Merge changes I7bbf776e,I1b827825

am: fe8b4a6579

* commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits)
  WIP: Changes after merge commit 'cb3f9bd'
  Make the nonlinear beamformer steerable
  Utilize bitrate above codec max to protect video.
  Enable VP9 internal resize by default.
  Filter overlapping RTP header extensions.
  Make VCMEncodedFrameCallback const.
  MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback.
  Remove redudant encoder rate calls.
  Create isolate files for nonparallel tests.
  Register header extensions in RtpRtcpObserver to avoid log spam.
  Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
  ACM: Move NACK functionality inside NetEq
  Fix chromium-style warnings in webrtc/sound/.
  Create a 'webrtc_nonparallel_tests' target.
  Update scalability structure data according to updates in the RTP payload profile.
  audio_coding: rename interface -> include
  Rewrote perform_action_on_all_files to be parallell.
  Update reference indices according to updates in the RTP payload profile.
  Disable P2PTransport...TestFailoverControlledSide on Memcheck
  pass clangcl compile options to ignore warnings in gflags.cc
  ...

1 files changed, 207 insertions, 0 deletions

diff --git a/talk/app/webrtc/rtpsender.cc b/talk/app/webrtc/rtpsender.cc
new file mode 100644
index 0000000000..3a78f4598a
--- /dev/null
+++ b/talk/app/webrtc/rtpsender.cc
@@ -0,0 +1,207 @@
+/*
+ * libjingle
+ * Copyright 2015 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/rtpsender.h"
+
+#include "talk/app/webrtc/localaudiosource.h"
+#include "talk/app/webrtc/videosourceinterface.h"
+
+namespace webrtc {
+
+LocalAudioSinkAdapter::LocalAudioSinkAdapter() : sink_(nullptr) {}
+
+LocalAudioSinkAdapter::~LocalAudioSinkAdapter() {
+  rtc::CritScope lock(&lock_);
+  if (sink_)
+    sink_->OnClose();
+}
+
+void LocalAudioSinkAdapter::OnData(const void* audio_data,
+                                   int bits_per_sample,
+                                   int sample_rate,
+                                   int number_of_channels,
+                                   size_t number_of_frames) {
+  rtc::CritScope lock(&lock_);
+  if (sink_) {
+    sink_->OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,
+                  number_of_frames);
+  }
+}
+
+void LocalAudioSinkAdapter::SetSink(cricket::AudioRenderer::Sink* sink) {
+  rtc::CritScope lock(&lock_);
+  ASSERT(!sink || !sink_);
+  sink_ = sink;
+}
+
+AudioRtpSender::AudioRtpSender(AudioTrackInterface* track,
+                               uint32_t ssrc,
+                               AudioProviderInterface* provider)
+    : id_(track->id()),
+      track_(track),
+      ssrc_(ssrc),
+      provider_(provider),
+      cached_track_enabled_(track->enabled()),
+      sink_adapter_(new LocalAudioSinkAdapter()) {
+  track_->RegisterObserver(this);
+  track_->AddSink(sink_adapter_.get());
+  Reconfigure();
+}
+
+AudioRtpSender::~AudioRtpSender() {
+  track_->RemoveSink(sink_adapter_.get());
+  track_->UnregisterObserver(this);
+  Stop();
+}
+
+void AudioRtpSender::OnChanged() {
+  if (cached_track_enabled_ != track_->enabled()) {
+    cached_track_enabled_ = track_->enabled();
+    Reconfigure();
+  }
+}
+
+bool AudioRtpSender::SetTrack(MediaStreamTrackInterface* track) {
+  if (track->kind() != "audio") {
+    LOG(LS_ERROR) << "SetTrack called on audio RtpSender with " << track->kind()
+                  << " track.";
+    return false;
+  }
+  AudioTrackInterface* audio_track = static_cast<AudioTrackInterface*>(track);
+
+  // Detach from old track.
+  track_->RemoveSink(sink_adapter_.get());
+  track_->UnregisterObserver(this);
+
+  // Attach to new track.
+  track_ = audio_track;
+  cached_track_enabled_ = track_->enabled();
+  track_->RegisterObserver(this);
+  track_->AddSink(sink_adapter_.get());
+  Reconfigure();
+  return true;
+}
+
+void AudioRtpSender::Stop() {
+  // TODO(deadbeef): Need to do more here to fully stop sending packets.
+  if (!provider_) {
+    return;
+  }
+  cricket::AudioOptions options;
+  provider_->SetAudioSend(ssrc_, false, options, nullptr);
+  provider_ = nullptr;
+}
+
+void AudioRtpSender::Reconfigure() {
+  if (!provider_) {
+    return;
+  }
+  cricket::AudioOptions options;
+  if (track_->enabled() && track_->GetSource()) {
+    // TODO(xians): Remove this static_cast since we should be able to connect
+    // a remote audio track to peer connection.
+    options = static_cast<LocalAudioSource*>(track_->GetSource())->options();
+  }
+
+  // Use the renderer if the audio track has one, otherwise use the sink
+  // adapter owned by this class.
+  cricket::AudioRenderer* renderer =
+      track_->GetRenderer() ? track_->GetRenderer() : sink_adapter_.get();
+  ASSERT(renderer != nullptr);
+  provider_->SetAudioSend(ssrc_, track_->enabled(), options, renderer);
+}
+
+VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
+                               uint32_t ssrc,
+                               VideoProviderInterface* provider)
+    : id_(track->id()),
+      track_(track),
+      ssrc_(ssrc),
+      provider_(provider),
+      cached_track_enabled_(track->enabled()) {
+  track_->RegisterObserver(this);
+  VideoSourceInterface* source = track_->GetSource();
+  if (source) {
+    provider_->SetCaptureDevice(ssrc_, source->GetVideoCapturer());
+  }
+  Reconfigure();
+}
+
+VideoRtpSender::~VideoRtpSender() {
+  track_->UnregisterObserver(this);
+  Stop();
+}
+
+void VideoRtpSender::OnChanged() {
+  if (cached_track_enabled_ != track_->enabled()) {
+    cached_track_enabled_ = track_->enabled();
+    Reconfigure();
+  }
+}
+
+bool VideoRtpSender::SetTrack(MediaStreamTrackInterface* track) {
+  if (track->kind() != "video") {
+    LOG(LS_ERROR) << "SetTrack called on video RtpSender with " << track->kind()
+                  << " track.";
+    return false;
+  }
+  VideoTrackInterface* video_track = static_cast<VideoTrackInterface*>(track);
+
+  // Detach from old track.
+  track_->UnregisterObserver(this);
+
+  // Attach to new track.
+  track_ = video_track;
+  cached_track_enabled_ = track_->enabled();
+  track_->RegisterObserver(this);
+  Reconfigure();
+  return true;
+}
+
+void VideoRtpSender::Stop() {
+  // TODO(deadbeef): Need to do more here to fully stop sending packets.
+  if (!provider_) {
+    return;
+  }
+  provider_->SetCaptureDevice(ssrc_, nullptr);
+  provider_->SetVideoSend(ssrc_, false, nullptr);
+  provider_ = nullptr;
+}
+
+void VideoRtpSender::Reconfigure() {
+  if (!provider_) {
+    return;
+  }
+  const cricket::VideoOptions* options = nullptr;
+  VideoSourceInterface* source = track_->GetSource();
+  if (track_->enabled() && source) {
+    options = source->options();
+  }
+  provider_->SetVideoSend(ssrc_, track_->enabled(), options);
+}
+
+}  // namespace webrtc