Chromium Code Reviews| Index: talk/app/webrtc/rtpsender.h |
| diff --git a/talk/app/webrtc/rtpsender.h b/talk/app/webrtc/rtpsender.h |
| index 374190932325bb9715ed4b477b61556177833935..934f8c3581c764a44f9fb1f1e97ba75f5ea5a20a 100644 |
| --- a/talk/app/webrtc/rtpsender.h |
| +++ b/talk/app/webrtc/rtpsender.h |
| @@ -36,6 +36,7 @@ |
| #include "talk/app/webrtc/mediastreamprovider.h" |
| #include "talk/app/webrtc/rtpsenderinterface.h" |
| +#include "talk/app/webrtc/statscollector.h" |
| #include "talk/media/base/audiorenderer.h" |
| #include "webrtc/base/basictypes.h" |
| #include "webrtc/base/criticalsection.h" |
| @@ -70,9 +71,15 @@ class LocalAudioSinkAdapter : public AudioTrackSinkInterface, |
| 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, |
| - uint32_t ssrc, |
| - AudioProviderInterface* provider); |
| + const std::string& stream_id, |
| + AudioProviderInterface* provider, |
| + StatsCollector* stats); |
| + |
| + // Randomly generates id and stream_id. |
| + AudioRtpSender(AudioProviderInterface* provider, StatsCollector* stats); |
| virtual ~AudioRtpSender(); |
| @@ -85,18 +92,34 @@ class AudioRtpSender : public ObserverInterface, |
| return track_.get(); |
| } |
| + void SetSsrc(uint32_t ssrc) override; |
| + |
| + uint32_t ssrc() const override { return ssrc_; } |
| + |
| + std::string kind() const override { return "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: |
| - void Reconfigure(); |
| + // Helper function to construct options for |
| + // AudioProviderInterface::SetAudioSend. |
| + void SetAudioSend(); |
| std::string id_; |
| - rtc::scoped_refptr<AudioTrackInterface> track_; |
| - uint32_t ssrc_; |
| + std::string stream_id_; |
| AudioProviderInterface* provider_; |
| - bool cached_track_enabled_; |
| + StatsCollector* stats_; |
| + rtc::scoped_refptr<AudioTrackInterface> track_; |
| + uint32_t ssrc_ = 0; |
|
pthatcher1
2015/10/20 17:42:49
Can you put a comment about how 0 is "unset" and w
Taylor Brandstetter
2015/10/21 00:22:08
It's in rtpsenderinterface.h.
|
| + bool cached_track_enabled_ = false; |
| + bool stopped_ = false; |
| // Used to pass the data callback from the |track_| to the other end of |
| // cricket::AudioRenderer. |
| @@ -107,9 +130,12 @@ class VideoRtpSender : public ObserverInterface, |
| public rtc::RefCountedObject<RtpSenderInterface> { |
| public: |
| VideoRtpSender(VideoTrackInterface* track, |
| - uint32_t ssrc, |
| + const std::string& stream_id, |
| VideoProviderInterface* provider); |
| + // Randomly generates id and stream_id. |
| + explicit VideoRtpSender(VideoProviderInterface* provider); |
| + |
| virtual ~VideoRtpSender(); |
| // ObserverInterface implementation |
| @@ -121,18 +147,33 @@ class VideoRtpSender : public ObserverInterface, |
| return track_.get(); |
| } |
| + void SetSsrc(uint32_t ssrc) override; |
| + |
| + uint32_t ssrc() const override { return ssrc_; } |
| + |
| + std::string kind() const override { return "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: |
| - void Reconfigure(); |
| + // Helper function to construct options for |
| + // VideoProviderInterface::SetVideoSend. |
| + void SetVideoSend(); |
| std::string id_; |
| - rtc::scoped_refptr<VideoTrackInterface> track_; |
| - uint32_t ssrc_; |
| + std::string stream_id_; |
| VideoProviderInterface* provider_; |
| - bool cached_track_enabled_; |
| + rtc::scoped_refptr<VideoTrackInterface> track_; |
| + uint32_t ssrc_ = 0; |
| + bool cached_track_enabled_ = false; |
| + bool stopped_ = false; |
| }; |
| } // namespace webrtc |