| Index: webrtc/modules/audio_coding/neteq/include/neteq.h
|
| diff --git a/webrtc/modules/audio_coding/neteq/include/neteq.h b/webrtc/modules/audio_coding/neteq/include/neteq.h
|
| index 9a1bc17b8bb05da97c00481c6ab75beb1979ac73..88677d83558735a7643e57f89d9971ec16537b9b 100644
|
| --- a/webrtc/modules/audio_coding/neteq/include/neteq.h
|
| +++ b/webrtc/modules/audio_coding/neteq/include/neteq.h
|
| @@ -251,6 +251,11 @@ class NetEq {
|
| // Returns true if the RTP timestamp is valid, otherwise false.
|
| virtual bool GetPlayoutTimestamp(uint32_t* timestamp) = 0;
|
|
|
| + // Returns the sample rate in Hz of the audio produced in the last GetAudio
|
| + // call. If GetAudio has not been called yet, the configured sample rate
|
| + // (Config::sample_rate_hz) is returned.
|
| + virtual int last_output_sample_rate_hz() const = 0;
|
| +
|
| // Not implemented.
|
| virtual int SetTargetNumberOfChannels() = 0;
|
|
|
|
|