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Add writing and parsing of the abs-capture-time RTP header extension.

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This change adds the writing and parsing of the `abs-capture-time` RTP header extension defined at:

  http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time

We are still missing the code to:

- Negotiate the header extension.
- Collect capture time for audio and video and have the info sent with the header extension.
- Receive the header extension and use its info.

Bug: webrtc:10739
Change-Id: I75af492e994367f45a5bdc110af199900327b126
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144221
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28468}
This commit is contained in:
Chen Xing
2019-07-01 10:56:51 +02:00
committed by Commit Bot
parent 53d45baa50
commit cd8a6e2f38
14 changed files with 250 additions and 3 deletions
Download Patch File Download Diff File Expand all files Collapse all files

19
modules/rtp_rtcp/source/rtp_utility_unittest.cc
View File

@ -149,23 +149,27 @@ TEST(RtpHeaderParser, ParseWithOverSizedExtension) {
EXPECT_EQ(sizeof(kPacket), header.headerLength);
}
TEST(RtpHeaderParser, ParseAll8Extensions) {
TEST(RtpHeaderParser, ParseAll9Extensions) {
const uint8_t kAudioLevel = 0x5a;
// clang-format off
const uint8_t kPacket[] = {
0x90, kPayloadType, 0x00, kSeqNum,
0x65, 0x43, 0x12, 0x78, // kTimestamp.
0x12, 0x34, 0x56, 0x78, // kSsrc.
0xbe, 0xde, 0x00, 0x08, // Extension of size 8x32bit words.
0xbe, 0xde, 0x00, 0x0c, // Extension of size 12x32bit words.
0x40, 0x80|kAudioLevel, // AudioLevel.
0x22, 0x01, 0x56, 0xce, // TransmissionOffset.
0x62, 0x12, 0x34, 0x56, // AbsoluteSendTime.
0x7f, 0x12, 0x34, 0x56, 0x78, // AbsoluteCaptureTime.
0x90, 0xab, 0xcd, 0xef, // AbsoluteCaptureTime. (cont.)
0x11, 0x22, 0x33, 0x44, // AbsoluteCaptureTime. (cont.)
0x55, 0x66, 0x77, 0x88, // AbsoluteCaptureTime. (cont.)
0x81, 0xce, 0xab, // TransportSequenceNumber.
0xa0, 0x03, // VideoRotation.
0xb2, 0x12, 0x48, 0x76, // PlayoutDelayLimits.
0xc2, 'r', 't', 'x', // RtpStreamId
0xd5, 's', 't', 'r', 'e', 'a', 'm', // RepairedRtpStreamId
0x00, 0x00, // Padding to 32bit boundary.
0x00, // Padding to 32bit boundary.
};
// clang-format on
ASSERT_EQ(sizeof(kPacket) % 4, 0u);
@ -174,6 +178,7 @@ TEST(RtpHeaderParser, ParseAll8Extensions) {
extensions.Register<TransmissionOffset>(2);
extensions.Register<AudioLevel>(4);
extensions.Register<AbsoluteSendTime>(6);
extensions.Register<AbsoluteCaptureTimeExtension>(7);
extensions.Register<TransportSequenceNumber>(8);
extensions.Register<VideoOrientation>(0xa);
extensions.Register<PlayoutDelayLimits>(0xb);
@ -194,6 +199,14 @@ TEST(RtpHeaderParser, ParseAll8Extensions) {
EXPECT_TRUE(header.extension.hasAbsoluteSendTime);
EXPECT_EQ(0x123456U, header.extension.absoluteSendTime);
ASSERT_TRUE(header.extension.absolute_capture_time.has_value());
EXPECT_EQ(0x1234567890abcdefULL,
header.extension.absolute_capture_time->absolute_capture_timestamp);
ASSERT_TRUE(header.extension.absolute_capture_time
->estimated_capture_clock_offset.has_value());
EXPECT_EQ(0x1122334455667788LL, header.extension.absolute_capture_time
->estimated_capture_clock_offset.value());
EXPECT_TRUE(header.extension.hasTransportSequenceNumber);
EXPECT_EQ(0xceab, header.extension.transportSequenceNumber);