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Unified Diff: webrtc/modules/audio_coding/neteq/include/neteq.h

Issue 1467163002: NetEq: Add new method last_output_sample_rate_hz (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 1 month ago
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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;

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