| Index: webrtc/modules/utility/source/coder.cc
|
| diff --git a/webrtc/modules/utility/source/coder.cc b/webrtc/modules/utility/source/coder.cc
|
| index f2ae43eb108a939f261ae17646a05c26fd307643..f72d03b887de9356df67d44bb20f7d8963f868c5 100644
|
| --- a/webrtc/modules/utility/source/coder.cc
|
| +++ b/webrtc/modules/utility/source/coder.cc
|
| @@ -54,7 +54,7 @@ int32_t AudioCoder::SetDecodeCodec(const CodecInst& codec_inst) {
|
| return 0;
|
| }
|
|
|
| -int32_t AudioCoder::Decode(AudioFrame& decoded_audio,
|
| +int32_t AudioCoder::Decode(AudioFrame* decoded_audio,
|
| uint32_t samp_freq_hz,
|
| const int8_t* incoming_payload,
|
| size_t payload_length) {
|
| @@ -68,22 +68,22 @@ int32_t AudioCoder::Decode(AudioFrame& decoded_audio,
|
| }
|
| bool muted;
|
| int32_t ret =
|
| - acm_->PlayoutData10Ms((uint16_t)samp_freq_hz, &decoded_audio, &muted);
|
| + acm_->PlayoutData10Ms((uint16_t)samp_freq_hz, decoded_audio, &muted);
|
| RTC_DCHECK(!muted);
|
| return ret;
|
| }
|
|
|
| -int32_t AudioCoder::PlayoutData(AudioFrame& decoded_audio,
|
| - uint16_t& samp_freq_hz) {
|
| +int32_t AudioCoder::PlayoutData(AudioFrame* decoded_audio,
|
| + uint16_t samp_freq_hz) {
|
| bool muted;
|
| - int32_t ret = acm_->PlayoutData10Ms(samp_freq_hz, &decoded_audio, &muted);
|
| + int32_t ret = acm_->PlayoutData10Ms(samp_freq_hz, decoded_audio, &muted);
|
| RTC_DCHECK(!muted);
|
| return ret;
|
| }
|
|
|
| int32_t AudioCoder::Encode(const AudioFrame& audio,
|
| int8_t* encoded_data,
|
| - size_t& encoded_length_in_bytes) {
|
| + size_t* encoded_length_in_bytes) {
|
| // Fake a timestamp in case audio doesn't contain a correct timestamp.
|
| // Make a local copy of the audio frame since audio is const
|
| AudioFrame audio_frame;
|
| @@ -98,7 +98,7 @@ int32_t AudioCoder::Encode(const AudioFrame& audio,
|
| return -1;
|
| }
|
| encoded_data_ = encoded_data;
|
| - encoded_length_in_bytes = encoded_length_in_bytes_;
|
| + *encoded_length_in_bytes = encoded_length_in_bytes_;
|
| return 0;
|
| }
|
|
|
|
|