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Use the clock to convert absolute capture timestamps to NTP times.

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This allows callers to use timestamps generated from their own clocks
without worrying about converting to webrtc time.

No-Try because of lack of infra lack of capacity on macs.

No-Try: True
Bug: webrtc:11327
Change-Id: I7b1935654a2b23cf844c7b3622ed68763ced9da5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219785
Commit-Queue: Paul Hallak <phallak@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34076}
This commit is contained in:
Paul Hallak
2021-05-21 10:34:34 +02:00
committed by WebRTC LUCI CQ
parent edc347c4e5
commit 47ed99872d
6 changed files with 23 additions and 12 deletions
Download Patch File Download Diff File Expand all files Collapse all files

3
modules/rtp_rtcp/source/rtp_sender_audio.cc
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@ -287,7 +287,8 @@ bool RTPSenderAudio::SendAudio(AudioFrameType frame_type,
// Replace missing value with 0 (invalid frequency), this will trigger
// absolute capture time sending.
encoder_rtp_timestamp_frequency.value_or(0),
Int64MsToUQ32x32(absolute_capture_timestamp_ms + NtpOffsetMs()),
Int64MsToUQ32x32(clock_->ConvertTimestampToNtpTimeInMilliseconds(
absolute_capture_timestamp_ms)),
/*estimated_capture_clock_offset=*/
include_capture_clock_offset_ ? absl::make_optional(0) : absl::nullopt);
if (absolute_capture_time) {

2
modules/rtp_rtcp/source/rtp_sender_audio.h
View File

@ -51,6 +51,8 @@ class RTPSenderAudio {
const uint8_t* payload_data,
size_t payload_size);
// `absolute_capture_timestamp_ms` and `Clock::CurrentTime`
// should be using the same epoch.
bool SendAudio(AudioFrameType frame_type,
int8_t payload_type,
uint32_t rtp_timestamp,

13
modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
View File

@ -19,7 +19,6 @@
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
#include "modules/rtp_rtcp/source/time_util.h"
#include "test/field_trial.h"
#include "test/gmock.h"
#include "test/gtest.h"
@ -167,8 +166,10 @@ TEST_F(RtpSenderAudioTest, SendAudioWithAbsoluteCaptureTime) {
transport_.last_sent_packet()
.GetExtension<AbsoluteCaptureTimeExtension>();
EXPECT_TRUE(absolute_capture_time);
EXPECT_EQ(absolute_capture_time->absolute_capture_timestamp,
Int64MsToUQ32x32(kAbsoluteCaptureTimestampMs + NtpOffsetMs()));
EXPECT_EQ(
absolute_capture_time->absolute_capture_timestamp,
Int64MsToUQ32x32(fake_clock_.ConvertTimestampToNtpTimeInMilliseconds(
kAbsoluteCaptureTimestampMs)));
EXPECT_FALSE(
absolute_capture_time->estimated_capture_clock_offset.has_value());
}
@ -201,8 +202,10 @@ TEST_F(RtpSenderAudioTest,
transport_.last_sent_packet()
.GetExtension<AbsoluteCaptureTimeExtension>();
EXPECT_TRUE(absolute_capture_time);
EXPECT_EQ(absolute_capture_time->absolute_capture_timestamp,
Int64MsToUQ32x32(kAbsoluteCaptureTimestampMs + NtpOffsetMs()));
EXPECT_EQ(
absolute_capture_time->absolute_capture_timestamp,
Int64MsToUQ32x32(fake_clock_.ConvertTimestampToNtpTimeInMilliseconds(
kAbsoluteCaptureTimestampMs)));
EXPECT_TRUE(
absolute_capture_time->estimated_capture_clock_offset.has_value());
EXPECT_EQ(0, *absolute_capture_time->estimated_capture_clock_offset);

3
modules/rtp_rtcp/source/rtp_sender_video.cc
View File

@ -523,7 +523,8 @@ bool RTPSenderVideo::SendVideo(
AbsoluteCaptureTimeSender::GetSource(single_packet->Ssrc(),
single_packet->Csrcs()),
single_packet->Timestamp(), kVideoPayloadTypeFrequency,
Int64MsToUQ32x32(single_packet->capture_time_ms() + NtpOffsetMs()),
Int64MsToUQ32x32(
clock_->ConvertTimestampToNtpTimeInMilliseconds(capture_time_ms)),
/*estimated_capture_clock_offset=*/
include_capture_clock_offset_ ? estimated_capture_clock_offset_ms
: absl::nullopt);

1
modules/rtp_rtcp/source/rtp_sender_video.h
View File

@ -89,6 +89,7 @@ class RTPSenderVideo {
virtual ~RTPSenderVideo();
// expected_retransmission_time_ms.has_value() -> retransmission allowed.
// `capture_time_ms` and `clock::CurrentTime` should be using the same epoch.
// Calls to this method is assumed to be externally serialized.
// |estimated_capture_clock_offset_ms| is an estimated clock offset between
// this sender and the original capturer, for this video packet. See

13
modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
View File

@ -34,7 +34,6 @@
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
#include "modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.h"
#include "modules/rtp_rtcp/source/time_util.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/rate_limiter.h"
#include "rtc_base/task_queue_for_test.h"
@ -1054,8 +1053,10 @@ TEST_P(RtpSenderVideoTest, AbsoluteCaptureTime) {
packet.GetExtension<AbsoluteCaptureTimeExtension>();
if (absolute_capture_time) {
++packets_with_abs_capture_time;
EXPECT_EQ(absolute_capture_time->absolute_capture_timestamp,
Int64MsToUQ32x32(kAbsoluteCaptureTimestampMs + NtpOffsetMs()));
EXPECT_EQ(
absolute_capture_time->absolute_capture_timestamp,
Int64MsToUQ32x32(fake_clock_.ConvertTimestampToNtpTimeInMilliseconds(
kAbsoluteCaptureTimestampMs)));
EXPECT_FALSE(
absolute_capture_time->estimated_capture_clock_offset.has_value());
}
@ -1092,8 +1093,10 @@ TEST_P(RtpSenderVideoTest, AbsoluteCaptureTimeWithCaptureClockOffset) {
packet.GetExtension<AbsoluteCaptureTimeExtension>();
if (absolute_capture_time) {
++packets_with_abs_capture_time;
EXPECT_EQ(absolute_capture_time->absolute_capture_timestamp,
Int64MsToUQ32x32(kAbsoluteCaptureTimestampMs + NtpOffsetMs()));
EXPECT_EQ(
absolute_capture_time->absolute_capture_timestamp,
Int64MsToUQ32x32(fake_clock_.ConvertTimestampToNtpTimeInMilliseconds(
kAbsoluteCaptureTimestampMs)));
EXPECT_EQ(kExpectedCaptureClockOffset,
absolute_capture_time->estimated_capture_clock_offset);
}