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Send absolute capture time through audio coding module.

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Bug: webrtc:10739
Change-Id: I44e0305be7c84b253172511c2977b1d700e40c1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167067
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30363}
This commit is contained in:
Minyue Li
2020-01-23 13:45:50 +01:00
committed by Commit Bot
parent cdd73e095c
commit 48655cfdbf
17 changed files with 69 additions and 30 deletions
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2
modules/audio_coding/test/opus_test.cc
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@ -337,7 +337,7 @@ void OpusTest::Run(TestPackStereo* channel,
// Send data to the channel. "channel" will handle the loss simulation.
channel->SendData(AudioFrameType::kAudioFrameSpeech, payload_type_,
rtp_timestamp_, bitstream, bitstream_len_byte);
rtp_timestamp_, bitstream, bitstream_len_byte, 0);
if (first_packet) {
first_packet = false;
start_time_stamp = rtp_timestamp_;