| Index: webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
|
| diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
|
| index ed496186469e1a7ccc4d48858fc1635528f6680e..905a7152dd40dda0dc523207d34e0072165a4c00 100644
|
| --- a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
|
| +++ b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
|
| @@ -60,7 +60,7 @@ int AudioEncoderPcm::NumChannels() const {
|
| }
|
|
|
| size_t AudioEncoderPcm::MaxEncodedBytes() const {
|
| - return full_frame_samples_;
|
| + return full_frame_samples_ * BytesPerSample();
|
| }
|
|
|
| int AudioEncoderPcm::Num10MsFramesInNextPacket() const {
|
| @@ -71,6 +71,10 @@ 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,
|
| @@ -104,12 +108,20 @@ int16_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
|
| 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) {
|
|
|