Remove AudioEncoder methods SetMaxBitrate and SetMaxPayloadSize

webrtc/modules/audio_coding/main/interface/audio_coding_module.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 702 matching lines @@
   //    0 if the function succeeds.
   //
   virtual int32_t PlayoutData10Ms(int32_t desired_freq_hz,
                                         AudioFrame* audio_frame) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   //   Codec specific
   //
 
   ///////////////////////////////////////////////////////////////////////////
-  // int32_t SetISACMaxRate()
-  // Set the maximum instantaneous rate of iSAC. For a payload of B bits
-  // with a frame-size of T sec the instantaneous rate is B/T bits per
-  // second. Therefore, (B/T < |max_rate_bps|) and
-  // (B < |max_payload_len_bytes| * 8) are always satisfied for iSAC payloads,
-  // c.f SetISACMaxPayloadSize().
-  //
-  // Input:
-  //   -max_rate_bps       : maximum instantaneous bit-rate given in bits/sec.
-  //
-  // Return value:
-  //   -1 if failed to set the maximum rate.
-  //    0 if the maximum rate is set successfully.
-  //
-  virtual int SetISACMaxRate(int max_rate_bps) = 0;
-
-  ///////////////////////////////////////////////////////////////////////////
-  // int32_t SetISACMaxPayloadSize()
-  // Set the maximum payload size of iSAC packets. No iSAC payload,
-  // regardless of its frame-size, may exceed the given limit. For
-  // an iSAC payload of size B bits and frame-size T seconds we have;
-  // (B < |max_payload_len_bytes| * 8) and (B/T < |max_rate_bps|), c.f.
-  // SetISACMaxRate().
-  //
-  // Input:
-  //   -max_payload_len_bytes : maximum payload size in bytes.
-  //
-  // Return value:
-  //   -1 if failed to set the maximum  payload-size.
-  //    0 if the given length is set successfully.
-  //
-  virtual int SetISACMaxPayloadSize(int max_payload_len_bytes) = 0;
-
-  ///////////////////////////////////////////////////////////////////////////
   // int SetOpusApplication()
   // Sets the intended application if current send codec is Opus. Opus uses this
   // to optimize the encoding for applications like VOIP and music. Currently,
   // two modes are supported: kVoip and kAudio.
   //
   // Input:
   //   - application            : intended application.
   //
   // Return value:
   //   -1 if current send codec is not Opus or error occurred in setting the
@@ skipping 275 matching lines @@
   virtual std::vector<uint16_t> GetNackList(int round_trip_time_ms) const = 0;
 
   // Returns the timing statistics for calls to Get10MsAudio.
   virtual void GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_