| Index: talk/app/webrtc/rtpsender.h
|
| diff --git a/talk/app/webrtc/rtpsender.h b/talk/app/webrtc/rtpsender.h
|
| index d5f88a941add3df7e8a2bcdc3efe0a0ab62eb073..374190932325bb9715ed4b477b61556177833935 100644
|
| --- a/talk/app/webrtc/rtpsender.h
|
| +++ b/talk/app/webrtc/rtpsender.h
|
| @@ -36,7 +36,6 @@
|
|
|
| #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"
|
| @@ -71,15 +70,9 @@
|
| 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 id and stream_id.
|
| - AudioRtpSender(AudioProviderInterface* provider, StatsCollector* stats);
|
| + uint32_t ssrc,
|
| + AudioProviderInterface* provider);
|
|
|
| virtual ~AudioRtpSender();
|
|
|
| @@ -92,37 +85,18 @@
|
| 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();
|
| + void Reconfigure();
|
|
|
| std::string id_;
|
| - std::string stream_id_;
|
| + rtc::scoped_refptr<AudioTrackInterface> track_;
|
| + uint32_t ssrc_;
|
| AudioProviderInterface* provider_;
|
| - StatsCollector* stats_;
|
| - rtc::scoped_refptr<AudioTrackInterface> track_;
|
| - uint32_t ssrc_ = 0;
|
| - bool cached_track_enabled_ = false;
|
| - bool stopped_ = false;
|
| + bool cached_track_enabled_;
|
|
|
| // Used to pass the data callback from the |track_| to the other end of
|
| // cricket::AudioRenderer.
|
| @@ -133,11 +107,8 @@
|
| public rtc::RefCountedObject<RtpSenderInterface> {
|
| public:
|
| VideoRtpSender(VideoTrackInterface* track,
|
| - const std::string& stream_id,
|
| + uint32_t ssrc,
|
| VideoProviderInterface* provider);
|
| -
|
| - // Randomly generates id and stream_id.
|
| - explicit VideoRtpSender(VideoProviderInterface* provider);
|
|
|
| virtual ~VideoRtpSender();
|
|
|
| @@ -150,36 +121,18 @@
|
| 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();
|
| + void Reconfigure();
|
|
|
| std::string id_;
|
| - std::string stream_id_;
|
| + rtc::scoped_refptr<VideoTrackInterface> track_;
|
| + uint32_t ssrc_;
|
| VideoProviderInterface* provider_;
|
| - rtc::scoped_refptr<VideoTrackInterface> track_;
|
| - uint32_t ssrc_ = 0;
|
| - bool cached_track_enabled_ = false;
|
| - bool stopped_ = false;
|
| + bool cached_track_enabled_;
|
| };
|
|
|
| } // namespace webrtc
|
|
|