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This is a reland of df5731e44d
with fixes
to avoid existing chromium tests to fail.
Instead of replacing the existing RtpSender::set_stream_ids() to
also fire OnRenegotiationNeeded(), this CL keeps the old
set_stream_ids() and adds the new RtpSender::SetStreams() which sets
the stream IDs and fires the callback.
This allows existing callsites to maintain behavior, and reserve
SetStreams() for the cases when we want OnRenegotiationNeeded() to fire.
Using the SetStreams() name instead of SetStreamIDs() to match the W3C
spec and to make it more different that the existing set_stream_ids().
Original change's description:
> Improve spec compliance of SetStreamIDs in RtpSenderInterface
>
> This CL makes RtpSender::SetStreamIDs fire fire negotiationneeded
> event if needed and exposes the method on RtpSenderInterface.
>
> This is a spec-compliance change.
>
> Bug: webrtc:10129
> Change-Id: I2b98b92665c847102838b094516a79b24de0e47e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135121
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27974}
Bug: webrtc:10129
Change-Id: Ic0b322bfa25c157e3a39465ef8b486f898eaf6bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137439
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27992}
344 lines
13 KiB
C++
344 lines
13 KiB
C++
/*
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* Copyright 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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// This file contains classes that implement RtpSenderInterface.
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// An RtpSender associates a MediaStreamTrackInterface with an underlying
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// transport (provided by AudioProviderInterface/VideoProviderInterface)
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#ifndef PC_RTP_SENDER_H_
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#define PC_RTP_SENDER_H_
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#include <memory>
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#include <string>
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#include <vector>
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#include "api/media_stream_interface.h"
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#include "api/rtp_sender_interface.h"
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#include "media/base/audio_source.h"
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#include "media/base/media_channel.h"
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#include "pc/dtmf_sender.h"
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#include "rtc_base/critical_section.h"
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namespace webrtc {
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class StatsCollector;
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bool UnimplementedRtpParameterHasValue(const RtpParameters& parameters);
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// Internal interface used by PeerConnection.
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class RtpSenderInternal : public RtpSenderInterface {
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public:
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// Sets the underlying MediaEngine channel associated with this RtpSender.
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// A VoiceMediaChannel should be used for audio RtpSenders and
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// a VideoMediaChannel should be used for video RtpSenders.
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// Must call SetMediaChannel(nullptr) before the media channel is destroyed.
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virtual void SetMediaChannel(cricket::MediaChannel* media_channel) = 0;
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// Used to set the SSRC of the sender, once a local description has been set.
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// If |ssrc| is 0, this indiates that the sender should disconnect from the
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// underlying transport (this occurs if the sender isn't seen in a local
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// description).
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virtual void SetSsrc(uint32_t ssrc) = 0;
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virtual void set_stream_ids(const std::vector<std::string>& stream_ids) = 0;
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virtual void set_init_send_encodings(
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const std::vector<RtpEncodingParameters>& init_send_encodings) = 0;
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virtual void set_transport(
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rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) = 0;
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virtual void Stop() = 0;
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// |GetParameters| and |SetParameters| operate with a transactional model.
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// Allow access to get/set parameters without invalidating transaction id.
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virtual RtpParameters GetParametersInternal() const = 0;
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virtual RTCError SetParametersInternal(const RtpParameters& parameters) = 0;
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// Returns an ID that changes every time SetTrack() is called, but
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// otherwise remains constant. Used to generate IDs for stats.
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// The special value zero means that no track is attached.
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virtual int AttachmentId() const = 0;
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// Disables the layers identified by the specified RIDs.
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// If the specified list is empty, this is a no-op.
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virtual RTCError DisableEncodingLayers(
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const std::vector<std::string>& rid) = 0;
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};
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// Shared implementation for RtpSenderInternal interface.
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class RtpSenderBase : public RtpSenderInternal, public ObserverInterface {
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public:
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class SetStreamsObserver {
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public:
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virtual ~SetStreamsObserver() = default;
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virtual void OnSetStreams() = 0;
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};
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// Sets the underlying MediaEngine channel associated with this RtpSender.
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// A VoiceMediaChannel should be used for audio RtpSenders and
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// a VideoMediaChannel should be used for video RtpSenders.
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// Must call SetMediaChannel(nullptr) before the media channel is destroyed.
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void SetMediaChannel(cricket::MediaChannel* media_channel) override;
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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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RtpParameters GetParameters() const override;
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RTCError SetParameters(const RtpParameters& parameters) override;
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// |GetParameters| and |SetParameters| operate with a transactional model.
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// Allow access to get/set parameters without invalidating transaction id.
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RtpParameters GetParametersInternal() const override;
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RTCError SetParametersInternal(const RtpParameters& parameters) override;
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// Used to set the SSRC of the sender, once a local description has been set.
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// If |ssrc| is 0, this indiates that the sender should disconnect from the
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// underlying transport (this occurs if the sender isn't seen in a local
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// description).
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void SetSsrc(uint32_t ssrc) override;
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uint32_t ssrc() const override { return ssrc_; }
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std::vector<std::string> stream_ids() const override { return stream_ids_; }
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void set_stream_ids(const std::vector<std::string>& stream_ids) override {
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stream_ids_ = stream_ids;
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}
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void SetStreams(const std::vector<std::string>& stream_ids) override;
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std::string id() const override { return id_; }
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void set_init_send_encodings(
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const std::vector<RtpEncodingParameters>& init_send_encodings) override {
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init_parameters_.encodings = init_send_encodings;
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}
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std::vector<RtpEncodingParameters> init_send_encodings() const override {
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return init_parameters_.encodings;
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}
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void set_transport(
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rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) override {
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dtls_transport_ = dtls_transport;
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}
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rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const override {
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return dtls_transport_;
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}
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void SetFrameEncryptor(
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rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) override;
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rtc::scoped_refptr<FrameEncryptorInterface> GetFrameEncryptor()
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const override {
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return frame_encryptor_;
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}
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void Stop() override;
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// Returns an ID that changes every time SetTrack() is called, but
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// otherwise remains constant. Used to generate IDs for stats.
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// The special value zero means that no track is attached.
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int AttachmentId() const override { return attachment_id_; }
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// Disables the layers identified by the specified RIDs.
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// If the specified list is empty, this is a no-op.
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RTCError DisableEncodingLayers(const std::vector<std::string>& rid) override;
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protected:
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// If |set_streams_observer| is not null, it is invoked when SetStreams()
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// is called. |set_streams_observer| is not owned by this object. If not
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// null, it must be valid at least until this sender becomes stopped.
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RtpSenderBase(rtc::Thread* worker_thread,
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const std::string& id,
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SetStreamsObserver* set_streams_observer);
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// TODO(nisse): Since SSRC == 0 is technically valid, figure out
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// 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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virtual std::string track_kind() const = 0;
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// Enable sending on the media channel.
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virtual void SetSend() = 0;
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// Disable sending on the media channel.
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virtual void ClearSend() = 0;
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// Template method pattern to allow subclasses to add custom behavior for
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// when tracks are attached, detached, and for adding tracks to statistics.
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virtual void AttachTrack() {}
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virtual void DetachTrack() {}
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virtual void AddTrackToStats() {}
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virtual void RemoveTrackFromStats() {}
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rtc::Thread* worker_thread_;
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uint32_t ssrc_ = 0;
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bool stopped_ = false;
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int attachment_id_ = 0;
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const std::string id_;
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std::vector<std::string> stream_ids_;
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RtpParameters init_parameters_;
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cricket::MediaChannel* media_channel_ = nullptr;
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rtc::scoped_refptr<MediaStreamTrackInterface> track_;
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rtc::scoped_refptr<DtlsTransportInterface> dtls_transport_;
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rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor_;
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// |last_transaction_id_| is used to verify that |SetParameters| is receiving
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// the parameters object that was last returned from |GetParameters|.
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// As such, it is used for internal verification and is not observable by the
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// the client. It is marked as mutable to enable |GetParameters| to be a
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// const method.
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mutable absl::optional<std::string> last_transaction_id_;
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std::vector<std::string> disabled_rids_;
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SetStreamsObserver* set_streams_observer_ = nullptr;
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};
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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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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,
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size_t number_of_channels,
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size_t number_of_frames) override;
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// cricket::AudioSource implementation.
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void SetSink(cricket::AudioSource::Sink* sink) override;
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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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class AudioRtpSender : public DtmfProviderInterface, public RtpSenderBase {
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public:
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// Construct an RtpSender for audio with the given sender ID.
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// The sender is initialized with no track to send and no associated streams.
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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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// If |set_streams_observer| is not null, it is invoked when SetStreams()
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// is called. |set_streams_observer| is not owned by this object. If not
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// null, it must be valid at least until this sender becomes stopped.
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static rtc::scoped_refptr<AudioRtpSender> Create(
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rtc::Thread* worker_thread,
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const std::string& id,
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StatsCollector* stats,
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SetStreamsObserver* set_streams_observer);
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virtual ~AudioRtpSender();
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// DtmfSenderProvider implementation.
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bool CanInsertDtmf() override;
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bool InsertDtmf(int code, int duration) override;
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sigslot::signal0<>* GetOnDestroyedSignal() override;
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// ObserverInterface implementation.
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void OnChanged() override;
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cricket::MediaType media_type() const override {
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return cricket::MEDIA_TYPE_AUDIO;
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}
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std::string track_kind() const override {
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return MediaStreamTrackInterface::kAudioKind;
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}
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rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const override;
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protected:
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AudioRtpSender(rtc::Thread* worker_thread,
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const std::string& id,
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StatsCollector* stats,
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SetStreamsObserver* set_streams_observer);
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void SetSend() override;
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void ClearSend() override;
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// Hooks to allow custom logic when tracks are attached and detached.
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void AttachTrack() override;
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void DetachTrack() override;
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void AddTrackToStats() override;
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void RemoveTrackFromStats() override;
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private:
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cricket::VoiceMediaChannel* voice_media_channel() {
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return static_cast<cricket::VoiceMediaChannel*>(media_channel_);
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}
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rtc::scoped_refptr<AudioTrackInterface> audio_track() const {
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return rtc::scoped_refptr<AudioTrackInterface>(
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static_cast<AudioTrackInterface*>(track_.get()));
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}
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sigslot::signal0<> SignalDestroyed;
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StatsCollector* stats_ = nullptr;
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rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender_proxy_;
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bool cached_track_enabled_ = false;
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// Used to pass the data callback from the |track_| to the other end of
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// cricket::AudioSource.
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std::unique_ptr<LocalAudioSinkAdapter> sink_adapter_;
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};
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class VideoRtpSender : public RtpSenderBase {
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public:
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// Construct an RtpSender for video with the given sender ID.
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// The sender is initialized with no track to send and no associated streams.
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// If |set_streams_observer| is not null, it is invoked when SetStreams()
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// is called. |set_streams_observer| is not owned by this object. If not
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// null, it must be valid at least until this sender becomes stopped.
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static rtc::scoped_refptr<VideoRtpSender> Create(
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rtc::Thread* worker_thread,
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const std::string& id,
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SetStreamsObserver* set_streams_observer);
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virtual ~VideoRtpSender();
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// ObserverInterface implementation
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void OnChanged() override;
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cricket::MediaType media_type() const override {
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return cricket::MEDIA_TYPE_VIDEO;
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}
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std::string track_kind() const override {
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return MediaStreamTrackInterface::kVideoKind;
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}
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rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const override;
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protected:
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VideoRtpSender(rtc::Thread* worker_thread,
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const std::string& id,
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SetStreamsObserver* set_streams_observer);
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void SetSend() override;
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void ClearSend() override;
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// Hook to allow custom logic when tracks are attached.
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void AttachTrack() override;
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private:
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cricket::VideoMediaChannel* video_media_channel() {
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return static_cast<cricket::VideoMediaChannel*>(media_channel_);
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}
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rtc::scoped_refptr<VideoTrackInterface> video_track() const {
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return rtc::scoped_refptr<VideoTrackInterface>(
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static_cast<VideoTrackInterface*>(track_.get()));
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}
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VideoTrackInterface::ContentHint cached_track_content_hint_ =
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VideoTrackInterface::ContentHint::kNone;
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};
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} // namespace webrtc
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#endif // PC_RTP_SENDER_H_
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