Index: talk/app/webrtc/rtpsender.h |
diff --git a/talk/app/webrtc/rtpsender.h b/talk/app/webrtc/rtpsender.h |
index aee77e173c9c438fed345afcd68adc0e7331e955..a0eae5dd6add152d29e51164e6635846fb3d3448 100644 |
--- a/talk/app/webrtc/rtpsender.h |
+++ b/talk/app/webrtc/rtpsender.h |
@@ -25,4 +25,116 @@ |
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
*/ |
-// This file is currently stubbed so that Chromium's build files can be updated. |
+// 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/media/base/audiorenderer.h" |
+#include "webrtc/base/basictypes.h" |
+#include "webrtc/base/criticalsection.h" |
+#include "webrtc/base/scoped_ptr.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, |
+ int 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: |
+ AudioRtpSender(AudioTrackInterface* track, |
+ uint32 ssrc, |
+ AudioProviderInterface* provider); |
+ |
+ virtual ~AudioRtpSender(); |
+ |
+ // ObserverInterface implementation |
+ void OnChanged() override; |
+ |
+ // RtpSenderInterface implementation |
+ bool SetTrack(MediaStreamTrackInterface* track) override; |
+ rtc::scoped_refptr<MediaStreamTrackInterface> track() const override { |
+ return track_.get(); |
+ } |
+ |
+ std::string id() const override { return id_; } |
+ |
+ void Stop() override; |
+ |
+ private: |
+ void Reconfigure(); |
+ |
+ std::string id_; |
+ rtc::scoped_refptr<AudioTrackInterface> track_; |
+ uint32 ssrc_; |
+ AudioProviderInterface* provider_; |
+ bool cached_track_enabled_; |
+ |
+ // 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, |
+ uint32 ssrc, |
+ 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(); |
+ } |
+ |
+ std::string id() const override { return id_; } |
+ |
+ void Stop() override; |
+ |
+ private: |
+ void Reconfigure(); |
+ |
+ std::string id_; |
+ rtc::scoped_refptr<VideoTrackInterface> track_; |
+ uint32 ssrc_; |
+ VideoProviderInterface* provider_; |
+ bool cached_track_enabled_; |
+}; |
+ |
+} // namespace webrtc |
+ |
+#endif // TALK_APP_WEBRTC_RTPSENDER_H_ |