c335b0e63b
The RemoteAudioSource has an AudioDataProxy that acts as a sink, passing along data from AudioRecvStreams to the RemoteAudioSource. If an SSRC is changed (or other reconfiguration happens) with SDP, the recv stream and proxy get recreated. In Plan B, because remote tracks maps 1:1 with SSRCs, it made sense to end remote track/audio source in response to this. In Plan B, a new receiver, with a new track and a new proxy would be created for the new SSRC. In Unified Plan however, remote tracks correspond to m= sections. The remote track should only end on port:0 (or RTCP BYE or timeout, etc), not because the recv stream of an m= section is recreated. The code already supports changing SSRC and this is working correctly, but because ~AudioDataProxy() would end the source this would cause the MediaStreamTrack of the receiver to end (even though the media engine is still processing the remote audio stream correctly under the hood). This issue only happened on audio tracks, and because of timing of PostTasks the track would kEnd in Chromium *after* promise.then(). This CL fixes that issue by not ending the source when the proxy is destroyed. Destroying a recv stream is a temporary action in Unified Plan, unless stopped. Tests are added ensuring tracks are kLive. I have manually verified that this CL fixes the issue and that both audio and video is flowing through the entire pipeline: https://jsfiddle.net/henbos/h21xec97/122/ Bug: chromium:1121454 Change-Id: Ic21ac8ea263ccf021b96a14d3e4e3b24eb756c86 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214136 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33645}
103 lines
3.3 KiB
C++
103 lines
3.3 KiB
C++
/*
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* Copyright 2014 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_REMOTE_AUDIO_SOURCE_H_
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#define PC_REMOTE_AUDIO_SOURCE_H_
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#include <stdint.h>
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#include <list>
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#include <string>
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#include "absl/types/optional.h"
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#include "api/call/audio_sink.h"
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#include "api/media_stream_interface.h"
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#include "api/notifier.h"
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#include "media/base/media_channel.h"
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#include "pc/channel.h"
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#include "rtc_base/message_handler.h"
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#include "rtc_base/synchronization/mutex.h"
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#include "rtc_base/thread.h"
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#include "rtc_base/thread_message.h"
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namespace rtc {
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struct Message;
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class Thread;
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} // namespace rtc
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namespace webrtc {
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// This class implements the audio source used by the remote audio track.
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// This class works by configuring itself as a sink with the underlying media
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// engine, then when receiving data will fan out to all added sinks.
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class RemoteAudioSource : public Notifier<AudioSourceInterface>,
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rtc::MessageHandler {
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public:
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// In Unified Plan, receivers map to m= sections and their tracks and sources
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// survive SSRCs being reconfigured. The life cycle of the remote audio source
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// is associated with the life cycle of the m= section, and thus even if an
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// audio channel is destroyed the RemoteAudioSource should kSurvive.
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//
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// In Plan B however, remote audio sources map 1:1 with an SSRCs and if an
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// audio channel is destroyed, the RemoteAudioSource should kEnd.
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enum class OnAudioChannelGoneAction {
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kSurvive,
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kEnd,
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};
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explicit RemoteAudioSource(
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rtc::Thread* worker_thread,
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OnAudioChannelGoneAction on_audio_channel_gone_action);
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// Register and unregister remote audio source with the underlying media
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// engine.
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void Start(cricket::VoiceMediaChannel* media_channel,
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absl::optional<uint32_t> ssrc);
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void Stop(cricket::VoiceMediaChannel* media_channel,
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absl::optional<uint32_t> ssrc);
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void SetState(SourceState new_state);
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// MediaSourceInterface implementation.
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MediaSourceInterface::SourceState state() const override;
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bool remote() const override;
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// AudioSourceInterface implementation.
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void SetVolume(double volume) override;
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void RegisterAudioObserver(AudioObserver* observer) override;
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void UnregisterAudioObserver(AudioObserver* observer) override;
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void AddSink(AudioTrackSinkInterface* sink) override;
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void RemoveSink(AudioTrackSinkInterface* sink) override;
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protected:
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~RemoteAudioSource() override;
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private:
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// These are callbacks from the media engine.
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class AudioDataProxy;
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void OnData(const AudioSinkInterface::Data& audio);
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void OnAudioChannelGone();
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void OnMessage(rtc::Message* msg) override;
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rtc::Thread* const main_thread_;
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rtc::Thread* const worker_thread_;
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const OnAudioChannelGoneAction on_audio_channel_gone_action_;
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std::list<AudioObserver*> audio_observers_;
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Mutex sink_lock_;
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std::list<AudioTrackSinkInterface*> sinks_;
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SourceState state_;
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};
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} // namespace webrtc
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#endif // PC_REMOTE_AUDIO_SOURCE_H_
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