Move talk/app/webrtc to webrtc/api

talk/app/webrtc/rtpsender.h

Index: talk/app/webrtc/rtpsender.h
diff --git a/talk/app/webrtc/rtpsender.h b/talk/app/webrtc/rtpsender.h
deleted file mode 100644
index c68f64be40b29ab673c2f507d15856d01f92434c..0000000000000000000000000000000000000000
--- a/talk/app/webrtc/rtpsender.h
+++ /dev/null
@@ -1,195 +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.
- */
-
-// This file contains classes that implement RtpSenderInterface.
-// An RtpSender associates a MediaStreamTrackInterface with an underlying
-// transport (provided by AudioProviderInterface/VideoProviderInterface)
-
-#ifndef TALK_APP_WEBRTC_RTPSENDER_H_
-#define TALK_APP_WEBRTC_RTPSENDER_H_
-
-#include <string>
-
-#include "talk/app/webrtc/mediastreamprovider.h"
-#include "talk/app/webrtc/rtpsenderinterface.h"
-#include "talk/app/webrtc/statscollector.h"
-#include "webrtc/base/basictypes.h"
-#include "webrtc/base/criticalsection.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/media/base/audiorenderer.h"
-
-namespace webrtc {
-
-// LocalAudioSinkAdapter receives data callback as a sink to the local
-// AudioTrack, and passes the data to the sink of AudioRenderer.
-class LocalAudioSinkAdapter : public AudioTrackSinkInterface,
-                              public cricket::AudioRenderer {
- public:
-  LocalAudioSinkAdapter();
-  virtual ~LocalAudioSinkAdapter();
-
- private:
-  // AudioSinkInterface implementation.
-  void OnData(const void* audio_data,
-              int bits_per_sample,
-              int sample_rate,
-              size_t number_of_channels,
-              size_t number_of_frames) override;
-
-  // cricket::AudioRenderer implementation.
-  void SetSink(cricket::AudioRenderer::Sink* sink) override;
-
-  cricket::AudioRenderer::Sink* sink_;
-  // Critical section protecting |sink_|.
-  rtc::CriticalSection lock_;
-};
-
-class AudioRtpSender : public ObserverInterface,
-                       public rtc::RefCountedObject<RtpSenderInterface> {
- public:
-  // StatsCollector provided so that Add/RemoveLocalAudioTrack can be called
-  // at the appropriate times.
-  AudioRtpSender(AudioTrackInterface* track,
-                 const std::string& stream_id,
-                 AudioProviderInterface* provider,
-                 StatsCollector* stats);
-
-  // Randomly generates stream_id.
-  AudioRtpSender(AudioTrackInterface* track,
-                 AudioProviderInterface* provider,
-                 StatsCollector* stats);
-
-  // Randomly generates id and stream_id.
-  AudioRtpSender(AudioProviderInterface* provider, StatsCollector* stats);
-
-  virtual ~AudioRtpSender();
-
-  // ObserverInterface implementation
-  void OnChanged() override;
-
-  // RtpSenderInterface implementation
-  bool SetTrack(MediaStreamTrackInterface* track) override;
-  rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
-    return track_.get();
-  }
-
-  void SetSsrc(uint32_t ssrc) override;
-
-  uint32_t ssrc() const override { return ssrc_; }
-
-  cricket::MediaType media_type() const override {
-    return cricket::MEDIA_TYPE_AUDIO;
-  }
-
-  std::string id() const override { return id_; }
-
-  void set_stream_id(const std::string& stream_id) override {
-    stream_id_ = stream_id;
-  }
-  std::string stream_id() const override { return stream_id_; }
-
-  void Stop() override;
-
- private:
-  bool can_send_track() const { return track_ && ssrc_; }
-  // Helper function to construct options for
-  // AudioProviderInterface::SetAudioSend.
-  void SetAudioSend();
-
-  std::string id_;
-  std::string stream_id_;
-  AudioProviderInterface* provider_;
-  StatsCollector* stats_;
-  rtc::scoped_refptr<AudioTrackInterface> track_;
-  uint32_t ssrc_ = 0;
-  bool cached_track_enabled_ = false;
-  bool stopped_ = false;
-
-  // Used to pass the data callback from the |track_| to the other end of
-  // cricket::AudioRenderer.
-  rtc::scoped_ptr<LocalAudioSinkAdapter> sink_adapter_;
-};
-
-class VideoRtpSender : public ObserverInterface,
-                       public rtc::RefCountedObject<RtpSenderInterface> {
- public:
-  VideoRtpSender(VideoTrackInterface* track,
-                 const std::string& stream_id,
-                 VideoProviderInterface* provider);
-
-  // Randomly generates stream_id.
-  VideoRtpSender(VideoTrackInterface* track, VideoProviderInterface* provider);
-
-  // Randomly generates id and stream_id.
-  explicit VideoRtpSender(VideoProviderInterface* provider);
-
-  virtual ~VideoRtpSender();
-
-  // ObserverInterface implementation
-  void OnChanged() override;
-
-  // RtpSenderInterface implementation
-  bool SetTrack(MediaStreamTrackInterface* track) override;
-  rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
-    return track_.get();
-  }
-
-  void SetSsrc(uint32_t ssrc) override;
-
-  uint32_t ssrc() const override { return ssrc_; }
-
-  cricket::MediaType media_type() const override {
-    return cricket::MEDIA_TYPE_VIDEO;
-  }
-
-  std::string id() const override { return id_; }
-
-  void set_stream_id(const std::string& stream_id) override {
-    stream_id_ = stream_id;
-  }
-  std::string stream_id() const override { return stream_id_; }
-
-  void Stop() override;
-
- private:
-  bool can_send_track() const { return track_ && ssrc_; }
-  // Helper function to construct options for
-  // VideoProviderInterface::SetVideoSend.
-  void SetVideoSend();
-
-  std::string id_;
-  std::string stream_id_;
-  VideoProviderInterface* provider_;
-  rtc::scoped_refptr<VideoTrackInterface> track_;
-  uint32_t ssrc_ = 0;
-  bool cached_track_enabled_ = false;
-  bool stopped_ = false;
-};
-
-}  // namespace webrtc
-
-#endif  // TALK_APP_WEBRTC_RTPSENDER_H_