Removed sync packet support from NetEq.

webrtc/modules/audio_coding/neteq/neteq_impl.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 90 matching lines @@
   ~NetEqImpl() override;
 
   // Inserts a new packet into NetEq. The |receive_timestamp| is an indication
   // of the time when the packet was received, and should be measured with
   // the same tick rate as the RTP timestamp of the current payload.
   // Returns 0 on success, -1 on failure.
   int InsertPacket(const WebRtcRTPHeader& rtp_header,
                    rtc::ArrayView<const uint8_t> payload,
                    uint32_t receive_timestamp) override;
 
-  // Inserts a sync-packet into packet queue. Sync-packets are decoded to
-  // silence and are intended to keep AV-sync intact in an event of long packet
-  // losses when Video NACK is enabled but Audio NACK is not. Clients of NetEq
-  // might insert sync-packet when they observe that buffer level of NetEq is
-  // decreasing below a certain threshold, defined by the application.
-  // Sync-packets should have the same payload type as the last audio payload
-  // type, i.e. they cannot have DTMF or CNG payload type, nor a codec change
-  // can be implied by inserting a sync-packet.
-  // Returns kOk on success, kFail on failure.
-  int InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
-                       uint32_t receive_timestamp) override;
-
   int GetAudio(AudioFrame* audio_frame, bool* muted) override;
 
   int RegisterPayloadType(NetEqDecoder codec,
                           const std::string& codec_name,
                           uint8_t rtp_payload_type) override;
 
   int RegisterExternalDecoder(AudioDecoder* decoder,
                               NetEqDecoder codec,
                               const std::string& codec_name,
                               uint8_t rtp_payload_type) override;
@@ skipping 83 matching lines @@
   // TODO(hlundin): Provide a better value for kSyncBufferSize.
   // Current value is kMaxFrameSize + 60 ms * 48 kHz, which is enough for
   // calculating correlations of current frame against history.
   static const size_t kSyncBufferSize = kMaxFrameSize + 60 * 48;
 
   // Inserts a new packet into NetEq. This is used by the InsertPacket method
   // above. Returns 0 on success, otherwise an error code.
   // TODO(hlundin): Merge this with InsertPacket above?
   int InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
                            rtc::ArrayView<const uint8_t> payload,
-                           uint32_t receive_timestamp,
-                           bool is_sync_packet)
+                           uint32_t receive_timestamp)
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   // Delivers 10 ms of audio data. The data is written to |audio_frame|.
   // Returns 0 on success, otherwise an error code.
   int GetAudioInternal(AudioFrame* audio_frame, bool* muted)
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   // Provides a decision to the GetAudioInternal method. The decision what to
   // do is written to |operation|. Packets to decode are written to
   // |packet_list|, and a DTMF event to play is written to |dtmf_event|. When
@@ skipping 176 matching lines @@
       AudioFrame::kVadPassive;
   std::unique_ptr<TickTimer::Stopwatch> generated_noise_stopwatch_
       GUARDED_BY(crit_sect_);
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_