| commit | bf47495979b8e938fd4f05ca13e6a3a6ab639387 | [log] [tgz] |
|---|---|---|
| author | Niels Möller <nisse@webrtc.org> | Mon Feb 18 12:00:06 2019 +0100 |
| committer | Commit Bot <commit-bot@chromium.org> | Mon Feb 18 13:29:35 2019 +0000 |
| tree | 9dc8ad343e4253a688988f94ff4bb50301c62b8c | |
| parent | a0b1fb9ac7698948736aa53c6d48972cb40779b5 [diff] [blame] |
Update remaining audio test code to not use WebRtcRTPHeader. Bug: webrtc:5876 Change-Id: I5b1abcec4a0ef52b6dd36d1fe94dbfd3f88f28a7 Reviewed-on: https://webrtc-review.googlesource.com/c/123235 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26736}
diff --git a/modules/audio_coding/test/EncodeDecodeTest.h b/modules/audio_coding/test/EncodeDecodeTest.h index df6ee5f..d9c22d7 100644 --- a/modules/audio_coding/test/EncodeDecodeTest.h +++ b/modules/audio_coding/test/EncodeDecodeTest.h
@@ -84,7 +84,7 @@ AudioCodingModule* _acm; uint8_t _incomingPayload[MAX_INCOMING_PAYLOAD]; RTPStream* _rtpStream; - WebRtcRTPHeader _rtpInfo; + RTPHeader _rtpHeader; size_t _realPayloadSizeBytes; size_t _payloadSizeBytes; uint32_t _nextTime;