Change rtp header extension AbsoluteSendTime::Write to take time in 24bit format
making it symmetric to AbsoluteSendTime::Parse function.
Bug: None
Change-Id: I9c71d840768064022ebebbbeb2962aeeecc68392
Reviewed-on: https://chromium-review.googlesource.com/531044
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18555}diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc b/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
index 07b394a..0a13be3 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
+++ b/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
@@ -126,7 +126,7 @@
rtp_packet.SetExtension<TransmissionOffset>(prng->Rand(0x00ffffff));
rtp_packet.SetExtension<AudioLevel>(prng->Rand<bool>(), prng->Rand(127));
- rtp_packet.SetExtension<AbsoluteSendTime>(prng->Rand<int32_t>());
+ rtp_packet.SetExtension<AbsoluteSendTime>(prng->Rand(0x00ffffff));
rtp_packet.SetExtension<VideoOrientation>(prng->Rand(2));
rtp_packet.SetExtension<TransportSequenceNumber>(prng->Rand<uint16_t>());
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc b/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc
index 3bb9bc5..2f3feb3 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc
@@ -43,8 +43,9 @@
return true;
}
-bool AbsoluteSendTime::Write(uint8_t* data, int64_t time_ms) {
- ByteWriter<uint32_t, 3>::WriteBigEndian(data, MsTo24Bits(time_ms));
+bool AbsoluteSendTime::Write(uint8_t* data, uint32_t time_24bits) {
+ RTC_DCHECK_LE(time_24bits, 0x00FFFFFF);
+ ByteWriter<uint32_t, 3>::WriteBigEndian(data, time_24bits);
return true;
}
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h b/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
index a1f69dd..d3e72eb 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
@@ -28,8 +28,8 @@
"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
static bool Parse(rtc::ArrayView<const uint8_t> data, uint32_t* time_24bits);
- static size_t ValueSize(int64_t time_ms) { return kValueSizeBytes; }
- static bool Write(uint8_t* data, int64_t time_ms);
+ static size_t ValueSize(uint32_t time_24bits) { return kValueSizeBytes; }
+ static bool Write(uint8_t* data, uint32_t time_24bits);
static constexpr uint32_t MsTo24Bits(int64_t time_ms) {
return static_cast<uint32_t>(((time_ms << 18) + 500) / 1000) & 0x00FFFFFF;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 1bdb804..00edd18 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -559,7 +559,8 @@
padding_packet.SetExtension<TransmissionOffset>(
(now_ms - capture_time_ms) * kTimestampTicksPerMs);
}
- padding_packet.SetExtension<AbsoluteSendTime>(now_ms);
+ padding_packet.SetExtension<AbsoluteSendTime>(
+ AbsoluteSendTime::MsTo24Bits(now_ms));
PacketOptions options;
bool has_transport_seq_num =
UpdateTransportSequenceNumber(&padding_packet, &options.packet_id);
@@ -739,7 +740,8 @@
int64_t diff_ms = now_ms - capture_time_ms;
packet_to_send->SetExtension<TransmissionOffset>(kTimestampTicksPerMs *
diff_ms);
- packet_to_send->SetExtension<AbsoluteSendTime>(now_ms);
+ packet_to_send->SetExtension<AbsoluteSendTime>(
+ AbsoluteSendTime::MsTo24Bits(now_ms));
PacketOptions options;
if (UpdateTransportSequenceNumber(packet_to_send, &options.packet_id)) {
@@ -829,7 +831,7 @@
packet->SetExtension<TransmissionOffset>(
kTimestampTicksPerMs * (now_ms - packet->capture_time_ms()));
}
- packet->SetExtension<AbsoluteSendTime>(now_ms);
+ packet->SetExtension<AbsoluteSendTime>(AbsoluteSendTime::MsTo24Bits(now_ms));
if (video_) {
BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoTotBitrate_kbps", now_ms,