rhubarb-lip-sync/rhubarb/lib/webrtc-8d2248ff/webrtc/media/engine/webrtcvideoengine2.h

/*
 *  Copyright (c) 2014 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 WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_
#define WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_

#include <map>
#include <memory>
#include <set>
#include <string>
#include <vector>

#include "webrtc/base/asyncinvoker.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/networkroute.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/media/base/videosinkinterface.h"
#include "webrtc/media/base/videosourceinterface.h"
#include "webrtc/call.h"
#include "webrtc/media/base/mediaengine.h"
#include "webrtc/media/engine/webrtcvideochannelfactory.h"
#include "webrtc/media/engine/webrtcvideodecoderfactory.h"
#include "webrtc/media/engine/webrtcvideoencoderfactory.h"
#include "webrtc/transport.h"
#include "webrtc/video_frame.h"
#include "webrtc/video_receive_stream.h"
#include "webrtc/video_send_stream.h"

namespace webrtc {
class VideoDecoder;
class VideoEncoder;
struct MediaConfig;
}

namespace rtc {
class Thread;
}  // namespace rtc

namespace cricket {

class VideoCapturer;
class VideoFrame;
class VideoProcessor;
class VideoRenderer;
class VoiceMediaChannel;
class WebRtcDecoderObserver;
class WebRtcEncoderObserver;
class WebRtcLocalStreamInfo;
class WebRtcRenderAdapter;
class WebRtcVideoChannelRecvInfo;
class WebRtcVideoChannelSendInfo;
class WebRtcVoiceEngine;
class WebRtcVoiceMediaChannel;

struct CapturedFrame;
struct Device;

// Exposed here for unittests.
std::vector<VideoCodec> DefaultVideoCodecList();

class UnsignalledSsrcHandler {
 public:
  enum Action {
    kDropPacket,
    kDeliverPacket,
  };
  virtual Action OnUnsignalledSsrc(WebRtcVideoChannel2* channel,
                                   uint32_t ssrc) = 0;
  virtual ~UnsignalledSsrcHandler() = default;
};

// TODO(pbos): Remove, use external handlers only.
class DefaultUnsignalledSsrcHandler : public UnsignalledSsrcHandler {
 public:
  DefaultUnsignalledSsrcHandler();
  Action OnUnsignalledSsrc(WebRtcVideoChannel2* channel,
                           uint32_t ssrc) override;

  rtc::VideoSinkInterface<VideoFrame>* GetDefaultSink() const;
  void SetDefaultSink(VideoMediaChannel* channel,
                      rtc::VideoSinkInterface<VideoFrame>* sink);
  virtual ~DefaultUnsignalledSsrcHandler() = default;

 private:
  uint32_t default_recv_ssrc_;
  rtc::VideoSinkInterface<VideoFrame>* default_sink_;
};

// WebRtcVideoEngine2 is used for the new native WebRTC Video API (webrtc:1667).
class WebRtcVideoEngine2 {
 public:
  WebRtcVideoEngine2();
  virtual ~WebRtcVideoEngine2();

  // Basic video engine implementation.
  void Init();

  WebRtcVideoChannel2* CreateChannel(webrtc::Call* call,
                                     const MediaConfig& config,
                                     const VideoOptions& options);

  const std::vector<VideoCodec>& codecs() const;
  RtpCapabilities GetCapabilities() const;

  // Set a WebRtcVideoDecoderFactory for external decoding. Video engine does
  // not take the ownership of |decoder_factory|. The caller needs to make sure
  // that |decoder_factory| outlives the video engine.
  void SetExternalDecoderFactory(WebRtcVideoDecoderFactory* decoder_factory);
  // Set a WebRtcVideoEncoderFactory for external encoding. Video engine does
  // not take the ownership of |encoder_factory|. The caller needs to make sure
  // that |encoder_factory| outlives the video engine.
  virtual void SetExternalEncoderFactory(
      WebRtcVideoEncoderFactory* encoder_factory);

 private:
  std::vector<VideoCodec> GetSupportedCodecs() const;

  std::vector<VideoCodec> video_codecs_;

  bool initialized_;

  WebRtcVideoDecoderFactory* external_decoder_factory_;
  WebRtcVideoEncoderFactory* external_encoder_factory_;
  std::unique_ptr<WebRtcVideoEncoderFactory> simulcast_encoder_factory_;
};

class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
 public:
  WebRtcVideoChannel2(webrtc::Call* call,
                      const MediaConfig& config,
                      const VideoOptions& options,
                      const std::vector<VideoCodec>& recv_codecs,
                      WebRtcVideoEncoderFactory* external_encoder_factory,
                      WebRtcVideoDecoderFactory* external_decoder_factory);
  ~WebRtcVideoChannel2() override;

  // VideoMediaChannel implementation
  rtc::DiffServCodePoint PreferredDscp() const override;

  bool SetSendParameters(const VideoSendParameters& params) override;
  bool SetRecvParameters(const VideoRecvParameters& params) override;
  webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override;
  bool SetRtpSendParameters(uint32_t ssrc,
                            const webrtc::RtpParameters& parameters) override;
  webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override;
  bool SetRtpReceiveParameters(
      uint32_t ssrc,
      const webrtc::RtpParameters& parameters) override;
  bool GetSendCodec(VideoCodec* send_codec) override;
  bool SetSend(bool send) override;
  bool SetVideoSend(
      uint32_t ssrc,
      bool enable,
      const VideoOptions* options,
      rtc::VideoSourceInterface<cricket::VideoFrame>* source) override;
  bool AddSendStream(const StreamParams& sp) override;
  bool RemoveSendStream(uint32_t ssrc) override;
  bool AddRecvStream(const StreamParams& sp) override;
  bool AddRecvStream(const StreamParams& sp, bool default_stream);
  bool RemoveRecvStream(uint32_t ssrc) override;
  bool SetSink(uint32_t ssrc,
               rtc::VideoSinkInterface<VideoFrame>* sink) override;
  bool GetStats(VideoMediaInfo* info) override;

  void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
                        const rtc::PacketTime& packet_time) override;
  void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
                      const rtc::PacketTime& packet_time) override;
  void OnReadyToSend(bool ready) override;
  void OnNetworkRouteChanged(const std::string& transport_name,
                             const rtc::NetworkRoute& network_route) override;
  void SetInterface(NetworkInterface* iface) override;

  // Implemented for VideoMediaChannelTest.
  bool sending() const { return sending_; }

  // AdaptReason is used for expressing why a WebRtcVideoSendStream request
  // a lower input frame size than the currently configured camera input frame
  // size. There can be more than one reason OR:ed together.
  enum AdaptReason {
    ADAPTREASON_NONE = 0,
    ADAPTREASON_CPU = 1,
    ADAPTREASON_BANDWIDTH = 2,
  };

 private:
  class WebRtcVideoReceiveStream;
  struct VideoCodecSettings {
    VideoCodecSettings();

    bool operator==(const VideoCodecSettings& other) const;
    bool operator!=(const VideoCodecSettings& other) const;

    VideoCodec codec;
    webrtc::FecConfig fec;
    int rtx_payload_type;
  };

  struct ChangedSendParameters {
    // These optionals are unset if not changed.
    rtc::Optional<VideoCodecSettings> codec;
    rtc::Optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions;
    rtc::Optional<int> max_bandwidth_bps;
    rtc::Optional<bool> conference_mode;
    rtc::Optional<webrtc::RtcpMode> rtcp_mode;
  };

  struct ChangedRecvParameters {
    // These optionals are unset if not changed.
    rtc::Optional<std::vector<VideoCodecSettings>> codec_settings;
    rtc::Optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions;
  };

  bool GetChangedSendParameters(const VideoSendParameters& params,
                                ChangedSendParameters* changed_params) const;
  bool GetChangedRecvParameters(const VideoRecvParameters& params,
                                ChangedRecvParameters* changed_params) const;

  void SetMaxSendBandwidth(int bps);

  void ConfigureReceiverRtp(webrtc::VideoReceiveStream::Config* config,
                            const StreamParams& sp) const;
  bool CodecIsExternallySupported(const std::string& name) const;
  bool ValidateSendSsrcAvailability(const StreamParams& sp) const
      EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
  bool ValidateReceiveSsrcAvailability(const StreamParams& sp) const
      EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
  void DeleteReceiveStream(WebRtcVideoReceiveStream* stream)
      EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);

  static std::string CodecSettingsVectorToString(
      const std::vector<VideoCodecSettings>& codecs);

  // Wrapper for the sender part, this is where the source is connected and
  // frames are then converted from cricket frames to webrtc frames.
  class WebRtcVideoSendStream
      : public rtc::VideoSinkInterface<cricket::VideoFrame>,
        public webrtc::LoadObserver {
   public:
    WebRtcVideoSendStream(
        webrtc::Call* call,
        const StreamParams& sp,
        const webrtc::VideoSendStream::Config& config,
        const VideoOptions& options,
        WebRtcVideoEncoderFactory* external_encoder_factory,
        bool enable_cpu_overuse_detection,
        int max_bitrate_bps,
        const rtc::Optional<VideoCodecSettings>& codec_settings,
        const rtc::Optional<std::vector<webrtc::RtpExtension>>& rtp_extensions,
        const VideoSendParameters& send_params);
    virtual ~WebRtcVideoSendStream();

    void SetSendParameters(const ChangedSendParameters& send_params);
    bool SetRtpParameters(const webrtc::RtpParameters& parameters);
    webrtc::RtpParameters GetRtpParameters() const;

    void OnFrame(const cricket::VideoFrame& frame) override;
    bool SetVideoSend(bool mute,
                      const VideoOptions* options,
                      rtc::VideoSourceInterface<cricket::VideoFrame>* source);
    void DisconnectSource();

    void SetSend(bool send);

    // Implements webrtc::LoadObserver.
    void OnLoadUpdate(Load load) override;

    const std::vector<uint32_t>& GetSsrcs() const;
    VideoSenderInfo GetVideoSenderInfo();
    void FillBandwidthEstimationInfo(BandwidthEstimationInfo* bwe_info);

   private:
    // Parameters needed to reconstruct the underlying stream.
    // webrtc::VideoSendStream doesn't support setting a lot of options on the
    // fly, so when those need to be changed we tear down and reconstruct with
    // similar parameters depending on which options changed etc.
    struct VideoSendStreamParameters {
      VideoSendStreamParameters(
          const webrtc::VideoSendStream::Config& config,
          const VideoOptions& options,
          int max_bitrate_bps,
          const rtc::Optional<VideoCodecSettings>& codec_settings);
      webrtc::VideoSendStream::Config config;
      VideoOptions options;
      int max_bitrate_bps;
      bool conference_mode;
      rtc::Optional<VideoCodecSettings> codec_settings;
      // Sent resolutions + bitrates etc. by the underlying VideoSendStream,
      // typically changes when setting a new resolution or reconfiguring
      // bitrates.
      webrtc::VideoEncoderConfig encoder_config;
    };

    struct AllocatedEncoder {
      AllocatedEncoder(webrtc::VideoEncoder* encoder,
                       webrtc::VideoCodecType type,
                       bool external);
      webrtc::VideoEncoder* encoder;
      webrtc::VideoEncoder* external_encoder;
      webrtc::VideoCodecType type;
      bool external;
    };

    struct VideoFrameInfo {
      // Initial encoder configuration (QCIF, 176x144) frame (to ensure that
      // hardware encoders can be initialized). This gives us low memory usage
      // but also makes it so configuration errors are discovered at the time we
      // apply the settings rather than when we get the first frame (waiting for
      // the first frame to know that you gave a bad codec parameter could make
      // debugging hard).
      // TODO(pbos): Consider setting up encoders lazily.
      VideoFrameInfo()
          : width(176),
            height(144),
            rotation(webrtc::kVideoRotation_0),
            is_texture(false) {}
      int width;
      int height;
      webrtc::VideoRotation rotation;
      bool is_texture;
    };

    union VideoEncoderSettings {
      webrtc::VideoCodecH264 h264;
      webrtc::VideoCodecVP8 vp8;
      webrtc::VideoCodecVP9 vp9;
    };

    static std::vector<webrtc::VideoStream> CreateVideoStreams(
        const VideoCodec& codec,
        const VideoOptions& options,
        int max_bitrate_bps,
        size_t num_streams);
    static std::vector<webrtc::VideoStream> CreateSimulcastVideoStreams(
        const VideoCodec& codec,
        const VideoOptions& options,
        int max_bitrate_bps,
        size_t num_streams);

    void* ConfigureVideoEncoderSettings(const VideoCodec& codec)
        EXCLUSIVE_LOCKS_REQUIRED(lock_);

    AllocatedEncoder CreateVideoEncoder(const VideoCodec& codec)
        EXCLUSIVE_LOCKS_REQUIRED(lock_);
    void DestroyVideoEncoder(AllocatedEncoder* encoder)
        EXCLUSIVE_LOCKS_REQUIRED(lock_);
    void SetCodec(const VideoCodecSettings& codec)
        EXCLUSIVE_LOCKS_REQUIRED(lock_);
    void RecreateWebRtcStream() EXCLUSIVE_LOCKS_REQUIRED(lock_);
    webrtc::VideoEncoderConfig CreateVideoEncoderConfig(
        const VideoCodec& codec) const EXCLUSIVE_LOCKS_REQUIRED(lock_);
    void ReconfigureEncoder() EXCLUSIVE_LOCKS_REQUIRED(lock_);
    bool ValidateRtpParameters(const webrtc::RtpParameters& parameters);

    // Calls Start or Stop according to whether or not |sending_| is true,
    // and whether or not the encoding in |rtp_parameters_| is active.
    void UpdateSendState() EXCLUSIVE_LOCKS_REQUIRED(lock_);

    rtc::ThreadChecker thread_checker_;
    rtc::AsyncInvoker invoker_;
    rtc::Thread* worker_thread_;
    const std::vector<uint32_t> ssrcs_;
    const std::vector<SsrcGroup> ssrc_groups_;
    webrtc::Call* const call_;
    rtc::VideoSinkWants sink_wants_;
    // Counter used for deciding if the video resolution is currently
    // restricted by CPU usage. It is reset if |source_| is changed.
    int cpu_restricted_counter_;
    // Total number of times resolution as been requested to be changed due to
    // CPU adaptation.
    int number_of_cpu_adapt_changes_;
    rtc::VideoSourceInterface<cricket::VideoFrame>* source_;
    WebRtcVideoEncoderFactory* const external_encoder_factory_
        GUARDED_BY(lock_);

    rtc::CriticalSection lock_;
    webrtc::VideoSendStream* stream_ GUARDED_BY(lock_);
    // Contains settings that are the same for all streams in the MediaChannel,
    // such as codecs, header extensions, and the global bitrate limit for the
    // entire channel.
    VideoSendStreamParameters parameters_ GUARDED_BY(lock_);
    // Contains settings that are unique for each stream, such as max_bitrate.
    // Does *not* contain codecs, however.
    // TODO(skvlad): Move ssrcs_ and ssrc_groups_ into rtp_parameters_.
    // TODO(skvlad): Combine parameters_ and rtp_parameters_ once we have only
    // one stream per MediaChannel.
    webrtc::RtpParameters rtp_parameters_ GUARDED_BY(lock_);
    bool pending_encoder_reconfiguration_ GUARDED_BY(lock_);
    VideoEncoderSettings encoder_settings_ GUARDED_BY(lock_);
    AllocatedEncoder allocated_encoder_ GUARDED_BY(lock_);
    VideoFrameInfo last_frame_info_ GUARDED_BY(lock_);

    bool sending_ GUARDED_BY(lock_);

    // The timestamp of the first frame received
    // Used to generate the timestamps of subsequent frames
    rtc::Optional<int64_t> first_frame_timestamp_ms_ GUARDED_BY(lock_);

    // The timestamp of the last frame received
    // Used to generate timestamp for the black frame when source is removed
    int64_t last_frame_timestamp_ms_ GUARDED_BY(lock_);
  };

  // Wrapper for the receiver part, contains configs etc. that are needed to
  // reconstruct the underlying VideoReceiveStream. Also serves as a wrapper
  // between rtc::VideoSinkInterface<webrtc::VideoFrame> and
  // rtc::VideoSinkInterface<cricket::VideoFrame>.
  class WebRtcVideoReceiveStream
      : public rtc::VideoSinkInterface<webrtc::VideoFrame> {
   public:
    WebRtcVideoReceiveStream(
        webrtc::Call* call,
        const StreamParams& sp,
        webrtc::VideoReceiveStream::Config config,
        WebRtcVideoDecoderFactory* external_decoder_factory,
        bool default_stream,
        const std::vector<VideoCodecSettings>& recv_codecs,
        bool red_disabled_by_remote_side);
    ~WebRtcVideoReceiveStream();

    const std::vector<uint32_t>& GetSsrcs() const;

    void SetLocalSsrc(uint32_t local_ssrc);
    // TODO(deadbeef): Move these feedback parameters into the recv parameters.
    void SetFeedbackParameters(bool nack_enabled,
                               bool remb_enabled,
                               bool transport_cc_enabled,
                               webrtc::RtcpMode rtcp_mode);
    void SetRecvParameters(const ChangedRecvParameters& recv_params);

    void OnFrame(const webrtc::VideoFrame& frame) override;
    bool IsDefaultStream() const;

    void SetSink(rtc::VideoSinkInterface<cricket::VideoFrame>* sink);

    VideoReceiverInfo GetVideoReceiverInfo();

    // Used to disable RED/FEC when the remote description doesn't contain those
    // codecs. This is needed to be able to work around an RTX bug which is only
    // happening if the remote side doesn't send RED, but the local side is
    // configured to receive RED.
    // TODO(holmer): Remove this after a couple of Chrome versions, M53-54
    // time frame.
    void SetFecDisabledRemotely(bool disable);

   private:
    struct AllocatedDecoder {
      AllocatedDecoder(webrtc::VideoDecoder* decoder,
                       webrtc::VideoCodecType type,
                       bool external);
      webrtc::VideoDecoder* decoder;
      // Decoder wrapped into a fallback decoder to permit software fallback.
      webrtc::VideoDecoder* external_decoder;
      webrtc::VideoCodecType type;
      bool external;
    };

    void RecreateWebRtcStream();

    void ConfigureCodecs(const std::vector<VideoCodecSettings>& recv_codecs,
                         std::vector<AllocatedDecoder>* old_codecs);
    AllocatedDecoder CreateOrReuseVideoDecoder(
        std::vector<AllocatedDecoder>* old_decoder,
        const VideoCodec& codec);
    void ClearDecoders(std::vector<AllocatedDecoder>* allocated_decoders);

    std::string GetCodecNameFromPayloadType(int payload_type);

    webrtc::Call* const call_;
    const std::vector<uint32_t> ssrcs_;
    const std::vector<SsrcGroup> ssrc_groups_;

    webrtc::VideoReceiveStream* stream_;
    const bool default_stream_;
    webrtc::VideoReceiveStream::Config config_;
    bool red_disabled_by_remote_side_;

    WebRtcVideoDecoderFactory* const external_decoder_factory_;
    std::vector<AllocatedDecoder> allocated_decoders_;

    rtc::CriticalSection sink_lock_;
    rtc::VideoSinkInterface<cricket::VideoFrame>* sink_ GUARDED_BY(sink_lock_);
    int last_width_ GUARDED_BY(sink_lock_);
    int last_height_ GUARDED_BY(sink_lock_);
    // Expands remote RTP timestamps to int64_t to be able to estimate how long
    // the stream has been running.
    rtc::TimestampWrapAroundHandler timestamp_wraparound_handler_
        GUARDED_BY(sink_lock_);
    int64_t first_frame_timestamp_ GUARDED_BY(sink_lock_);
    // Start NTP time is estimated as current remote NTP time (estimated from
    // RTCP) minus the elapsed time, as soon as remote NTP time is available.
    int64_t estimated_remote_start_ntp_time_ms_ GUARDED_BY(sink_lock_);
  };

  void Construct(webrtc::Call* call, WebRtcVideoEngine2* engine);

  bool SendRtp(const uint8_t* data,
               size_t len,
               const webrtc::PacketOptions& options) override;
  bool SendRtcp(const uint8_t* data, size_t len) override;

  static std::vector<VideoCodecSettings> MapCodecs(
      const std::vector<VideoCodec>& codecs);
  std::vector<VideoCodecSettings> FilterSupportedCodecs(
      const std::vector<VideoCodecSettings>& mapped_codecs) const;
  static bool ReceiveCodecsHaveChanged(std::vector<VideoCodecSettings> before,
                                       std::vector<VideoCodecSettings> after);

  void FillSenderStats(VideoMediaInfo* info);
  void FillReceiverStats(VideoMediaInfo* info);
  void FillBandwidthEstimationStats(const webrtc::Call::Stats& stats,
                                    VideoMediaInfo* info);

  rtc::ThreadChecker thread_checker_;

  uint32_t rtcp_receiver_report_ssrc_;
  bool sending_;
  webrtc::Call* const call_;

  DefaultUnsignalledSsrcHandler default_unsignalled_ssrc_handler_;
  UnsignalledSsrcHandler* const unsignalled_ssrc_handler_;

  const MediaConfig::Video video_config_;

  rtc::CriticalSection stream_crit_;
  // Using primary-ssrc (first ssrc) as key.
  std::map<uint32_t, WebRtcVideoSendStream*> send_streams_
      GUARDED_BY(stream_crit_);
  std::map<uint32_t, WebRtcVideoReceiveStream*> receive_streams_
      GUARDED_BY(stream_crit_);
  std::set<uint32_t> send_ssrcs_ GUARDED_BY(stream_crit_);
  std::set<uint32_t> receive_ssrcs_ GUARDED_BY(stream_crit_);

  rtc::Optional<VideoCodecSettings> send_codec_;
  rtc::Optional<std::vector<webrtc::RtpExtension>> send_rtp_extensions_;

  WebRtcVideoEncoderFactory* const external_encoder_factory_;
  WebRtcVideoDecoderFactory* const external_decoder_factory_;
  std::vector<VideoCodecSettings> recv_codecs_;
  std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
  webrtc::Call::Config::BitrateConfig bitrate_config_;
  // TODO(deadbeef): Don't duplicate information between
  // send_params/recv_params, rtp_extensions, options, etc.
  VideoSendParameters send_params_;
  VideoOptions default_send_options_;
  VideoRecvParameters recv_params_;
  bool red_disabled_by_remote_side_;
};

}  // namespace cricket

#endif  // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_