Files
platform-external-webrtc/webrtc/api/remoteaudiosource.cc
Taylor Brandstetter ba29c6aac7 Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
Relanding again after fixing issue with RTC_DCHECKs.

This CL eliminates the need for the extra layer of indirection provided by
mediastreamprovider.h. It will thus make it easier to implement new
functionality in RtpSender/RtpReceiver.

It also brings us one step closer to the end goal of combining "senders"
and "send streams". Currently the sender still needs to go through the
BaseChannel and MediaChannel, using an SSRC as a key.

R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/2046173002 .

Cr-Commit-Position: refs/heads/master@{#13305}
2016-06-27 23:30:45 +00:00

161 lines
4.7 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 "webrtc/api/remoteaudiosource.h"
#include <algorithm>
#include <functional>
#include <memory>
#include <utility>
#include "webrtc/base/checks.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/thread.h"
namespace webrtc {
class RemoteAudioSource::MessageHandler : public rtc::MessageHandler {
public:
explicit MessageHandler(RemoteAudioSource* source) : source_(source) {}
private:
~MessageHandler() override {}
void OnMessage(rtc::Message* msg) override {
source_->OnMessage(msg);
delete this;
}
const rtc::scoped_refptr<RemoteAudioSource> source_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MessageHandler);
};
class RemoteAudioSource::Sink : public AudioSinkInterface {
public:
explicit Sink(RemoteAudioSource* source) : source_(source) {}
~Sink() override { source_->OnAudioChannelGone(); }
private:
void OnData(const AudioSinkInterface::Data& audio) override {
if (source_)
source_->OnData(audio);
}
const rtc::scoped_refptr<RemoteAudioSource> source_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Sink);
};
rtc::scoped_refptr<RemoteAudioSource> RemoteAudioSource::Create(
uint32_t ssrc,
cricket::VoiceChannel* channel) {
rtc::scoped_refptr<RemoteAudioSource> ret(
new rtc::RefCountedObject<RemoteAudioSource>());
ret->Initialize(ssrc, channel);
return ret;
}
RemoteAudioSource::RemoteAudioSource()
: main_thread_(rtc::Thread::Current()),
state_(MediaSourceInterface::kLive) {
RTC_DCHECK(main_thread_);
}
RemoteAudioSource::~RemoteAudioSource() {
RTC_DCHECK(main_thread_->IsCurrent());
RTC_DCHECK(audio_observers_.empty());
RTC_DCHECK(sinks_.empty());
}
void RemoteAudioSource::Initialize(uint32_t ssrc,
cricket::VoiceChannel* channel) {
RTC_DCHECK(main_thread_->IsCurrent());
// To make sure we always get notified when the channel goes out of scope,
// we register for callbacks here and not on demand in AddSink.
if (channel) { // May be null in tests.
channel->SetRawAudioSink(
ssrc, std::unique_ptr<AudioSinkInterface>(new Sink(this)));
}
}
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(volume >= 0 && volume <= 10);
for (auto* observer : audio_observers_)
observer->OnSetVolume(volume);
}
void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) {
RTC_DCHECK(observer != NULL);
RTC_DCHECK(std::find(audio_observers_.begin(), audio_observers_.end(),
observer) == audio_observers_.end());
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) {
LOG(LS_ERROR) << "Can't register sink as the source isn't live.";
return;
}
rtc::CritScope lock(&sink_lock_);
RTC_DCHECK(std::find(sinks_.begin(), sinks_.end(), sink) == sinks_.end());
sinks_.push_back(sink);
}
void RemoteAudioSource::RemoveSink(AudioTrackSinkInterface* sink) {
RTC_DCHECK(main_thread_->IsCurrent());
RTC_DCHECK(sink);
rtc::CritScope lock(&sink_lock_);
sinks_.remove(sink);
}
void RemoteAudioSource::OnData(const AudioSinkInterface::Data& audio) {
// Called on the externally-owned audio callback thread, via/from webrtc.
rtc::CritScope lock(&sink_lock_);
for (auto* sink : sinks_) {
sink->OnData(audio.data, 16, audio.sample_rate, audio.channels,
audio.samples_per_channel);
}
}
void RemoteAudioSource::OnAudioChannelGone() {
// Called when the audio channel is deleted. It may be the worker thread
// in libjingle or may be a different worker thread.
main_thread_->Post(RTC_FROM_HERE, new MessageHandler(this));
}
void RemoteAudioSource::OnMessage(rtc::Message* msg) {
RTC_DCHECK(main_thread_->IsCurrent());
sinks_.clear();
state_ = MediaSourceInterface::kEnded;
FireOnChanged();
}
} // namespace webrtc