Move the decoder thread into VideoReceiveStream.

webrtc/video/vie_channel.h

 /*
  *  Copyright (c) 2012 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 28 matching lines @@
 class PacedSender;
 class PacketRouter;
 class PayloadRouter;
 class ProcessThread;
 class ReceiveStatisticsProxy;
 class ReportBlockStats;
 class RtcpRttStats;
 class ViEChannelProtectionCallback;
 class ViERTPObserver;
 class VideoCodingModule;
-class VideoDecoder;
 class VideoRenderCallback;
 class VoEVideoSync;
 
 enum StreamType {
   kViEStreamTypeNormal = 0,  // Normal media stream
   kViEStreamTypeRtx = 1      // Retransmission media stream
 };
 
 class ViEChannel : public VCMFrameTypeCallback,
                    public VCMReceiveCallback,
@@ skipping 96 matching lines @@
   int32_t OnInitializeDecoder(const int8_t payload_type,
                               const char payload_name[RTP_PAYLOAD_NAME_SIZE],
                               const int frequency,
                               const size_t channels,
                               const uint32_t rate) override;
   void OnIncomingSSRCChanged(const uint32_t ssrc) override;
   void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) override;
 
   int32_t StartSend();
   int32_t StopSend();
-  bool Sending();
-  void StartReceive();
-  void StopReceive();
 
   // Sets the maximum transfer unit size for the network link, i.e. including
   // IP, UDP and RTP headers.
   int32_t SetMTU(uint16_t mtu);
 
   // Gets the modules used by the channel.
   RtpRtcp* rtp_rtcp();
   ViEReceiver* vie_receiver();
   VCMProtectionCallback* vcm_protection_callback();
 
@@ skipping 43 matching lines @@
   void RegisterPreRenderCallback(I420FrameCallback* pre_render_callback);
 
   void RegisterSendFrameCountObserver(FrameCountObserver* observer);
   void RegisterRtcpPacketTypeCounterObserver(
       RtcpPacketTypeCounterObserver* observer);
   void RegisterReceiveStatisticsProxy(
       ReceiveStatisticsProxy* receive_statistics_proxy);
   void SetIncomingVideoStream(IncomingVideoStream* incoming_video_stream);
 
  protected:
-  static bool ChannelDecodeThreadFunction(void* obj);
-  bool ChannelDecodeProcess();
-
   void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms);
 
   int ProtectionRequest(const FecProtectionParams* delta_fec_params,
                         const FecProtectionParams* key_fec_params,
                         uint32_t* sent_video_rate_bps,
                         uint32_t* sent_nack_rate_bps,
                         uint32_t* sent_fec_rate_bps);
 
  private:
   static std::vector<RtpRtcp*> CreateRtpRtcpModules(
@@ skipping 131 matching lines @@
   FrameCounts receive_frame_counts_ GUARDED_BY(crit_);
   IncomingVideoStream* incoming_video_stream_ GUARDED_BY(crit_);
   RtcpIntraFrameObserver* const intra_frame_observer_;
   RtcpRttStats* const rtt_stats_;
   PacedSender* const paced_sender_;
   PacketRouter* const packet_router_;
 
   const rtc::scoped_ptr<RtcpBandwidthObserver> bandwidth_observer_;
   TransportFeedbackObserver* const transport_feedback_observer_;
 
-  rtc::PlatformThread decode_thread_;
-
   int nack_history_size_sender_;
   int max_nack_reordering_threshold_;
   I420FrameCallback* pre_render_callback_ GUARDED_BY(crit_);
 
   const rtc::scoped_ptr<ReportBlockStats> report_block_stats_sender_;
 
   int64_t time_of_first_rtt_ms_ GUARDED_BY(crit_);
   int64_t rtt_sum_ms_ GUARDED_BY(crit_);
   int64_t last_rtt_ms_ GUARDED_BY(crit_);
   size_t num_rtts_ GUARDED_BY(crit_);
 
   // RtpRtcp modules, declared last as they use other members on construction.
   const std::vector<RtpRtcp*> rtp_rtcp_modules_;
   size_t num_active_rtp_rtcp_modules_ GUARDED_BY(crit_);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_VIE_CHANNEL_H_