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Delete WebRtcRTPHeader, this struct is no longer used.

Browse Source 
Bug: webrtc:10397
Change-Id: I1b7acd9c89b9e14d1d8e1914c8c12c51fe4c643f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134203
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27773}
This commit is contained in:
Niels Möller
2019-04-25 13:25:29 +02:00
committed by Commit Bot
parent f204fafdb4
commit 0fb0bd8e9f
4 changed files with 0 additions and 34 deletions
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13
modules/include/module_common_types.h
View File

@ -24,19 +24,6 @@
namespace webrtc {
struct WebRtcRTPHeader {
RTPVideoHeader& video_header() { return video; }
const RTPVideoHeader& video_header() const { return video; }
RTPVideoHeader video;
RTPHeader header;
// Used for video only.
// TODO(nisse): Delete, now included on RTPVideoHeader.
VideoFrameType frameType;
// NTP time of the capture time in local timebase in milliseconds.
int64_t ntp_time_ms;
};
class RTC_EXPORT RTPFragmentationHeader {
public:
RTPFragmentationHeader();

5
modules/video_coding/include/video_coding.h
View File

@ -137,11 +137,6 @@ class VideoCodingModule : public Module {
const RTPHeader& rtp_header,
const RTPVideoHeader& video_header) = 0;
// DEPRECATED
virtual int32_t IncomingPacket(const uint8_t* incomingPayload,
size_t payloadLength,
const WebRtcRTPHeader& rtpInfo) = 0;
// Robustness APIs
// DEPRECATED.

7
modules/video_coding/video_coding_impl.cc
View File

@ -90,13 +90,6 @@ class VideoCodingModuleImpl : public VideoCodingModule {
return receiver_.Decode(maxWaitTimeMs);
}
int32_t IncomingPacket(const uint8_t* incomingPayload,
size_t payloadLength,
const WebRtcRTPHeader& rtpInfo) override {
return IncomingPacket(incomingPayload, payloadLength, rtpInfo.header,
rtpInfo.video_header());
}
int32_t IncomingPacket(const uint8_t* incomingPayload,
size_t payloadLength,
const RTPHeader& rtp_header,

9
video/rtp_video_stream_receiver_unittest.cc
View File

@ -839,13 +839,4 @@ TEST_F(RtpVideoStreamReceiverTest, RepeatedSecondarySinkDisallowed) {
}
#endif
// Initialization of WebRtcRTPHeader is a bit convoluted, with some fields
// zero-initialized. RtpVideoStreamReceiver depends on proper default values for
// the playout delay.
TEST(WebRtcRTPHeader, DefaultPlayoutDelayIsUnspecified) {
WebRtcRTPHeader webrtc_rtp_header = {};
EXPECT_EQ(webrtc_rtp_header.video_header().playout_delay.min_ms, -1);
EXPECT_EQ(webrtc_rtp_header.video_header().playout_delay.max_ms, -1);
}
} // namespace webrtc