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Add default values for PlayoutDelay in RTPVideoHeader.

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There have been several bugs where the members of PlayoutDelay were
zero initialized when handling RTP packets without the corresponding
extensions. Initializing to {-1, -1} (meaning not provided) is less
brittle.

Bug: None
Change-Id: I196850377128d5e67a19bdaf9298403b2e9f5a6e
Reviewed-on: https://webrtc-review.googlesource.com/c/111181
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25670}
This commit is contained in:
Niels Möller
2018-11-16 12:35:26 +01:00
committed by Commit Bot
parent 7dbb7c311f
commit a32d7e2a2f
4 changed files with 11 additions and 4 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
modules/rtp_rtcp/source/rtp_video_header.cc
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@ -12,7 +12,7 @@
namespace webrtc { namespace webrtc {
RTPVideoHeader::RTPVideoHeader() : playout_delay(), video_timing() {} RTPVideoHeader::RTPVideoHeader() : video_timing() {}
RTPVideoHeader::RTPVideoHeader(const RTPVideoHeader& other) = default; RTPVideoHeader::RTPVideoHeader(const RTPVideoHeader& other) = default;
RTPVideoHeader::~RTPVideoHeader() = default; RTPVideoHeader::~RTPVideoHeader() = default;

2
modules/rtp_rtcp/source/rtp_video_header.h
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@ -59,7 +59,7 @@ struct RTPVideoHeader {
uint8_t simulcastIdx = 0; uint8_t simulcastIdx = 0;
VideoCodecType codec = VideoCodecType::kVideoCodecGeneric; VideoCodecType codec = VideoCodecType::kVideoCodecGeneric;
PlayoutDelay playout_delay; PlayoutDelay playout_delay = {-1, -1};
VideoSendTiming video_timing; VideoSendTiming video_timing;
FrameMarking frame_marking; FrameMarking frame_marking;
RTPVideoTypeHeader video_type_header; RTPVideoTypeHeader video_type_header;

2
video/rtp_video_stream_receiver.cc
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@ -530,8 +530,6 @@ void RtpVideoStreamReceiver::ReceivePacket(const RtpPacketReceived& packet) {
webrtc_rtp_header.video_header().content_type = VideoContentType::UNSPECIFIED; webrtc_rtp_header.video_header().content_type = VideoContentType::UNSPECIFIED;
webrtc_rtp_header.video_header().video_timing.flags = webrtc_rtp_header.video_header().video_timing.flags =
VideoSendTiming::kInvalid; VideoSendTiming::kInvalid;
webrtc_rtp_header.video_header().playout_delay.min_ms = -1;
webrtc_rtp_header.video_header().playout_delay.max_ms = -1;
webrtc_rtp_header.video_header().is_last_packet_in_frame = webrtc_rtp_header.video_header().is_last_packet_in_frame =
webrtc_rtp_header.header.markerBit; webrtc_rtp_header.header.markerBit;

9
video/rtp_video_stream_receiver_unittest.cc
View File

@ -626,4 +626,13 @@ TEST_F(RtpVideoStreamReceiverTest, RepeatedSecondarySinkDisallowed) {
} }
#endif #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 } // namespace webrtc