[EXPERIMENTAL] Generic stereo codec with index header sending merged frames

webrtc/media/engine/webrtcvideoengine.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.
  */
 
@@ skipping 187 matching lines @@
     // of |other|.
     bool operator==(const VideoCodecSettings& other) const;
     bool operator!=(const VideoCodecSettings& other) const;
 
     // Checks if all members of |a|, except |flexfec_payload_type|, are equal
     // to the corresponding members of |b|.
     static bool EqualsDisregardingFlexfec(const VideoCodecSettings& a,
                                           const VideoCodecSettings& b);
 
     VideoCodec codec;
+    rtc::Optional<VideoCodec> stereo_codec;
     webrtc::UlpfecConfig ulpfec;
     int flexfec_payload_type;
     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;
@@ skipping 103 matching lines @@
       bool external;
     };
 
     rtc::scoped_refptr<webrtc::VideoEncoderConfig::EncoderSpecificSettings>
     ConfigureVideoEncoderSettings(const VideoCodec& codec);
     // If force_encoder_allocation is true, a new AllocatedEncoder is always
     // created. If false, the allocated encoder may be reused, if the type
     // matches.
     AllocatedEncoder CreateVideoEncoder(const VideoCodec& codec,
                                         bool force_encoder_allocation);
+    AllocatedEncoder CreateStereoVideoEncoder(const VideoCodec& codec);
     void DestroyVideoEncoder(AllocatedEncoder* encoder);
     void SetCodec(const VideoCodecSettings& codec,
                   bool force_encoder_allocation);
     void RecreateWebRtcStream();
     webrtc::VideoEncoderConfig CreateVideoEncoderConfig(
         const VideoCodec& codec) const;
     void ReconfigureEncoder();
     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();
 
     webrtc::VideoSendStream::DegradationPreference GetDegradationPreference()
         const EXCLUSIVE_LOCKS_REQUIRED(&thread_checker_);
 
     rtc::ThreadChecker thread_checker_;
     rtc::AsyncInvoker invoker_;
     rtc::Thread* worker_thread_;
     const std::vector<uint32_t> ssrcs_ ACCESS_ON(&thread_checker_);
     const std::vector<SsrcGroup> ssrc_groups_ ACCESS_ON(&thread_checker_);
     webrtc::Call* const call_;
     const bool enable_cpu_overuse_detection_;
     rtc::VideoSourceInterface<webrtc::VideoFrame>* source_
         ACCESS_ON(&thread_checker_);
     WebRtcVideoEncoderFactory* const external_encoder_factory_
         ACCESS_ON(&thread_checker_);
     const std::unique_ptr<WebRtcVideoEncoderFactory> internal_encoder_factory_
         ACCESS_ON(&thread_checker_);
+    std::unique_ptr<WebRtcVideoEncoderFactory> stereo_encoder_factory_
+        ACCESS_ON(&thread_checker_);
 
     webrtc::VideoSendStream* stream_ ACCESS_ON(&thread_checker_);
     rtc::VideoSinkInterface<webrtc::VideoFrame>* encoder_sink_
         ACCESS_ON(&thread_checker_);
     // 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_ ACCESS_ON(&thread_checker_);
     // Contains settings that are unique for each stream, such as max_bitrate.
     // Does *not* contain codecs, however.
@@ skipping 53 matching lines @@
 
     void RecreateWebRtcVideoStream();
     void MaybeRecreateWebRtcFlexfecStream();
 
     void ConfigureCodecs(const std::vector<VideoCodecSettings>& recv_codecs,
                          std::vector<AllocatedDecoder>* old_codecs);
     void ConfigureFlexfecCodec(int flexfec_payload_type);
     AllocatedDecoder CreateOrReuseVideoDecoder(
         std::vector<AllocatedDecoder>* old_decoder,
         const VideoCodec& codec);
+    AllocatedDecoder CreateStereoVideoDecoder(const VideoCodec& codec);
     void ClearDecoders(std::vector<AllocatedDecoder>* allocated_decoders);
 
     std::string GetCodecNameFromPayloadType(int payload_type);
 
     webrtc::Call* const call_;
     StreamParams stream_params_;
 
     // Both |stream_| and |flexfec_stream_| are managed by |this|. They are
     // destroyed by calling call_->DestroyVideoReceiveStream and
     // call_->DestroyFlexfecReceiveStream, respectively.
     webrtc::VideoReceiveStream* stream_;
     const bool default_stream_;
     webrtc::VideoReceiveStream::Config config_;
     webrtc::FlexfecReceiveStream::Config flexfec_config_;
     webrtc::FlexfecReceiveStream* flexfec_stream_;
 
     WebRtcVideoDecoderFactory* const external_decoder_factory_;
+    const std::unique_ptr<WebRtcVideoDecoderFactory> internal_decoder_factory_;
+    std::unique_ptr<WebRtcVideoDecoderFactory> stereo_decoder_factory_;
     std::vector<AllocatedDecoder> allocated_decoders_;
 
     rtc::CriticalSection sink_lock_;
     rtc::VideoSinkInterface<webrtc::VideoFrame>* sink_ 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, WebRtcVideoEngine* 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);
   // Select what video codec will be used for sending, i.e. what codec is used
   // for local encoding, based on supported remote codecs. The first remote
   // codec that is supported locally will be selected.
   rtc::Optional<VideoCodecSettings> SelectSendVideoCodec(
       const std::vector<VideoCodecSettings>& remote_mapped_codecs) const;
+  rtc::Optional<WebRtcVideoChannel::VideoCodecSettings>
+  SelectStereoAssociatedVideoCodec(
+      const std::vector<VideoCodecSettings>& remote_mapped_codecs) const;
 
   static bool NonFlexfecReceiveCodecsHaveChanged(
       std::vector<VideoCodecSettings> before,
       std::vector<VideoCodecSettings> after);
 
   void FillSenderStats(VideoMediaInfo* info, bool log_stats);
   void FillReceiverStats(VideoMediaInfo* info, bool log_stats);
   void FillBandwidthEstimationStats(const webrtc::Call::Stats& stats,
                                     VideoMediaInfo* info);
   void FillSendAndReceiveCodecStats(VideoMediaInfo* video_media_info);
@@ skipping 55 matching lines @@
   const std::string codec_name_;
   const int max_qp_;
   const int max_framerate_;
   const bool is_screencast_;
   const bool conference_mode_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE_H_