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Diffstat (limited to 'third_party/libwebrtc/media/base/media_channel_impl.h')
| -rw-r--r-- | third_party/libwebrtc/media/base/media_channel_impl.h | 683 |
1 files changed, 683 insertions, 0 deletions
diff --git a/third_party/libwebrtc/media/base/media_channel_impl.h b/third_party/libwebrtc/media/base/media_channel_impl.h new file mode 100644 index 0000000000..8142dd45b6 --- /dev/null +++ b/third_party/libwebrtc/media/base/media_channel_impl.h @@ -0,0 +1,683 @@ +/* + * Copyright 2022 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef MEDIA_BASE_MEDIA_CHANNEL_IMPL_H_ +#define MEDIA_BASE_MEDIA_CHANNEL_IMPL_H_ + +#include <stddef.h> +#include <stdint.h> + +#include <functional> +#include <memory> +#include <string> +#include <utility> +#include <vector> + +#include "absl/strings/string_view.h" +#include "absl/types/optional.h" +#include "api/audio_options.h" +#include "api/call/audio_sink.h" +#include "api/call/transport.h" +#include "api/crypto/frame_decryptor_interface.h" +#include "api/crypto/frame_encryptor_interface.h" +#include "api/frame_transformer_interface.h" +#include "api/media_types.h" +#include "api/rtc_error.h" +#include "api/rtp_parameters.h" +#include "api/rtp_sender_interface.h" +#include "api/scoped_refptr.h" +#include "api/sequence_checker.h" +#include "api/task_queue/pending_task_safety_flag.h" +#include "api/task_queue/task_queue_base.h" +#include "api/transport/rtp/rtp_source.h" +#include "api/video/recordable_encoded_frame.h" +#include "api/video/video_frame.h" +#include "api/video/video_sink_interface.h" +#include "api/video/video_source_interface.h" +#include "api/video_codecs/video_encoder_factory.h" +#include "media/base/codec.h" +#include "media/base/media_channel.h" +#include "media/base/stream_params.h" +#include "modules/rtp_rtcp/source/rtp_packet_received.h" +#include "rtc_base/async_packet_socket.h" +#include "rtc_base/checks.h" +#include "rtc_base/copy_on_write_buffer.h" +#include "rtc_base/dscp.h" +#include "rtc_base/logging.h" +#include "rtc_base/network/sent_packet.h" +#include "rtc_base/network_route.h" +#include "rtc_base/socket.h" +#include "rtc_base/thread_annotations.h" +// This file contains the base classes for classes that implement +// the MediaChannel interfaces. +// These implementation classes used to be the exposed interface names, +// but this is in the process of being changed. +// TODO(bugs.webrtc.org/13931): Consider removing these classes. + +// The target + +namespace cricket { + +class VoiceMediaChannel; +class VideoMediaChannel; + +class MediaChannel : public MediaSendChannelInterface, + public MediaReceiveChannelInterface { + public: + explicit MediaChannel(webrtc::TaskQueueBase* network_thread, + bool enable_dscp = false); + virtual ~MediaChannel(); + + // Downcasting to the subclasses. + virtual VideoMediaChannel* AsVideoChannel() { + RTC_CHECK_NOTREACHED(); + return nullptr; + } + + virtual VoiceMediaChannel* AsVoiceChannel() { + RTC_CHECK_NOTREACHED(); + return nullptr; + } + + // Must declare the methods inherited from the base interface template, + // even when abstract, to tell the compiler that all instances of the name + // referred to by subclasses of this share the same implementation. + cricket::MediaType media_type() const override = 0; + void OnPacketReceived(const webrtc::RtpPacketReceived& packet) override = 0; + void OnPacketSent(const rtc::SentPacket& sent_packet) override = 0; + void OnReadyToSend(bool ready) override = 0; + void OnNetworkRouteChanged(absl::string_view transport_name, + const rtc::NetworkRoute& network_route) override = + 0; + + // Sets the abstract interface class for sending RTP/RTCP data. + virtual void SetInterface(MediaChannelNetworkInterface* iface); + // Returns the absolute sendtime extension id value from media channel. + virtual int GetRtpSendTimeExtnId() const; + // Base method to send packet using MediaChannelNetworkInterface. + bool SendPacket(rtc::CopyOnWriteBuffer* packet, + const rtc::PacketOptions& options); + + bool SendRtcp(rtc::CopyOnWriteBuffer* packet, + const rtc::PacketOptions& options); + + int SetOption(MediaChannelNetworkInterface::SocketType type, + rtc::Socket::Option opt, + int option); + + // Corresponds to the SDP attribute extmap-allow-mixed, see RFC8285. + // Set to true if it's allowed to mix one- and two-byte RTP header extensions + // in the same stream. The setter and getter must only be called from + // worker_thread. + void SetExtmapAllowMixed(bool extmap_allow_mixed) override; + bool ExtmapAllowMixed() const override; + + // Returns `true` if a non-null MediaChannelNetworkInterface pointer is held. + // Must be called on the network thread. + bool HasNetworkInterface() const; + + void SetFrameEncryptor(uint32_t ssrc, + rtc::scoped_refptr<webrtc::FrameEncryptorInterface> + frame_encryptor) override; + void SetFrameDecryptor(uint32_t ssrc, + rtc::scoped_refptr<webrtc::FrameDecryptorInterface> + frame_decryptor) override; + + void SetEncoderToPacketizerFrameTransformer( + uint32_t ssrc, + rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) + override; + void SetDepacketizerToDecoderFrameTransformer( + uint32_t ssrc, + rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) + override; + + protected: + int SetOptionLocked(MediaChannelNetworkInterface::SocketType type, + rtc::Socket::Option opt, + int option) RTC_RUN_ON(network_thread_); + + bool DscpEnabled() const; + + // This is the DSCP value used for both RTP and RTCP channels if DSCP is + // enabled. It can be changed at any time via `SetPreferredDscp`. + rtc::DiffServCodePoint PreferredDscp() const; + void SetPreferredDscp(rtc::DiffServCodePoint new_dscp); + + rtc::scoped_refptr<webrtc::PendingTaskSafetyFlag> network_safety(); + + // Utility implementation for derived classes (video/voice) that applies + // the packet options and passes the data onwards to `SendPacket`. + void SendRtp(const uint8_t* data, + size_t len, + const webrtc::PacketOptions& options); + + void SendRtcp(const uint8_t* data, size_t len); + + private: + // Apply the preferred DSCP setting to the underlying network interface RTP + // and RTCP channels. If DSCP is disabled, then apply the default DSCP value. + void UpdateDscp() RTC_RUN_ON(network_thread_); + + bool DoSendPacket(rtc::CopyOnWriteBuffer* packet, + bool rtcp, + const rtc::PacketOptions& options); + + const bool enable_dscp_; + const rtc::scoped_refptr<webrtc::PendingTaskSafetyFlag> network_safety_ + RTC_PT_GUARDED_BY(network_thread_); + webrtc::TaskQueueBase* const network_thread_; + MediaChannelNetworkInterface* network_interface_ + RTC_GUARDED_BY(network_thread_) = nullptr; + rtc::DiffServCodePoint preferred_dscp_ RTC_GUARDED_BY(network_thread_) = + rtc::DSCP_DEFAULT; + bool extmap_allow_mixed_ = false; +}; + +// Base class for implementation classes + +class VideoMediaChannel : public MediaChannel, + public VideoMediaSendChannelInterface, + public VideoMediaReceiveChannelInterface { + public: + explicit VideoMediaChannel(webrtc::TaskQueueBase* network_thread, + bool enable_dscp = false) + : MediaChannel(network_thread, enable_dscp) {} + ~VideoMediaChannel() override {} + + // Downcasting to the implemented interfaces. + VideoMediaSendChannelInterface* AsVideoSendChannel() override { return this; } + VoiceMediaSendChannelInterface* AsVoiceSendChannel() override { + RTC_CHECK_NOTREACHED(); + return nullptr; + } + + VideoMediaReceiveChannelInterface* AsVideoReceiveChannel() override { + return this; + } + cricket::MediaType media_type() const override; + + // Downcasting to the subclasses. + VideoMediaChannel* AsVideoChannel() override { return this; } + + void SetExtmapAllowMixed(bool mixed) override { + MediaChannel::SetExtmapAllowMixed(mixed); + } + bool ExtmapAllowMixed() const override { + return MediaChannel::ExtmapAllowMixed(); + } + // This fills the "bitrate parts" (rtx, video bitrate) of the + // BandwidthEstimationInfo, since that part that isn't possible to get + // through webrtc::Call::GetStats, as they are statistics of the send + // streams. + // TODO(holmer): We should change this so that either BWE graphs doesn't + // need access to bitrates of the streams, or change the (RTC)StatsCollector + // so that it's getting the send stream stats separately by calling + // GetStats(), and merges with BandwidthEstimationInfo by itself. + void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) override = 0; + // Gets quality stats for the channel. + virtual bool GetSendStats(VideoMediaSendInfo* info) = 0; + virtual bool GetReceiveStats(VideoMediaReceiveInfo* info) = 0; + // Enable network condition based codec switching. + void SetVideoCodecSwitchingEnabled(bool enabled) override; + + private: + // Functions not implemented on this interface + bool GetStats(VideoMediaSendInfo* info) override { + RTC_CHECK_NOTREACHED(); + return false; + } + bool GetStats(VideoMediaReceiveInfo* info) override { + RTC_CHECK_NOTREACHED(); + return false; + } +}; + +// Base class for implementation classes +class VoiceMediaChannel : public MediaChannel, + public VoiceMediaSendChannelInterface, + public VoiceMediaReceiveChannelInterface { + public: + MediaType media_type() const override; + VoiceMediaChannel(webrtc::TaskQueueBase* network_thread, + bool enable_dscp = false) + : MediaChannel(network_thread, enable_dscp) {} + ~VoiceMediaChannel() override {} + + // Downcasting to the implemented interfaces. + VoiceMediaSendChannelInterface* AsVoiceSendChannel() override { return this; } + + VoiceMediaReceiveChannelInterface* AsVoiceReceiveChannel() override { + return this; + } + + VoiceMediaChannel* AsVoiceChannel() override { return this; } + + VideoMediaSendChannelInterface* AsVideoSendChannel() override { + RTC_CHECK_NOTREACHED(); + return nullptr; + } + + void SetExtmapAllowMixed(bool mixed) override { + MediaChannel::SetExtmapAllowMixed(mixed); + } + bool ExtmapAllowMixed() const override { + return MediaChannel::ExtmapAllowMixed(); + } + + // Gets quality stats for the channel. + virtual bool GetSendStats(VoiceMediaSendInfo* info) = 0; + virtual bool GetReceiveStats(VoiceMediaReceiveInfo* info, + bool get_and_clear_legacy_stats) = 0; + + private: + // Functions not implemented on this interface + bool GetStats(VoiceMediaSendInfo* info) override { + RTC_CHECK_NOTREACHED(); + return false; + } + bool GetStats(VoiceMediaReceiveInfo* info, + bool get_and_clear_legacy_stats) override { + RTC_CHECK_NOTREACHED(); + return false; + } +}; + +// The externally exposed objects that support the Send and Receive interfaces. +// These dispatch their functions to the underlying MediaChannel objects. + +class VoiceMediaSendChannel : public VoiceMediaSendChannelInterface { + public: + explicit VoiceMediaSendChannel(VoiceMediaChannel* impl) : impl_(impl) {} + virtual ~VoiceMediaSendChannel() {} + VoiceMediaSendChannelInterface* AsVoiceSendChannel() override { return this; } + VideoMediaSendChannelInterface* AsVideoSendChannel() override { + RTC_CHECK_NOTREACHED(); + return nullptr; + } + + // Implementation of MediaBaseChannelInterface + cricket::MediaType media_type() const override { return MEDIA_TYPE_AUDIO; } + void OnPacketReceived(const webrtc::RtpPacketReceived& packet) override { + impl()->OnPacketReceived(packet); + } + void OnPacketSent(const rtc::SentPacket& sent_packet) override { + impl()->OnPacketSent(sent_packet); + } + void OnReadyToSend(bool ready) override { impl()->OnReadyToSend(ready); } + void OnNetworkRouteChanged(absl::string_view transport_name, + const rtc::NetworkRoute& network_route) override { + impl()->OnNetworkRouteChanged(transport_name, network_route); + } + void SetExtmapAllowMixed(bool extmap_allow_mixed) override { + impl()->SetExtmapAllowMixed(extmap_allow_mixed); + } + bool ExtmapAllowMixed() const override { return impl()->ExtmapAllowMixed(); } + // Implementation of MediaSendChannelInterface + bool AddSendStream(const StreamParams& sp) override { + return impl()->AddSendStream(sp); + } + bool RemoveSendStream(uint32_t ssrc) override { + return impl()->RemoveSendStream(ssrc); + } + void SetFrameEncryptor(uint32_t ssrc, + rtc::scoped_refptr<webrtc::FrameEncryptorInterface> + frame_encryptor) override { + impl()->SetFrameEncryptor(ssrc, frame_encryptor); + } + webrtc::RTCError SetRtpSendParameters( + uint32_t ssrc, + const webrtc::RtpParameters& parameters, + webrtc::SetParametersCallback callback = nullptr) override { + return impl()->SetRtpSendParameters(ssrc, parameters, std::move(callback)); + } + + void SetEncoderToPacketizerFrameTransformer( + uint32_t ssrc, + rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) + override { + return impl()->SetEncoderToPacketizerFrameTransformer(ssrc, + frame_transformer); + } + void SetEncoderSelector(uint32_t ssrc, + webrtc::VideoEncoderFactory::EncoderSelectorInterface* + encoder_selector) override { + impl()->SetEncoderSelector(ssrc, encoder_selector); + } + webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override { + return impl()->GetRtpSendParameters(ssrc); + } + // Implementation of VoiceMediaSendChannel + bool SetSendParameters(const AudioSendParameters& params) override { + return impl()->SetSendParameters(params); + } + void SetSend(bool send) override { return impl()->SetSend(send); } + bool SetAudioSend(uint32_t ssrc, + bool enable, + const AudioOptions* options, + AudioSource* source) override { + return impl()->SetAudioSend(ssrc, enable, options, source); + } + bool CanInsertDtmf() override { return impl()->CanInsertDtmf(); } + bool InsertDtmf(uint32_t ssrc, int event, int duration) override { + return impl()->InsertDtmf(ssrc, event, duration); + } + bool GetStats(VoiceMediaSendInfo* info) override { + return impl_->GetSendStats(info); + } + + private: + VoiceMediaSendChannelInterface* impl() { return impl_; } + const VoiceMediaSendChannelInterface* impl() const { return impl_; } + VoiceMediaChannel* impl_; +}; + +class VoiceMediaReceiveChannel : public VoiceMediaReceiveChannelInterface { + public: + explicit VoiceMediaReceiveChannel(VoiceMediaChannel* impl) : impl_(impl) {} + virtual ~VoiceMediaReceiveChannel() {} + // Implementation of MediaBaseChannelInterface + cricket::MediaType media_type() const override { return MEDIA_TYPE_AUDIO; } + void OnPacketReceived(const webrtc::RtpPacketReceived& packet) override { + impl()->OnPacketReceived(packet); + } + void OnPacketSent(const rtc::SentPacket& sent_packet) override { + impl()->OnPacketSent(sent_packet); + } + void OnReadyToSend(bool ready) override { impl()->OnReadyToSend(ready); } + void OnNetworkRouteChanged(absl::string_view transport_name, + const rtc::NetworkRoute& network_route) override { + impl()->OnNetworkRouteChanged(transport_name, network_route); + } + void SetExtmapAllowMixed(bool extmap_allow_mixed) override { + impl()->SetExtmapAllowMixed(extmap_allow_mixed); + } + bool ExtmapAllowMixed() const override { return impl()->ExtmapAllowMixed(); } + // Implementation of Delayable + bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override { + return impl()->SetBaseMinimumPlayoutDelayMs(ssrc, delay_ms); + } + absl::optional<int> GetBaseMinimumPlayoutDelayMs( + uint32_t ssrc) const override { + return impl()->GetBaseMinimumPlayoutDelayMs(ssrc); + } + // Implementation of MediaReceiveChannelInterface + bool AddRecvStream(const StreamParams& sp) override { + return impl()->AddRecvStream(sp); + } + bool RemoveRecvStream(uint32_t ssrc) override { + return impl()->RemoveRecvStream(ssrc); + } + void ResetUnsignaledRecvStream() override { + return impl()->ResetUnsignaledRecvStream(); + } + absl::optional<uint32_t> GetUnsignaledSsrc() const override { + return impl()->GetUnsignaledSsrc(); + } + void OnDemuxerCriteriaUpdatePending() override { + impl()->OnDemuxerCriteriaUpdatePending(); + } + void OnDemuxerCriteriaUpdateComplete() override { + impl()->OnDemuxerCriteriaUpdateComplete(); + } + void SetFrameDecryptor(uint32_t ssrc, + rtc::scoped_refptr<webrtc::FrameDecryptorInterface> + frame_decryptor) override { + impl()->SetFrameDecryptor(ssrc, frame_decryptor); + } + void SetDepacketizerToDecoderFrameTransformer( + uint32_t ssrc, + rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) + override { + impl()->SetDepacketizerToDecoderFrameTransformer(ssrc, frame_transformer); + } + // Implementation of VoiceMediaReceiveChannelInterface + bool SetRecvParameters(const AudioRecvParameters& params) override { + return impl()->SetRecvParameters(params); + } + webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override { + return impl()->GetRtpReceiveParameters(ssrc); + } + std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override { + return impl()->GetSources(ssrc); + } + webrtc::RtpParameters GetDefaultRtpReceiveParameters() const override { + return impl()->GetDefaultRtpReceiveParameters(); + } + void SetPlayout(bool playout) override { return impl()->SetPlayout(playout); } + bool SetOutputVolume(uint32_t ssrc, double volume) override { + return impl()->SetOutputVolume(ssrc, volume); + } + bool SetDefaultOutputVolume(double volume) override { + return impl()->SetDefaultOutputVolume(volume); + } + void SetRawAudioSink( + uint32_t ssrc, + std::unique_ptr<webrtc::AudioSinkInterface> sink) override { + return impl()->SetRawAudioSink(ssrc, std::move(sink)); + } + void SetDefaultRawAudioSink( + std::unique_ptr<webrtc::AudioSinkInterface> sink) override { + return impl()->SetDefaultRawAudioSink(std::move(sink)); + } + bool GetStats(VoiceMediaReceiveInfo* info, bool reset_legacy) override { + return impl_->GetReceiveStats(info, reset_legacy); + } + + private: + VoiceMediaReceiveChannelInterface* impl() { return impl_; } + const VoiceMediaReceiveChannelInterface* impl() const { return impl_; } + VoiceMediaChannel* impl_; +}; + +class VideoMediaSendChannel : public VideoMediaSendChannelInterface { + public: + explicit VideoMediaSendChannel(VideoMediaChannel* impl) : impl_(impl) {} + + VideoMediaSendChannelInterface* AsVideoSendChannel() override { return this; } + VoiceMediaSendChannelInterface* AsVoiceSendChannel() override { + RTC_CHECK_NOTREACHED(); + return nullptr; + } + + // Implementation of MediaBaseChannelInterface + cricket::MediaType media_type() const override { return MEDIA_TYPE_VIDEO; } + void OnPacketReceived(const webrtc::RtpPacketReceived& packet) override { + impl()->OnPacketReceived(packet); + } + void OnPacketSent(const rtc::SentPacket& sent_packet) override { + impl()->OnPacketSent(sent_packet); + } + void OnReadyToSend(bool ready) override { impl()->OnReadyToSend(ready); } + void OnNetworkRouteChanged(absl::string_view transport_name, + const rtc::NetworkRoute& network_route) override { + impl()->OnNetworkRouteChanged(transport_name, network_route); + } + void SetExtmapAllowMixed(bool extmap_allow_mixed) override { + impl()->SetExtmapAllowMixed(extmap_allow_mixed); + } + bool ExtmapAllowMixed() const override { return impl()->ExtmapAllowMixed(); } + // Implementation of MediaSendChannelInterface + bool AddSendStream(const StreamParams& sp) override { + return impl()->AddSendStream(sp); + } + bool RemoveSendStream(uint32_t ssrc) override { + return impl()->RemoveSendStream(ssrc); + } + void SetFrameEncryptor(uint32_t ssrc, + rtc::scoped_refptr<webrtc::FrameEncryptorInterface> + frame_encryptor) override { + impl()->SetFrameEncryptor(ssrc, frame_encryptor); + } + webrtc::RTCError SetRtpSendParameters( + uint32_t ssrc, + const webrtc::RtpParameters& parameters, + webrtc::SetParametersCallback callback = nullptr) override { + return impl()->SetRtpSendParameters(ssrc, parameters, std::move(callback)); + } + + void SetEncoderToPacketizerFrameTransformer( + uint32_t ssrc, + rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) + override { + return impl()->SetEncoderToPacketizerFrameTransformer(ssrc, + frame_transformer); + } + void SetEncoderSelector(uint32_t ssrc, + webrtc::VideoEncoderFactory::EncoderSelectorInterface* + encoder_selector) override { + impl()->SetEncoderSelector(ssrc, encoder_selector); + } + webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override { + return impl()->GetRtpSendParameters(ssrc); + } + // Implementation of VideoMediaSendChannelInterface + bool SetSendParameters(const VideoSendParameters& params) override { + return impl()->SetSendParameters(params); + } + bool GetSendCodec(VideoCodec* send_codec) override { + return impl()->GetSendCodec(send_codec); + } + bool SetSend(bool send) override { return impl()->SetSend(send); } + bool SetVideoSend( + uint32_t ssrc, + const VideoOptions* options, + rtc::VideoSourceInterface<webrtc::VideoFrame>* source) override { + return impl()->SetVideoSend(ssrc, options, source); + } + void GenerateSendKeyFrame(uint32_t ssrc, + const std::vector<std::string>& rids) override { + return impl()->GenerateSendKeyFrame(ssrc, rids); + } + void SetVideoCodecSwitchingEnabled(bool enabled) override { + return impl()->SetVideoCodecSwitchingEnabled(enabled); + } + bool GetStats(VideoMediaSendInfo* info) override { + return impl_->GetSendStats(info); + } + void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) override { + return impl_->FillBitrateInfo(bwe_info); + } + + private: + VideoMediaSendChannelInterface* impl() { return impl_; } + const VideoMediaSendChannelInterface* impl() const { return impl_; } + VideoMediaChannel* const impl_; +}; + +class VideoMediaReceiveChannel : public VideoMediaReceiveChannelInterface { + public: + explicit VideoMediaReceiveChannel(VideoMediaChannel* impl) : impl_(impl) {} + // Implementation of MediaBaseChannelInterface + cricket::MediaType media_type() const override { return MEDIA_TYPE_VIDEO; } + void OnPacketReceived(const webrtc::RtpPacketReceived& packet) override { + impl()->OnPacketReceived(packet); + } + void OnPacketSent(const rtc::SentPacket& sent_packet) override { + impl()->OnPacketSent(sent_packet); + } + void OnReadyToSend(bool ready) override { impl()->OnReadyToSend(ready); } + void OnNetworkRouteChanged(absl::string_view transport_name, + const rtc::NetworkRoute& network_route) override { + impl()->OnNetworkRouteChanged(transport_name, network_route); + } + void SetExtmapAllowMixed(bool extmap_allow_mixed) override { + impl()->SetExtmapAllowMixed(extmap_allow_mixed); + } + bool ExtmapAllowMixed() const override { return impl()->ExtmapAllowMixed(); } + // Implementation of Delayable + bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override { + return impl()->SetBaseMinimumPlayoutDelayMs(ssrc, delay_ms); + } + absl::optional<int> GetBaseMinimumPlayoutDelayMs( + uint32_t ssrc) const override { + return impl()->GetBaseMinimumPlayoutDelayMs(ssrc); + } + // Implementation of MediaReceiveChannelInterface + VideoMediaReceiveChannelInterface* AsVideoReceiveChannel() override { + return this; + } + bool AddRecvStream(const StreamParams& sp) override { + return impl()->AddRecvStream(sp); + } + bool RemoveRecvStream(uint32_t ssrc) override { + return impl()->RemoveRecvStream(ssrc); + } + void ResetUnsignaledRecvStream() override { + return impl()->ResetUnsignaledRecvStream(); + } + absl::optional<uint32_t> GetUnsignaledSsrc() const override { + return impl()->GetUnsignaledSsrc(); + } + void OnDemuxerCriteriaUpdatePending() override { + impl()->OnDemuxerCriteriaUpdatePending(); + } + void OnDemuxerCriteriaUpdateComplete() override { + impl()->OnDemuxerCriteriaUpdateComplete(); + } + void SetFrameDecryptor(uint32_t ssrc, + rtc::scoped_refptr<webrtc::FrameDecryptorInterface> + frame_decryptor) override { + impl()->SetFrameDecryptor(ssrc, frame_decryptor); + } + void SetDepacketizerToDecoderFrameTransformer( + uint32_t ssrc, + rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) + override { + impl()->SetDepacketizerToDecoderFrameTransformer(ssrc, frame_transformer); + } + // Implementation on videoMediaReceiveChannelInterface + bool SetRecvParameters(const VideoRecvParameters& params) override { + return impl()->SetRecvParameters(params); + } + webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override { + return impl()->GetRtpReceiveParameters(ssrc); + } + webrtc::RtpParameters GetDefaultRtpReceiveParameters() const override { + return impl()->GetDefaultRtpReceiveParameters(); + } + bool SetSink(uint32_t ssrc, + rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override { + return impl()->SetSink(ssrc, sink); + } + void SetDefaultSink( + rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override { + return impl()->SetDefaultSink(sink); + } + void RequestRecvKeyFrame(uint32_t ssrc) override { + return impl()->RequestRecvKeyFrame(ssrc); + } + std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override { + return impl()->GetSources(ssrc); + } + // Set recordable encoded frame callback for `ssrc` + void SetRecordableEncodedFrameCallback( + uint32_t ssrc, + std::function<void(const webrtc::RecordableEncodedFrame&)> callback) + override { + return impl()->SetRecordableEncodedFrameCallback(ssrc, std::move(callback)); + } + // Clear recordable encoded frame callback for `ssrc` + void ClearRecordableEncodedFrameCallback(uint32_t ssrc) override { + impl()->ClearRecordableEncodedFrameCallback(ssrc); + } + bool GetStats(VideoMediaReceiveInfo* info) override { + return impl_->GetReceiveStats(info); + } + + private: + VideoMediaReceiveChannelInterface* impl() { return impl_; } + const VideoMediaReceiveChannelInterface* impl() const { return impl_; } + VideoMediaChannel* const impl_; +}; + +} // namespace cricket + +#endif // MEDIA_BASE_MEDIA_CHANNEL_IMPL_H_ |