Add AudioEncoder::GetTargetBitrate

webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.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 64 matching lines @@
 }
 
 int AudioEncoderG722::Num10MsFramesInNextPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
 int AudioEncoderG722::Max10MsFramesInAPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
+int AudioEncoderG722::GetTargetBitrate() const {
+  // 4 bits/sample, 16000 samples/s/channel.
+  return 64000 * NumChannels();
+}
+
 AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal(
     uint32_t rtp_timestamp,
     const int16_t* audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
   CHECK_GE(max_encoded_bytes, MaxEncodedBytes());
 
   if (num_10ms_frames_buffered_ == 0)
     first_timestamp_in_buffer_ = rtp_timestamp;
 
@@ skipping 52 matching lines @@
   config.payload_type = codec_inst.pltype;
   return config;
 }
 }  // namespace
 
 AudioEncoderMutableG722::AudioEncoderMutableG722(const CodecInst& codec_inst)
     : AudioEncoderMutableImpl<AudioEncoderG722>(CreateConfig(codec_inst)) {
 }
 
 }  // namespace webrtc