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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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3
modules/audio_coding/test/EncodeDecodeTest.cc
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@ -33,7 +33,8 @@ int32_t TestPacketization::SendData(const AudioFrameType /* frameType */,
const uint8_t payloadType,
const uint32_t timeStamp,
const uint8_t* payloadData,
const size_t payloadSize) {
const size_t payloadSize,
int64_t absolute_capture_timestamp_ms) {
_rtpStream->Write(payloadType, timeStamp, _seqNo++, payloadData, payloadSize,
_frequency);
return 1;