zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Make AudioFrameType an enum class, and move to audio_coding_module_typedefs.h

Browse Source 
Bug: webrtc:5876
Change-Id: I0c92f9410fcf0832bfa321229b3437134255dba6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128085
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27190}
This commit is contained in:
Niels Möller
2019-03-19 14:10:16 +01:00
committed by Commit Bot
parent 1d14af908e
commit c936cb6a86
21 changed files with 91 additions and 102 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
modules/audio_coding/test/opus_test.cc
View File

@ -315,8 +315,8 @@ void OpusTest::Run(TestPackStereo* channel,
}
// Send data to the channel. "channel" will handle the loss simulation.
channel->SendData(kAudioFrameSpeech, payload_type_, rtp_timestamp_,
bitstream, bitstream_len_byte, NULL);
channel->SendData(AudioFrameType::kAudioFrameSpeech, payload_type_,
rtp_timestamp_, bitstream, bitstream_len_byte, NULL);
if (first_packet) {
first_packet = false;
start_time_stamp = rtp_timestamp_;