameters.encodings[0].codec = vp9;
parameters.encodings[0].scalability_mode = "L3T3";
EXPECT_TRUE(video_transceiver->sender()->SetParameters(parameters).ok());
parameters = video_transceiver->sender()->GetParameters();
EXPECT_EQ(parameters.encodings[0].codec, vp9);
EXPECT_EQ(parameters.encodings[0].scalability_mode, "L3T3");
EXPECT_TRUE_WAIT(IsCodecIdDifferentWithScalabilityMode(local_pc_wrapper, 0,
last_codec_id, "L3T3"),
kDefaultTimeout.ms());
report = GetStats(local_pc_wrapper);
outbound_rtps = report->GetStatsOfType<RTCOutboundRtpStreamStats>();
ASSERT_EQ(outbound_rtps.size(), 1u);
codec_name = GetCurrentCodecMimeType(report, *outbound_rtps[0]);
EXPECT_STRCASEEQ(("video/" + vp9->name).c_str(), codec_name.c_str());
EXPECT_EQ(outbound_rtps[0]->scalability_mode.value(), "L3T3");
}
TEST_F(PeerConnectionEncodingsIntegrationTest,
AddTransceiverRejectsUnknownCodecParameterAudio) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
webrtc::RtpCodec dummy_codec;
dummy_codec.kind = cricket::MEDIA_TYPE_AUDIO;
dummy_codec.name = "FOOBAR";
dummy_codec.clock_rate = 90000;
dummy_codec.num_channels = 2;
webrtc::RtpTransceiverInit init;
init.direction = webrtc::RtpTransceiverDirection::kSendOnly;
webrtc::RtpEncodingParameters encoding_parameters;
encoding_parameters.codec = dummy_codec;
init.send_encodings.push_back(encoding_parameters);
auto transceiver_or_error =
local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init);
EXPECT_FALSE(transceiver_or_error.ok());
EXPECT_EQ(transceiver_or_error.error().type(),
RTCErrorType::UNSUPPORTED_OPERATION);
}
TEST_F(PeerConnectionEncodingsIntegrationTest,
AddTransceiverRejectsUnknownCodecParameterVideo) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
webrtc::RtpCodec dummy_codec;
dummy_codec.kind = cricket::MEDIA_TYPE_VIDEO;
dummy_codec.name = "FOOBAR";
dummy_codec.clock_rate = 90000;
webrtc::RtpTransceiverInit init;
init.direction = webrtc::RtpTransceiverDirection::kSendOnly;
webrtc::RtpEncodingParameters encoding_parameters;
encoding_parameters.codec = dummy_codec;
init.send_encodings.push_back(encoding_parameters);
auto transceiver_or_error =
local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init);
EXPECT_FALSE(transceiver_or_error.ok());
EXPECT_EQ(transceiver_or_error.error().type(),
RTCErrorType::UNSUPPORTED_OPERATION);
}
TEST_F(PeerConnectionEncodingsIntegrationTest,
SetParametersRejectsUnknownCodecParameterAudio) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
webrtc::RtpCodec dummy_codec;
dummy_codec.kind = cricket::MEDIA_TYPE_AUDIO;
dummy_codec.name = "FOOBAR";
dummy_codec.clock_rate = 90000;
dummy_codec.num_channels = 2;
auto transceiver_or_error =
local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
ASSERT_TRUE(transceiver_or_error.ok());
rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver =
transceiver_or_error.MoveValue();
webrtc::RtpParameters parameters =
audio_transceiver->sender()->GetParameters();
parameters.encodings[0].codec = dummy_codec;
RTCError error = audio_transceiver->sender()->SetParameters(parameters);
EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION);
}
TEST_F(PeerConnectionEncodingsIntegrationTest,
SetParametersRejectsUnknownCodecParameterVideo) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
webrtc::RtpCodec dummy_codec;
dummy_codec.kind = cricket::MEDIA_TYPE_VIDEO;
dummy_codec.name = "FOOBAR";
dummy_codec.clock_rate = 90000;
auto transceiver_or_error =
local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
ASSERT_TRUE(transceiver_or_error.ok());
rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver =
transceiver_or_error.MoveValue();
webrtc::RtpParameters parameters =
video_transceiver->sender()->GetParameters();
parameters.encodings[0].codec = dummy_codec;
RTCError error = video_transceiver->sender()->SetParameters(parameters);
EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION);
}
TEST_F(PeerConnectionEncodingsIntegrationTest,
SetParametersRejectsNonPreferredCodecParameterAudio) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
absl::optional<webrtc::RtpCodecCapability> opus =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
"opus");
ASSERT_TRUE(opus);
std::vector<webrtc::RtpCodecCapability> not_opus_codecs =
local_pc_wrapper->pc_factory()
->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO)
.codecs;
not_opus_codecs.erase(
std::remove_if(not_opus_codecs.begin(), not_opus_codecs.end(),
[&](const auto& codec) {
return absl::EqualsIgnoreCase(codec.name, opus->name);
}),
not_opus_codecs.end());
auto transceiver_or_error =
local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
ASSERT_TRUE(transceiver_or_error.ok());
rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver =
transceiver_or_error.MoveValue();
ASSERT_TRUE(audio_transceiver->SetCodecPreferences(not_opus_codecs).ok());
webrtc::RtpParameters parameters =
audio_transceiver->sender()->GetParameters();
parameters.encodings[0].codec = opus;
RTCError error = audio_transceiver->sender()->SetParameters(parameters);
EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION);
}
TEST_F(PeerConnectionEncodingsIntegrationTest,
SetParametersRejectsNonPreferredCodecParameterVideo) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
absl::optional<webrtc::RtpCodecCapability> vp8 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp8");
ASSERT_TRUE(vp8);
std::vector<webrtc::RtpCodecCapability> not_vp8_codecs =
local_pc_wrapper->pc_factory()
->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO)
.codecs;
not_vp8_codecs.erase(
std::remove_if(not_vp8_codecs.begin(), not_vp8_codecs.end(),
[&](const auto& codec) {
return absl::EqualsIgnoreCase(codec.name, vp8->name);
}),
not_vp8_codecs.end());
auto transceiver_or_error =
local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
ASSERT_TRUE(transceiver_or_error.ok());
rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver =
transceiver_or_error.MoveValue();
ASSERT_TRUE(video_transceiver->SetCodecPreferences(not_vp8_codecs).ok());
webrtc::RtpParameters parameters =
video_transceiver->sender()->GetParameters();
parameters.encodings[0].codec = vp8;
RTCError error = video_transceiver->sender()->SetParameters(parameters);
EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION);
}
TEST_F(PeerConnectionEncodingsIntegrationTest,
SetParametersRejectsNonNegotiatedCodecParameterAudio) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
absl::optional<webrtc::RtpCodecCapability> opus =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
"opus");
ASSERT_TRUE(opus);
std::vector<webrtc::RtpCodecCapability> not_opus_codecs =
local_pc_wrapper->pc_factory()
->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO)
.codecs;
not_opus_codecs.erase(
std::remove_if(not_opus_codecs.begin(), not_opus_codecs.end(),
[&](const auto& codec) {
return absl::EqualsIgnoreCase(codec.name, opus->name);
}),
not_opus_codecs.end());
auto transceiver_or_error =
local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
ASSERT_TRUE(transceiver_or_error.ok());
rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver =
transceiver_or_error.MoveValue();
ASSERT_TRUE(audio_transceiver->SetCodecPreferences(not_opus_codecs).ok());
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
webrtc::RtpParameters parameters =
audio_transceiver->sender()->GetParameters();
parameters.encodings[0].codec = opus;
RTCError error = audio_transceiver->sender()->SetParameters(parameters);
EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION);
}
TEST_F(PeerConnectionEncodingsIntegrationTest,
SetParametersRejectsNonNegotiatedCodecParameterVideo) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
absl::optional<webrtc::RtpCodecCapability> vp8 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp8");
ASSERT_TRUE(vp8);
std::vector<webrtc::RtpCodecCapability> not_vp8_codecs =
local_pc_wrapper->pc_factory()
->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO)
.codecs;
not_vp8_codecs.erase(
std::remove_if(not_vp8_codecs.begin(), not_vp8_codecs.end(),
[&](const auto& codec) {
return absl::EqualsIgnoreCase(codec.name, vp8->name);
}),
not_vp8_codecs.end());
auto transceiver_or_error =
local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
ASSERT_TRUE(transceiver_or_error.ok());
rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver =
transceiver_or_error.MoveValue();
ASSERT_TRUE(video_transceiver->SetCodecPreferences(not_vp8_codecs).ok());
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
webrtc::RtpParameters parameters =
video_transceiver->sender()->GetParameters();
parameters.encodings[0].codec = vp8;
RTCError error = video_transceiver->sender()->SetParameters(parameters);
EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION);
}
TEST_F(PeerConnectionEncodingsIntegrationTest,
EncodingParametersCodecRemovedAfterNegotiationAudio) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
absl::optional<webrtc::RtpCodecCapability> opus =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
"opus");
ASSERT_TRUE(opus);
std::vector<webrtc::RtpCodecCapability> not_opus_codecs =
local_pc_wrapper->pc_factory()
->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO)
.codecs;
not_opus_codecs.erase(
std::remove_if(not_opus_codecs.begin(), not_opus_codecs.end(),
[&](const auto& codec) {
return absl::EqualsIgnoreCase(codec.name, opus->name);
}),
not_opus_codecs.end());
webrtc::RtpTransceiverInit init;
init.direction = webrtc::RtpTransceiverDirection::kSendOnly;
webrtc::RtpEncodingParameters encoding_parameters;
encoding_parameters.codec = opus;
init.send_encodings.push_back(encoding_parameters);
auto transceiver_or_error =
local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init);
ASSERT_TRUE(transceiver_or_error.ok());
rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver =
transceiver_or_error.MoveValue();
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
webrtc::RtpParameters parameters =
audio_transceiver->sender()->GetParameters();
EXPECT_EQ(parameters.encodings[0].codec, opus);
ASSERT_TRUE(audio_transceiver->SetCodecPreferences(not_opus_codecs).ok());
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
parameters = audio_transceiver->sender()->GetParameters();
EXPECT_FALSE(parameters.encodings[0].codec);
}
TEST_F(PeerConnectionEncodingsIntegrationTest,
SetParametersRejectsScalabilityModeForSelectedCodec) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
absl::optional<webrtc::RtpCodecCapability> vp8 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp8");
ASSERT_TRUE(vp8);
webrtc::RtpTransceiverInit init;
init.direction = webrtc::RtpTransceiverDirection::kSendOnly;
webrtc::RtpEncodingParameters encoding_parameters;
encoding_parameters.codec = vp8;
encoding_parameters.scalability_mode = "L1T3";
init.send_encodings.push_back(encoding_parameters);
auto transceiver_or_error =
local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init);
ASSERT_TRUE(transceiver_or_error.ok());
rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver =
transceiver_or_error.MoveValue();
webrtc::RtpParameters parameters =
video_transceiver->sender()->GetParameters();
parameters.encodings[0].scalability_mode = "L3T3";
RTCError error = video_transceiver->sender()->SetParameters(parameters);
EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION);
}
TEST_F(PeerConnectionEncodingsIntegrationTest,
EncodingParametersCodecRemovedByNegotiationVideo) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
absl::optional<webrtc::RtpCodecCapability> vp8 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp8");
ASSERT_TRUE(vp8);
std::vector<webrtc::RtpCodecCapability> not_vp8_codecs =
local_pc_wrapper->pc_factory()
->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO)
.codecs;
not_vp8_codecs.erase(
std::remove_if(not_vp8_codecs.begin(), not_vp8_codecs.end(),
[&](const auto& codec) {
return absl::EqualsIgnoreCase(codec.name, vp8->name);
}),
not_vp8_codecs.end());
webrtc::RtpTransceiverInit init;
init.direction = webrtc::RtpTransceiverDirection::kSendOnly;
webrtc::RtpEncodingParameters encoding_parameters;
encoding_parameters.rid = "h";
encoding_parameters.codec = vp8;
encoding_parameters.scale_resolution_down_by = 2;
init.send_encodings.push_back(encoding_parameters);
encoding_parameters.rid = "f";
encoding_parameters.scale_resolution_down_by = 1;
init.send_encodings.push_back(encoding_parameters);
auto transceiver_or_error =
local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init);
ASSERT_TRUE(transceiver_or_error.ok());
rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver =
transceiver_or_error.MoveValue();
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
webrtc::RtpParameters parameters =
video_transceiver->sender()->GetParameters();
ASSERT_EQ(parameters.encodings.size(), 2u);
EXPECT_EQ(parameters.encodings[0].codec, vp8);
EXPECT_EQ(parameters.encodings[1].codec, vp8);
ASSERT_TRUE(video_transceiver->SetCodecPreferences(not_vp8_codecs).ok());
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
parameters = video_transceiver->sender()->GetParameters();
EXPECT_FALSE(parameters.encodings[0].codec);
EXPECT_FALSE(parameters.encodings[1].codec);
}
TEST_F(PeerConnectionEncodingsIntegrationTest,
AddTransceiverRejectsMixedCodecSimulcast) {
// Mixed Codec Simulcast is not yet supported, so we ensure that we reject
// such parameters.
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
absl::optional<webrtc::RtpCodecCapability> vp8 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp8");
ASSERT_TRUE(vp8);
absl::optional<webrtc::RtpCodecCapability> vp9 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp9");
webrtc::RtpTransceiverInit init;
init.direction = webrtc::RtpTransceiverDirection::kSendOnly;
webrtc::RtpEncodingParameters encoding_parameters;
encoding_parameters.rid = "h";
encoding_parameters.codec = vp8;
encoding_parameters.scale_resolution_down_by = 2;
init.send_encodings.push_back(encoding_parameters);
encoding_parameters.rid = "f";
encoding_parameters.codec = vp9;
encoding_parameters.scale_resolution_down_by = 1;
init.send_encodings.push_back(encoding_parameters);
auto transceiver_or_error =
local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init);
ASSERT_FALSE(transceiver_or_error.ok());
EXPECT_EQ(transceiver_or_error.error().type(),
RTCErrorType::UNSUPPORTED_OPERATION);
}
// Tests that use the standard path (specifying both `scalability_mode` and
// `scale_resolution_down_by`) should pass for all codecs.
class PeerConnectionEncodingsIntegrationParameterizedTest
@ -952,7 +1688,7 @@ TEST_P(PeerConnectionEncodingsIntegrationParameterizedTest, AllLayersInactive) {
parameters.encodings[2].active = false;
sender->SetParameters(parameters);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper, layers);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
@ -995,7 +1731,7 @@ TEST_P(PeerConnectionEncodingsIntegrationParameterizedTest, Simulcast) {
parameters.encodings[2].scale_resolution_down_by = 1;
sender->SetParameters(parameters);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper, layers);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();

2
pc/peer_connection_media_unittest.cc
View File

@ -1521,7 +1521,7 @@ bool CompareCodecs(const std::vector<webrtc::RtpCodecCapability>& capabilities,
absl::c_equal(
capabilities_no_rtx, codecs_no_rtx,
[](const webrtc::RtpCodecCapability& capability, const C& codec) {
return codec.MatchesCapability(capability);
return codec.MatchesRtpCodec(capability);
});
}

6
pc/peer_connection_svc_integrationtest.cc
View File

@ -139,7 +139,7 @@ TEST_F(PeerConnectionSVCIntegrationTest, SetParametersRejectsL3T3WithVP8) {
parameters.encodings[0].scalability_mode = "L3T3";
auto result = transceiver->sender()->SetParameters(parameters);
EXPECT_FALSE(result.ok());
EXPECT_EQ(result.type(), webrtc::RTCErrorType::UNSUPPORTED_OPERATION);
EXPECT_EQ(result.type(), webrtc::RTCErrorType::INVALID_MODIFICATION);
}
TEST_F(PeerConnectionSVCIntegrationTest,
@ -212,7 +212,7 @@ TEST_F(PeerConnectionSVCIntegrationTest,
parameters.encodings[0].scalability_mode = "L3T3";
auto result = transceiver->sender()->SetParameters(parameters);
EXPECT_FALSE(result.ok());
EXPECT_EQ(result.type(), webrtc::RTCErrorType::UNSUPPORTED_OPERATION);
EXPECT_EQ(result.type(), webrtc::RTCErrorType::INVALID_MODIFICATION);
}
TEST_F(PeerConnectionSVCIntegrationTest,
@ -237,7 +237,7 @@ TEST_F(PeerConnectionSVCIntegrationTest,
parameters.encodings[0].scalability_mode = "FOOBAR";
auto result = transceiver->sender()->SetParameters(parameters);
EXPECT_FALSE(result.ok());
EXPECT_EQ(result.type(), webrtc::RTCErrorType::UNSUPPORTED_OPERATION);
EXPECT_EQ(result.type(), webrtc::RTCErrorType::INVALID_MODIFICATION);
}
TEST_F(PeerConnectionSVCIntegrationTest, FallbackToL1Tx) {

45
pc/rtp_sender.cc
View File

@ -248,7 +248,7 @@ void RtpSenderBase::SetParametersInternal(const RtpParameters& parameters,
}
if (!media_channel_ || !ssrc_) {
auto result = cricket::CheckRtpParametersInvalidModificationAndValues(
init_parameters_, parameters, video_codec_preferences_);
init_parameters_, parameters, codec_preferences_, absl::nullopt);
if (result.ok()) {
init_parameters_ = parameters;
}
@ -272,7 +272,7 @@ void RtpSenderBase::SetParametersInternal(const RtpParameters& parameters,
return;
}
result = CheckSVCParameters(rtp_parameters);
result = CheckCodecParameters(rtp_parameters);
if (!result.ok()) {
webrtc::InvokeSetParametersCallback(callback, result);
return;
@ -299,7 +299,7 @@ RTCError RtpSenderBase::SetParametersInternalWithAllLayers(
}
if (!media_channel_ || !ssrc_) {
auto result = cricket::CheckRtpParametersInvalidModificationAndValues(
init_parameters_, parameters, video_codec_preferences_);
init_parameters_, parameters, codec_preferences_, absl::nullopt);
if (result.ok()) {
init_parameters_ = parameters;
}
@ -338,6 +338,26 @@ RTCError RtpSenderBase::CheckSetParameters(const RtpParameters& parameters) {
return RTCError::OK();
}
RTCError RtpSenderBase::CheckCodecParameters(const RtpParameters& parameters) {
absl::optional<cricket::Codec> send_codec = media_channel_->GetSendCodec();
// Match the currently used codec against the codec preferences to gather
// the SVC capabilities.
absl::optional<cricket::Codec> send_codec_with_svc_info;
if (send_codec && send_codec->type == cricket::Codec::Type::kVideo) {
auto codec_match =
absl::c_find_if(codec_preferences_, [&](auto& codec_preference) {
return send_codec->Matches(codec_preference);
});
if (codec_match != codec_preferences_.end()) {
send_codec_with_svc_info = *codec_match;
}
}
return cricket::CheckScalabilityModeValues(parameters, codec_preferences_,
send_codec_with_svc_info);
}
RTCError RtpSenderBase::SetParameters(const RtpParameters& parameters) {
RTC_DCHECK_RUN_ON(signaling_thread_);
TRACE_EVENT0("webrtc", "RtpSenderBase::SetParameters");
@ -876,23 +896,4 @@ void VideoRtpSender::ClearSend() {
[&] { video_media_channel()->SetVideoSend(ssrc_, nullptr, nullptr); });
}
RTCError VideoRtpSender::CheckSVCParameters(const RtpParameters& parameters) {
absl::optional<cricket::VideoCodec> send_codec =
video_media_channel()->GetSendCodec();
// Match the currently used codec against the codec preferences to gather
// the SVC capabilities.
std::vector<cricket::VideoCodec> codecs;
if (send_codec) {
for (const auto& codec_preference : video_codec_preferences_) {
if (send_codec->Matches(codec_preference)) {
codecs.push_back(codec_preference);
break;
}
}
}
return cricket::CheckScalabilityModeValues(parameters, codecs);
}
} // namespace webrtc

19
pc/rtp_sender.h
View File

@ -85,8 +85,8 @@ class RtpSenderInternal : public RtpSenderInterface {
virtual RTCError SetParametersInternalWithAllLayers(
const RtpParameters& parameters) = 0;
// Additional checks that are specific to the Sender type
virtual RTCError CheckSVCParameters(const RtpParameters& parameters) {
// Additional checks that are specific to the current codec settings
virtual RTCError CheckCodecParameters(const RtpParameters& parameters) {
return webrtc::RTCError::OK();
}
@ -104,8 +104,8 @@ class RtpSenderInternal : public RtpSenderInterface {
// Used by the owning transceiver to inform the sender on the currently
// selected codecs.
virtual void SetVideoCodecPreferences(
std::vector<cricket::VideoCodec> codec_preferences) = 0;
virtual void SetCodecPreferences(
std::vector<cricket::Codec> codec_preferences) = 0;
};
// Shared implementation for RtpSenderInternal interface.
@ -144,6 +144,7 @@ class RtpSenderBase : public RtpSenderInternal, public ObserverInterface {
SetParametersCallback callback = nullptr,
bool blocking = true) override;
RTCError CheckSetParameters(const RtpParameters& parameters);
RTCError CheckCodecParameters(const RtpParameters& parameters) override;
RtpParameters GetParametersInternalWithAllLayers() const override;
RTCError SetParametersInternalWithAllLayers(
const RtpParameters& parameters) override;
@ -222,9 +223,9 @@ class RtpSenderBase : public RtpSenderInternal, public ObserverInterface {
is_transceiver_stopped_ = true;
}
void SetVideoCodecPreferences(
std::vector<cricket::VideoCodec> codec_preferences) override {
video_codec_preferences_ = codec_preferences;
void SetCodecPreferences(
std::vector<cricket::Codec> codec_preferences) override {
codec_preferences_ = codec_preferences;
}
protected:
@ -262,7 +263,7 @@ class RtpSenderBase : public RtpSenderInternal, public ObserverInterface {
std::vector<std::string> stream_ids_;
RtpParameters init_parameters_;
std::vector<cricket::VideoCodec> video_codec_preferences_;
std::vector<cricket::Codec> codec_preferences_;
// TODO(tommi): `media_channel_` and several other member variables in this
// class (ssrc_, stopped_, etc) are accessed from more than one thread without
@ -422,8 +423,6 @@ class VideoRtpSender : public RtpSenderBase {
rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const override;
RTCError GenerateKeyFrame(const std::vector<std::string>& rids) override;
RTCError CheckSVCParameters(const RtpParameters& parameters) override;
protected:
VideoRtpSender(rtc::Thread* worker_thread,
const std::string& id,

56
pc/rtp_transceiver.cc
View File

@ -56,7 +56,7 @@ RTCError VerifyCodecPreferences(
codec.name != cricket::kFlexfecCodecName &&
absl::c_any_of(recv_codecs,
[&codec](const cricket::Codec& recv_codec) {
return recv_codec.MatchesCapability(codec);
return recv_codec.MatchesRtpCodec(codec);
});
})) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
@ -70,7 +70,7 @@ RTCError VerifyCodecPreferences(
codec.name != cricket::kFlexfecCodecName &&
absl::c_any_of(send_codecs,
[&codec](const cricket::Codec& send_codec) {
return send_codec.MatchesCapability(codec);
return send_codec.MatchesRtpCodec(codec);
});
})) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
@ -85,12 +85,12 @@ RTCError VerifyCodecPreferences(
for (const auto& codec_preference : codecs) {
bool is_recv_codec = absl::c_any_of(
recv_codecs, [&codec_preference](const cricket::Codec& codec) {
return codec.MatchesCapability(codec_preference);
return codec.MatchesRtpCodec(codec_preference);
});
bool is_send_codec = absl::c_any_of(
send_codecs, [&codec_preference](const cricket::Codec& codec) {
return codec.MatchesCapability(codec_preference);
return codec.MatchesRtpCodec(codec_preference);
});
if (!is_recv_codec && !is_send_codec) {
@ -126,7 +126,7 @@ std::vector<cricket::VideoCodec> MatchCodecPreferences(
for (const auto& codec_preference : codecs) {
for (const cricket::VideoCodec& send_codec : send_codecs) {
if (send_codec.MatchesCapability(codec_preference)) {
if (send_codec.MatchesRtpCodec(codec_preference)) {
result.push_back(send_codec);
}
}
@ -171,9 +171,10 @@ RtpTransceiver::RtpTransceiver(
RTC_DCHECK(media_type_ == cricket::MEDIA_TYPE_AUDIO ||
media_type_ == cricket::MEDIA_TYPE_VIDEO);
RTC_DCHECK_EQ(sender->media_type(), receiver->media_type());
if (sender->media_type() == cricket::MEDIA_TYPE_VIDEO)
sender->internal()->SetVideoCodecPreferences(
media_engine()->video().send_codecs(false));
sender->internal()->SetCodecPreferences(
sender->media_type() == cricket::MEDIA_TYPE_VIDEO
? media_engine()->video().send_codecs(false)
: media_engine()->voice().send_codecs());
senders_.push_back(sender);
receivers_.push_back(receiver);
}
@ -412,14 +413,15 @@ void RtpTransceiver::AddSender(
RTC_DCHECK(sender);
RTC_DCHECK_EQ(media_type(), sender->media_type());
RTC_DCHECK(!absl::c_linear_search(senders_, sender));
if (media_type() == cricket::MEDIA_TYPE_VIDEO) {
std::vector<cricket::VideoCodec> send_codecs =
media_engine()->video().send_codecs(false);
sender->internal()->SetVideoCodecPreferences(
std::vector<cricket::Codec> send_codecs =
media_type() == cricket::MEDIA_TYPE_VIDEO
? media_engine()->video().send_codecs(false)
: media_engine()->voice().send_codecs();
sender->internal()->SetCodecPreferences(
codec_preferences_.empty()
? send_codecs
: MatchCodecPreferences(codec_preferences_, send_codecs));
}
senders_.push_back(sender);
}
@ -668,9 +670,10 @@ RTCError RtpTransceiver::SetCodecPreferences(
// to codecs and abort these steps.
if (codec_capabilities.empty()) {
codec_preferences_.clear();
if (media_type() == cricket::MEDIA_TYPE_VIDEO)
senders_.front()->internal()->SetVideoCodecPreferences(
media_engine()->video().send_codecs(false));
senders_.front()->internal()->SetCodecPreferences(
media_type() == cricket::MEDIA_TYPE_VIDEO
? media_engine()->video().send_codecs(false)
: media_engine()->voice().send_codecs());
return RTCError::OK();
}
@ -683,24 +686,19 @@ RTCError RtpTransceiver::SetCodecPreferences(
// 6. to 8.
RTCError result;
std::vector<cricket::Codec> recv_codecs, send_codecs;
if (media_type_ == cricket::MEDIA_TYPE_AUDIO) {
result =
VerifyCodecPreferences(codecs, media_engine()->voice().send_codecs(),
media_engine()->voice().recv_codecs());
send_codecs = media_engine()->voice().send_codecs();
recv_codecs = media_engine()->voice().recv_codecs();
} else if (media_type_ == cricket::MEDIA_TYPE_VIDEO) {
std::vector<cricket::Codec> send_codecs =
media_engine()->video().send_codecs(context()->use_rtx());
result = VerifyCodecPreferences(
codecs, send_codecs,
media_engine()->video().recv_codecs(context()->use_rtx()));
send_codecs = media_engine()->video().send_codecs(context()->use_rtx());
recv_codecs = media_engine()->video().recv_codecs(context()->use_rtx());
}
result = VerifyCodecPreferences(codecs, send_codecs, recv_codecs);
if (result.ok()) {
senders_.front()->internal()->SetVideoCodecPreferences(
senders_.front()->internal()->SetCodecPreferences(
MatchCodecPreferences(codecs, send_codecs));
}
}
if (result.ok()) {
codec_preferences_ = codecs;
}

7
pc/test/mock_rtp_sender_internal.h
View File

@ -64,9 +64,12 @@ class MockRtpSenderInternal : public RtpSenderInternal {
SetParametersInternalWithAllLayers,
(const RtpParameters&),
(override));
MOCK_METHOD(RTCError, CheckSVCParameters, (const RtpParameters&), (override));
MOCK_METHOD(RTCError,
CheckCodecParameters,
(const RtpParameters&),
(override));
MOCK_METHOD(void,
SetVideoCodecPreferences,
SetCodecPreferences,
(std::vector<cricket::VideoCodec>),
(override));
MOCK_METHOD(rtc::scoped_refptr<DtmfSenderInterface>,

16
pc/test/peer_connection_test_wrapper.cc
View File

@ -17,9 +17,11 @@
#include <utility>
#include <vector>
#include "absl/strings/match.h"
#include "absl/types/optional.h"
#include "api/audio/audio_mixer.h"
#include "api/create_peerconnection_factory.h"
#include "api/media_types.h"
#include "api/sequence_checker.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_decoder_factory_template.h"
@ -201,6 +203,20 @@ PeerConnectionTestWrapper::CreateDataChannel(
return result.MoveValue();
}
absl::optional<webrtc::RtpCodecCapability>
PeerConnectionTestWrapper::FindFirstSendCodecWithName(
cricket::MediaType media_type,
const std::string& name) const {
std::vector<webrtc::RtpCodecCapability> codecs =
peer_connection_factory_->GetRtpSenderCapabilities(media_type).codecs;
for (const auto& codec : codecs) {
if (absl::EqualsIgnoreCase(codec.name, name)) {
return codec;
}
}
return absl::nullopt;
}
void PeerConnectionTestWrapper::WaitForNegotiation() {
EXPECT_TRUE_WAIT(!pending_negotiation_, kMaxWait);
}

5
pc/test/peer_connection_test_wrapper.h
View File

@ -23,6 +23,7 @@
#include "api/media_stream_interface.h"
#include "api/peer_connection_interface.h"
#include "api/rtc_error.h"
#include "api/rtp_parameters.h"
#include "api/rtp_receiver_interface.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
@ -64,6 +65,10 @@ class PeerConnectionTestWrapper
const std::string& label,
const webrtc::DataChannelInit& init);
absl::optional<webrtc::RtpCodecCapability> FindFirstSendCodecWithName(
cricket::MediaType media_type,
const std::string& name) const;
void WaitForNegotiation();
// Implements PeerConnectionObserver.