Delete AcmReceiver::SetInitialDelay

webrtc/modules/audio_coding/main/acm2/acm_receiver.h

 /*
  *  Copyright (c) 2013 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 133 matching lines @@
   int SetMaximumDelay(int delay_ms);
 
   //
   // Get least required delay computed based on channel conditions. Note that
   // this is before applying any user-defined limits (specified by calling
   // (SetMinimumDelay() and/or SetMaximumDelay()).
   //
   int LeastRequiredDelayMs() const;
 
   //
-  // Sets an initial delay of |delay_ms| milliseconds. This introduces a playout
-  // delay. Silence (zero signal) is played out until equivalent of |delay_ms|
-  // millisecond of audio is buffered. Then, NetEq maintains the delay.
-  //
-  // Input:
-  //   - delay_ms             : initial delay in milliseconds.
-  //
-  // Return value             : 0 if OK.
-  //                           <0 if NetEq returned an error.
-  //
-  int SetInitialDelay(int delay_ms);
-
-  //
   // Resets the initial delay to zero.
   //
   void ResetInitialDelay();
 
   //
   // Get the current sampling frequency in Hz.
   //
   // Return value             : Sampling frequency in Hz.
   //
   int current_sample_rate_hz() const;
@@ skipping 107 matching lines @@
   //    -round_trip_time_ms : estimate of the round-trip-time (in milliseconds).
   // Return value           : list of packets to be retransmitted.
   //
   std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const;
 
   //
   // Get statistics of calls to GetAudio().
   void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
 
  private:
-  bool GetSilence(int desired_sample_rate_hz, AudioFrame* frame)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
-
-  int GetNumSyncPacketToInsert(uint16_t received_squence_number);
-
   const Decoder* RtpHeaderToDecoder(const RTPHeader& rtp_header,
                                     const uint8_t* payload) const
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   uint32_t NowInTimestamp(int decoder_sampling_rate) const;
 
-  void InsertStreamOfSyncPackets(InitialDelayManager::SyncStream* sync_stream);

[kwiberg-webrtc, 2015/11/02 14:45:49]

I presume InsertStreamOfConsciousness() was already gone?

[hlundin-webrtc, 2015/11/02 15:31:54]

Yes, together with MuteInnerVoices().

-
   rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
   int id_;  // TODO(henrik.lundin) Make const.
   const Decoder* last_audio_decoder_ GUARDED_BY(crit_sect_);
   AudioFrame::VADActivity previous_audio_activity_ GUARDED_BY(crit_sect_);
   int current_sample_rate_hz_ GUARDED_BY(crit_sect_);
   ACMResampler resampler_ GUARDED_BY(crit_sect_);
   // Used in GetAudio, declared as member to avoid allocating every 10ms.
   // TODO(henrik.lundin) Stack-allocate in GetAudio instead?
   rtc::scoped_ptr<int16_t[]> audio_buffer_ GUARDED_BY(crit_sect_);
   rtc::scoped_ptr<int16_t[]> last_audio_buffer_ GUARDED_BY(crit_sect_);
   CallStatistics call_stats_ GUARDED_BY(crit_sect_);
   NetEq* neteq_;
   // Decoders map is keyed by payload type
   std::map<uint8_t, Decoder> decoders_ GUARDED_BY(crit_sect_);
   bool vad_enabled_;
   Clock* clock_;  // TODO(henrik.lundin) Make const if possible.
   bool resampled_last_output_frame_ GUARDED_BY(crit_sect_);
-
-  // Indicates if a non-zero initial delay is set, and the receiver is in
-  // AV-sync mode.
-  bool av_sync_;
-  rtc::scoped_ptr<InitialDelayManager> initial_delay_manager_;
-
-  // The following are defined as members to avoid creating them in every
-  // iteration. |missing_packets_sync_stream_| is *ONLY* used in InsertPacket().
-  // |late_packets_sync_stream_| is only used in GetAudio(). Both of these
-  // member variables are allocated only when we AV-sync is enabled, i.e.
-  // initial delay is set.
-  rtc::scoped_ptr<InitialDelayManager::SyncStream> missing_packets_sync_stream_;
-  rtc::scoped_ptr<InitialDelayManager::SyncStream> late_packets_sync_stream_;
 };
 
 }  // namespace acm2
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_