| Index: webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
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| diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
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| index ed496186469e1a7ccc4d48858fc1635528f6680e..905a7152dd40dda0dc523207d34e0072165a4c00 100644
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| --- a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
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| +++ b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
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| @@ -60,7 +60,7 @@ int AudioEncoderPcm::NumChannels() const {
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|  }
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|  
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|  size_t AudioEncoderPcm::MaxEncodedBytes() const {
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| -  return full_frame_samples_;
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| +  return full_frame_samples_ * BytesPerSample();
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|  }
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|  
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|  int AudioEncoderPcm::Num10MsFramesInNextPacket() const {
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| @@ -71,6 +71,10 @@ int AudioEncoderPcm::Max10MsFramesInAPacket() const {
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|    return num_10ms_frames_per_packet_;
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|  }
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|  
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| +int AudioEncoderPcm::GetTargetBitrate() const {
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| +  return 8 * BytesPerSample() * SampleRateHz() * NumChannels();
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| +}
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| +
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|  AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeInternal(
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|      uint32_t rtp_timestamp,
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|      const int16_t* audio,
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| @@ -104,12 +108,20 @@ int16_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
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|    return WebRtcG711_EncodeA(audio, static_cast<int16_t>(input_len), encoded);
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|  }
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|  
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| +int AudioEncoderPcmA::BytesPerSample() const {
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| +  return 1;
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| +}
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| +
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|  int16_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
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|                                       size_t input_len,
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|                                       uint8_t* encoded) {
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|    return WebRtcG711_EncodeU(audio, static_cast<int16_t>(input_len), encoded);
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|  }
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|  
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| +int AudioEncoderPcmU::BytesPerSample() const {
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| +  return 1;
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| +}
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| +
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|  namespace {
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|  template <typename T>
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|  typename T::Config CreateConfig(const CodecInst& codec_inst) {
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| 
 |