webrtc_m130/pc/channel_manager.h

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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.
*/
#ifndef PC_CHANNEL_MANAGER_H_
#define PC_CHANNEL_MANAGER_H_
#include <stdint.h>
#include <memory>
#include <string>
#include <vector>
#include "api/audio_options.h"
#include "api/crypto/crypto_options.h"
#include "api/transport/media/media_transport_config.h"
#include "call/call.h"
#include "media/base/codec.h"
#include "media/base/media_channel.h"
#include "media/base/media_config.h"
#include "media/base/media_engine.h"
#include "pc/channel.h"
#include "pc/rtp_transport_internal.h"
#include "pc/session_description.h"
#include "rtc_base/system/file_wrapper.h"
#include "rtc_base/thread.h"
namespace cricket {
// ChannelManager allows the MediaEngine to run on a separate thread, and takes
// care of marshalling calls between threads. It also creates and keeps track of
// voice and video channels; by doing so, it can temporarily pause all the
// channels when a new audio or video device is chosen. The voice and video
// channels are stored in separate vectors, to easily allow operations on just
// voice or just video channels.
// ChannelManager also allows the application to discover what devices it has
// using device manager.
class ChannelManager final {
public:
// Construct a ChannelManager with the specified media engine and data engine.
ChannelManager(std::unique_ptr<MediaEngineInterface> media_engine,
std::unique_ptr<DataEngineInterface> data_engine,
rtc::Thread* worker_thread,
rtc::Thread* network_thread);
~ChannelManager();
// Accessors for the worker thread, allowing it to be set after construction,
// but before Init. set_worker_thread will return false if called after Init.
rtc::Thread* worker_thread() const { return worker_thread_; }
bool set_worker_thread(rtc::Thread* thread) {
if (initialized_) {
return false;
}
worker_thread_ = thread;
return true;
}
rtc::Thread* network_thread() const { return network_thread_; }
bool set_network_thread(rtc::Thread* thread) {
if (initialized_) {
return false;
}
network_thread_ = thread;
return true;
}
MediaEngineInterface* media_engine() { return media_engine_.get(); }
// Retrieves the list of supported audio & video codec types.
// Can be called before starting the media engine.
void GetSupportedAudioSendCodecs(std::vector<AudioCodec>* codecs) const;
void GetSupportedAudioReceiveCodecs(std::vector<AudioCodec>* codecs) const;
Revert "Reland "Reland "Reland "Distinguish between send and receive codecs"""" This reverts commit 184ea66aed43161f05d80fbb74183a2efccca352. Reason for revert: Breaks downstream projects. TBR=steveanton@webrtc.org Original change's description: > Reland "Reland "Reland "Distinguish between send and receive codecs""" > > This reverts commit a104ceb0ceec0f95e199e6d6704f41ec88a51fc5. > > Reason for revert: Keep logic as is. > > Original change's description: > > Revert "Reland "Reland "Distinguish between send and receive codecs""" > > > > This reverts commit 9bac68c0cc4444b852416396f0e0f31ea66a9cfe. > > > > Reason for revert: Breaks perf test on iOS. > > > > Original change's description: > > > Reland "Reland "Distinguish between send and receive codecs"" > > > > > > This reverts commit 00a30873c415d717af8dcdf21c2df7fd4b6d1ed2. > > > > > > Reason for revert: Flaky test in Chromium fixed. > > > > > > Original change's description: > > > > Revert "Reland "Distinguish between send and receive codecs"" > > > > > > > > This reverts commit 133bf2bd28596aab5c7684e0ea3da99b1fece77f. > > > > > > > > Reason for revert: Breaks Chromium import due to flaky test in Chromium. > > > > > > > > Original change's description: > > > > > Reland "Distinguish between send and receive codecs" > > > > > > > > > > This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8. > > > > > > > > > > Reason for revert: Fixed negotiation of send-only clients. > > > > > > > > > > Original change's description: > > > > > > Revert "Distinguish between send and receive codecs" > > > > > > > > > > > > This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d. > > > > > > > > > > > > Reason for revert: breaks negotiation with send-only clients > > > > > > > > > > > > (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264] > > > > > > (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER) > > > > > > (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters. > > > > > > > > > > > > Original change's description: > > > > > > > Distinguish between send and receive codecs > > > > > > > > > > > > > > Even though send and receive codecs may be the same, they might have > > > > > > > different support in HW. Distinguish between send and receive codecs > > > > > > > to be able to keep track of which codecs have HW support. > > > > > > > > > > > > > > Bug: chromium:1029737 > > > > > > > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d > > > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763 > > > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > > > > > Cr-Commit-Position: refs/heads/master@{#30284} > > > > > > > > > > > > TBR=steveanton@webrtc.org,kron@webrtc.org > > > > > > > > > > > > Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa > > > > > > No-Presubmit: true > > > > > > No-Tree-Checks: true > > > > > > No-Try: true > > > > > > Bug: chromium:1029737 > > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420 > > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > > > > Commit-Queue: Steve Anton <steveanton@webrtc.org> > > > > > > Cr-Commit-Position: refs/heads/master@{#30292} > > > > > > > > > > TBR=steveanton@webrtc.org,kron@webrtc.org > > > > > > > > > > > > > > > Bug: chromium:1029737 > > > > > Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604 > > > > > Reviewed-by: Johannes Kron <kron@webrtc.org> > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > > > > Cr-Commit-Position: refs/heads/master@{#30348} > > > > > > > > TBR=steveanton@webrtc.org,kron@webrtc.org > > > > > > > > Change-Id: I9f8731309749e07ce7e651e1550ecfabddb1735f > > > > No-Presubmit: true > > > > No-Tree-Checks: true > > > > No-Try: true > > > > Bug: chromium:1029737 > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167205 > > > > Reviewed-by: Johannes Kron <kron@webrtc.org> > > > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > > > Cr-Commit-Position: refs/heads/master@{#30360} > > > > > > TBR=steveanton@webrtc.org,kron@webrtc.org > > > > > > Change-Id: I1cc2d83bd884f10685503a9c31288f96c935d6a3 > > > No-Presubmit: true > > > No-Tree-Checks: true > > > No-Try: true > > > Bug: chromium:1029737 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167206 > > > Reviewed-by: Johannes Kron <kron@webrtc.org> > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#30367} > > > > TBR=steveanton@webrtc.org,kron@webrtc.org > > > > Change-Id: I0a9b0b58922ce7c558b3d31b64cc12086b2a6a55 > > No-Presubmit: true > > No-Tree-Checks: true > > No-Try: true > > Bug: chromium:1029737 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167364 > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > Reviewed-by: Johannes Kron <kron@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#30373} > > TBR=steveanton@webrtc.org,kron@webrtc.org > > > Bug: chromium:1029737 > Change-Id: Id381cb6d8e03b0fca941e392978362af6fdab0b6 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167531 > Reviewed-by: Johannes Kron <kron@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Johannes Kron <kron@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30415} TBR=steveanton@webrtc.org,kron@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: chromium:1029737 Change-Id: Ice25339e7dfb9fc75049bd207d097b0910bd4446 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168341 Commit-Queue: Johannes Kron <kron@webrtc.org> Reviewed-by: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30484}
2020-02-07 14:23:45 +00:00
void GetSupportedVideoCodecs(std::vector<VideoCodec>* codecs) const;
void GetSupportedDataCodecs(std::vector<DataCodec>* codecs) const;
RtpHeaderExtensions GetDefaultEnabledAudioRtpHeaderExtensions() const;
std::vector<webrtc::RtpHeaderExtensionCapability>
GetSupportedAudioRtpHeaderExtensions() const;
RtpHeaderExtensions GetDefaultEnabledVideoRtpHeaderExtensions() const;
std::vector<webrtc::RtpHeaderExtensionCapability>
GetSupportedVideoRtpHeaderExtensions() const;
// Indicates whether the media engine is started.
bool initialized() const { return initialized_; }
// Starts up the media engine.
bool Init();
// Shuts down the media engine.
void Terminate();
// The operations below all occur on the worker thread.
// ChannelManager retains ownership of the created channels, so clients should
// call the appropriate Destroy*Channel method when done.
// Creates a voice channel, to be associated with the specified session.
Reland "Reland "Propagate media transport to media channel."" This is a reland of da65ed2adcfa57ff3288ce01c1602c973fcab00d Original change's description: > Reland "Propagate media transport to media channel." > > This reverts commit 37cf2455a420124b341ad06ac27fa3c4dbd29d3c. > > Reason for revert: <INSERT REASONING HERE> > > Original change's description: > > Revert "Propagate media transport to media channel." > > > > This reverts commit 8c16f745ab92cb6d305283e87fa8a661ae500ce4. > > > > Reason for revert: Breaks downstream project > > > > Original change's description: > > > Propagate media transport to media channel. > > > > > > 1. Pass media transport factory to JSEP transport controller. > > > 2. Pass media transport to voice media channel. > > > 3. Add basic unit test that make sure if peer connection is created with media transport, it is propagated to voice media channel. > > > > > > Change-Id: Ie922db78ade0efd893e019cd2b4441a9947a2f71 > > > Bug: webrtc:9719 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/105542 > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > > Reviewed-by: Peter Slatala <psla@webrtc.org> > > > Commit-Queue: Anton Sukhanov <sukhanov@google.com> > > > Cr-Commit-Position: refs/heads/master@{#25152} > > > > TBR=steveanton@webrtc.org,nisse@webrtc.org,psla@webrtc.org,sukhanov@google.com > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: webrtc:9719 > > Change-Id: Ic78cdc142a2145682ad74eac8b72c71c50f0a5c1 > > Reviewed-on: https://webrtc-review.googlesource.com/c/105840 > > Reviewed-by: Oleh Prypin <oprypin@webrtc.org> > > Commit-Queue: Oleh Prypin <oprypin@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25154} > > TBR=steveanton@webrtc.org,oprypin@webrtc.org,nisse@webrtc.org,sukhanov@webrtc.org,psla@webrtc.org,sukhanov@google.com > > Change-Id: I505ff3451eae81573531faef155ff35d7f894022 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9719 > Reviewed-on: https://webrtc-review.googlesource.com/c/106500 > Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org> > Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25220} Bug: webrtc:9719 Tbr: Steve Anton <steveanton@webrtc.org> Tbr: Niels Moller <nisse@webrtc.org> Change-Id: Ib45691ba8be9abb89ff8c6dac1861bdf59be4c8d Reviewed-on: https://webrtc-review.googlesource.com/c/106561 Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org> Reviewed-by: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25240}
2018-10-17 13:15:42 -07:00
VoiceChannel* CreateVoiceChannel(
webrtc::Call* call,
const cricket::MediaConfig& media_config,
webrtc::RtpTransportInternal* rtp_transport,
const webrtc::MediaTransportConfig& media_transport_config,
Reland "Reland "Propagate media transport to media channel."" This is a reland of da65ed2adcfa57ff3288ce01c1602c973fcab00d Original change's description: > Reland "Propagate media transport to media channel." > > This reverts commit 37cf2455a420124b341ad06ac27fa3c4dbd29d3c. > > Reason for revert: <INSERT REASONING HERE> > > Original change's description: > > Revert "Propagate media transport to media channel." > > > > This reverts commit 8c16f745ab92cb6d305283e87fa8a661ae500ce4. > > > > Reason for revert: Breaks downstream project > > > > Original change's description: > > > Propagate media transport to media channel. > > > > > > 1. Pass media transport factory to JSEP transport controller. > > > 2. Pass media transport to voice media channel. > > > 3. Add basic unit test that make sure if peer connection is created with media transport, it is propagated to voice media channel. > > > > > > Change-Id: Ie922db78ade0efd893e019cd2b4441a9947a2f71 > > > Bug: webrtc:9719 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/105542 > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > > Reviewed-by: Peter Slatala <psla@webrtc.org> > > > Commit-Queue: Anton Sukhanov <sukhanov@google.com> > > > Cr-Commit-Position: refs/heads/master@{#25152} > > > > TBR=steveanton@webrtc.org,nisse@webrtc.org,psla@webrtc.org,sukhanov@google.com > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: webrtc:9719 > > Change-Id: Ic78cdc142a2145682ad74eac8b72c71c50f0a5c1 > > Reviewed-on: https://webrtc-review.googlesource.com/c/105840 > > Reviewed-by: Oleh Prypin <oprypin@webrtc.org> > > Commit-Queue: Oleh Prypin <oprypin@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25154} > > TBR=steveanton@webrtc.org,oprypin@webrtc.org,nisse@webrtc.org,sukhanov@webrtc.org,psla@webrtc.org,sukhanov@google.com > > Change-Id: I505ff3451eae81573531faef155ff35d7f894022 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9719 > Reviewed-on: https://webrtc-review.googlesource.com/c/106500 > Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org> > Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25220} Bug: webrtc:9719 Tbr: Steve Anton <steveanton@webrtc.org> Tbr: Niels Moller <nisse@webrtc.org> Change-Id: Ib45691ba8be9abb89ff8c6dac1861bdf59be4c8d Reviewed-on: https://webrtc-review.googlesource.com/c/106561 Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org> Reviewed-by: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25240}
2018-10-17 13:15:42 -07:00
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
const webrtc::CryptoOptions& crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator,
Reland "Reland "Propagate media transport to media channel."" This is a reland of da65ed2adcfa57ff3288ce01c1602c973fcab00d Original change's description: > Reland "Propagate media transport to media channel." > > This reverts commit 37cf2455a420124b341ad06ac27fa3c4dbd29d3c. > > Reason for revert: <INSERT REASONING HERE> > > Original change's description: > > Revert "Propagate media transport to media channel." > > > > This reverts commit 8c16f745ab92cb6d305283e87fa8a661ae500ce4. > > > > Reason for revert: Breaks downstream project > > > > Original change's description: > > > Propagate media transport to media channel. > > > > > > 1. Pass media transport factory to JSEP transport controller. > > > 2. Pass media transport to voice media channel. > > > 3. Add basic unit test that make sure if peer connection is created with media transport, it is propagated to voice media channel. > > > > > > Change-Id: Ie922db78ade0efd893e019cd2b4441a9947a2f71 > > > Bug: webrtc:9719 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/105542 > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > > Reviewed-by: Peter Slatala <psla@webrtc.org> > > > Commit-Queue: Anton Sukhanov <sukhanov@google.com> > > > Cr-Commit-Position: refs/heads/master@{#25152} > > > > TBR=steveanton@webrtc.org,nisse@webrtc.org,psla@webrtc.org,sukhanov@google.com > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: webrtc:9719 > > Change-Id: Ic78cdc142a2145682ad74eac8b72c71c50f0a5c1 > > Reviewed-on: https://webrtc-review.googlesource.com/c/105840 > > Reviewed-by: Oleh Prypin <oprypin@webrtc.org> > > Commit-Queue: Oleh Prypin <oprypin@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25154} > > TBR=steveanton@webrtc.org,oprypin@webrtc.org,nisse@webrtc.org,sukhanov@webrtc.org,psla@webrtc.org,sukhanov@google.com > > Change-Id: I505ff3451eae81573531faef155ff35d7f894022 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9719 > Reviewed-on: https://webrtc-review.googlesource.com/c/106500 > Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org> > Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25220} Bug: webrtc:9719 Tbr: Steve Anton <steveanton@webrtc.org> Tbr: Niels Moller <nisse@webrtc.org> Change-Id: Ib45691ba8be9abb89ff8c6dac1861bdf59be4c8d Reviewed-on: https://webrtc-review.googlesource.com/c/106561 Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org> Reviewed-by: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25240}
2018-10-17 13:15:42 -07:00
const AudioOptions& options);
// Destroys a voice channel created by CreateVoiceChannel.
void DestroyVoiceChannel(VoiceChannel* voice_channel);
// Creates a video channel, synced with the specified voice channel, and
// associated with the specified session.
// Version of the above that takes PacketTransportInternal.
VideoChannel* CreateVideoChannel(
webrtc::Call* call,
const cricket::MediaConfig& media_config,
webrtc::RtpTransportInternal* rtp_transport,
const webrtc::MediaTransportConfig& media_transport_config,
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
const webrtc::CryptoOptions& crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator,
const VideoOptions& options,
webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory);
// Destroys a video channel created by CreateVideoChannel.
void DestroyVideoChannel(VideoChannel* video_channel);
RtpDataChannel* CreateRtpDataChannel(
const cricket::MediaConfig& media_config,
webrtc::RtpTransportInternal* rtp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
const webrtc::CryptoOptions& crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator);
// Destroys a data channel created by CreateRtpDataChannel.
void DestroyRtpDataChannel(RtpDataChannel* data_channel);
// Indicates whether any channels exist.
bool has_channels() const {
return (!voice_channels_.empty() || !video_channels_.empty() ||
!data_channels_.empty());
}
// RTX will be enabled/disabled in engines that support it. The supporting
// engines will start offering an RTX codec. Must be called before Init().
bool SetVideoRtxEnabled(bool enable);
// Starts/stops the local microphone and enables polling of the input level.
bool capturing() const { return capturing_; }
// The operations below occur on the main thread.
// Starts AEC dump using existing file, with a specified maximum file size in
// bytes. When the limit is reached, logging will stop and the file will be
// closed. If max_size_bytes is set to <= 0, no limit will be used.
bool StartAecDump(webrtc::FileWrapper file, int64_t max_size_bytes);
// Stops recording AEC dump.
void StopAecDump();
private:
std::unique_ptr<MediaEngineInterface> media_engine_; // Nullable.
std::unique_ptr<DataEngineInterface> data_engine_; // Non-null.
bool initialized_ = false;
rtc::Thread* main_thread_;
rtc::Thread* worker_thread_;
rtc::Thread* network_thread_;
// Vector contents are non-null.
std::vector<std::unique_ptr<VoiceChannel>> voice_channels_;
std::vector<std::unique_ptr<VideoChannel>> video_channels_;
std::vector<std::unique_ptr<RtpDataChannel>> data_channels_;
bool enable_rtx_ = false;
bool capturing_ = false;
};
} // namespace cricket
#endif // PC_CHANNEL_MANAGER_H_