Merge "Merge upstream SHA 04cb763"android-cts_7.1_r1android-cts-7.1_r9android-cts-7.1_r8android-cts-7.1_r7android-cts-7.1_r6android-cts-7.1_r5android-cts-7.1_r4android-cts-7.1_r3android-cts-7.1_r29android-cts-7.1_r28android-cts-7.1_r27android-cts-7.1_r26android-cts-7.1_r25android-cts-7.1_r24android-cts-7.1_r23android-cts-7.1_r22android-cts-7.1_r21android-cts-7.1_r20android-cts-7.1_r2android-cts-7.1_r19android-cts-7.1_r18android-cts-7.1_r17android-cts-7.1_r16android-cts-7.1_r15android-cts-7.1_r14android-cts-7.1_r13android-cts-7.1_r12android-cts-7.1_r11android-cts-7.1_r10android-cts-7.1_r1android-cts-7.0_r9android-cts-7.0_r8android-cts-7.0_r7android-cts-7.0_r6android-cts-7.0_r5android-cts-7.0_r4android-cts-7.0_r33android-cts-7.0_r32android-cts-7.0_r31android-cts-7.0_r30android-cts-7.0_r3android-cts-7.0_r29android-cts-7.0_r28android-cts-7.0_r27android-cts-7.0_r26android-cts-7.0_r25android-cts-7.0_r24android-cts-7.0_r23android-cts-7.0_r22android-cts-7.0_r21android-cts-7.0_r20android-cts-7.0_r2android-cts-7.0_r19android-cts-7.0_r18android-cts-7.0_r17android-cts-7.0_r16android-cts-7.0_r15android-cts-7.0_r14android-cts-7.0_r13android-cts-7.0_r12android-cts-7.0_r11android-cts-7.0_r10android-cts-7.0_r1android-7.1.2_r9android-7.1.2_r8android-7.1.2_r6android-7.1.2_r5android-7.1.2_r4android-7.1.2_r39android-7.1.2_r38android-7.1.2_r37android-7.1.2_r36android-7.1.2_r33android-7.1.2_r32android-7.1.2_r30android-7.1.2_r3android-7.1.2_r29android-7.1.2_r28android-7.1.2_r27android-7.1.2_r25android-7.1.2_r24android-7.1.2_r23android-7.1.2_r2android-7.1.2_r19android-7.1.2_r18android-7.1.2_r17android-7.1.2_r16android-7.1.2_r15android-7.1.2_r14android-7.1.2_r13android-7.1.2_r12android-7.1.2_r11android-7.1.2_r10android-7.1.2_r1android-7.1.1_r9android-7.1.1_r8android-7.1.1_r7android-7.1.1_r61android-7.1.1_r60android-7.1.1_r6android-7.1.1_r59android-7.1.1_r58android-7.1.1_r57android-7.1.1_r56android-7.1.1_r55android-7.1.1_r54android-7.1.1_r53android-7.1.1_r52android-7.1.1_r51android-7.1.1_r50android-7.1.1_r49android-7.1.1_r48android-7.1.1_r47android-7.1.1_r46android-7.1.1_r45android-7.1.1_r44android-7.1.1_r43android-7.1.1_r42android-7.1.1_r41android-7.1.1_r40android-7.1.1_r4android-7.1.1_r39android-7.1.1_r38android-7.1.1_r35android-7.1.1_r33android-7.1.1_r32android-7.1.1_r31android-7.1.1_r3android-7.1.1_r28android-7.1.1_r27android-7.1.1_r26android-7.1.1_r25android-7.1.1_r24android-7.1.1_r23android-7.1.1_r22android-7.1.1_r21android-7.1.1_r20android-7.1.1_r2android-7.1.1_r17android-7.1.1_r16android-7.1.1_r15android-7.1.1_r14android-7.1.1_r13android-7.1.1_r12android-7.1.1_r11android-7.1.1_r10android-7.1.1_r1android-7.1.0_r7android-7.1.0_r6android-7.1.0_r5android-7.1.0_r4android-7.1.0_r3android-7.1.0_r2android-7.1.0_r1android-7.0.0_r9android-7.0.0_r8android-7.0.0_r7android-7.0.0_r6android-7.0.0_r5android-7.0.0_r4android-7.0.0_r36android-7.0.0_r35android-7.0.0_r34android-7.0.0_r33android-7.0.0_r32android-7.0.0_r31android-7.0.0_r30android-7.0.0_r3android-7.0.0_r29android-7.0.0_r28android-7.0.0_r27android-7.0.0_r24android-7.0.0_r21android-7.0.0_r19android-7.0.0_r17android-7.0.0_r15android-7.0.0_r14android-7.0.0_r13android-7.0.0_r12android-7.0.0_r11android-7.0.0_r10android-7.0.0_r1nougat-releasenougat-mr2.3-releasenougat-mr2.2-releasenougat-mr2.1-releasenougat-mr2-security-releasenougat-mr2-releasenougat-mr2-pixel-releasenougat-mr2-devnougat-mr1.8-releasenougat-mr1.7-releasenougat-mr1.6-releasenougat-mr1.5-releasenougat-mr1.4-releasenougat-mr1.3-releasenougat-mr1.2-releasenougat-mr1.1-releasenougat-mr1-volantis-releasenougat-mr1-security-releasenougat-mr1-releasenougat-mr1-flounder-releasenougat-mr1-devnougat-mr1-cts-releasenougat-mr0.5-releasenougat-dr1-releasenougat-devnougat-cts-releasenougat-bugfix-release

am: 9a337512d9

* commit '9a337512d97e37afc142dee4fd50a41b741a87d2': (797 commits)
  Add tests for verifying transport feedback for audio and video.
  Eliminate defines in talk/
  Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ )
  Remove assert which was incorrectly added to TcpPort::OnSentPacket.
  Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Update with new default boringssl no-aes cipher suites. Re-enable tests.
  Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ )
  Re-land: "Use an explicit identifier in Config"
  Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ )
  Remove libfuzzer trybot from default trybot set.
  Add ramp-up tests for transport sequence number with and w/o audio.
  Delete remnants of non-square pixel support from cricket::VideoFrame.
  Fix IPAddress::ToSensitiveString() to avoid dependency on inet_ntop().
  Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
  Re-enable tests that failed under Linux_Msan.
  Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ )
  Roll chromium_revision 346fea9..099be58 (369082:369139)
  Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan
  Add build_protobuf variable.
  ...

1 files changed, 118 insertions, 14 deletions

diff --git a/talk/app/webrtc/remoteaudiosource.cc b/talk/app/webrtc/remoteaudiosource.cc
index 41f3d8798a..e904dd9192 100644
--- a/talk/app/webrtc/remoteaudiosource.cc
+++ b/talk/app/webrtc/remoteaudiosource.cc
@@ -29,44 +29,148 @@
 
 #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 {
 
-rtc::scoped_refptr<RemoteAudioSource> RemoteAudioSource::Create() {
-  return new rtc::RefCountedObject<RemoteAudioSource>();
+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() {
+RemoteAudioSource::RemoteAudioSource()
+    : main_thread_(rtc::Thread::Current()),
+      state_(MediaSourceInterface::kLive) {
+  RTC_DCHECK(main_thread_);
 }
 
 RemoteAudioSource::~RemoteAudioSource() {
-  ASSERT(audio_observers_.empty());
+  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 {
-  return MediaSourceInterface::kLive;
+  RTC_DCHECK(main_thread_->IsCurrent());
+  return state_;
+}
+
+bool RemoteAudioSource::remote() const {
+  RTC_DCHECK(main_thread_->IsCurrent());
+  return true;
 }
 
 void RemoteAudioSource::SetVolume(double volume) {
-  ASSERT(volume >= 0 && volume <= 10);
-  for (AudioObserverList::iterator it = audio_observers_.begin();
-       it != audio_observers_.end(); ++it) {
-    (*it)->OnSetVolume(volume);
-  }
+  RTC_DCHECK(volume >= 0 && volume <= 10);
+  for (auto* observer : audio_observers_)
+    observer->OnSetVolume(volume);
 }
 
 void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) {
-  ASSERT(observer != NULL);
-  ASSERT(std::find(audio_observers_.begin(), audio_observers_.end(),
-                   observer) == audio_observers_.end());
+  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) {
-  ASSERT(observer != NULL);
+  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