| commit | c35b6e675a878996c61e89f2b472f6f71bd74d67 | [log] [tgz] |
|---|---|---|
| author | Niels Möller <nisse@webrtc.org> | Thu Apr 25 16:31:18 2019 +0200 |
| committer | Commit Bot <commit-bot@chromium.org> | Fri Apr 26 12:58:14 2019 +0000 |
| tree | 2edb2df58ddcb4a17a5c9a8a6754c06cfda352c9 | |
| parent | e670fd97951a967283f74ff9b313dc16aa19e7af [diff] [blame] |
Deprecate RTPFragmentationHeader argument to AudioPacketizationCallback::SendData It appears unused everywhere. It will be deleted in a followup cl. Bug: webrtc:6471 Change-Id: Ief992db6e52aee3cf1bc77ffd659ffbc072672ba Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134212 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27787}
diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc index ad742ce..a3f0964 100644 --- a/modules/audio_coding/test/TestAllCodecs.cc +++ b/modules/audio_coding/test/TestAllCodecs.cc
@@ -64,8 +64,7 @@ uint8_t payload_type, uint32_t timestamp, const uint8_t* payload_data, - size_t payload_size, - const RTPFragmentationHeader* fragmentation) { + size_t payload_size) { RTPHeader rtp_header; int32_t status;