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// Copyright 2017 The Chromium Authors
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#ifndef THIRD_PARTY_BLINK_RENDERER_MODULES_PEERCONNECTION_RTC_RTP_SENDER_H_
#define THIRD_PARTY_BLINK_RENDERER_MODULES_PEERCONNECTION_RTC_RTP_SENDER_H_
#include <memory>
#include "base/task/single_thread_task_runner.h"
#include "base/threading/thread_checker.h"
#include "third_party/blink/renderer/bindings/core/v8/script_promise.h"
#include "third_party/blink/renderer/bindings/modules/v8/v8_rtc_rtp_encoding_parameters.h"
#include "third_party/blink/renderer/bindings/modules/v8/v8_rtc_rtp_send_parameters.h"
#include "third_party/blink/renderer/bindings/modules/v8/v8_rtc_set_parameter_options.h"
#include "third_party/blink/renderer/core/execution_context/execution_context_lifecycle_observer.h"
#include "third_party/blink/renderer/modules/mediastream/media_stream.h"
#include "third_party/blink/renderer/modules/peerconnection/rtc_rtp_script_transform.h"
#include "third_party/blink/renderer/modules/peerconnection/rtc_rtp_send_stream.h"
#include "third_party/blink/renderer/platform/bindings/script_wrappable.h"
#include "third_party/blink/renderer/platform/heap/garbage_collected.h"
#include "third_party/blink/renderer/platform/heap/member.h"
#include "third_party/blink/renderer/platform/heap/visitor.h"
#include "third_party/blink/renderer/platform/peerconnection/rtc_encoded_audio_stream_transformer.h"
#include "third_party/blink/renderer/platform/peerconnection/rtc_encoded_video_stream_transformer.h"
#include "third_party/blink/renderer/platform/peerconnection/rtc_rtp_sender_platform.h"
#include "third_party/blink/renderer/platform/wtf/text/wtf_string.h"
#include "third_party/webrtc/api/rtp_transceiver_interface.h"
namespace blink {
class ExceptionState;
class MediaStreamTrack;
class RTCDtlsTransport;
class RTCDTMFSender;
class RTCEncodedAudioUnderlyingSink;
class RTCEncodedAudioUnderlyingSource;
class RTCEncodedVideoUnderlyingSink;
class RTCEncodedVideoUnderlyingSource;
class RTCInsertableStreams;
class RTCPeerConnection;
class RTCRtpCapabilities;
class RTCRtpTransceiver;
class RTCInsertableStreams;
class RTCStatsReport;
class RTCRtpScriptTransform;
webrtc::RtpEncodingParameters ToRtpEncodingParameters(
ExecutionContext* context,
const RTCRtpEncodingParameters*,
const String& kind);
RTCRtpHeaderExtensionParameters* ToRtpHeaderExtensionParameters(
const webrtc::RtpExtension& headers);
RTCRtpCodecParameters* ToRtpCodecParameters(
const webrtc::RtpCodecParameters& codecs);
// https://w3c.github.io/webrtc-pc/#rtcrtpsender-interface
class RTCRtpSender final : public ScriptWrappable,
public ExecutionContextLifecycleObserver {
DEFINE_WRAPPERTYPEINFO();
public:
// If |require_encoded_insertable_streams| is true, no received frames will be
// passed to the packetizer until |createEncodedStreams()| has been called and
// the frames have been transformed and passed back to the returned
// WritableStream. If it's false, during construction a task will be posted to
// |encoded_transform_shortcircuit_runner| to check if
// |createEncodedStreams()| has been called yet and if not will tell the
// underlying WebRTC sender to 'short circuit' the transform, so frames will
// flow directly to the packetizer.
// TODO(hbos): Get rid of sender's reference to RTCPeerConnection?
// https://github.com/w3c/webrtc-pc/issues/1712
RTCRtpSender(RTCPeerConnection*,
std::unique_ptr<RTCRtpSenderPlatform>,
String kind,
MediaStreamTrack*,
MediaStreamVector streams,
bool require_encoded_insertable_streams,
scoped_refptr<base::SequencedTaskRunner>
encoded_transform_shortcircuit_runner);
MediaStreamTrack* track();
RTCDtlsTransport* transport();
RTCDtlsTransport* rtcpTransport();
ScriptPromise<IDLUndefined> replaceTrack(ScriptState*,
MediaStreamTrack*,
ExceptionState&);
RTCDTMFSender* dtmf();
static RTCRtpCapabilities* getCapabilities(ScriptState* state,
const String& kind);
RTCRtpSendParameters* getParameters();
ScriptPromise<IDLUndefined> setParameters(ScriptState*,
const RTCRtpSendParameters*,
const RTCSetParameterOptions*,
ExceptionState&);
ScriptPromise<RTCStatsReport> getStats(ScriptState*);
void setStreams(HeapVector<Member<MediaStream>> streams, ExceptionState&);
RTCInsertableStreams* createEncodedStreams(ScriptState*, ExceptionState&);
RTCRtpScriptTransform* transform() { return transform_; }
void setTransform(RTCRtpScriptTransform*, ExceptionState& exception_state);
RTCRtpSenderPlatform* web_sender();
// Sets the track. This must be called when the |RTCRtpSenderPlatform| has its
// track updated, and the |track| must match the
// |RTCRtpSenderPlatform::Track|.
void SetTrack(MediaStreamTrack*);
void ClearLastReturnedParameters();
MediaStreamVector streams() const;
void set_streams(MediaStreamVector streams);
void set_transceiver(RTCRtpTransceiver*);
void set_transport(RTCDtlsTransport*);
ScriptPromise<IDLSequence<RTCRtpSendStream>> replaceSendStreams() {
// TODO(crbug.com/345101934): Implement me.
return {};
}
// ExecutionContextLifecycleObserver
void ContextDestroyed() override;
void Trace(Visitor*) const override;
private:
// Insertable Streams audio support methods
RTCInsertableStreams* CreateEncodedAudioStreams(ScriptState*);
void RegisterEncodedAudioStreamCallback();
void UnregisterEncodedAudioStreamCallback();
void SetAudioUnderlyingSource(
RTCEncodedAudioUnderlyingSource* new_underlying_source,
scoped_refptr<base::SingleThreadTaskRunner> task_runner);
void SetAudioUnderlyingSink(
RTCEncodedAudioUnderlyingSink* new_underlying_sink);
void OnAudioFrameFromEncoder(
std::unique_ptr<webrtc::TransformableAudioFrameInterface> frame);
// Insertable Streams video support methods
RTCInsertableStreams* CreateEncodedVideoStreams(ScriptState*);
void RegisterEncodedVideoStreamCallback();
void UnregisterEncodedVideoStreamCallback();
void SetVideoUnderlyingSource(
RTCEncodedVideoUnderlyingSource* new_underlying_source,
scoped_refptr<base::SingleThreadTaskRunner> task_runner);
void SetVideoUnderlyingSink(
RTCEncodedVideoUnderlyingSink* new_underlying_sink);
void OnVideoFrameFromEncoder(
std::unique_ptr<webrtc::TransformableVideoFrameInterface> frame);
void LogMessage(const std::string& message);
// If createEncodedStreams has not yet been called, instead tell the webrtc
// encoded transform to 'short circuit', skipping calling the transform.
void MaybeShortCircuitEncodedStreams();
Member<RTCPeerConnection> pc_;
std::unique_ptr<RTCRtpSenderPlatform> sender_;
String kind_;
Member<MediaStreamTrack> track_;
Member<RTCDtlsTransport> transport_;
Member<RTCDTMFSender> dtmf_;
MediaStreamVector streams_;
Member<RTCRtpSendParameters> last_returned_parameters_;
Member<RTCRtpTransceiver> transceiver_;
bool transform_shortcircuited_ = false;
Member<RTCInsertableStreams> encoded_streams_;
Member<RTCRtpScriptTransform> transform_;
// Insertable Streams audio support
base::Lock audio_underlying_source_lock_;
CrossThreadPersistent<RTCEncodedAudioUnderlyingSource>
audio_from_encoder_underlying_source_
GUARDED_BY(audio_underlying_source_lock_);
base::Lock audio_underlying_sink_lock_;
CrossThreadPersistent<RTCEncodedAudioUnderlyingSink>
audio_to_packetizer_underlying_sink_
GUARDED_BY(audio_underlying_sink_lock_);
const scoped_refptr<blink::RTCEncodedAudioStreamTransformer::Broker>
encoded_audio_transformer_;
// Insertable Streams video support
base::Lock video_underlying_source_lock_;
CrossThreadPersistent<RTCEncodedVideoUnderlyingSource>
video_from_encoder_underlying_source_
GUARDED_BY(video_underlying_source_lock_);
base::Lock video_underlying_sink_lock_;
CrossThreadPersistent<RTCEncodedVideoUnderlyingSink>
video_to_packetizer_underlying_sink_
GUARDED_BY(video_underlying_sink_lock_);
const scoped_refptr<blink::RTCEncodedVideoStreamTransformer::Broker>
encoded_video_transformer_;
THREAD_CHECKER(thread_checker_);
};
} // namespace blink
#endif // THIRD_PARTY_BLINK_RENDERER_MODULES_PEERCONNECTION_RTC_RTP_SENDER_H_
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