| Index: webrtc/api/audio_codecs/audio_encoder.cc
|
| diff --git a/webrtc/api/audio_codecs/audio_encoder.cc b/webrtc/api/audio_codecs/audio_encoder.cc
|
| index 3ee371f1cee7f4b5d5390bd8c3340d7d1abd42a3..e38fe9c2e5eee1bdc683a03b4033bfe3466fa9b6 100644
|
| --- a/webrtc/api/audio_codecs/audio_encoder.cc
|
| +++ b/webrtc/api/audio_codecs/audio_encoder.cc
|
| @@ -24,6 +24,11 @@ AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(
|
| AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(EncodedInfo&&) =
|
| default;
|
|
|
| +AudioEncoder::AudioEncoderStats::AudioEncoderStats() = default;
|
| +AudioEncoder::AudioEncoderStats::~AudioEncoderStats() = default;
|
| +AudioEncoder::AudioEncoderStats::AudioEncoderStats(const AudioEncoderStats&) =
|
| + default;
|
| +
|
| int AudioEncoder::RtpTimestampRateHz() const {
|
| return SampleRateHz();
|
| }
|
| @@ -95,4 +100,8 @@ void AudioEncoder::OnReceivedOverhead(size_t overhead_bytes_per_packet) {}
|
| void AudioEncoder::SetReceiverFrameLengthRange(int min_frame_length_ms,
|
| int max_frame_length_ms) {}
|
|
|
| +AudioEncoder::AudioEncoderStats AudioEncoder::GetStats() const {
|
| + return AudioEncoder::AudioEncoderStats();
|
| +}
|
| +
|
| } // namespace webrtc
|
|
|