Chromium Code Reviews| Index: webrtc/api/rtpsender.h |
| diff --git a/webrtc/api/rtpsender.h b/webrtc/api/rtpsender.h |
| index 067ae5e5b8720219c268b182292f287e5f0ecd3d..3b7faec631b081ad91f080ad5a701f4d67db51ed 100644 |
| --- a/webrtc/api/rtpsender.h |
| +++ b/webrtc/api/rtpsender.h |
| @@ -8,226 +8,11 @@ |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| -// This file contains classes that implement RtpSenderInterface. |
| -// An RtpSender associates a MediaStreamTrackInterface with an underlying |
| -// transport (provided by AudioProviderInterface/VideoProviderInterface) |
| - |
| #ifndef WEBRTC_API_RTPSENDER_H_ |
| #define WEBRTC_API_RTPSENDER_H_ |
| -#include <memory> |
| -#include <string> |
| - |
| -#include "webrtc/api/mediastreaminterface.h" |
| -#include "webrtc/api/rtpsenderinterface.h" |
| -#include "webrtc/api/statscollector.h" |
| -#include "webrtc/base/basictypes.h" |
| -#include "webrtc/base/criticalsection.h" |
| -#include "webrtc/media/base/audiosource.h" |
| -#include "webrtc/pc/channel.h" |
| - |
| -namespace webrtc { |
| - |
| -// Internal interface used by PeerConnection. |
| -class RtpSenderInternal : public RtpSenderInterface { |
| - public: |
| - // Used to set the SSRC of the sender, once a local description has been set. |
| - // If |ssrc| is 0, this indiates that the sender should disconnect from the |
| - // underlying transport (this occurs if the sender isn't seen in a local |
| - // description). |
| - virtual void SetSsrc(uint32_t ssrc) = 0; |
| - |
| - // TODO(deadbeef): Support one sender having multiple stream ids. |
| - virtual void set_stream_id(const std::string& stream_id) = 0; |
| - virtual std::string stream_id() const = 0; |
| - |
| - virtual void Stop() = 0; |
| -}; |
| - |
| -// LocalAudioSinkAdapter receives data callback as a sink to the local |
| -// AudioTrack, and passes the data to the sink of AudioSource. |
| -class LocalAudioSinkAdapter : public AudioTrackSinkInterface, |
| - public cricket::AudioSource { |
| - 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::AudioSource implementation. |
| - void SetSink(cricket::AudioSource::Sink* sink) override; |
| - |
| - cricket::AudioSource::Sink* sink_; |
| - // Critical section protecting |sink_|. |
| - rtc::CriticalSection lock_; |
| -}; |
| - |
| -class AudioRtpSender : public ObserverInterface, |
| - public rtc::RefCountedObject<RtpSenderInternal> { |
| - public: |
| - // StatsCollector provided so that Add/RemoveLocalAudioTrack can be called |
| - // at the appropriate times. |
| - // |channel| can be null if one does not exist yet. |
| - AudioRtpSender(AudioTrackInterface* track, |
| - const std::string& stream_id, |
| - cricket::VoiceChannel* channel, |
| - StatsCollector* stats); |
| - |
| - // Randomly generates stream_id. |
| - // |channel| can be null if one does not exist yet. |
| - AudioRtpSender(AudioTrackInterface* track, |
| - cricket::VoiceChannel* channel, |
| - StatsCollector* stats); |
| - |
| - // Randomly generates id and stream_id. |
| - // |channel| can be null if one does not exist yet. |
| - AudioRtpSender(cricket::VoiceChannel* channel, 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_; |
| - } |
| - |
| - 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_; } |
| - |
| - std::vector<std::string> stream_ids() const override { |
| - std::vector<std::string> ret = {stream_id_}; |
| - return ret; |
| - } |
| - |
| - RtpParameters GetParameters() const override; |
| - bool SetParameters(const RtpParameters& parameters) override; |
| - |
| - // RtpSenderInternal implementation. |
| - void SetSsrc(uint32_t ssrc) override; |
| - |
| - 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; |
| - |
| - // Does not take ownership. |
| - // Should call SetChannel(nullptr) before |channel| is destroyed. |
| - void SetChannel(cricket::VoiceChannel* channel) { channel_ = channel; } |
| - |
| - private: |
| - // TODO(nisse): Since SSRC == 0 is technically valid, figure out |
| - // some other way to test if we have a valid SSRC. |
| - bool can_send_track() const { return track_ && ssrc_; } |
| - // Helper function to construct options for |
| - // AudioProviderInterface::SetAudioSend. |
| - void SetAudioSend(); |
| - // Helper function to call SetAudioSend with "stop sending" parameters. |
| - void ClearAudioSend(); |
| - |
| - std::string id_; |
| - std::string stream_id_; |
| - cricket::VoiceChannel* channel_ = nullptr; |
| - 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::AudioSource. |
| - std::unique_ptr<LocalAudioSinkAdapter> sink_adapter_; |
| -}; |
| - |
| -class VideoRtpSender : public ObserverInterface, |
| - public rtc::RefCountedObject<RtpSenderInternal> { |
| - public: |
| - // |channel| can be null if one does not exist yet. |
| - VideoRtpSender(VideoTrackInterface* track, |
| - const std::string& stream_id, |
| - cricket::VideoChannel* channel); |
| - |
| - // Randomly generates stream_id. |
| - // |channel| can be null if one does not exist yet. |
| - VideoRtpSender(VideoTrackInterface* track, cricket::VideoChannel* channel); |
| - |
| - // Randomly generates id and stream_id. |
| - // |channel| can be null if one does not exist yet. |
| - explicit VideoRtpSender(cricket::VideoChannel* channel); |
| - |
| - virtual ~VideoRtpSender(); |
| - |
| - // ObserverInterface implementation |
| - void OnChanged() override; |
| - |
| - // RtpSenderInterface implementation |
| - bool SetTrack(MediaStreamTrackInterface* track) override; |
| - rtc::scoped_refptr<MediaStreamTrackInterface> track() const override { |
| - return track_; |
| - } |
| - |
| - 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_; } |
| - |
| - std::vector<std::string> stream_ids() const override { |
| - std::vector<std::string> ret = {stream_id_}; |
| - return ret; |
| - } |
| - |
| - RtpParameters GetParameters() const override; |
| - bool SetParameters(const RtpParameters& parameters) override; |
| - |
| - // RtpSenderInternal implementation. |
| - void SetSsrc(uint32_t ssrc) override; |
| - |
| - 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; |
| - |
| - // Does not take ownership. |
| - // Should call SetChannel(nullptr) before |channel| is destroyed. |
| - void SetChannel(cricket::VideoChannel* channel) { channel_ = channel; } |
| - |
| - private: |
| - bool can_send_track() const { return track_ && ssrc_; } |
| - // Helper function to construct options for |
| - // VideoProviderInterface::SetVideoSend. |
| - void SetVideoSend(); |
| - // Helper function to call SetVideoSend with "stop sending" parameters. |
| - void ClearVideoSend(); |
| - |
| - std::string id_; |
| - std::string stream_id_; |
| - cricket::VideoChannel* channel_ = nullptr; |
| - rtc::scoped_refptr<VideoTrackInterface> track_; |
| - uint32_t ssrc_ = 0; |
| - bool cached_track_enabled_ = false; |
| - bool stopped_ = false; |
| -}; |
| - |
| -} // namespace webrtc |
| +// Including this file is deprecated. It is no longer part of the public API. |
| +// This only includes the file in its new location for backwards compatibility. |
| +#include "webrtc/pc/rtpsender.h" |
|
ossu
2016/12/09 15:15:29
Added for backwards-compatibility.
|
| #endif // WEBRTC_API_RTPSENDER_H_ |