/*
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* libjingle
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* Copyright 2014 Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#include "talk/app/webrtc/remoteaudiosource.h"
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#include <algorithm>
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#include <functional>
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#include <utility>
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#include "talk/app/webrtc/mediastreamprovider.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/base/logging.h"
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#include "webrtc/base/thread.h"
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namespace webrtc {
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class RemoteAudioSource::MessageHandler : public rtc::MessageHandler {
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public:
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explicit MessageHandler(RemoteAudioSource* source) : source_(source) {}
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private:
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~MessageHandler() override {}
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void OnMessage(rtc::Message* msg) override {
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source_->OnMessage(msg);
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delete this;
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}
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const rtc::scoped_refptr<RemoteAudioSource> source_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MessageHandler);
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};
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class RemoteAudioSource::Sink : public AudioSinkInterface {
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public:
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explicit Sink(RemoteAudioSource* source) : source_(source) {}
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~Sink() override { source_->OnAudioProviderGone(); }
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private:
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void OnData(const AudioSinkInterface::Data& audio) override {
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if (source_)
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source_->OnData(audio);
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}
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const rtc::scoped_refptr<RemoteAudioSource> source_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Sink);
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};
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rtc::scoped_refptr<RemoteAudioSource> RemoteAudioSource::Create(
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uint32_t ssrc,
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AudioProviderInterface* provider) {
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rtc::scoped_refptr<RemoteAudioSource> ret(
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new rtc::RefCountedObject<RemoteAudioSource>());
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ret->Initialize(ssrc, provider);
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return ret;
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}
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RemoteAudioSource::RemoteAudioSource()
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: main_thread_(rtc::Thread::Current()),
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state_(MediaSourceInterface::kLive) {
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RTC_DCHECK(main_thread_);
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}
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RemoteAudioSource::~RemoteAudioSource() {
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RTC_DCHECK(main_thread_->IsCurrent());
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RTC_DCHECK(audio_observers_.empty());
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RTC_DCHECK(sinks_.empty());
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}
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void RemoteAudioSource::Initialize(uint32_t ssrc,
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AudioProviderInterface* provider) {
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RTC_DCHECK(main_thread_->IsCurrent());
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// To make sure we always get notified when the provider goes out of scope,
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// we register for callbacks here and not on demand in AddSink.
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if (provider) { // May be null in tests.
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provider->SetRawAudioSink(
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ssrc, rtc::scoped_ptr<AudioSinkInterface>(new Sink(this)));
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}
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}
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MediaSourceInterface::SourceState RemoteAudioSource::state() const {
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RTC_DCHECK(main_thread_->IsCurrent());
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return state_;
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}
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bool RemoteAudioSource::remote() const {
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RTC_DCHECK(main_thread_->IsCurrent());
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return true;
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}
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void RemoteAudioSource::SetVolume(double volume) {
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RTC_DCHECK(volume >= 0 && volume <= 10);
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for (auto* observer : audio_observers_)
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observer->OnSetVolume(volume);
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}
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void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) {
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RTC_DCHECK(observer != NULL);
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RTC_DCHECK(std::find(audio_observers_.begin(), audio_observers_.end(),
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observer) == audio_observers_.end());
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audio_observers_.push_back(observer);
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}
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void RemoteAudioSource::UnregisterAudioObserver(AudioObserver* observer) {
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RTC_DCHECK(observer != NULL);
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audio_observers_.remove(observer);
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}
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void RemoteAudioSource::AddSink(AudioTrackSinkInterface* sink) {
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RTC_DCHECK(main_thread_->IsCurrent());
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RTC_DCHECK(sink);
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if (state_ != MediaSourceInterface::kLive) {
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LOG(LS_ERROR) << "Can't register sink as the source isn't live.";
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return;
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}
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rtc::CritScope lock(&sink_lock_);
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RTC_DCHECK(std::find(sinks_.begin(), sinks_.end(), sink) == sinks_.end());
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sinks_.push_back(sink);
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}
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void RemoteAudioSource::RemoveSink(AudioTrackSinkInterface* sink) {
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RTC_DCHECK(main_thread_->IsCurrent());
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RTC_DCHECK(sink);
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rtc::CritScope lock(&sink_lock_);
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sinks_.remove(sink);
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}
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void RemoteAudioSource::OnData(const AudioSinkInterface::Data& audio) {
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// Called on the externally-owned audio callback thread, via/from webrtc.
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rtc::CritScope lock(&sink_lock_);
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for (auto* sink : sinks_) {
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sink->OnData(audio.data, 16, audio.sample_rate, audio.channels,
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audio.samples_per_channel);
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}
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}
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void RemoteAudioSource::OnAudioProviderGone() {
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// Called when the data provider is deleted. It may be the worker thread
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// in libjingle or may be a different worker thread.
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main_thread_->Post(new MessageHandler(this));
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}
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void RemoteAudioSource::OnMessage(rtc::Message* msg) {
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RTC_DCHECK(main_thread_->IsCurrent());
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sinks_.clear();
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state_ = MediaSourceInterface::kEnded;
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FireOnChanged();
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}
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} // namespace webrtc
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