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/*
* 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 <cstdint>
#include <memory>
#include <optional>
#include <utility>
#include "absl/algorithm/container.h"
#include "api/call/audio_sink.h"
#include "api/media_stream_interface.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/task_queue/task_queue_base.h"
#include "media/base/media_channel.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/strings/string_format.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/trace_event.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 scoped_refptr<RemoteAudioSource> source_;
};
RemoteAudioSource::RemoteAudioSource(
TaskQueueBase* worker_thread,
OnAudioChannelGoneAction on_audio_channel_gone_action)
: main_thread_(TaskQueueBase::Current()),
worker_thread_(worker_thread),
on_audio_channel_gone_action_(on_audio_channel_gone_action),
state_(MediaSourceInterface::kInitializing) {
RTC_DCHECK(main_thread_);
RTC_DCHECK(worker_thread_);
}
RemoteAudioSource::~RemoteAudioSource() {
RTC_DCHECK(audio_observers_.empty());
if (!sinks_.empty()) {
RTC_LOG(LS_WARNING)
<< "RemoteAudioSource destroyed while sinks_ is non-empty.";
}
}
void RemoteAudioSource::Start(VoiceMediaReceiveChannelInterface* media_channel,
std::optional<uint32_t> ssrc) {
RTC_DCHECK_RUN_ON(worker_thread_);
// Register for callbacks immediately before AddSink so that we always get
// notified when a channel goes out of scope (signaled when "AudioDataProxy"
// is destroyed).
RTC_DCHECK(media_channel);
ssrc ? media_channel->SetRawAudioSink(*ssrc,
std::make_unique<AudioDataProxy>(this))
: media_channel->SetDefaultRawAudioSink(
std::make_unique<AudioDataProxy>(this));
}
void RemoteAudioSource::Stop(VoiceMediaReceiveChannelInterface* media_channel,
std::optional<uint32_t> ssrc) {
RTC_DCHECK_RUN_ON(worker_thread_);
RTC_DCHECK(media_channel);
ssrc ? media_channel->SetRawAudioSink(*ssrc, nullptr)
: media_channel->SetDefaultRawAudioSink(nullptr);
}
void RemoteAudioSource::SetState(SourceState new_state) {
RTC_DCHECK_RUN_ON(main_thread_);
if (state_ != new_state) {
state_ = new_state;
FireOnChanged();
}
}
MediaSourceInterface::SourceState RemoteAudioSource::state() const {
RTC_DCHECK_RUN_ON(main_thread_);
return state_;
}
bool RemoteAudioSource::remote() const {
RTC_DCHECK_RUN_ON(main_thread_);
return true;
}
void RemoteAudioSource::SetVolume(double volume) {
RTC_DCHECK_GE(volume, 0);
RTC_DCHECK_LE(volume, 10);
RTC_LOG(LS_INFO) << StringFormat("RAS::%s({volume=%.2f})", __func__, volume);
for (auto* observer : audio_observers_) {
observer->OnSetVolume(volume);
}
}
void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) {
RTC_DCHECK(observer != nullptr);
RTC_DCHECK(!absl::c_linear_search(audio_observers_, observer));
audio_observers_.push_back(observer);
}
void RemoteAudioSource::UnregisterAudioObserver(AudioObserver* observer) {
RTC_DCHECK(observer != nullptr);
audio_observers_.remove(observer);
}
void RemoteAudioSource::AddSink(AudioTrackSinkInterface* sink) {
RTC_DCHECK_RUN_ON(main_thread_);
RTC_DCHECK(sink);
MutexLock lock(&sink_lock_);
RTC_DCHECK(!absl::c_linear_search(sinks_, sink));
sinks_.push_back(sink);
}
void RemoteAudioSource::RemoveSink(AudioTrackSinkInterface* sink) {
RTC_DCHECK_RUN_ON(main_thread_);
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.
TRACE_EVENT0("webrtc", "RemoteAudioSource::OnData");
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=*/std::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 or
// may be a different task queue.
// This object needs to live long enough for the cleanup logic in the posted
// task to run, so take a reference to it. Sometimes the task may not be
// processed (because the task queue was destroyed shortly after this call),
// but that is fine because the task queue destructor will take care of
// destroying task which will release the reference on RemoteAudioSource.
scoped_refptr<RemoteAudioSource> thiz(this);
main_thread_->PostTask([thiz = std::move(thiz)] {
thiz->sinks_.clear();
thiz->SetState(MediaSourceInterface::kEnded);
});
}
} // namespace webrtc
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