Fix bug in how send timestamps are converted to 24 bits.
BUG=webrtc:4173
R=sprang@webrtc.org
Review URL: https://codereview.webrtc.org/1412683004 .
Cr-Commit-Position: refs/heads/master@{#10356}
diff --git a/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc b/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc
index 9d86ba3..cb8d0db 100644
--- a/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc
+++ b/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc
@@ -36,7 +36,7 @@
VerboseLogging(true);
}
- Random random_;
+ test::Random random_;
private:
RTC_DISALLOW_COPY_AND_ASSIGN(BweSimulation);
diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
index 56a309c..522489e 100644
--- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
+++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
@@ -64,6 +64,15 @@
return keys;
}
+uint32_t ConvertMsTo24Bits(int64_t time_ms) {
+ uint32_t time_24_bits =
+ static_cast<uint32_t>(
+ ((static_cast<uint64_t>(time_ms) << kAbsSendTimeFraction) + 500) /
+ 1000) &
+ 0x00FFFFFF;
+ return time_24_bits;
+}
+
bool RemoteBitrateEstimatorAbsSendTime::IsWithinClusterBounds(
int send_delta_ms,
const Cluster& cluster_aggregate) {
@@ -219,12 +228,8 @@
void RemoteBitrateEstimatorAbsSendTime::IncomingPacketFeedbackVector(
const std::vector<PacketInfo>& packet_feedback_vector) {
for (const auto& packet_info : packet_feedback_vector) {
- // TODO(holmer): We should get rid of this conversion if possible as we may
- // lose precision.
- uint32_t send_time_32bits = (packet_info.send_time_ms) / kTimestampToMs;
- uint32_t send_time_24bits =
- send_time_32bits >> kAbsSendTimeInterArrivalUpshift;
- IncomingPacketInfo(packet_info.arrival_time_ms, send_time_24bits,
+ IncomingPacketInfo(packet_info.arrival_time_ms,
+ ConvertMsTo24Bits(packet_info.send_time_ms),
packet_info.payload_size, 0, packet_info.was_paced);
}
}
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 8dc8ca6..b21f115 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -26,8 +26,9 @@
namespace webrtc {
// Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
-const size_t kMaxPaddingLength = 224;
-const int kSendSideDelayWindowMs = 1000;
+static const size_t kMaxPaddingLength = 224;
+static const int kSendSideDelayWindowMs = 1000;
+static const uint32_t kAbsSendTimeFraction = 18;
namespace {
@@ -45,6 +46,16 @@
return "";
}
+// TODO(holmer): Merge this with the implementation in
+// remote_bitrate_estimator_abs_send_time.cc.
+uint32_t ConvertMsTo24Bits(int64_t time_ms) {
+ uint32_t time_24_bits =
+ static_cast<uint32_t>(
+ ((static_cast<uint64_t>(time_ms) << kAbsSendTimeFraction) + 500) /
+ 1000) &
+ 0x00FFFFFF;
+ return time_24_bits;
+}
} // namespace
class BitrateAggregator {
@@ -1596,7 +1607,7 @@
// Update absolute send time field (convert ms to 24-bit unsigned with 18 bit
// fractional part).
ByteWriter<uint32_t, 3>::WriteBigEndian(rtp_packet + offset + 1,
- ((now_ms << 18) / 1000) & 0x00ffffff);
+ ConvertMsTo24Bits(now_ms));
}
uint16_t RTPSender::UpdateTransportSequenceNumber(
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index e4ace67..9be888c 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -64,7 +64,7 @@
}
uint64_t ConvertMsToAbsSendTime(int64_t time_ms) {
- return 0x00fffffful & ((time_ms << 18) / 1000);
+ return (((time_ms << 18) + 500) / 1000) & 0x00ffffff;
}
class LoopbackTransportTest : public webrtc::Transport {
