webrtc/modules/audio_coding/neteq/include/neteq.h - Issue 1863993002: Move setting of AudioFrame::timestamp_ into NetEq

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

Issue 1863993002: Move setting of AudioFrame::timestamp_ into NetEq (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@neteq-empty-playout-ts

Patch Set: Created 4 years, 8 months 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 5b52424bee71609d2b34e9d8f20ff27e8495f842..89b0c543244536084d15bf4df08db51eebda6509 100644

--- a/webrtc/modules/audio_coding/neteq/include/neteq.h

+++ b/webrtc/modules/audio_coding/neteq/include/neteq.h

@@ -246,7 +246,7 @@ class NetEq {

// Returns the RTP timestamp for the last sample delivered by GetAudio().

// The return value will be empty if no valid timestamp is available.

- virtual rtc::Optional<uint32_t> GetPlayoutTimestamp() = 0;

+ virtual rtc::Optional<uint32_t> GetPlayoutTimestamp() const = 0;

hlundin-webrtc 2016/04/06 09:34:07 This was always essentially a const method, but it

// 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