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Add timestamps to AudioDeviceBuffer::SetRecordedBuffer

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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}
This commit is contained in:
Olov Brändström
2022-01-28 15:07:39 +01:00
committed by WebRTC LUCI CQ
parent 9897649336
commit b732bd5fb5
9 changed files with 121 additions and 17 deletions
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4
sdk/android/native_unittests/audio_device/audio_device_unittest.cc
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@ -893,7 +893,7 @@ TEST_F(AudioDeviceTest, StartRecordingVerifyCallbacks) {
EXPECT_CALL(
mock, RecordedDataIsAvailable(NotNull(), record_frames_per_10ms_buffer(),
kBytesPerSample, record_channels(),
record_sample_rate(), _, 0, 0, false, _))
record_sample_rate(), _, 0, 0, false, _, _))
.Times(AtLeast(kNumCallbacks));
EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
@ -914,7 +914,7 @@ TEST_F(AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) {
EXPECT_CALL(
mock, RecordedDataIsAvailable(NotNull(), record_frames_per_10ms_buffer(),
kBytesPerSample, record_channels(),
record_sample_rate(), _, 0, 0, false, _))
record_sample_rate(), _, 0, 0, false, _, _))
.Times(AtLeast(kNumCallbacks));
EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
StartPlayout();