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}
182 lines
7.0 KiB
C++
182 lines
7.0 KiB
C++
/*
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* Copyright 2004 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef PC_CHANNEL_MANAGER_H_
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#define PC_CHANNEL_MANAGER_H_
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#include <stdint.h>
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#include <memory>
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#include <string>
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#include <vector>
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#include "api/audio_options.h"
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#include "api/crypto/crypto_options.h"
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#include "api/transport/media/media_transport_config.h"
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#include "call/call.h"
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#include "media/base/codec.h"
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#include "media/base/media_channel.h"
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#include "media/base/media_config.h"
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#include "media/base/media_engine.h"
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#include "pc/channel.h"
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#include "pc/rtp_transport_internal.h"
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#include "pc/session_description.h"
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#include "rtc_base/system/file_wrapper.h"
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#include "rtc_base/thread.h"
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namespace cricket {
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// ChannelManager allows the MediaEngine to run on a separate thread, and takes
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// care of marshalling calls between threads. It also creates and keeps track of
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// voice and video channels; by doing so, it can temporarily pause all the
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// channels when a new audio or video device is chosen. The voice and video
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// channels are stored in separate vectors, to easily allow operations on just
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// voice or just video channels.
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// ChannelManager also allows the application to discover what devices it has
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// using device manager.
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class ChannelManager final {
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public:
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// Construct a ChannelManager with the specified media engine and data engine.
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ChannelManager(std::unique_ptr<MediaEngineInterface> media_engine,
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std::unique_ptr<DataEngineInterface> data_engine,
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rtc::Thread* worker_thread,
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rtc::Thread* network_thread);
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~ChannelManager();
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// Accessors for the worker thread, allowing it to be set after construction,
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// but before Init. set_worker_thread will return false if called after Init.
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rtc::Thread* worker_thread() const { return worker_thread_; }
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bool set_worker_thread(rtc::Thread* thread) {
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if (initialized_) {
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return false;
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}
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worker_thread_ = thread;
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return true;
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}
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rtc::Thread* network_thread() const { return network_thread_; }
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bool set_network_thread(rtc::Thread* thread) {
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if (initialized_) {
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return false;
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}
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network_thread_ = thread;
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return true;
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}
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MediaEngineInterface* media_engine() { return media_engine_.get(); }
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// Retrieves the list of supported audio & video codec types.
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// Can be called before starting the media engine.
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void GetSupportedAudioSendCodecs(std::vector<AudioCodec>* codecs) const;
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void GetSupportedAudioReceiveCodecs(std::vector<AudioCodec>* codecs) const;
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void GetSupportedAudioRtpHeaderExtensions(RtpHeaderExtensions* ext) const;
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void GetSupportedVideoSendCodecs(std::vector<VideoCodec>* codecs) const;
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void GetSupportedVideoReceiveCodecs(std::vector<VideoCodec>* codecs) const;
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void GetSupportedVideoRtpHeaderExtensions(RtpHeaderExtensions* ext) const;
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void GetSupportedDataCodecs(std::vector<DataCodec>* codecs) const;
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// Indicates whether the media engine is started.
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bool initialized() const { return initialized_; }
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// Starts up the media engine.
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bool Init();
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// Shuts down the media engine.
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void Terminate();
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// The operations below all occur on the worker thread.
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// ChannelManager retains ownership of the created channels, so clients should
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// call the appropriate Destroy*Channel method when done.
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// Creates a voice channel, to be associated with the specified session.
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VoiceChannel* CreateVoiceChannel(
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webrtc::Call* call,
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const cricket::MediaConfig& media_config,
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webrtc::RtpTransportInternal* rtp_transport,
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const webrtc::MediaTransportConfig& media_transport_config,
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rtc::Thread* signaling_thread,
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const std::string& content_name,
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bool srtp_required,
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const webrtc::CryptoOptions& crypto_options,
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rtc::UniqueRandomIdGenerator* ssrc_generator,
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const AudioOptions& options);
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// Destroys a voice channel created by CreateVoiceChannel.
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void DestroyVoiceChannel(VoiceChannel* voice_channel);
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// Creates a video channel, synced with the specified voice channel, and
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// associated with the specified session.
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// Version of the above that takes PacketTransportInternal.
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VideoChannel* CreateVideoChannel(
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webrtc::Call* call,
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const cricket::MediaConfig& media_config,
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webrtc::RtpTransportInternal* rtp_transport,
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const webrtc::MediaTransportConfig& media_transport_config,
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rtc::Thread* signaling_thread,
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const std::string& content_name,
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bool srtp_required,
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const webrtc::CryptoOptions& crypto_options,
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rtc::UniqueRandomIdGenerator* ssrc_generator,
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const VideoOptions& options,
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webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory);
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// Destroys a video channel created by CreateVideoChannel.
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void DestroyVideoChannel(VideoChannel* video_channel);
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RtpDataChannel* CreateRtpDataChannel(
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const cricket::MediaConfig& media_config,
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webrtc::RtpTransportInternal* rtp_transport,
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rtc::Thread* signaling_thread,
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const std::string& content_name,
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bool srtp_required,
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const webrtc::CryptoOptions& crypto_options,
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rtc::UniqueRandomIdGenerator* ssrc_generator);
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// Destroys a data channel created by CreateRtpDataChannel.
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void DestroyRtpDataChannel(RtpDataChannel* data_channel);
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// Indicates whether any channels exist.
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bool has_channels() const {
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return (!voice_channels_.empty() || !video_channels_.empty() ||
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!data_channels_.empty());
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}
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// RTX will be enabled/disabled in engines that support it. The supporting
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// engines will start offering an RTX codec. Must be called before Init().
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bool SetVideoRtxEnabled(bool enable);
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// Starts/stops the local microphone and enables polling of the input level.
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bool capturing() const { return capturing_; }
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// The operations below occur on the main thread.
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// Starts AEC dump using existing file, with a specified maximum file size in
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// bytes. When the limit is reached, logging will stop and the file will be
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// closed. If max_size_bytes is set to <= 0, no limit will be used.
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bool StartAecDump(webrtc::FileWrapper file, int64_t max_size_bytes);
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// Stops recording AEC dump.
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void StopAecDump();
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private:
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std::unique_ptr<MediaEngineInterface> media_engine_; // Nullable.
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std::unique_ptr<DataEngineInterface> data_engine_; // Non-null.
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bool initialized_ = false;
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rtc::Thread* main_thread_;
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rtc::Thread* worker_thread_;
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rtc::Thread* network_thread_;
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// Vector contents are non-null.
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std::vector<std::unique_ptr<VoiceChannel>> voice_channels_;
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std::vector<std::unique_ptr<VideoChannel>> video_channels_;
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std::vector<std::unique_ptr<RtpDataChannel>> data_channels_;
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bool enable_rtx_ = false;
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bool capturing_ = false;
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};
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} // namespace cricket
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#endif // PC_CHANNEL_MANAGER_H_
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