Add `AbsoluteCaptureTime` to `RtpPacketInfo`.
This change stores the optional `AbsoluteCaptureTime` header extension in `RtpPacketInfo` so that we later can consume it in `SourceTracker`.
Bug: webrtc:10739
Change-Id: I975e8863117fcda134535cd49ad71079a7ff38ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148068
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28790}
diff --git a/api/BUILD.gn b/api/BUILD.gn
index 5c681bc..ffaa4aa 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -92,6 +92,7 @@
":rtp_headers",
":scoped_refptr",
"..:webrtc_common",
+ "../rtc_base:deprecation",
"../rtc_base:rtc_base_approved",
"//third_party/abseil-cpp/absl/types:optional",
]
diff --git a/api/rtp_headers.h b/api/rtp_headers.h
index e5155f0..4415bd3 100644
--- a/api/rtp_headers.h
+++ b/api/rtp_headers.h
@@ -79,6 +79,18 @@
absl::optional<int64_t> estimated_capture_clock_offset;
};
+inline bool operator==(const AbsoluteCaptureTime& lhs,
+ const AbsoluteCaptureTime& rhs) {
+ return (lhs.absolute_capture_timestamp == rhs.absolute_capture_timestamp) &&
+ (lhs.estimated_capture_clock_offset ==
+ rhs.estimated_capture_clock_offset);
+}
+
+inline bool operator!=(const AbsoluteCaptureTime& lhs,
+ const AbsoluteCaptureTime& rhs) {
+ return !(lhs == rhs);
+}
+
struct RTPHeaderExtension {
RTPHeaderExtension();
RTPHeaderExtension(const RTPHeaderExtension& other);
diff --git a/api/rtp_packet_info.cc b/api/rtp_packet_info.cc
index efb7838..54e26b4 100644
--- a/api/rtp_packet_info.cc
+++ b/api/rtp_packet_info.cc
@@ -18,16 +18,31 @@
RtpPacketInfo::RtpPacketInfo()
: ssrc_(0), rtp_timestamp_(0), receive_time_ms_(-1) {}
+RtpPacketInfo::RtpPacketInfo(
+ uint32_t ssrc,
+ std::vector<uint32_t> csrcs,
+ uint32_t rtp_timestamp,
+ absl::optional<uint8_t> audio_level,
+ absl::optional<AbsoluteCaptureTime> absolute_capture_time,
+ int64_t receive_time_ms)
+ : ssrc_(ssrc),
+ csrcs_(std::move(csrcs)),
+ rtp_timestamp_(rtp_timestamp),
+ audio_level_(audio_level),
+ absolute_capture_time_(absolute_capture_time),
+ receive_time_ms_(receive_time_ms) {}
+
RtpPacketInfo::RtpPacketInfo(uint32_t ssrc,
std::vector<uint32_t> csrcs,
uint32_t rtp_timestamp,
absl::optional<uint8_t> audio_level,
int64_t receive_time_ms)
- : ssrc_(ssrc),
- csrcs_(std::move(csrcs)),
- rtp_timestamp_(rtp_timestamp),
- audio_level_(audio_level),
- receive_time_ms_(receive_time_ms) {}
+ : RtpPacketInfo(ssrc,
+ std::move(csrcs),
+ rtp_timestamp,
+ audio_level,
+ /*absolute_capture_time=*/absl::nullopt,
+ receive_time_ms) {}
RtpPacketInfo::RtpPacketInfo(const RTPHeader& rtp_header,
int64_t receive_time_ms)
@@ -42,12 +57,15 @@
if (extension.hasAudioLevel) {
audio_level_ = extension.audioLevel;
}
+
+ absolute_capture_time_ = extension.absolute_capture_time;
}
bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs) {
return (lhs.ssrc() == rhs.ssrc()) && (lhs.csrcs() == rhs.csrcs()) &&
(lhs.rtp_timestamp() == rhs.rtp_timestamp()) &&
(lhs.audio_level() == rhs.audio_level()) &&
+ (lhs.absolute_capture_time() == rhs.absolute_capture_time()) &&
(lhs.receive_time_ms() == rhs.receive_time_ms());
}
diff --git a/api/rtp_packet_info.h b/api/rtp_packet_info.h
index a9e8655..6973027 100644
--- a/api/rtp_packet_info.h
+++ b/api/rtp_packet_info.h
@@ -17,6 +17,7 @@
#include "absl/types/optional.h"
#include "api/rtp_headers.h"
+#include "rtc_base/deprecation.h"
namespace webrtc {
@@ -33,6 +34,15 @@
std::vector<uint32_t> csrcs,
uint32_t rtp_timestamp,
absl::optional<uint8_t> audio_level,
+ absl::optional<AbsoluteCaptureTime> absolute_capture_time,
+ int64_t receive_time_ms);
+
+ // TODO(bugs.webrtc.org/10739): Will be removed sometime after 2019-09-19.
+ RTC_DEPRECATED
+ RtpPacketInfo(uint32_t ssrc,
+ std::vector<uint32_t> csrcs,
+ uint32_t rtp_timestamp,
+ absl::optional<uint8_t> audio_level,
int64_t receive_time_ms);
RtpPacketInfo(const RTPHeader& rtp_header, int64_t receive_time_ms);
@@ -54,6 +64,14 @@
absl::optional<uint8_t> audio_level() const { return audio_level_; }
void set_audio_level(absl::optional<uint8_t> value) { audio_level_ = value; }
+ const absl::optional<AbsoluteCaptureTime>& absolute_capture_time() const {
+ return absolute_capture_time_;
+ }
+ void set_absolute_capture_time(
+ const absl::optional<AbsoluteCaptureTime>& value) {
+ absolute_capture_time_ = value;
+ }
+
int64_t receive_time_ms() const { return receive_time_ms_; }
void set_receive_time_ms(int64_t value) { receive_time_ms_ = value; }
@@ -68,6 +86,10 @@
// https://tools.ietf.org/html/rfc6464#section-3
absl::optional<uint8_t> audio_level_;
+ // Fields from the Absolute Capture Time header extension:
+ // http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time
+ absl::optional<AbsoluteCaptureTime> absolute_capture_time_;
+
// Local |webrtc::Clock|-based timestamp of when the packet was received.
int64_t receive_time_ms_;
};
diff --git a/api/rtp_packet_info_unittest.cc b/api/rtp_packet_info_unittest.cc
index 66cc2ed..fe79f6d 100644
--- a/api/rtp_packet_info_unittest.cc
+++ b/api/rtp_packet_info_unittest.cc
@@ -37,7 +37,7 @@
rhs = RtpPacketInfo();
EXPECT_NE(rhs.ssrc(), value);
- rhs = RtpPacketInfo(value, {}, {}, {}, {});
+ rhs = RtpPacketInfo(value, {}, {}, {}, {}, {});
EXPECT_EQ(rhs.ssrc(), value);
}
@@ -64,7 +64,7 @@
rhs = RtpPacketInfo();
EXPECT_NE(rhs.csrcs(), value);
- rhs = RtpPacketInfo({}, value, {}, {}, {});
+ rhs = RtpPacketInfo({}, value, {}, {}, {}, {});
EXPECT_EQ(rhs.csrcs(), value);
}
@@ -91,7 +91,7 @@
rhs = RtpPacketInfo();
EXPECT_NE(rhs.rtp_timestamp(), value);
- rhs = RtpPacketInfo({}, {}, value, {}, {});
+ rhs = RtpPacketInfo({}, {}, value, {}, {}, {});
EXPECT_EQ(rhs.rtp_timestamp(), value);
}
@@ -118,10 +118,37 @@
rhs = RtpPacketInfo();
EXPECT_NE(rhs.audio_level(), value);
- rhs = RtpPacketInfo({}, {}, {}, value, {});
+ rhs = RtpPacketInfo({}, {}, {}, value, {}, {});
EXPECT_EQ(rhs.audio_level(), value);
}
+TEST(RtpPacketInfoTest, AbsoluteCaptureTime) {
+ const absl::optional<AbsoluteCaptureTime> value = AbsoluteCaptureTime{12, 34};
+
+ RtpPacketInfo lhs;
+ RtpPacketInfo rhs;
+
+ EXPECT_TRUE(lhs == rhs);
+ EXPECT_FALSE(lhs != rhs);
+
+ rhs.set_absolute_capture_time(value);
+ EXPECT_EQ(rhs.absolute_capture_time(), value);
+
+ EXPECT_FALSE(lhs == rhs);
+ EXPECT_TRUE(lhs != rhs);
+
+ lhs = rhs;
+
+ EXPECT_TRUE(lhs == rhs);
+ EXPECT_FALSE(lhs != rhs);
+
+ rhs = RtpPacketInfo();
+ EXPECT_NE(rhs.absolute_capture_time(), value);
+
+ rhs = RtpPacketInfo({}, {}, {}, {}, value, {});
+ EXPECT_EQ(rhs.absolute_capture_time(), value);
+}
+
TEST(RtpPacketInfoTest, ReceiveTimeMs) {
const int64_t value = 8868963877546349045LL;
@@ -145,7 +172,7 @@
rhs = RtpPacketInfo();
EXPECT_NE(rhs.receive_time_ms(), value);
- rhs = RtpPacketInfo({}, {}, {}, {}, value);
+ rhs = RtpPacketInfo({}, {}, {}, {}, {}, value);
EXPECT_EQ(rhs.receive_time_ms(), value);
}
diff --git a/api/rtp_packet_infos_unittest.cc b/api/rtp_packet_infos_unittest.cc
index a14d448..ce502ac 100644
--- a/api/rtp_packet_infos_unittest.cc
+++ b/api/rtp_packet_infos_unittest.cc
@@ -27,9 +27,9 @@
} // namespace
TEST(RtpPacketInfosTest, BasicFunctionality) {
- RtpPacketInfo p0(123, {1, 2}, 89, 5, 7);
- RtpPacketInfo p1(456, {3, 4}, 89, 4, 1);
- RtpPacketInfo p2(789, {5, 6}, 88, 1, 7);
+ RtpPacketInfo p0(123, {1, 2}, 89, 5, AbsoluteCaptureTime{45, 78}, 7);
+ RtpPacketInfo p1(456, {3, 4}, 89, 4, AbsoluteCaptureTime{13, 21}, 1);
+ RtpPacketInfo p2(789, {5, 6}, 88, 1, AbsoluteCaptureTime{99, 78}, 7);
RtpPacketInfos x({p0, p1, p2});
@@ -52,9 +52,9 @@
}
TEST(RtpPacketInfosTest, CopyShareData) {
- RtpPacketInfo p0(123, {1, 2}, 89, 5, 7);
- RtpPacketInfo p1(456, {3, 4}, 89, 4, 1);
- RtpPacketInfo p2(789, {5, 6}, 88, 1, 7);
+ RtpPacketInfo p0(123, {1, 2}, 89, 5, AbsoluteCaptureTime{45, 78}, 7);
+ RtpPacketInfo p1(456, {3, 4}, 89, 4, AbsoluteCaptureTime{13, 21}, 1);
+ RtpPacketInfo p2(789, {5, 6}, 88, 1, AbsoluteCaptureTime{99, 78}, 7);
RtpPacketInfos lhs({p0, p1, p2});
RtpPacketInfos rhs = lhs;
diff --git a/modules/audio_coding/neteq/red_payload_splitter.cc b/modules/audio_coding/neteq/red_payload_splitter.cc
index 7ff5679..1343690 100644
--- a/modules/audio_coding/neteq/red_payload_splitter.cc
+++ b/modules/audio_coding/neteq/red_payload_splitter.cc
@@ -123,6 +123,7 @@
/*csrcs=*/std::vector<uint32_t>(),
/*rtp_timestamp=*/new_packet.timestamp,
/*audio_level=*/absl::nullopt,
+ /*absolute_capture_time=*/absl::nullopt,
/*receive_time_ms=*/red_packet.packet_info.receive_time_ms());
new_packets.push_front(std::move(new_packet));
payload_ptr += payload_length;
diff --git a/modules/rtp_rtcp/source/source_tracker_unittest.cc b/modules/rtp_rtcp/source/source_tracker_unittest.cc
index 2342697..55ae4d1 100644
--- a/modules/rtp_rtcp/source/source_tracker_unittest.cc
+++ b/modules/rtp_rtcp/source/source_tracker_unittest.cc
@@ -109,6 +109,7 @@
for (size_t i = 0; i < count; ++i) {
packet_infos.emplace_back(GenerateSsrc(), GenerateCsrcs(),
GenerateRtpTimestamp(), GenerateAudioLevel(),
+ GenerateAbsoluteCaptureTime(),
GenerateReceiveTimeMs());
}
@@ -170,6 +171,26 @@
std::uniform_int_distribution<uint16_t>()(generator_));
}
+ absl::optional<AbsoluteCaptureTime> GenerateAbsoluteCaptureTime() {
+ if (std::bernoulli_distribution(0.25)(generator_)) {
+ return absl::nullopt;
+ }
+
+ AbsoluteCaptureTime value;
+
+ value.absolute_capture_timestamp =
+ std::uniform_int_distribution<uint64_t>()(generator_);
+
+ if (std::bernoulli_distribution(0.5)(generator_)) {
+ value.estimated_capture_clock_offset = absl::nullopt;
+ } else {
+ value.estimated_capture_clock_offset =
+ std::uniform_int_distribution<int64_t>()(generator_);
+ }
+
+ return value;
+ }
+
int64_t GenerateReceiveTimeMs() {
return std::uniform_int_distribution<int64_t>()(generator_);
}
@@ -223,13 +244,15 @@
constexpr uint32_t kCsrcs1 = 21;
constexpr uint32_t kRtpTimestamp = 40;
constexpr absl::optional<uint8_t> kAudioLevel = 50;
+ constexpr absl::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime = {};
constexpr int64_t kReceiveTimeMs = 60;
SimulatedClock clock(1000000000000ULL);
SourceTracker tracker(&clock);
- tracker.OnFrameDelivered(RtpPacketInfos({RtpPacketInfo(
- kSsrc, {kCsrcs0, kCsrcs1}, kRtpTimestamp, kAudioLevel, kReceiveTimeMs)}));
+ tracker.OnFrameDelivered(RtpPacketInfos(
+ {RtpPacketInfo(kSsrc, {kCsrcs0, kCsrcs1}, kRtpTimestamp, kAudioLevel,
+ kAbsoluteCaptureTime, kReceiveTimeMs)}));
int64_t timestamp_ms = clock.TimeInMilliseconds();
@@ -251,23 +274,24 @@
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 int64_t kReceiveTimeMs0 = 60;
constexpr int64_t kReceiveTimeMs1 = 61;
SimulatedClock clock(1000000000000ULL);
SourceTracker tracker(&clock);
- tracker.OnFrameDelivered(
- RtpPacketInfos({RtpPacketInfo(kSsrc, {kCsrcs0, kCsrcs1}, kRtpTimestamp0,
- kAudioLevel0, kReceiveTimeMs0)}));
+ tracker.OnFrameDelivered(RtpPacketInfos(
+ {RtpPacketInfo(kSsrc, {kCsrcs0, kCsrcs1}, kRtpTimestamp0, kAudioLevel0,
+ kAbsoluteCaptureTime, kReceiveTimeMs0)}));
int64_t timestamp_ms_0 = clock.TimeInMilliseconds();
clock.AdvanceTimeMilliseconds(17);
- tracker.OnFrameDelivered(
- RtpPacketInfos({RtpPacketInfo(kSsrc, {kCsrcs0, kCsrcs2}, kRtpTimestamp1,
- kAudioLevel1, kReceiveTimeMs1)}));
+ tracker.OnFrameDelivered(RtpPacketInfos(
+ {RtpPacketInfo(kSsrc, {kCsrcs0, kCsrcs2}, kRtpTimestamp1, kAudioLevel1,
+ kAbsoluteCaptureTime, kReceiveTimeMs1)}));
int64_t timestamp_ms_1 = clock.TimeInMilliseconds();
@@ -292,21 +316,22 @@
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 int64_t kReceiveTimeMs0 = 60;
constexpr int64_t kReceiveTimeMs1 = 61;
SimulatedClock clock(1000000000000ULL);
SourceTracker tracker(&clock);
- tracker.OnFrameDelivered(
- RtpPacketInfos({RtpPacketInfo(kSsrc, {kCsrcs0, kCsrcs1}, kRtpTimestamp0,
- kAudioLevel0, kReceiveTimeMs0)}));
+ tracker.OnFrameDelivered(RtpPacketInfos(
+ {RtpPacketInfo(kSsrc, {kCsrcs0, kCsrcs1}, kRtpTimestamp0, kAudioLevel0,
+ kAbsoluteCaptureTime, kReceiveTimeMs0)}));
clock.AdvanceTimeMilliseconds(17);
- tracker.OnFrameDelivered(
- RtpPacketInfos({RtpPacketInfo(kSsrc, {kCsrcs0, kCsrcs2}, kRtpTimestamp1,
- kAudioLevel1, kReceiveTimeMs1)}));
+ tracker.OnFrameDelivered(RtpPacketInfos(
+ {RtpPacketInfo(kSsrc, {kCsrcs0, kCsrcs2}, kRtpTimestamp1, kAudioLevel1,
+ kAbsoluteCaptureTime, kReceiveTimeMs1)}));
int64_t timestamp_ms_1 = clock.TimeInMilliseconds();
