webrtc/pc/remote_audio_source.cc
Henrik Boström fa8a9465d5 Remove obsolete DCHECK in remote_audio_source.cc.
When fixing so that RemoteAudioSource does not end the track just
because the audio channel is gone in Unified Plan[1], this made it
possible for ~PeerConnection to delete all objects, including deleting
the MediaStreamTrack and its RemoteAudioSource, when all tracks are not
in an ended state.

In a real application or Chromium, the PeerConnection would not be
destroyed prior to closing and not hit this DCHECK. But in upstream
dependent projects' unit tests, it would be possible for ref counted
tracks to be destroyed when the track are still kLive, and as a
side-effect hit this DCHECK.

sinks_ is just a list of raw pointers, and whether or not we have done
sinks_.clear() prior to destruction is irrelevant going forward.

[1] https://webrtc-review.googlesource.com/c/src/+/214136

Bug: chromium:1121454
Change-Id: If6cf3dffcd3cb47d46694755b5dc45fa381285fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215226
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33739}
2021-04-15 10:18:40 +00:00

195 lines
6.3 KiB
C++

/*
* Copyright 2014 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "pc/remote_audio_source.h"
#include <stddef.h>
#include <memory>
#include "absl/algorithm/container.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "rtc_base/checks.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/strings/string_format.h"
#include "rtc_base/thread.h"
namespace webrtc {
// This proxy is passed to the underlying media engine to receive audio data as
// they come in. The data will then be passed back up to the RemoteAudioSource
// which will fan it out to all the sinks that have been added to it.
class RemoteAudioSource::AudioDataProxy : public AudioSinkInterface {
public:
explicit AudioDataProxy(RemoteAudioSource* source) : source_(source) {
RTC_DCHECK(source);
}
AudioDataProxy() = delete;
AudioDataProxy(const AudioDataProxy&) = delete;
AudioDataProxy& operator=(const AudioDataProxy&) = delete;
~AudioDataProxy() override { source_->OnAudioChannelGone(); }
// AudioSinkInterface implementation.
void OnData(const AudioSinkInterface::Data& audio) override {
source_->OnData(audio);
}
private:
const rtc::scoped_refptr<RemoteAudioSource> source_;
};
RemoteAudioSource::RemoteAudioSource(
rtc::Thread* worker_thread,
OnAudioChannelGoneAction on_audio_channel_gone_action)
: main_thread_(rtc::Thread::Current()),
worker_thread_(worker_thread),
on_audio_channel_gone_action_(on_audio_channel_gone_action),
state_(MediaSourceInterface::kLive) {
RTC_DCHECK(main_thread_);
RTC_DCHECK(worker_thread_);
}
RemoteAudioSource::~RemoteAudioSource() {
RTC_DCHECK(main_thread_->IsCurrent());
RTC_DCHECK(audio_observers_.empty());
if (!sinks_.empty()) {
RTC_LOG(LS_WARNING)
<< "RemoteAudioSource destroyed while sinks_ is non-empty.";
}
}
void RemoteAudioSource::Start(cricket::VoiceMediaChannel* media_channel,
absl::optional<uint32_t> ssrc) {
RTC_DCHECK_RUN_ON(main_thread_);
RTC_DCHECK(media_channel);
// Register for callbacks immediately before AddSink so that we always get
// notified when a channel goes out of scope (signaled when "AudioDataProxy"
// is destroyed).
worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
ssrc ? media_channel->SetRawAudioSink(
*ssrc, std::make_unique<AudioDataProxy>(this))
: media_channel->SetDefaultRawAudioSink(
std::make_unique<AudioDataProxy>(this));
});
}
void RemoteAudioSource::Stop(cricket::VoiceMediaChannel* media_channel,
absl::optional<uint32_t> ssrc) {
RTC_DCHECK_RUN_ON(main_thread_);
RTC_DCHECK(media_channel);
worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
ssrc ? media_channel->SetRawAudioSink(*ssrc, nullptr)
: media_channel->SetDefaultRawAudioSink(nullptr);
});
}
void RemoteAudioSource::SetState(SourceState new_state) {
if (state_ != new_state) {
state_ = new_state;
FireOnChanged();
}
}
MediaSourceInterface::SourceState RemoteAudioSource::state() const {
RTC_DCHECK(main_thread_->IsCurrent());
return state_;
}
bool RemoteAudioSource::remote() const {
RTC_DCHECK(main_thread_->IsCurrent());
return true;
}
void RemoteAudioSource::SetVolume(double volume) {
RTC_DCHECK_GE(volume, 0);
RTC_DCHECK_LE(volume, 10);
RTC_LOG(LS_INFO) << rtc::StringFormat("RAS::%s({volume=%.2f})", __func__,
volume);
for (auto* observer : audio_observers_) {
observer->OnSetVolume(volume);
}
}
void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) {
RTC_DCHECK(observer != NULL);
RTC_DCHECK(!absl::c_linear_search(audio_observers_, observer));
audio_observers_.push_back(observer);
}
void RemoteAudioSource::UnregisterAudioObserver(AudioObserver* observer) {
RTC_DCHECK(observer != NULL);
audio_observers_.remove(observer);
}
void RemoteAudioSource::AddSink(AudioTrackSinkInterface* sink) {
RTC_DCHECK(main_thread_->IsCurrent());
RTC_DCHECK(sink);
if (state_ != MediaSourceInterface::kLive) {
RTC_LOG(LS_ERROR) << "Can't register sink as the source isn't live.";
return;
}
MutexLock lock(&sink_lock_);
RTC_DCHECK(!absl::c_linear_search(sinks_, sink));
sinks_.push_back(sink);
}
void RemoteAudioSource::RemoveSink(AudioTrackSinkInterface* sink) {
RTC_DCHECK(main_thread_->IsCurrent());
RTC_DCHECK(sink);
MutexLock lock(&sink_lock_);
sinks_.remove(sink);
}
void RemoteAudioSource::OnData(const AudioSinkInterface::Data& audio) {
// Called on the externally-owned audio callback thread, via/from webrtc.
MutexLock lock(&sink_lock_);
for (auto* sink : sinks_) {
// When peerconnection acts as an audio source, it should not provide
// absolute capture timestamp.
sink->OnData(audio.data, 16, audio.sample_rate, audio.channels,
audio.samples_per_channel,
/*absolute_capture_timestamp_ms=*/absl::nullopt);
}
}
void RemoteAudioSource::OnAudioChannelGone() {
if (on_audio_channel_gone_action_ != OnAudioChannelGoneAction::kEnd) {
return;
}
// Called when the audio channel is deleted. It may be the worker thread
// in libjingle or may be a different worker thread.
// This object needs to live long enough for the cleanup logic in OnMessage to
// run, so take a reference to it as the data. Sometimes the message may not
// be processed (because the thread was destroyed shortly after this call),
// but that is fine because the thread destructor will take care of destroying
// the message data which will release the reference on RemoteAudioSource.
main_thread_->Post(RTC_FROM_HERE, this, 0,
new rtc::ScopedRefMessageData<RemoteAudioSource>(this));
}
void RemoteAudioSource::OnMessage(rtc::Message* msg) {
RTC_DCHECK(main_thread_->IsCurrent());
sinks_.clear();
SetState(MediaSourceInterface::kEnded);
// Will possibly delete this RemoteAudioSource since it is reference counted
// in the message.
delete msg->pdata;
}
} // namespace webrtc