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Unified Diff: talk/app/webrtc/rtpsender.h

Issue 1426443007: Revert of Adding the ability to create an RtpSender without a track. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 2 months ago
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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
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