Adding absolute capture timestamp to AudioTrackSinkInterface.
Bug: webrtc:10739
Change-Id: I8c134cbe82452ac71625cd0c810c783a73f17822
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167532
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30408}
diff --git a/pc/rtp_sender.cc b/pc/rtp_sender.cc
index 402ad97..73cfcd0 100644
--- a/pc/rtp_sender.cc
+++ b/pc/rtp_sender.cc
@@ -372,15 +372,17 @@
sink_->OnClose();
}
-void LocalAudioSinkAdapter::OnData(const void* audio_data,
- int bits_per_sample,
- int sample_rate,
- size_t number_of_channels,
- size_t number_of_frames) {
+void LocalAudioSinkAdapter::OnData(
+ const void* audio_data,
+ int bits_per_sample,
+ int sample_rate,
+ size_t number_of_channels,
+ size_t number_of_frames,
+ absl::optional<int64_t> absolute_capture_timestamp_ms) {
rtc::CritScope lock(&lock_);
if (sink_) {
sink_->OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,
- number_of_frames);
+ number_of_frames, absolute_capture_timestamp_ms);
}
}