Add AudioEncoder::GetTargetBitrate

webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.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 42 matching lines @@
 
 int AudioEncoderPcm::SampleRateHz() const {
   return sample_rate_hz_;
 }
 
 int AudioEncoderPcm::NumChannels() const {
   return num_channels_;
 }
 
 size_t AudioEncoderPcm::MaxEncodedBytes() const {
-  return full_frame_samples_;
+  return full_frame_samples_ * BytesPerSample();
 }
 
 int AudioEncoderPcm::Num10MsFramesInNextPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
 int AudioEncoderPcm::Max10MsFramesInAPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
+int AudioEncoderPcm::GetTargetBitrate() const {
+  return 8 * BytesPerSample() * SampleRateHz() * NumChannels();
+}
+
 AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeInternal(
     uint32_t rtp_timestamp,
     const int16_t* audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
   const int num_samples = SampleRateHz() / 100 * NumChannels();
   if (speech_buffer_.empty()) {
     first_timestamp_in_buffer_ = rtp_timestamp;
   }
   for (int i = 0; i < num_samples; ++i) {
@@ skipping 13 matching lines @@
   speech_buffer_.clear();
   return info;
 }
 
 int16_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
                                      size_t input_len,
                                      uint8_t* encoded) {
   return WebRtcG711_EncodeA(audio, static_cast<int16_t>(input_len), encoded);
 }
 
+int AudioEncoderPcmA::BytesPerSample() const {
+  return 1;
+}
+
 int16_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
                                      size_t input_len,
                                      uint8_t* encoded) {
   return WebRtcG711_EncodeU(audio, static_cast<int16_t>(input_len), encoded);
 }
 
+int AudioEncoderPcmU::BytesPerSample() const {
+  return 1;
+}
+
 namespace {
 template <typename T>
 typename T::Config CreateConfig(const CodecInst& codec_inst) {
   typename T::Config config;
   config.frame_size_ms = codec_inst.pacsize / 8;
   config.num_channels = codec_inst.channels;
   config.payload_type = codec_inst.pltype;
   return config;
 }
 }  // namespace
 
 AudioEncoderMutablePcmU::AudioEncoderMutablePcmU(const CodecInst& codec_inst)
     : AudioEncoderMutableImpl<AudioEncoderPcmU>(
           CreateConfig<AudioEncoderPcmU>(codec_inst)) {
 }
 
 AudioEncoderMutablePcmA::AudioEncoderMutablePcmA(const CodecInst& codec_inst)
     : AudioEncoderMutableImpl<AudioEncoderPcmA>(
           CreateConfig<AudioEncoderPcmA>(codec_inst)) {
 }
 
 }  // namespace webrtc