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/RTPFile.h b/modules/audio_coding/test/RTPFile.h
index 141075b..1c555ed 100644
--- a/modules/audio_coding/test/RTPFile.h
+++ b/modules/audio_coding/test/RTPFile.h
@@ -33,7 +33,7 @@
// 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 @@
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 @@
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 @@
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;