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Update remaining audio test code to not use WebRtcRTPHeader.

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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}
This commit is contained in:
Niels Möller
2019-02-18 12:00:06 +01:00
committed by Commit Bot
parent a0b1fb9ac7
commit bf47495979
7 changed files with 56 additions and 58 deletions
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2
modules/audio_coding/test/EncodeDecodeTest.h
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@ -84,7 +84,7 @@ class Receiver {
AudioCodingModule* _acm;
uint8_t _incomingPayload[MAX_INCOMING_PAYLOAD];
RTPStream* _rtpStream;
WebRtcRTPHeader _rtpInfo;
RTPHeader _rtpHeader;
size_t _realPayloadSizeBytes;
size_t _payloadSizeBytes;
uint32_t _nextTime;