| Index: webrtc/modules/audio_coding/codecs/audio_encoder.cc
|
| diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.cc b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
|
| index 72e4265e987bda014badf8c7f615686731542c64..eb553a7e7d1017d49bd5d3e2301452b3d803e4c0 100644
|
| --- a/webrtc/modules/audio_coding/codecs/audio_encoder.cc
|
| +++ b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
|
| @@ -13,10 +13,12 @@
|
|
|
| namespace webrtc {
|
|
|
| -AudioEncoder::EncodedInfo::EncodedInfo() : EncodedInfoLeaf() {
|
| -}
|
| +AudioEncoder::EncodedInfo::EncodedInfo() = default;
|
| +
|
| +AudioEncoder::EncodedInfo::~EncodedInfo() = default;
|
|
|
| -AudioEncoder::EncodedInfo::~EncodedInfo() {
|
| +int AudioEncoder::RtpTimestampRateHz() const {
|
| + return SampleRateHz();
|
| }
|
|
|
| AudioEncoder::EncodedInfo AudioEncoder::Encode(uint32_t rtp_timestamp,
|
| @@ -32,8 +34,28 @@ AudioEncoder::EncodedInfo AudioEncoder::Encode(uint32_t rtp_timestamp,
|
| return info;
|
| }
|
|
|
| -int AudioEncoder::RtpTimestampRateHz() const {
|
| - return SampleRateHz();
|
| +bool AudioEncoder::SetFec(bool enable) {
|
| + return !enable;
|
| +}
|
| +
|
| +bool AudioEncoder::SetDtx(bool enable) {
|
| + return !enable;
|
| }
|
|
|
| +bool AudioEncoder::SetApplication(Application application) {
|
| + return false;
|
| +}
|
| +
|
| +bool AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {
|
| + return true;
|
| +}
|
| +
|
| +void AudioEncoder::SetProjectedPacketLossRate(double fraction) {}
|
| +
|
| +void AudioEncoder::SetTargetBitrate(int target_bps) {}
|
| +
|
| +void AudioEncoder::SetMaxBitrate(int max_bps) {}
|
| +
|
| +void AudioEncoder::SetMaxPayloadSize(int max_payload_size_bytes) {}
|
| +
|
| } // namespace webrtc
|
|
|