Add absolute capture time property to rtp sources.
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}
diff --git a/api/transport/rtp/BUILD.gn b/api/transport/rtp/BUILD.gn
index 04298e3..777ff22 100644
--- a/api/transport/rtp/BUILD.gn
+++ b/api/transport/rtp/BUILD.gn
@@ -14,6 +14,7 @@
"rtp_source.h",
]
deps = [
+ "../../../api:rtp_headers",
"../../../rtc_base:checks",
"//third_party/abseil-cpp/absl/types:optional",
]
diff --git a/api/transport/rtp/rtp_source.h b/api/transport/rtp/rtp_source.h
index d26572c..8c543ca 100644
--- a/api/transport/rtp/rtp_source.h
+++ b/api/transport/rtp/rtp_source.h
@@ -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 @@
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 @@
// 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 @@
int64_t timestamp_ms_;
uint32_t source_id_;
RtpSourceType source_type_;
- absl::optional<uint8_t> audio_level_;
+ RtpSource::Extensions extensions_;
uint32_t rtp_timestamp_;
};
diff --git a/modules/rtp_rtcp/source/source_tracker.cc b/modules/rtp_rtcp/source/source_tracker.cc
index 2878b11..22b887c 100644
--- a/modules/rtp_rtcp/source/source_tracker.cc
+++ b/modules/rtp_rtcp/source/source_tracker.cc
@@ -34,6 +34,7 @@
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 @@
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 @@
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;
diff --git a/modules/rtp_rtcp/source/source_tracker.h b/modules/rtp_rtcp/source/source_tracker.h
index 3ea16e9..fcf99bf 100644
--- a/modules/rtp_rtcp/source/source_tracker.h
+++ b/modules/rtp_rtcp/source/source_tracker.h
@@ -90,6 +90,11 @@
// 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;
diff --git a/modules/rtp_rtcp/source/source_tracker_unittest.cc b/modules/rtp_rtcp/source/source_tracker_unittest.cc
index 8f6a512..32f9f4b 100644
--- a/modules/rtp_rtcp/source/source_tracker_unittest.cc
+++ b/modules/rtp_rtcp/source/source_tracker_unittest.cc
@@ -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 @@
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 @@
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 @@
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 @@
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 @@
tracker.OnFrameDelivered(RtpPacketInfos(
{RtpPacketInfo(kSsrc, {kCsrcs0, kCsrcs1}, kRtpTimestamp0, kAudioLevel0,
- kAbsoluteCaptureTime, kReceiveTimeMs0)}));
+ kAbsoluteCaptureTime0, kReceiveTimeMs0)}));
int64_t timestamp_ms_0 = clock.TimeInMilliseconds();
@@ -290,20 +299,25 @@
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 @@
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 @@
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