Add AudioEncoder::GetTargetBitrate

webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc

 /*
  *  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 48 matching lines @@
 }
 
 int AudioEncoderIlbc::Num10MsFramesInNextPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
 int AudioEncoderIlbc::Max10MsFramesInAPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
+int AudioEncoderIlbc::SetTargetBitrate(int bits_per_second) {
+  switch (num_10ms_frames_per_packet_) {
+    case 2: case 4:  return 15200;
+    case 3: case 6:  return 13333;

[kwiberg-webrtc, 2015/06/16 19:22:54]

I'd recommend an explanation here. Otherwise these will be very hard to verify
or change.

[hlundin-webrtc, 2015/06/16 21:38:50]

Done.

+    default:  FATAL();
+  }
+}
+
 AudioEncoder::EncodedInfo AudioEncoderIlbc::EncodeInternal(
     uint32_t rtp_timestamp,
     const int16_t* audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
   DCHECK_GE(max_encoded_bytes, RequiredOutputSizeBytes());
 
   // Save timestamp if starting a new packet.
   if (num_10ms_frames_buffered_ == 0)
     first_timestamp_in_buffer_ = rtp_timestamp;
@@ skipping 43 matching lines @@
   config.payload_type = codec_inst.pltype;
   return config;
 }
 }  // namespace
 
 AudioEncoderMutableIlbc::AudioEncoderMutableIlbc(const CodecInst& codec_inst)
     : AudioEncoderMutableImpl<AudioEncoderIlbc>(CreateConfig(codec_inst)) {
 }
 
 }  // namespace webrtc