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Add absolute capture time property to rtp sources.

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This part of the effort to implement A/V sync metric.

Bug: webrtc:10739
Change-Id: I4adba1b99b37b31868168e37d9aa8e03f8ea6d4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159886
Commit-Queue: Ruslan Burakov <kuddai@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ruslan Burakov <kuddai@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29849}
This commit is contained in:
Ruslan Burakov
2019-11-20 16:48:34 +01:00
committed by Commit Bot
parent bb55e0bc72
commit d51cc7bd71
5 changed files with 86 additions and 28 deletions
Download Patch File Download Diff File Expand all files Collapse all files

1
api/transport/rtp/BUILD.gn
View File

@ -14,6 +14,7 @@ rtc_source_set("rtp_source") {
"rtp_source.h",
]
deps = [
"../../../api:rtp_headers",
"../../../rtc_base:checks",
"//third_party/abseil-cpp/absl/types:optional",
]

39
api/transport/rtp/rtp_source.h
View File

@ -14,6 +14,7 @@
#include <stdint.h>
#include "absl/types/optional.h"
#include "api/rtp_headers.h"
#include "rtc_base/checks.h"
namespace webrtc {
@ -25,17 +26,35 @@ enum class RtpSourceType {
class RtpSource {
public:
struct Extensions {
absl::optional<uint8_t> audio_level;
absl::optional<AbsoluteCaptureTime> absolute_capture_time;
};
RtpSource() = delete;
// TODO(bugs.webrtc.org/10739): Remove this constructor once all clients
// migrate to the version with absolute capture time.
RtpSource(int64_t timestamp_ms,
uint32_t source_id,
RtpSourceType source_type,
absl::optional<uint8_t> audio_level,
uint32_t rtp_timestamp)
: RtpSource(timestamp_ms,
source_id,
source_type,
rtp_timestamp,
{audio_level, absl::nullopt}) {}
RtpSource(int64_t timestamp_ms,
uint32_t source_id,
RtpSourceType source_type,
uint32_t rtp_timestamp,
const RtpSource::Extensions& extensions)
: timestamp_ms_(timestamp_ms),
source_id_(source_id),
source_type_(source_type),
audio_level_(audio_level),
extensions_(extensions),
rtp_timestamp_(rtp_timestamp) {}
RtpSource(const RtpSource&) = default;
@ -54,16 +73,26 @@ class RtpSource {
// The source can be either a contributing source or a synchronization source.
RtpSourceType source_type() const { return source_type_; }
absl::optional<uint8_t> audio_level() const { return audio_level_; }
absl::optional<uint8_t> audio_level() const {
return extensions_.audio_level;
}
void set_audio_level(const absl::optional<uint8_t>& level) {
audio_level_ = level;
extensions_.audio_level = level;
}
uint32_t rtp_timestamp() const { return rtp_timestamp_; }
absl::optional<AbsoluteCaptureTime> absolute_capture_time() const {
return extensions_.absolute_capture_time;
}
bool operator==(const RtpSource& o) const {
return timestamp_ms_ == o.timestamp_ms() && source_id_ == o.source_id() &&
source_type_ == o.source_type() && audio_level_ == o.audio_level_ &&
source_type_ == o.source_type() &&
extensions_.audio_level == o.extensions_.audio_level &&
extensions_.absolute_capture_time ==
o.extensions_.absolute_capture_time &&
rtp_timestamp_ == o.rtp_timestamp();
}
@ -71,7 +100,7 @@ class RtpSource {
int64_t timestamp_ms_;
uint32_t source_id_;
RtpSourceType source_type_;
absl::optional<uint8_t> audio_level_;
RtpSource::Extensions extensions_;
uint32_t rtp_timestamp_;
};

7
modules/rtp_rtcp/source/source_tracker.cc
View File

@ -34,6 +34,7 @@ void SourceTracker::OnFrameDelivered(const RtpPacketInfos& packet_infos) {
entry.timestamp_ms = now_ms;
entry.audio_level = packet_info.audio_level();
entry.absolute_capture_time = packet_info.absolute_capture_time();
entry.rtp_timestamp = packet_info.rtp_timestamp();
}
@ -42,6 +43,7 @@ void SourceTracker::OnFrameDelivered(const RtpPacketInfos& packet_infos) {
entry.timestamp_ms = now_ms;
entry.audio_level = packet_info.audio_level();
entry.absolute_capture_time = packet_info.absolute_capture_time();
entry.rtp_timestamp = packet_info.rtp_timestamp();
}
@ -60,8 +62,9 @@ std::vector<RtpSource> SourceTracker::GetSources() const {
const SourceKey& key = pair.first;
const SourceEntry& entry = pair.second;
sources.emplace_back(entry.timestamp_ms, key.source, key.source_type,
entry.audio_level, entry.rtp_timestamp);
sources.emplace_back(
entry.timestamp_ms, key.source, key.source_type, entry.rtp_timestamp,
RtpSource::Extensions{entry.audio_level, entry.absolute_capture_time});
}
return sources;

5
modules/rtp_rtcp/source/source_tracker.h
View File

@ -90,6 +90,11 @@ class SourceTracker {
// specs for `RTCRtpContributingSource` for more info.
absl::optional<uint8_t> audio_level;
// Absolute capture time header extension received or interpolated from the
// most recent packet used to assemble the frame. For more info see
// https://webrtc.org/experiments/rtp-hdrext/abs-capture-time/
absl::optional<AbsoluteCaptureTime> absolute_capture_time;
// RTP timestamp of the most recent packet used to assemble the frame
// associated with |timestamp_ms|.
uint32_t rtp_timestamp;

62
modules/rtp_rtcp/source/source_tracker_unittest.cc
View File

@ -18,6 +18,7 @@
#include <utility>
#include <vector>
#include "absl/types/optional.h"
#include "api/rtp_headers.h"
#include "api/rtp_packet_info.h"
#include "api/rtp_packet_infos.h"
@ -46,15 +47,16 @@ class ExpectedSourceTracker {
const int64_t now_ms = clock_->TimeInMilliseconds();
for (const auto& packet_info : packet_infos) {
RtpSource::Extensions extensions = {packet_info.audio_level(),
packet_info.absolute_capture_time()};
for (const auto& csrc : packet_info.csrcs()) {
entries_.emplace_front(now_ms, csrc, RtpSourceType::CSRC,
packet_info.audio_level(),
packet_info.rtp_timestamp());
packet_info.rtp_timestamp(), extensions);
}
entries_.emplace_front(now_ms, packet_info.ssrc(), RtpSourceType::SSRC,
packet_info.audio_level(),
packet_info.rtp_timestamp());
packet_info.rtp_timestamp(), extensions);
}
PruneEntries(now_ms);
@ -243,7 +245,9 @@ TEST(SourceTrackerTest, OnFrameDeliveredRecordsSources) {
constexpr uint32_t kCsrcs1 = 21;
constexpr uint32_t kRtpTimestamp = 40;
constexpr absl::optional<uint8_t> kAudioLevel = 50;
constexpr absl::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime = {};
constexpr absl::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime =
AbsoluteCaptureTime{/*absolute_capture_timestamp=*/12,
/*estimated_capture_clock_offset=*/absl::nullopt};
constexpr int64_t kReceiveTimeMs = 60;
SimulatedClock clock(1000000000000ULL);
@ -254,14 +258,16 @@ TEST(SourceTrackerTest, OnFrameDeliveredRecordsSources) {
kAbsoluteCaptureTime, kReceiveTimeMs)}));
int64_t timestamp_ms = clock.TimeInMilliseconds();
constexpr RtpSource::Extensions extensions = {kAudioLevel,
kAbsoluteCaptureTime};
EXPECT_THAT(tracker.GetSources(),
ElementsAre(RtpSource(timestamp_ms, kSsrc, RtpSourceType::SSRC,
kAudioLevel, kRtpTimestamp),
kRtpTimestamp, extensions),
RtpSource(timestamp_ms, kCsrcs1, RtpSourceType::CSRC,
kAudioLevel, kRtpTimestamp),
kRtpTimestamp, extensions),
RtpSource(timestamp_ms, kCsrcs0, RtpSourceType::CSRC,
kAudioLevel, kRtpTimestamp)));
kRtpTimestamp, extensions)));
}
TEST(SourceTrackerTest, OnFrameDeliveredUpdatesSources) {
@ -273,7 +279,10 @@ TEST(SourceTrackerTest, OnFrameDeliveredUpdatesSources) {
constexpr uint32_t kRtpTimestamp1 = 41;
constexpr absl::optional<uint8_t> kAudioLevel0 = 50;
constexpr absl::optional<uint8_t> kAudioLevel1 = absl::nullopt;
constexpr absl::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime = {};
constexpr absl::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime0 =
AbsoluteCaptureTime{12, 34};
constexpr absl::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime1 =
AbsoluteCaptureTime{56, 78};
constexpr int64_t kReceiveTimeMs0 = 60;
constexpr int64_t kReceiveTimeMs1 = 61;
@ -282,7 +291,7 @@ TEST(SourceTrackerTest, OnFrameDeliveredUpdatesSources) {
tracker.OnFrameDelivered(RtpPacketInfos(
{RtpPacketInfo(kSsrc, {kCsrcs0, kCsrcs1}, kRtpTimestamp0, kAudioLevel0,
kAbsoluteCaptureTime, kReceiveTimeMs0)}));
kAbsoluteCaptureTime0, kReceiveTimeMs0)}));
int64_t timestamp_ms_0 = clock.TimeInMilliseconds();
@ -290,20 +299,25 @@ TEST(SourceTrackerTest, OnFrameDeliveredUpdatesSources) {
tracker.OnFrameDelivered(RtpPacketInfos(
{RtpPacketInfo(kSsrc, {kCsrcs0, kCsrcs2}, kRtpTimestamp1, kAudioLevel1,
kAbsoluteCaptureTime, kReceiveTimeMs1)}));
kAbsoluteCaptureTime1, kReceiveTimeMs1)}));
int64_t timestamp_ms_1 = clock.TimeInMilliseconds();
constexpr RtpSource::Extensions extensions0 = {kAudioLevel0,
kAbsoluteCaptureTime0};
constexpr RtpSource::Extensions extensions1 = {kAudioLevel1,
kAbsoluteCaptureTime1};
EXPECT_THAT(
tracker.GetSources(),
ElementsAre(RtpSource(timestamp_ms_1, kSsrc, RtpSourceType::SSRC,
kAudioLevel1, kRtpTimestamp1),
kRtpTimestamp1, extensions1),
RtpSource(timestamp_ms_1, kCsrcs2, RtpSourceType::CSRC,
kAudioLevel1, kRtpTimestamp1),
kRtpTimestamp1, extensions1),
RtpSource(timestamp_ms_1, kCsrcs0, RtpSourceType::CSRC,
kAudioLevel1, kRtpTimestamp1),
kRtpTimestamp1, extensions1),
RtpSource(timestamp_ms_0, kCsrcs1, RtpSourceType::CSRC,
kAudioLevel0, kRtpTimestamp0)));
kRtpTimestamp0, extensions0)));
}
TEST(SourceTrackerTest, TimedOutSourcesAreRemoved) {
@ -315,7 +329,10 @@ TEST(SourceTrackerTest, TimedOutSourcesAreRemoved) {
constexpr uint32_t kRtpTimestamp1 = 41;
constexpr absl::optional<uint8_t> kAudioLevel0 = 50;
constexpr absl::optional<uint8_t> kAudioLevel1 = absl::nullopt;
constexpr absl::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime = {};
constexpr absl::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime0 =
AbsoluteCaptureTime{12, 34};
constexpr absl::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime1 =
AbsoluteCaptureTime{56, 78};
constexpr int64_t kReceiveTimeMs0 = 60;
constexpr int64_t kReceiveTimeMs1 = 61;
@ -324,26 +341,29 @@ TEST(SourceTrackerTest, TimedOutSourcesAreRemoved) {
tracker.OnFrameDelivered(RtpPacketInfos(
{RtpPacketInfo(kSsrc, {kCsrcs0, kCsrcs1}, kRtpTimestamp0, kAudioLevel0,
kAbsoluteCaptureTime, kReceiveTimeMs0)}));
kAbsoluteCaptureTime0, kReceiveTimeMs0)}));
clock.AdvanceTimeMilliseconds(17);
tracker.OnFrameDelivered(RtpPacketInfos(
{RtpPacketInfo(kSsrc, {kCsrcs0, kCsrcs2}, kRtpTimestamp1, kAudioLevel1,
kAbsoluteCaptureTime, kReceiveTimeMs1)}));
kAbsoluteCaptureTime1, kReceiveTimeMs1)}));
int64_t timestamp_ms_1 = clock.TimeInMilliseconds();
clock.AdvanceTimeMilliseconds(SourceTracker::kTimeoutMs);
constexpr RtpSource::Extensions extensions1 = {kAudioLevel1,
kAbsoluteCaptureTime1};
EXPECT_THAT(
tracker.GetSources(),
ElementsAre(RtpSource(timestamp_ms_1, kSsrc, RtpSourceType::SSRC,
kAudioLevel1, kRtpTimestamp1),
kRtpTimestamp1, extensions1),
RtpSource(timestamp_ms_1, kCsrcs2, RtpSourceType::CSRC,
kAudioLevel1, kRtpTimestamp1),
kRtpTimestamp1, extensions1),
RtpSource(timestamp_ms_1, kCsrcs0, RtpSourceType::CSRC,
kAudioLevel1, kRtpTimestamp1)));
kRtpTimestamp1, extensions1)));
}
} // namespace webrtc