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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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8
modules/audio_coding/test/RTPFile.h
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@ -33,7 +33,7 @@ class RTPStream {
// Returns the packet's payload size. Zero should be treated as an
// end-of-stream (in the case that EndOfFile() is true) or an error.
virtual size_t Read(WebRtcRTPHeader* rtpInfo,
virtual size_t Read(RTPHeader* rtp_Header,
uint8_t* payloadData,
size_t payloadSize,
uint32_t* offset) = 0;
@ -46,7 +46,7 @@ class RTPStream {
uint32_t timeStamp,
uint32_t ssrc);
void ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const uint8_t* rtpHeader);
void ParseRTPHeader(RTPHeader* rtp_header, const uint8_t* rtpHeader);
};
class RTPPacket {
@ -81,7 +81,7 @@ class RTPBuffer : public RTPStream {
const size_t payloadSize,
uint32_t frequency) override;
size_t Read(WebRtcRTPHeader* rtpInfo,
size_t Read(RTPHeader* rtp_header,
uint8_t* payloadData,
size_t payloadSize,
uint32_t* offset) override;
@ -114,7 +114,7 @@ class RTPFile : public RTPStream {
const size_t payloadSize,
uint32_t frequency) override;
size_t Read(WebRtcRTPHeader* rtpInfo,
size_t Read(RTPHeader* rtp_header,
uint8_t* payloadData,
size_t payloadSize,
uint32_t* offset) override;