[rtp_rtcp] time helper functions
RTP timestams helper functions moved from rtp_utility
added functions to deal with CompactNtp timestamps
R=åsapersson
BUG=webrtc:5260
Review URL: https://codereview.webrtc.org/1535113002
Cr-Commit-Position: refs/heads/master@{#11106}
diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp
index edaa2f8..06a8822 100644
--- a/webrtc/modules/modules.gyp
+++ b/webrtc/modules/modules.gyp
@@ -328,6 +328,7 @@
'rtp_rtcp/source/rtp_rtcp_impl_unittest.cc',
'rtp_rtcp/source/rtp_header_extension_unittest.cc',
'rtp_rtcp/source/rtp_sender_unittest.cc',
+ 'rtp_rtcp/source/time_util_unittest.cc',
'rtp_rtcp/source/vp8_partition_aggregator_unittest.cc',
'rtp_rtcp/test/testAPI/test_api.cc',
'rtp_rtcp/test/testAPI/test_api.h',
diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
index 8ac7e0a..f089d36 100644
--- a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
@@ -14,7 +14,7 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/rtp_rtcp/source/bitrate.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/modules/rtp_rtcp/source/time_util.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
namespace webrtc {
@@ -37,8 +37,6 @@
cumulative_loss_(0),
jitter_q4_transmission_time_offset_(0),
last_receive_time_ms_(0),
- last_receive_time_secs_(0),
- last_receive_time_frac_(0),
last_received_timestamp_(0),
last_received_transmission_time_offset_(0),
received_seq_first_(0),
@@ -79,9 +77,7 @@
// are received, 4 will be ignored.
if (in_order) {
// Current time in samples.
- uint32_t receive_time_secs;
- uint32_t receive_time_frac;
- clock_->CurrentNtp(receive_time_secs, receive_time_frac);
+ NtpTime receive_time(*clock_);
// Wrong if we use RetransmitOfOldPacket.
if (receive_counters_.transmitted.packets > 1 &&
@@ -97,11 +93,10 @@
if (header.timestamp != last_received_timestamp_ &&
(receive_counters_.transmitted.packets -
receive_counters_.retransmitted.packets) > 1) {
- UpdateJitter(header, receive_time_secs, receive_time_frac);
+ UpdateJitter(header, receive_time);
}
last_received_timestamp_ = header.timestamp;
- last_receive_time_secs_ = receive_time_secs;
- last_receive_time_frac_ = receive_time_frac;
+ last_receive_time_ntp_ = receive_time;
last_receive_time_ms_ = clock_->TimeInMilliseconds();
}
@@ -113,14 +108,11 @@
}
void StreamStatisticianImpl::UpdateJitter(const RTPHeader& header,
- uint32_t receive_time_secs,
- uint32_t receive_time_frac) {
- uint32_t receive_time_rtp = RtpUtility::ConvertNTPTimeToRTP(
- receive_time_secs, receive_time_frac, header.payload_type_frequency);
+ NtpTime receive_time) {
+ uint32_t receive_time_rtp =
+ NtpToRtp(receive_time, header.payload_type_frequency);
uint32_t last_receive_time_rtp =
- RtpUtility::ConvertNTPTimeToRTP(last_receive_time_secs_,
- last_receive_time_frac_,
- header.payload_type_frequency);
+ NtpToRtp(last_receive_time_ntp_, header.payload_type_frequency);
int32_t time_diff_samples = (receive_time_rtp - last_receive_time_rtp) -
(header.timestamp - last_received_timestamp_);
@@ -319,8 +311,8 @@
void StreamStatisticianImpl::LastReceiveTimeNtp(uint32_t* secs,
uint32_t* frac) const {
CriticalSectionScoped cs(stream_lock_.get());
- *secs = last_receive_time_secs_;
- *frac = last_receive_time_frac_;
+ *secs = last_receive_time_ntp_.seconds();
+ *frac = last_receive_time_ntp_.fractions();
}
bool StreamStatisticianImpl::IsRetransmitOfOldPacket(
diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
index b3ef287..025dcd4 100644
--- a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
@@ -19,6 +19,7 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/rtp_rtcp/source/bitrate.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/ntp_time.h"
namespace webrtc {
@@ -52,9 +53,7 @@
private:
bool InOrderPacketInternal(uint16_t sequence_number) const;
RtcpStatistics CalculateRtcpStatistics();
- void UpdateJitter(const RTPHeader& header,
- uint32_t receive_time_secs,
- uint32_t receive_time_frac);
+ void UpdateJitter(const RTPHeader& header, NtpTime receive_time);
void UpdateCounters(const RTPHeader& rtp_header,
size_t packet_length,
bool retransmitted);
@@ -73,8 +72,7 @@
uint32_t jitter_q4_transmission_time_offset_;
int64_t last_receive_time_ms_;
- uint32_t last_receive_time_secs_;
- uint32_t last_receive_time_frac_;
+ NtpTime last_receive_time_ntp_;
uint32_t last_received_timestamp_;
int32_t last_received_transmission_time_offset_;
uint16_t received_seq_first_;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
index fb9b206..93b115a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
@@ -22,7 +22,6 @@
namespace webrtc {
-using RtpUtility::GetCurrentRTP;
using RtpUtility::Payload;
using RtpUtility::StringCompare;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 17fbbac..940d12b 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -21,6 +21,7 @@
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
+#include "webrtc/modules/rtp_rtcp/source/time_util.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/tick_util.h"
@@ -1643,7 +1644,7 @@
void RTPSender::SetSendingStatus(bool enabled) {
if (enabled) {
uint32_t frequency_hz = SendPayloadFrequency();
- uint32_t RTPtime = RtpUtility::GetCurrentRTP(clock_, frequency_hz);
+ uint32_t RTPtime = CurrentRtp(*clock_, frequency_hz);
// Will be ignored if it's already configured via API.
SetStartTimestamp(RTPtime, false);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_utility.cc b/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
index fcd43ad..bd7df42 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
@@ -77,38 +77,6 @@
};
/*
- * Time routines.
- */
-
-uint32_t GetCurrentRTP(Clock* clock, uint32_t freq) {
- const bool use_global_clock = (clock == NULL);
- Clock* local_clock = clock;
- if (use_global_clock) {
- local_clock = Clock::GetRealTimeClock();
- }
- uint32_t secs = 0, frac = 0;
- local_clock->CurrentNtp(secs, frac);
- if (use_global_clock) {
- delete local_clock;
- }
- return ConvertNTPTimeToRTP(secs, frac, freq);
-}
-
-uint32_t ConvertNTPTimeToRTP(uint32_t NTPsec, uint32_t NTPfrac, uint32_t freq) {
- float ftemp = (float)NTPfrac / (float)NTP_FRAC;
- uint32_t tmp = (uint32_t)(ftemp * freq);
- return NTPsec * freq + tmp;
-}
-
-uint32_t ConvertNTPTimeToMS(uint32_t NTPsec, uint32_t NTPfrac) {
- int freq = 1000;
- float ftemp = (float)NTPfrac / (float)NTP_FRAC;
- uint32_t tmp = (uint32_t)(ftemp * freq);
- uint32_t MStime = NTPsec * freq + tmp;
- return MStime;
-}
-
-/*
* Misc utility routines
*/
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_utility.h b/webrtc/modules/rtp_rtcp/source/rtp_utility.h
index 7582f3a..bdcb11c 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_utility.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_utility.h
@@ -46,15 +46,6 @@
typedef std::map<int8_t, Payload*> PayloadTypeMap;
- // Return the current RTP timestamp from the NTP timestamp
- // returned by the specified clock.
- uint32_t GetCurrentRTP(Clock* clock, uint32_t freq);
-
- // Return the current RTP absolute timestamp.
- uint32_t ConvertNTPTimeToRTP(uint32_t NTPsec,
- uint32_t NTPfrac,
- uint32_t freq);
-
uint32_t pow2(uint8_t exp);
// Returns true if |newTimestamp| is older than |existingTimestamp|.
diff --git a/webrtc/modules/rtp_rtcp/source/time_util.h b/webrtc/modules/rtp_rtcp/source/time_util.h
new file mode 100644
index 0000000..5b544dd
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/time_util.h
@@ -0,0 +1,48 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_TIME_UTIL_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_TIME_UTIL_H_
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/system_wrappers/include/ntp_time.h"
+
+namespace webrtc {
+
+// Converts NTP timestamp to RTP timestamp.
+inline uint32_t NtpToRtp(NtpTime ntp, uint32_t freq) {
+ uint32_t tmp = (static_cast<uint64_t>(ntp.fractions()) * freq) >> 32;
+ return ntp.seconds() * freq + tmp;
+}
+// Return the current RTP timestamp from the NTP timestamp
+// returned by the specified clock.
+inline uint32_t CurrentRtp(const Clock& clock, uint32_t freq) {
+ return NtpToRtp(NtpTime(clock), freq);
+}
+
+// Helper function for compact ntp representation:
+// RFC 3550, Section 4. Time Format.
+// Wallclock time is represented using the timestamp format of
+// the Network Time Protocol (NTP).
+// ...
+// In some fields where a more compact representation is
+// appropriate, only the middle 32 bits are used; that is, the low 16
+// bits of the integer part and the high 16 bits of the fractional part.
+inline uint32_t CompactNtp(NtpTime ntp) {
+ return (ntp.seconds() << 16) | (ntp.fractions() >> 16);
+}
+// Converts interval between compact ntp timestamps to milliseconds.
+// This interval can be upto ~18.2 hours (2^16 seconds).
+inline uint32_t CompactNtpIntervalToMs(uint32_t compact_ntp_interval) {
+ return static_cast<uint64_t>(compact_ntp_interval) * 1000 / (1 << 16);
+}
+
+} // namespace webrtc
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_TIME_UTIL_H_
diff --git a/webrtc/modules/rtp_rtcp/source/time_util_unittest.cc b/webrtc/modules/rtp_rtcp/source/time_util_unittest.cc
new file mode 100644
index 0000000..7efb83c
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/time_util_unittest.cc
@@ -0,0 +1,62 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+#include "webrtc/modules/rtp_rtcp/source/time_util.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+namespace webrtc {
+
+TEST(TimeUtilTest, CompactNtp) {
+ const uint32_t kNtpSec = 0x12345678;
+ const uint32_t kNtpFrac = 0x23456789;
+ const NtpTime kNtp(kNtpSec, kNtpFrac);
+ const uint32_t kNtpMid = 0x56782345;
+ EXPECT_EQ(kNtpMid, CompactNtp(kNtp));
+}
+
+TEST(TimeUtilTest, CompactNtpToMs) {
+ const NtpTime ntp1(0x12345, 0x23456);
+ const NtpTime ntp2(0x12654, 0x64335);
+ uint32_t ms_diff = ntp2.ToMs() - ntp1.ToMs();
+ uint32_t ntp_diff = CompactNtp(ntp2) - CompactNtp(ntp1);
+
+ uint32_t ntp_to_ms_diff = CompactNtpIntervalToMs(ntp_diff);
+
+ EXPECT_NEAR(ms_diff, ntp_to_ms_diff, 1);
+}
+
+TEST(TimeUtilTest, CompactNtpToMsWithWrap) {
+ const NtpTime ntp1(0x1ffff, 0x23456);
+ const NtpTime ntp2(0x20000, 0x64335);
+ uint32_t ms_diff = ntp2.ToMs() - ntp1.ToMs();
+
+ // While ntp2 > ntp1, there compact ntp presentation happen to be opposite.
+ // That shouldn't be a problem as long as unsigned arithmetic is used.
+ ASSERT_GT(ntp2.ToMs(), ntp1.ToMs());
+ ASSERT_LT(CompactNtp(ntp2), CompactNtp(ntp1));
+
+ uint32_t ntp_diff = CompactNtp(ntp2) - CompactNtp(ntp1);
+ uint32_t ntp_to_ms_diff = CompactNtpIntervalToMs(ntp_diff);
+
+ EXPECT_NEAR(ms_diff, ntp_to_ms_diff, 1);
+}
+
+TEST(TimeUtilTest, CompactNtpToMsLarge) {
+ const NtpTime ntp1(0x10000, 0x23456);
+ const NtpTime ntp2(0x1ffff, 0x64335);
+ uint32_t ms_diff = ntp2.ToMs() - ntp1.ToMs();
+ // Ntp difference close to maximum of ~18 hours should convert correctly too.
+ ASSERT_GT(ms_diff, 18u * 3600 * 1000);
+ uint32_t ntp_diff = CompactNtp(ntp2) - CompactNtp(ntp1);
+ uint32_t ntp_to_ms_diff = CompactNtpIntervalToMs(ntp_diff);
+
+ EXPECT_NEAR(ms_diff, ntp_to_ms_diff, 1);
+}
+} // namespace webrtc
