| Index: webrtc/api/rtpsender.h
|
| diff --git a/webrtc/api/rtpsender.h b/webrtc/api/rtpsender.h
|
| index 6c0592148bb06d264bdb84b3bdff51b262b03e08..3b7faec631b081ad91f080ad5a701f4d67db51ed 100644
|
| --- a/webrtc/api/rtpsender.h
|
| +++ b/webrtc/api/rtpsender.h
|
| @@ -8,228 +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.
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| - virtual void set_stream_id(const std::string& stream_id) = 0;
|
| - virtual std::string stream_id() const = 0;
|
| -
|
| - virtual void Stop() = 0;
|
| -};
|
| -
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| -// LocalAudioSinkAdapter receives data callback as a sink to the local
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| -// AudioTrack, and passes the data to the sink of AudioSource.
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| -class LocalAudioSinkAdapter : public AudioTrackSinkInterface,
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| - public cricket::AudioSource {
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| - public:
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| - LocalAudioSinkAdapter();
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| - virtual ~LocalAudioSinkAdapter();
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| -
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| - private:
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| - // AudioSinkInterface implementation.
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| - void OnData(const void* audio_data,
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| - int bits_per_sample,
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| - int sample_rate,
|
| - size_t number_of_channels,
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| - size_t number_of_frames) override;
|
| -
|
| - // cricket::AudioSource implementation.
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| - void SetSink(cricket::AudioSource::Sink* sink) override;
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| -
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| - cricket::AudioSource::Sink* sink_;
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| - // Critical section protecting |sink_|.
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| - rtc::CriticalSection lock_;
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| -};
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| -
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| -class AudioRtpSender : public ObserverInterface,
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| - public rtc::RefCountedObject<RtpSenderInternal> {
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| - public:
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| - // StatsCollector provided so that Add/RemoveLocalAudioTrack can be called
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| - // at the appropriate times.
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| - // |channel| can be null if one does not exist yet.
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| - AudioRtpSender(AudioTrackInterface* track,
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| - const std::string& stream_id,
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| - cricket::VoiceChannel* channel,
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| - StatsCollector* stats);
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| -
|
| - // Randomly generates stream_id.
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| - // |channel| can be null if one does not exist yet.
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| - AudioRtpSender(AudioTrackInterface* track,
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| - cricket::VoiceChannel* channel,
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| - StatsCollector* stats);
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| -
|
| - // Randomly generates id and stream_id.
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| - // |channel| can be null if one does not exist yet.
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| - AudioRtpSender(cricket::VoiceChannel* channel, StatsCollector* stats);
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| -
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| - virtual ~AudioRtpSender();
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| -
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| - // ObserverInterface implementation
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| - void OnChanged() override;
|
| -
|
| - // RtpSenderInterface implementation
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| - bool SetTrack(MediaStreamTrackInterface* track) override;
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| - rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
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| - return track_;
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| - }
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| -
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| - uint32_t ssrc() const override { return ssrc_; }
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| -
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| - cricket::MediaType media_type() const override {
|
| - return cricket::MEDIA_TYPE_AUDIO;
|
| - }
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| -
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| - std::string id() const override { return id_; }
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| -
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| - std::vector<std::string> stream_ids() const override {
|
| - std::vector<std::string> ret = {stream_id_};
|
| - return ret;
|
| - }
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| -
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| - RtpParameters GetParameters() const override;
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| - bool SetParameters(const RtpParameters& parameters) override;
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| -
|
| - // RtpSenderInternal implementation.
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| - void SetSsrc(uint32_t ssrc) override;
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| -
|
| - void set_stream_id(const std::string& stream_id) override {
|
| - stream_id_ = stream_id;
|
| - }
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| - std::string stream_id() const override { return stream_id_; }
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| -
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| - void Stop() override;
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| -
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| - // Does not take ownership.
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| - // Should call SetChannel(nullptr) before |channel| is destroyed.
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| - void SetChannel(cricket::VoiceChannel* channel) { channel_ = channel; }
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| -
|
| - private:
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| - // TODO(nisse): Since SSRC == 0 is technically valid, figure out
|
| - // some other way to test if we have a valid SSRC.
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| - bool can_send_track() const { return track_ && ssrc_; }
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| - // Helper function to construct options for
|
| - // AudioProviderInterface::SetAudioSend.
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| - void SetAudioSend();
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| - // Helper function to call SetAudioSend with "stop sending" parameters.
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| - void ClearAudioSend();
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| -
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| - std::string id_;
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| - std::string stream_id_;
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| - cricket::VoiceChannel* channel_ = nullptr;
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| - StatsCollector* stats_;
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| - 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
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| - // cricket::AudioSource.
|
| - std::unique_ptr<LocalAudioSinkAdapter> sink_adapter_;
|
| -};
|
| -
|
| -class VideoRtpSender : public ObserverInterface,
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| - public rtc::RefCountedObject<RtpSenderInternal> {
|
| - public:
|
| - // |channel| can be null if one does not exist yet.
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| - 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;
|
| - VideoTrackInterface::ContentHint cached_track_content_hint_ =
|
| - VideoTrackInterface::ContentHint::kNone;
|
| - 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"
|
|
|
| #endif // WEBRTC_API_RTPSENDER_H_
|
|
|