Move talk/app/webrtc to webrtc/api

talk/app/webrtc/remoteaudiosource.cc

Index: talk/app/webrtc/remoteaudiosource.cc
diff --git a/talk/app/webrtc/remoteaudiosource.cc b/talk/app/webrtc/remoteaudiosource.cc
deleted file mode 100644
index e904dd91925e48654832a5bb769f2e971c8b513e..0000000000000000000000000000000000000000
--- a/talk/app/webrtc/remoteaudiosource.cc
+++ /dev/null
@@ -1,176 +0,0 @@
-/*
- * libjingle
- * Copyright 2014 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "talk/app/webrtc/remoteaudiosource.h"
-
-#include <algorithm>
-#include <functional>
-#include <utility>
-
-#include "talk/app/webrtc/mediastreamprovider.h"
-#include "webrtc/base/checks.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_->OnAudioProviderGone(); }
-
- 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,
-    AudioProviderInterface* provider) {
-  rtc::scoped_refptr<RemoteAudioSource> ret(
-      new rtc::RefCountedObject<RemoteAudioSource>());
-  ret->Initialize(ssrc, provider);
-  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,
-                                   AudioProviderInterface* provider) {
-  RTC_DCHECK(main_thread_->IsCurrent());
-  // To make sure we always get notified when the provider goes out of scope,
-  // we register for callbacks here and not on demand in AddSink.
-  if (provider) {  // May be null in tests.
-    provider->SetRawAudioSink(
-        ssrc, rtc::scoped_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::OnAudioProviderGone() {
-  // Called when the data provider is deleted.  It may be the worker thread
-  // in libjingle or may be a different worker thread.
-  main_thread_->Post(new MessageHandler(this));
-}
-
-void RemoteAudioSource::OnMessage(rtc::Message* msg) {
-  RTC_DCHECK(main_thread_->IsCurrent());
-  sinks_.clear();
-  state_ = MediaSourceInterface::kEnded;
-  FireOnChanged();
-}
-
-}  // namespace webrtc