Move talk/app/webrtc to webrtc/api

talk/app/webrtc/rtpsender.cc

Index: talk/app/webrtc/rtpsender.cc
diff --git a/talk/app/webrtc/rtpsender.cc b/talk/app/webrtc/rtpsender.cc
deleted file mode 100644
index a30bf0b163616f7022530f7c993d9ad153561969..0000000000000000000000000000000000000000
--- a/talk/app/webrtc/rtpsender.cc
+++ /dev/null
@@ -1,348 +0,0 @@
-/*
- * 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"
-#include "webrtc/base/helpers.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,
-                                   size_t 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,
-                               const std::string& stream_id,
-                               AudioProviderInterface* provider,
-                               StatsCollector* stats)
-    : id_(track->id()),
-      stream_id_(stream_id),
-      provider_(provider),
-      stats_(stats),
-      track_(track),
-      cached_track_enabled_(track->enabled()),
-      sink_adapter_(new LocalAudioSinkAdapter()) {
-  RTC_DCHECK(provider != nullptr);
-  track_->RegisterObserver(this);
-  track_->AddSink(sink_adapter_.get());
-}
-
-AudioRtpSender::AudioRtpSender(AudioTrackInterface* track,
-                               AudioProviderInterface* provider,
-                               StatsCollector* stats)
-    : id_(track->id()),
-      stream_id_(rtc::CreateRandomUuid()),
-      provider_(provider),
-      stats_(stats),
-      track_(track),
-      cached_track_enabled_(track->enabled()),
-      sink_adapter_(new LocalAudioSinkAdapter()) {
-  RTC_DCHECK(provider != nullptr);
-  track_->RegisterObserver(this);
-  track_->AddSink(sink_adapter_.get());
-}
-
-AudioRtpSender::AudioRtpSender(AudioProviderInterface* provider,
-                               StatsCollector* stats)
-    : id_(rtc::CreateRandomUuid()),
-      stream_id_(rtc::CreateRandomUuid()),
-      provider_(provider),
-      stats_(stats),
-      sink_adapter_(new LocalAudioSinkAdapter()) {}
-
-AudioRtpSender::~AudioRtpSender() {
-  Stop();
-}
-
-void AudioRtpSender::OnChanged() {
-  RTC_DCHECK(!stopped_);
-  if (cached_track_enabled_ != track_->enabled()) {
-    cached_track_enabled_ = track_->enabled();
-    if (can_send_track()) {
-      SetAudioSend();
-    }
-  }
-}
-
-bool AudioRtpSender::SetTrack(MediaStreamTrackInterface* track) {
-  if (stopped_) {
-    LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender.";
-    return false;
-  }
-  if (track && track->kind() != MediaStreamTrackInterface::kAudioKind) {
-    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.
-  if (track_) {
-    track_->RemoveSink(sink_adapter_.get());
-    track_->UnregisterObserver(this);
-  }
-
-  if (can_send_track() && stats_) {
-    stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
-  }
-
-  // Attach to new track.
-  bool prev_can_send_track = can_send_track();
-  track_ = audio_track;
-  if (track_) {
-    cached_track_enabled_ = track_->enabled();
-    track_->RegisterObserver(this);
-    track_->AddSink(sink_adapter_.get());
-  }
-
-  // Update audio provider.
-  if (can_send_track()) {
-    SetAudioSend();
-    if (stats_) {
-      stats_->AddLocalAudioTrack(track_.get(), ssrc_);
-    }
-  } else if (prev_can_send_track) {
-    cricket::AudioOptions options;
-    provider_->SetAudioSend(ssrc_, false, options, nullptr);
-  }
-  return true;
-}
-
-void AudioRtpSender::SetSsrc(uint32_t ssrc) {
-  if (stopped_ || ssrc == ssrc_) {
-    return;
-  }
-  // If we are already sending with a particular SSRC, stop sending.
-  if (can_send_track()) {
-    cricket::AudioOptions options;
-    provider_->SetAudioSend(ssrc_, false, options, nullptr);
-    if (stats_) {
-      stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
-    }
-  }
-  ssrc_ = ssrc;
-  if (can_send_track()) {
-    SetAudioSend();
-    if (stats_) {
-      stats_->AddLocalAudioTrack(track_.get(), ssrc_);
-    }
-  }
-}
-
-void AudioRtpSender::Stop() {
-  // TODO(deadbeef): Need to do more here to fully stop sending packets.
-  if (stopped_) {
-    return;
-  }
-  if (track_) {
-    track_->RemoveSink(sink_adapter_.get());
-    track_->UnregisterObserver(this);
-  }
-  if (can_send_track()) {
-    cricket::AudioOptions options;
-    provider_->SetAudioSend(ssrc_, false, options, nullptr);
-    if (stats_) {
-      stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
-    }
-  }
-  stopped_ = true;
-}
-
-void AudioRtpSender::SetAudioSend() {
-  RTC_DCHECK(!stopped_ && can_send_track());
-  cricket::AudioOptions options;
-#if !defined(WEBRTC_CHROMIUM_BUILD)
-  // TODO(tommi): Remove this hack when we move CreateAudioSource out of
-  // PeerConnection.  This is a bit of a strange way to apply local audio
-  // options since it is also applied to all streams/channels, local or remote.
-  if (track_->enabled() && track_->GetSource() &&
-      !track_->GetSource()->remote()) {
-    // TODO(xians): Remove this static_cast since we should be able to connect
-    // a remote audio track to a peer connection.
-    options = static_cast<LocalAudioSource*>(track_->GetSource())->options();
-  }
-#endif
-
-  cricket::AudioRenderer* renderer = sink_adapter_.get();
-  ASSERT(renderer != nullptr);
-  provider_->SetAudioSend(ssrc_, track_->enabled(), options, renderer);
-}
-
-VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
-                               const std::string& stream_id,
-                               VideoProviderInterface* provider)
-    : id_(track->id()),
-      stream_id_(stream_id),
-      provider_(provider),
-      track_(track),
-      cached_track_enabled_(track->enabled()) {
-  RTC_DCHECK(provider != nullptr);
-  track_->RegisterObserver(this);
-}
-
-VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
-                               VideoProviderInterface* provider)
-    : id_(track->id()),
-      stream_id_(rtc::CreateRandomUuid()),
-      provider_(provider),
-      track_(track),
-      cached_track_enabled_(track->enabled()) {
-  RTC_DCHECK(provider != nullptr);
-  track_->RegisterObserver(this);
-}
-
-VideoRtpSender::VideoRtpSender(VideoProviderInterface* provider)
-    : id_(rtc::CreateRandomUuid()),
-      stream_id_(rtc::CreateRandomUuid()),
-      provider_(provider) {}
-
-VideoRtpSender::~VideoRtpSender() {
-  Stop();
-}
-
-void VideoRtpSender::OnChanged() {
-  RTC_DCHECK(!stopped_);
-  if (cached_track_enabled_ != track_->enabled()) {
-    cached_track_enabled_ = track_->enabled();
-    if (can_send_track()) {
-      SetVideoSend();
-    }
-  }
-}
-
-bool VideoRtpSender::SetTrack(MediaStreamTrackInterface* track) {
-  if (stopped_) {
-    LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender.";
-    return false;
-  }
-  if (track && track->kind() != MediaStreamTrackInterface::kVideoKind) {
-    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.
-  if (track_) {
-    track_->UnregisterObserver(this);
-  }
-
-  // Attach to new track.
-  bool prev_can_send_track = can_send_track();
-  track_ = video_track;
-  if (track_) {
-    cached_track_enabled_ = track_->enabled();
-    track_->RegisterObserver(this);
-  }
-
-  // Update video provider.
-  if (can_send_track()) {
-    VideoSourceInterface* source = track_->GetSource();
-    // TODO(deadbeef): If SetTrack is called with a disabled track, and the
-    // previous track was enabled, this could cause a frame from the new track
-    // to slip out. Really, what we need is for SetCaptureDevice and
-    // SetVideoSend
-    // to be combined into one atomic operation, all the way down to
-    // WebRtcVideoSendStream.
-    provider_->SetCaptureDevice(ssrc_,
-                                source ? source->GetVideoCapturer() : nullptr);
-    SetVideoSend();
-  } else if (prev_can_send_track) {
-    provider_->SetCaptureDevice(ssrc_, nullptr);
-    provider_->SetVideoSend(ssrc_, false, nullptr);
-  }
-  return true;
-}
-
-void VideoRtpSender::SetSsrc(uint32_t ssrc) {
-  if (stopped_ || ssrc == ssrc_) {
-    return;
-  }
-  // If we are already sending with a particular SSRC, stop sending.
-  if (can_send_track()) {
-    provider_->SetCaptureDevice(ssrc_, nullptr);
-    provider_->SetVideoSend(ssrc_, false, nullptr);
-  }
-  ssrc_ = ssrc;
-  if (can_send_track()) {
-    VideoSourceInterface* source = track_->GetSource();
-    provider_->SetCaptureDevice(ssrc_,
-                                source ? source->GetVideoCapturer() : nullptr);
-    SetVideoSend();
-  }
-}
-
-void VideoRtpSender::Stop() {
-  // TODO(deadbeef): Need to do more here to fully stop sending packets.
-  if (stopped_) {
-    return;
-  }
-  if (track_) {
-    track_->UnregisterObserver(this);
-  }
-  if (can_send_track()) {
-    provider_->SetCaptureDevice(ssrc_, nullptr);
-    provider_->SetVideoSend(ssrc_, false, nullptr);
-  }
-  stopped_ = true;
-}
-
-void VideoRtpSender::SetVideoSend() {
-  RTC_DCHECK(!stopped_ && can_send_track());
-  const cricket::VideoOptions* options = nullptr;
-  VideoSourceInterface* source = track_->GetSource();
-  if (track_->enabled() && source) {
-    options = source->options();
-  }
-  provider_->SetVideoSend(ssrc_, track_->enabled(), options);
-}
-
-}  // namespace webrtc