561 lines
22 KiB
C++
561 lines
22 KiB
C++
/*
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* Copyright (c) 2014 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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#ifndef WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_
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#define WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_
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#include <map>
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#include <memory>
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#include <set>
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#include <string>
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#include <vector>
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#include "webrtc/base/asyncinvoker.h"
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/base/networkroute.h"
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#include "webrtc/base/thread_annotations.h"
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#include "webrtc/base/thread_checker.h"
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#include "webrtc/media/base/videosinkinterface.h"
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#include "webrtc/media/base/videosourceinterface.h"
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#include "webrtc/call.h"
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#include "webrtc/media/base/mediaengine.h"
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#include "webrtc/media/engine/webrtcvideochannelfactory.h"
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#include "webrtc/media/engine/webrtcvideodecoderfactory.h"
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#include "webrtc/media/engine/webrtcvideoencoderfactory.h"
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#include "webrtc/transport.h"
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#include "webrtc/video_frame.h"
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#include "webrtc/video_receive_stream.h"
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#include "webrtc/video_send_stream.h"
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namespace webrtc {
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class VideoDecoder;
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class VideoEncoder;
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struct MediaConfig;
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}
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namespace rtc {
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class Thread;
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} // namespace rtc
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namespace cricket {
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class VideoCapturer;
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class VideoFrame;
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class VideoProcessor;
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class VideoRenderer;
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class VoiceMediaChannel;
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class WebRtcDecoderObserver;
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class WebRtcEncoderObserver;
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class WebRtcLocalStreamInfo;
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class WebRtcRenderAdapter;
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class WebRtcVideoChannelRecvInfo;
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class WebRtcVideoChannelSendInfo;
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class WebRtcVoiceEngine;
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class WebRtcVoiceMediaChannel;
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struct CapturedFrame;
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struct Device;
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// Exposed here for unittests.
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std::vector<VideoCodec> DefaultVideoCodecList();
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class UnsignalledSsrcHandler {
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public:
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enum Action {
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kDropPacket,
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kDeliverPacket,
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};
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virtual Action OnUnsignalledSsrc(WebRtcVideoChannel2* channel,
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uint32_t ssrc) = 0;
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virtual ~UnsignalledSsrcHandler() = default;
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};
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// TODO(pbos): Remove, use external handlers only.
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class DefaultUnsignalledSsrcHandler : public UnsignalledSsrcHandler {
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public:
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DefaultUnsignalledSsrcHandler();
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Action OnUnsignalledSsrc(WebRtcVideoChannel2* channel,
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uint32_t ssrc) override;
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rtc::VideoSinkInterface<VideoFrame>* GetDefaultSink() const;
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void SetDefaultSink(VideoMediaChannel* channel,
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rtc::VideoSinkInterface<VideoFrame>* sink);
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virtual ~DefaultUnsignalledSsrcHandler() = default;
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private:
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uint32_t default_recv_ssrc_;
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rtc::VideoSinkInterface<VideoFrame>* default_sink_;
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};
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// WebRtcVideoEngine2 is used for the new native WebRTC Video API (webrtc:1667).
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class WebRtcVideoEngine2 {
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public:
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WebRtcVideoEngine2();
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virtual ~WebRtcVideoEngine2();
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// Basic video engine implementation.
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void Init();
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WebRtcVideoChannel2* CreateChannel(webrtc::Call* call,
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const MediaConfig& config,
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const VideoOptions& options);
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const std::vector<VideoCodec>& codecs() const;
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RtpCapabilities GetCapabilities() const;
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// Set a WebRtcVideoDecoderFactory for external decoding. Video engine does
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// not take the ownership of |decoder_factory|. The caller needs to make sure
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// that |decoder_factory| outlives the video engine.
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void SetExternalDecoderFactory(WebRtcVideoDecoderFactory* decoder_factory);
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// Set a WebRtcVideoEncoderFactory for external encoding. Video engine does
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// not take the ownership of |encoder_factory|. The caller needs to make sure
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// that |encoder_factory| outlives the video engine.
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virtual void SetExternalEncoderFactory(
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WebRtcVideoEncoderFactory* encoder_factory);
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private:
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std::vector<VideoCodec> GetSupportedCodecs() const;
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std::vector<VideoCodec> video_codecs_;
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bool initialized_;
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WebRtcVideoDecoderFactory* external_decoder_factory_;
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WebRtcVideoEncoderFactory* external_encoder_factory_;
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std::unique_ptr<WebRtcVideoEncoderFactory> simulcast_encoder_factory_;
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};
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class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
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public:
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WebRtcVideoChannel2(webrtc::Call* call,
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const MediaConfig& config,
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const VideoOptions& options,
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const std::vector<VideoCodec>& recv_codecs,
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WebRtcVideoEncoderFactory* external_encoder_factory,
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WebRtcVideoDecoderFactory* external_decoder_factory);
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~WebRtcVideoChannel2() override;
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// VideoMediaChannel implementation
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rtc::DiffServCodePoint PreferredDscp() const override;
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bool SetSendParameters(const VideoSendParameters& params) override;
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bool SetRecvParameters(const VideoRecvParameters& params) override;
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webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override;
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bool SetRtpSendParameters(uint32_t ssrc,
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const webrtc::RtpParameters& parameters) override;
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webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override;
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bool SetRtpReceiveParameters(
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uint32_t ssrc,
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const webrtc::RtpParameters& parameters) override;
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bool GetSendCodec(VideoCodec* send_codec) override;
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bool SetSend(bool send) override;
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bool SetVideoSend(
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uint32_t ssrc,
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bool enable,
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const VideoOptions* options,
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rtc::VideoSourceInterface<cricket::VideoFrame>* source) override;
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bool AddSendStream(const StreamParams& sp) override;
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bool RemoveSendStream(uint32_t ssrc) override;
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bool AddRecvStream(const StreamParams& sp) override;
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bool AddRecvStream(const StreamParams& sp, bool default_stream);
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bool RemoveRecvStream(uint32_t ssrc) override;
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bool SetSink(uint32_t ssrc,
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rtc::VideoSinkInterface<VideoFrame>* sink) override;
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bool GetStats(VideoMediaInfo* info) override;
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void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketTime& packet_time) override;
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void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketTime& packet_time) override;
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void OnReadyToSend(bool ready) override;
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void OnNetworkRouteChanged(const std::string& transport_name,
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const rtc::NetworkRoute& network_route) override;
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void SetInterface(NetworkInterface* iface) override;
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// Implemented for VideoMediaChannelTest.
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bool sending() const { return sending_; }
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// AdaptReason is used for expressing why a WebRtcVideoSendStream request
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// a lower input frame size than the currently configured camera input frame
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// size. There can be more than one reason OR:ed together.
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enum AdaptReason {
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ADAPTREASON_NONE = 0,
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ADAPTREASON_CPU = 1,
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ADAPTREASON_BANDWIDTH = 2,
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};
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private:
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class WebRtcVideoReceiveStream;
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struct VideoCodecSettings {
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VideoCodecSettings();
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bool operator==(const VideoCodecSettings& other) const;
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bool operator!=(const VideoCodecSettings& other) const;
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VideoCodec codec;
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webrtc::FecConfig fec;
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int rtx_payload_type;
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};
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struct ChangedSendParameters {
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// These optionals are unset if not changed.
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rtc::Optional<VideoCodecSettings> codec;
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rtc::Optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions;
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rtc::Optional<int> max_bandwidth_bps;
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rtc::Optional<bool> conference_mode;
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rtc::Optional<webrtc::RtcpMode> rtcp_mode;
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};
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struct ChangedRecvParameters {
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// These optionals are unset if not changed.
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rtc::Optional<std::vector<VideoCodecSettings>> codec_settings;
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rtc::Optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions;
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};
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bool GetChangedSendParameters(const VideoSendParameters& params,
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ChangedSendParameters* changed_params) const;
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bool GetChangedRecvParameters(const VideoRecvParameters& params,
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ChangedRecvParameters* changed_params) const;
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void SetMaxSendBandwidth(int bps);
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void ConfigureReceiverRtp(webrtc::VideoReceiveStream::Config* config,
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const StreamParams& sp) const;
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bool CodecIsExternallySupported(const std::string& name) const;
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bool ValidateSendSsrcAvailability(const StreamParams& sp) const
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EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
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bool ValidateReceiveSsrcAvailability(const StreamParams& sp) const
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EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
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void DeleteReceiveStream(WebRtcVideoReceiveStream* stream)
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EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
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static std::string CodecSettingsVectorToString(
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const std::vector<VideoCodecSettings>& codecs);
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// Wrapper for the sender part, this is where the source is connected and
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// frames are then converted from cricket frames to webrtc frames.
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class WebRtcVideoSendStream
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: public rtc::VideoSinkInterface<cricket::VideoFrame>,
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public webrtc::LoadObserver {
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public:
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WebRtcVideoSendStream(
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webrtc::Call* call,
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const StreamParams& sp,
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const webrtc::VideoSendStream::Config& config,
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const VideoOptions& options,
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WebRtcVideoEncoderFactory* external_encoder_factory,
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bool enable_cpu_overuse_detection,
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int max_bitrate_bps,
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const rtc::Optional<VideoCodecSettings>& codec_settings,
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const rtc::Optional<std::vector<webrtc::RtpExtension>>& rtp_extensions,
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const VideoSendParameters& send_params);
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virtual ~WebRtcVideoSendStream();
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void SetSendParameters(const ChangedSendParameters& send_params);
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bool SetRtpParameters(const webrtc::RtpParameters& parameters);
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webrtc::RtpParameters GetRtpParameters() const;
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void OnFrame(const cricket::VideoFrame& frame) override;
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bool SetVideoSend(bool mute,
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const VideoOptions* options,
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rtc::VideoSourceInterface<cricket::VideoFrame>* source);
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void DisconnectSource();
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void SetSend(bool send);
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// Implements webrtc::LoadObserver.
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void OnLoadUpdate(Load load) override;
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const std::vector<uint32_t>& GetSsrcs() const;
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VideoSenderInfo GetVideoSenderInfo();
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void FillBandwidthEstimationInfo(BandwidthEstimationInfo* bwe_info);
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private:
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// Parameters needed to reconstruct the underlying stream.
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// webrtc::VideoSendStream doesn't support setting a lot of options on the
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// fly, so when those need to be changed we tear down and reconstruct with
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// similar parameters depending on which options changed etc.
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struct VideoSendStreamParameters {
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VideoSendStreamParameters(
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const webrtc::VideoSendStream::Config& config,
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const VideoOptions& options,
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int max_bitrate_bps,
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const rtc::Optional<VideoCodecSettings>& codec_settings);
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webrtc::VideoSendStream::Config config;
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VideoOptions options;
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int max_bitrate_bps;
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bool conference_mode;
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rtc::Optional<VideoCodecSettings> codec_settings;
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// Sent resolutions + bitrates etc. by the underlying VideoSendStream,
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// typically changes when setting a new resolution or reconfiguring
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// bitrates.
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webrtc::VideoEncoderConfig encoder_config;
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};
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struct AllocatedEncoder {
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AllocatedEncoder(webrtc::VideoEncoder* encoder,
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webrtc::VideoCodecType type,
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bool external);
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webrtc::VideoEncoder* encoder;
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webrtc::VideoEncoder* external_encoder;
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webrtc::VideoCodecType type;
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bool external;
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};
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struct VideoFrameInfo {
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// Initial encoder configuration (QCIF, 176x144) frame (to ensure that
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// hardware encoders can be initialized). This gives us low memory usage
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// but also makes it so configuration errors are discovered at the time we
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// apply the settings rather than when we get the first frame (waiting for
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// the first frame to know that you gave a bad codec parameter could make
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// debugging hard).
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// TODO(pbos): Consider setting up encoders lazily.
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VideoFrameInfo()
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: width(176),
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height(144),
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rotation(webrtc::kVideoRotation_0),
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is_texture(false) {}
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int width;
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int height;
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webrtc::VideoRotation rotation;
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bool is_texture;
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};
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union VideoEncoderSettings {
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webrtc::VideoCodecH264 h264;
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webrtc::VideoCodecVP8 vp8;
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webrtc::VideoCodecVP9 vp9;
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};
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static std::vector<webrtc::VideoStream> CreateVideoStreams(
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const VideoCodec& codec,
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const VideoOptions& options,
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int max_bitrate_bps,
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size_t num_streams);
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static std::vector<webrtc::VideoStream> CreateSimulcastVideoStreams(
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const VideoCodec& codec,
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const VideoOptions& options,
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int max_bitrate_bps,
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size_t num_streams);
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void* ConfigureVideoEncoderSettings(const VideoCodec& codec)
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EXCLUSIVE_LOCKS_REQUIRED(lock_);
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AllocatedEncoder CreateVideoEncoder(const VideoCodec& codec)
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EXCLUSIVE_LOCKS_REQUIRED(lock_);
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void DestroyVideoEncoder(AllocatedEncoder* encoder)
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EXCLUSIVE_LOCKS_REQUIRED(lock_);
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void SetCodec(const VideoCodecSettings& codec)
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EXCLUSIVE_LOCKS_REQUIRED(lock_);
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void RecreateWebRtcStream() EXCLUSIVE_LOCKS_REQUIRED(lock_);
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webrtc::VideoEncoderConfig CreateVideoEncoderConfig(
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const VideoCodec& codec) const EXCLUSIVE_LOCKS_REQUIRED(lock_);
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void ReconfigureEncoder() EXCLUSIVE_LOCKS_REQUIRED(lock_);
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bool ValidateRtpParameters(const webrtc::RtpParameters& parameters);
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// Calls Start or Stop according to whether or not |sending_| is true,
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// and whether or not the encoding in |rtp_parameters_| is active.
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void UpdateSendState() EXCLUSIVE_LOCKS_REQUIRED(lock_);
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rtc::ThreadChecker thread_checker_;
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rtc::AsyncInvoker invoker_;
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rtc::Thread* worker_thread_;
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const std::vector<uint32_t> ssrcs_;
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const std::vector<SsrcGroup> ssrc_groups_;
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webrtc::Call* const call_;
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rtc::VideoSinkWants sink_wants_;
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// Counter used for deciding if the video resolution is currently
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// restricted by CPU usage. It is reset if |source_| is changed.
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int cpu_restricted_counter_;
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// Total number of times resolution as been requested to be changed due to
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// CPU adaptation.
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int number_of_cpu_adapt_changes_;
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rtc::VideoSourceInterface<cricket::VideoFrame>* source_;
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WebRtcVideoEncoderFactory* const external_encoder_factory_
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GUARDED_BY(lock_);
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rtc::CriticalSection lock_;
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webrtc::VideoSendStream* stream_ GUARDED_BY(lock_);
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// Contains settings that are the same for all streams in the MediaChannel,
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// such as codecs, header extensions, and the global bitrate limit for the
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// entire channel.
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VideoSendStreamParameters parameters_ GUARDED_BY(lock_);
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// Contains settings that are unique for each stream, such as max_bitrate.
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// Does *not* contain codecs, however.
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// TODO(skvlad): Move ssrcs_ and ssrc_groups_ into rtp_parameters_.
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// TODO(skvlad): Combine parameters_ and rtp_parameters_ once we have only
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// one stream per MediaChannel.
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webrtc::RtpParameters rtp_parameters_ GUARDED_BY(lock_);
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bool pending_encoder_reconfiguration_ GUARDED_BY(lock_);
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VideoEncoderSettings encoder_settings_ GUARDED_BY(lock_);
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AllocatedEncoder allocated_encoder_ GUARDED_BY(lock_);
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VideoFrameInfo last_frame_info_ GUARDED_BY(lock_);
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bool sending_ GUARDED_BY(lock_);
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// The timestamp of the first frame received
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// Used to generate the timestamps of subsequent frames
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rtc::Optional<int64_t> first_frame_timestamp_ms_ GUARDED_BY(lock_);
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// The timestamp of the last frame received
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// Used to generate timestamp for the black frame when source is removed
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int64_t last_frame_timestamp_ms_ GUARDED_BY(lock_);
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};
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// Wrapper for the receiver part, contains configs etc. that are needed to
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// reconstruct the underlying VideoReceiveStream. Also serves as a wrapper
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// between rtc::VideoSinkInterface<webrtc::VideoFrame> and
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// rtc::VideoSinkInterface<cricket::VideoFrame>.
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class WebRtcVideoReceiveStream
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: public rtc::VideoSinkInterface<webrtc::VideoFrame> {
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public:
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WebRtcVideoReceiveStream(
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webrtc::Call* call,
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const StreamParams& sp,
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webrtc::VideoReceiveStream::Config config,
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WebRtcVideoDecoderFactory* external_decoder_factory,
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bool default_stream,
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const std::vector<VideoCodecSettings>& recv_codecs,
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bool red_disabled_by_remote_side);
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~WebRtcVideoReceiveStream();
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const std::vector<uint32_t>& GetSsrcs() const;
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void SetLocalSsrc(uint32_t local_ssrc);
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// TODO(deadbeef): Move these feedback parameters into the recv parameters.
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void SetFeedbackParameters(bool nack_enabled,
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bool remb_enabled,
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bool transport_cc_enabled,
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webrtc::RtcpMode rtcp_mode);
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void SetRecvParameters(const ChangedRecvParameters& recv_params);
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void OnFrame(const webrtc::VideoFrame& frame) override;
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bool IsDefaultStream() const;
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void SetSink(rtc::VideoSinkInterface<cricket::VideoFrame>* sink);
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VideoReceiverInfo GetVideoReceiverInfo();
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// Used to disable RED/FEC when the remote description doesn't contain those
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// codecs. This is needed to be able to work around an RTX bug which is only
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// happening if the remote side doesn't send RED, but the local side is
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// configured to receive RED.
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// TODO(holmer): Remove this after a couple of Chrome versions, M53-54
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// time frame.
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void SetFecDisabledRemotely(bool disable);
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private:
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struct AllocatedDecoder {
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AllocatedDecoder(webrtc::VideoDecoder* decoder,
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webrtc::VideoCodecType type,
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bool external);
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webrtc::VideoDecoder* decoder;
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// Decoder wrapped into a fallback decoder to permit software fallback.
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webrtc::VideoDecoder* external_decoder;
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webrtc::VideoCodecType type;
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bool external;
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};
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void RecreateWebRtcStream();
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void ConfigureCodecs(const std::vector<VideoCodecSettings>& recv_codecs,
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std::vector<AllocatedDecoder>* old_codecs);
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AllocatedDecoder CreateOrReuseVideoDecoder(
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std::vector<AllocatedDecoder>* old_decoder,
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const VideoCodec& codec);
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void ClearDecoders(std::vector<AllocatedDecoder>* allocated_decoders);
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std::string GetCodecNameFromPayloadType(int payload_type);
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webrtc::Call* const call_;
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const std::vector<uint32_t> ssrcs_;
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const std::vector<SsrcGroup> ssrc_groups_;
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webrtc::VideoReceiveStream* stream_;
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const bool default_stream_;
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webrtc::VideoReceiveStream::Config config_;
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bool red_disabled_by_remote_side_;
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WebRtcVideoDecoderFactory* const external_decoder_factory_;
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std::vector<AllocatedDecoder> allocated_decoders_;
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rtc::CriticalSection sink_lock_;
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rtc::VideoSinkInterface<cricket::VideoFrame>* sink_ GUARDED_BY(sink_lock_);
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int last_width_ GUARDED_BY(sink_lock_);
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int last_height_ GUARDED_BY(sink_lock_);
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// Expands remote RTP timestamps to int64_t to be able to estimate how long
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// the stream has been running.
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rtc::TimestampWrapAroundHandler timestamp_wraparound_handler_
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GUARDED_BY(sink_lock_);
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int64_t first_frame_timestamp_ GUARDED_BY(sink_lock_);
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// Start NTP time is estimated as current remote NTP time (estimated from
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// RTCP) minus the elapsed time, as soon as remote NTP time is available.
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int64_t estimated_remote_start_ntp_time_ms_ GUARDED_BY(sink_lock_);
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};
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void Construct(webrtc::Call* call, WebRtcVideoEngine2* engine);
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bool SendRtp(const uint8_t* data,
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size_t len,
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const webrtc::PacketOptions& options) override;
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bool SendRtcp(const uint8_t* data, size_t len) override;
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static std::vector<VideoCodecSettings> MapCodecs(
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const std::vector<VideoCodec>& codecs);
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std::vector<VideoCodecSettings> FilterSupportedCodecs(
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const std::vector<VideoCodecSettings>& mapped_codecs) const;
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static bool ReceiveCodecsHaveChanged(std::vector<VideoCodecSettings> before,
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std::vector<VideoCodecSettings> after);
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void FillSenderStats(VideoMediaInfo* info);
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void FillReceiverStats(VideoMediaInfo* info);
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void FillBandwidthEstimationStats(const webrtc::Call::Stats& stats,
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VideoMediaInfo* info);
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rtc::ThreadChecker thread_checker_;
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uint32_t rtcp_receiver_report_ssrc_;
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bool sending_;
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webrtc::Call* const call_;
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DefaultUnsignalledSsrcHandler default_unsignalled_ssrc_handler_;
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UnsignalledSsrcHandler* const unsignalled_ssrc_handler_;
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const MediaConfig::Video video_config_;
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rtc::CriticalSection stream_crit_;
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// Using primary-ssrc (first ssrc) as key.
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std::map<uint32_t, WebRtcVideoSendStream*> send_streams_
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GUARDED_BY(stream_crit_);
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std::map<uint32_t, WebRtcVideoReceiveStream*> receive_streams_
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GUARDED_BY(stream_crit_);
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std::set<uint32_t> send_ssrcs_ GUARDED_BY(stream_crit_);
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std::set<uint32_t> receive_ssrcs_ GUARDED_BY(stream_crit_);
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rtc::Optional<VideoCodecSettings> send_codec_;
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rtc::Optional<std::vector<webrtc::RtpExtension>> send_rtp_extensions_;
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WebRtcVideoEncoderFactory* const external_encoder_factory_;
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WebRtcVideoDecoderFactory* const external_decoder_factory_;
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std::vector<VideoCodecSettings> recv_codecs_;
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std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
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|
webrtc::Call::Config::BitrateConfig bitrate_config_;
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|
// TODO(deadbeef): Don't duplicate information between
|
|
// send_params/recv_params, rtp_extensions, options, etc.
|
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VideoSendParameters send_params_;
|
|
VideoOptions default_send_options_;
|
|
VideoRecvParameters recv_params_;
|
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bool red_disabled_by_remote_side_;
|
|
};
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} // namespace cricket
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#endif // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_
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