Add timestamps to AudioDeviceBuffer::SetRecordedBuffer
Add timestamps to the function AudioDeviceBuffer::SetRecordedBuffer. This will
be used to store audio timestaps in future changes.
This is a part of the A/V sync metric metric feature for mobile. The metric
have already launched for web clients.
Bug: webrtc:13609
Change-Id: I0031843476ff1b573b262308fca52d587fae30b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249085
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Minyue Li <minyue@google.com>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#35851}
diff --git a/audio/audio_transport_impl.h b/audio/audio_transport_impl.h
index a946e2b..0b1406f 100644
--- a/audio/audio_transport_impl.h
+++ b/audio/audio_transport_impl.h
@@ -41,6 +41,7 @@
~AudioTransportImpl() override;
+ // TODO(bugs.webrtc.org/13620) Deprecate this function
int32_t RecordedDataIsAvailable(const void* audioSamples,
size_t nSamples,
size_t nBytesPerSample,
@@ -52,6 +53,18 @@
bool keyPressed,
uint32_t& newMicLevel) override;
+ int32_t RecordedDataIsAvailable(const void* audioSamples,
+ size_t nSamples,
+ size_t nBytesPerSample,
+ size_t nChannels,
+ uint32_t samplesPerSec,
+ uint32_t totalDelayMS,
+ int32_t clockDrift,
+ uint32_t currentMicLevel,
+ bool keyPressed,
+ uint32_t& newMicLevel,
+ int64_t estimated_capture_time_ns) override;
+
int32_t NeedMorePlayData(size_t nSamples,
size_t nBytesPerSample,
size_t nChannels,
