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; |