Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine.
BUG=3111
TEST=try bots
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6074 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp
index 444c665..d7e7cf8 100644
--- a/webrtc/modules/modules.gyp
+++ b/webrtc/modules/modules.gyp
@@ -180,7 +180,6 @@
'remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc',
'remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h',
'remote_bitrate_estimator/remote_bitrate_estimators_test.cc',
- 'remote_bitrate_estimator/rtp_to_ntp_unittest.cc',
'remote_bitrate_estimator/test/bwe_test_baselinefile.cc',
'remote_bitrate_estimator/test/bwe_test_baselinefile.h',
'remote_bitrate_estimator/test/bwe_test_fileutils.cc',
diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi
index 810da46..7292cae 100644
--- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi
+++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi
@@ -21,10 +21,8 @@
'sources': [
'include/bwe_defines.h',
'include/remote_bitrate_estimator.h',
- 'include/rtp_to_ntp.h',
'rate_statistics.cc',
'rate_statistics.h',
- 'rtp_to_ntp.cc',
], # source
},
{
diff --git a/webrtc/modules/remote_bitrate_estimator/rtp_to_ntp_unittest.cc b/webrtc/modules/remote_bitrate_estimator/rtp_to_ntp_unittest.cc
deleted file mode 100644
index aff314a..0000000
--- a/webrtc/modules/remote_bitrate_estimator/rtp_to_ntp_unittest.cc
+++ /dev/null
@@ -1,163 +0,0 @@
-/*
- * Copyright (c) 2012 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 "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/remote_bitrate_estimator/include/rtp_to_ntp.h"
-
-namespace webrtc {
-
-TEST(WrapAroundTests, NoWrap) {
- EXPECT_EQ(0, synchronization::CheckForWrapArounds(0xFFFFFFFF, 0xFFFFFFFE));
- EXPECT_EQ(0, synchronization::CheckForWrapArounds(1, 0));
- EXPECT_EQ(0, synchronization::CheckForWrapArounds(0x00010000, 0x0000FFFF));
-}
-
-TEST(WrapAroundTests, ForwardWrap) {
- EXPECT_EQ(1, synchronization::CheckForWrapArounds(0, 0xFFFFFFFF));
- EXPECT_EQ(1, synchronization::CheckForWrapArounds(0, 0xFFFF0000));
- EXPECT_EQ(1, synchronization::CheckForWrapArounds(0x0000FFFF, 0xFFFFFFFF));
- EXPECT_EQ(1, synchronization::CheckForWrapArounds(0x0000FFFF, 0xFFFF0000));
-}
-
-TEST(WrapAroundTests, BackwardWrap) {
- EXPECT_EQ(-1, synchronization::CheckForWrapArounds(0xFFFFFFFF, 0));
- EXPECT_EQ(-1, synchronization::CheckForWrapArounds(0xFFFF0000, 0));
- EXPECT_EQ(-1, synchronization::CheckForWrapArounds(0xFFFFFFFF, 0x0000FFFF));
- EXPECT_EQ(-1, synchronization::CheckForWrapArounds(0xFFFF0000, 0x0000FFFF));
-}
-
-TEST(WrapAroundTests, OldRtcpWrapped) {
- synchronization::RtcpList rtcp;
- uint32_t ntp_sec = 0;
- uint32_t ntp_frac = 0;
- uint32_t timestamp = 0;
- const uint32_t kOneMsInNtpFrac = 4294967;
- const uint32_t kTimestampTicksPerMs = 90;
- rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac,
- timestamp));
- ntp_frac += kOneMsInNtpFrac;
- timestamp -= kTimestampTicksPerMs;
- rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac,
- timestamp));
- ntp_frac += kOneMsInNtpFrac;
- timestamp -= kTimestampTicksPerMs;
- int64_t timestamp_in_ms = -1;
- // This expected to fail since it's highly unlikely that the older RTCP
- // has a much smaller RTP timestamp than the newer.
- EXPECT_FALSE(synchronization::RtpToNtpMs(timestamp, rtcp, ×tamp_in_ms));
-}
-
-TEST(WrapAroundTests, NewRtcpWrapped) {
- synchronization::RtcpList rtcp;
- uint32_t ntp_sec = 0;
- uint32_t ntp_frac = 0;
- uint32_t timestamp = 0xFFFFFFFF;
- const uint32_t kOneMsInNtpFrac = 4294967;
- const uint32_t kTimestampTicksPerMs = 90;
- rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac,
- timestamp));
- ntp_frac += kOneMsInNtpFrac;
- timestamp += kTimestampTicksPerMs;
- rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac,
- timestamp));
- int64_t timestamp_in_ms = -1;
- EXPECT_TRUE(synchronization::RtpToNtpMs(rtcp.back().rtp_timestamp, rtcp,
- ×tamp_in_ms));
- // Since this RTP packet has the same timestamp as the RTCP packet constructed
- // at time 0 it should be mapped to 0 as well.
- EXPECT_EQ(0, timestamp_in_ms);
-}
-
-TEST(WrapAroundTests, RtpWrapped) {
- const uint32_t kOneMsInNtpFrac = 4294967;
- const uint32_t kTimestampTicksPerMs = 90;
- synchronization::RtcpList rtcp;
- uint32_t ntp_sec = 0;
- uint32_t ntp_frac = 0;
- uint32_t timestamp = 0xFFFFFFFF - 2 * kTimestampTicksPerMs;
- rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac,
- timestamp));
- ntp_frac += kOneMsInNtpFrac;
- timestamp += kTimestampTicksPerMs;
- rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac,
- timestamp));
- ntp_frac += kOneMsInNtpFrac;
- timestamp += kTimestampTicksPerMs;
- int64_t timestamp_in_ms = -1;
- EXPECT_TRUE(synchronization::RtpToNtpMs(timestamp, rtcp,
- ×tamp_in_ms));
- // Since this RTP packet has the same timestamp as the RTCP packet constructed
- // at time 0 it should be mapped to 0 as well.
- EXPECT_EQ(2, timestamp_in_ms);
-}
-
-TEST(WrapAroundTests, OldRtp_RtcpsWrapped) {
- const uint32_t kOneMsInNtpFrac = 4294967;
- const uint32_t kTimestampTicksPerMs = 90;
- synchronization::RtcpList rtcp;
- uint32_t ntp_sec = 0;
- uint32_t ntp_frac = 0;
- uint32_t timestamp = 0;
- rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac,
- timestamp));
- ntp_frac += kOneMsInNtpFrac;
- timestamp += kTimestampTicksPerMs;
- rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac,
- timestamp));
- ntp_frac += kOneMsInNtpFrac;
- timestamp -= 2*kTimestampTicksPerMs;
- int64_t timestamp_in_ms = -1;
- EXPECT_FALSE(synchronization::RtpToNtpMs(timestamp, rtcp,
- ×tamp_in_ms));
-}
-
-TEST(WrapAroundTests, OldRtp_NewRtcpWrapped) {
- const uint32_t kOneMsInNtpFrac = 4294967;
- const uint32_t kTimestampTicksPerMs = 90;
- synchronization::RtcpList rtcp;
- uint32_t ntp_sec = 0;
- uint32_t ntp_frac = 0;
- uint32_t timestamp = 0xFFFFFFFF;
- rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac,
- timestamp));
- ntp_frac += kOneMsInNtpFrac;
- timestamp += kTimestampTicksPerMs;
- rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac,
- timestamp));
- ntp_frac += kOneMsInNtpFrac;
- timestamp -= kTimestampTicksPerMs;
- int64_t timestamp_in_ms = -1;
- EXPECT_TRUE(synchronization::RtpToNtpMs(timestamp, rtcp,
- ×tamp_in_ms));
- // Constructed at the same time as the first RTCP and should therefore be
- // mapped to zero.
- EXPECT_EQ(0, timestamp_in_ms);
-}
-
-TEST(WrapAroundTests, OldRtp_OldRtcpWrapped) {
- const uint32_t kOneMsInNtpFrac = 4294967;
- const uint32_t kTimestampTicksPerMs = 90;
- synchronization::RtcpList rtcp;
- uint32_t ntp_sec = 0;
- uint32_t ntp_frac = 0;
- uint32_t timestamp = 0;
- rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac,
- timestamp));
- ntp_frac += kOneMsInNtpFrac;
- timestamp -= kTimestampTicksPerMs;
- rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac,
- timestamp));
- ntp_frac += kOneMsInNtpFrac;
- timestamp += 2*kTimestampTicksPerMs;
- int64_t timestamp_in_ms = -1;
- EXPECT_FALSE(synchronization::RtpToNtpMs(timestamp, rtcp,
- ×tamp_in_ms));
-}
-}; // namespace webrtc
diff --git a/webrtc/modules/video_coding/main/source/Android.mk b/webrtc/modules/video_coding/main/source/Android.mk
index 9ebdbed..a8cf2d0 100644
--- a/webrtc/modules/video_coding/main/source/Android.mk
+++ b/webrtc/modules/video_coding/main/source/Android.mk
@@ -37,7 +37,6 @@
receiver.cc \
rtt_filter.cc \
session_info.cc \
- timestamp_extrapolator.cc \
timestamp_map.cc \
timing.cc \
video_coding_impl.cc
@@ -56,7 +55,7 @@
$(LOCAL_PATH)/../../../../common_video/vplib/main/interface \
$(LOCAL_PATH)/../../../../common_video/interface \
$(LOCAL_PATH)/../../utility/include \
- $(LOCAL_PATH)/../../../../system_wrappers/interface
+ $(LOCAL_PATH)/../../../../system_wrappers/interface
LOCAL_SHARED_LIBRARIES := \
libcutils \
diff --git a/webrtc/modules/video_coding/main/source/timing.cc b/webrtc/modules/video_coding/main/source/timing.cc
index dd82187..af0e35c 100644
--- a/webrtc/modules/video_coding/main/source/timing.cc
+++ b/webrtc/modules/video_coding/main/source/timing.cc
@@ -10,12 +10,10 @@
#include "webrtc/modules/video_coding/main/source/timing.h"
-
#include "webrtc/modules/video_coding/main/source/internal_defines.h"
#include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
-#include "webrtc/modules/video_coding/main/source/timestamp_extrapolator.h"
#include "webrtc/system_wrappers/interface/clock.h"
-
+#include "webrtc/system_wrappers/interface/timestamp_extrapolator.h"
namespace webrtc {
@@ -35,8 +33,7 @@
prev_frame_timestamp_(0) {
if (master_timing == NULL) {
master_ = true;
- ts_extrapolator_ =
- new VCMTimestampExtrapolator(clock_->TimeInMilliseconds());
+ ts_extrapolator_ = new TimestampExtrapolator(clock_->TimeInMilliseconds());
} else {
ts_extrapolator_ = master_timing->ts_extrapolator_;
}
diff --git a/webrtc/modules/video_coding/main/source/timing.h b/webrtc/modules/video_coding/main/source/timing.h
index fcb0402..1dca5e6 100644
--- a/webrtc/modules/video_coding/main/source/timing.h
+++ b/webrtc/modules/video_coding/main/source/timing.h
@@ -18,7 +18,7 @@
namespace webrtc {
class Clock;
-class VCMTimestampExtrapolator;
+class TimestampExtrapolator;
class VCMTiming {
public:
@@ -101,7 +101,7 @@
CriticalSectionWrapper* crit_sect_;
Clock* clock_;
bool master_;
- VCMTimestampExtrapolator* ts_extrapolator_;
+ TimestampExtrapolator* ts_extrapolator_;
VCMCodecTimer codec_timer_;
uint32_t render_delay_ms_;
uint32_t min_playout_delay_ms_;
diff --git a/webrtc/modules/video_coding/main/source/video_coding.gypi b/webrtc/modules/video_coding/main/source/video_coding.gypi
index b4f6cb7..f19a585 100644
--- a/webrtc/modules/video_coding/main/source/video_coding.gypi
+++ b/webrtc/modules/video_coding/main/source/video_coding.gypi
@@ -48,7 +48,6 @@
'receiver.h',
'rtt_filter.h',
'session_info.h',
- 'timestamp_extrapolator.h',
'timestamp_map.h',
'timing.h',
'video_coding_impl.h',
@@ -72,7 +71,6 @@
'receiver.cc',
'rtt_filter.cc',
'session_info.cc',
- 'timestamp_extrapolator.cc',
'timestamp_map.cc',
'timing.cc',
'video_coding_impl.cc',
diff --git a/webrtc/modules/remote_bitrate_estimator/include/rtp_to_ntp.h b/webrtc/system_wrappers/interface/rtp_to_ntp.h
similarity index 84%
rename from webrtc/modules/remote_bitrate_estimator/include/rtp_to_ntp.h
rename to webrtc/system_wrappers/interface/rtp_to_ntp.h
index 08a4d46..dfc25cd 100644
--- a/webrtc/modules/remote_bitrate_estimator/include/rtp_to_ntp.h
+++ b/webrtc/system_wrappers/interface/rtp_to_ntp.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_RTP_TO_NTP_H_
-#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_RTP_TO_NTP_H_
+#ifndef SYSTEM_WRAPPERS_INTERFACE_RTP_TO_NTP_H_
+#define SYSTEM_WRAPPERS_INTERFACE_RTP_TO_NTP_H_
#include <list>
@@ -17,8 +17,6 @@
namespace webrtc {
-namespace synchronization {
-
struct RtcpMeasurement {
RtcpMeasurement();
RtcpMeasurement(uint32_t ntp_secs, uint32_t ntp_frac, uint32_t timestamp);
@@ -46,7 +44,7 @@
// Returns 1 there has been a forward wrap around, 0 if there has been no wrap
// around and -1 if there has been a backwards wrap around (i.e. reordering).
int CheckForWrapArounds(uint32_t rtp_timestamp, uint32_t rtcp_rtp_timestamp);
-} // namespace synchronization
+
} // namespace webrtc
-#endif // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_RTP_TO_NTP_H_
+#endif // SYSTEM_WRAPPERS_INTERFACE_RTP_TO_NTP_H_
diff --git a/webrtc/modules/video_coding/main/source/timestamp_extrapolator.h b/webrtc/system_wrappers/interface/timestamp_extrapolator.h
similarity index 82%
rename from webrtc/modules/video_coding/main/source/timestamp_extrapolator.h
rename to webrtc/system_wrappers/interface/timestamp_extrapolator.h
index 151e4de..d067198 100644
--- a/webrtc/modules/video_coding/main/source/timestamp_extrapolator.h
+++ b/webrtc/system_wrappers/interface/timestamp_extrapolator.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_EXTRAPOLATOR_H_
-#define WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_EXTRAPOLATOR_H_
+#ifndef SYSTEM_WRAPPERS_INTERFACE_TIMESTAMP_EXTRAPOLATOR_H_
+#define SYSTEM_WRAPPERS_INTERFACE_TIMESTAMP_EXTRAPOLATOR_H_
#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
#include "webrtc/typedefs.h"
@@ -17,11 +17,11 @@
namespace webrtc
{
-class VCMTimestampExtrapolator
+class TimestampExtrapolator
{
public:
- explicit VCMTimestampExtrapolator(int64_t start_ms);
- ~VCMTimestampExtrapolator();
+ explicit TimestampExtrapolator(int64_t start_ms);
+ ~TimestampExtrapolator();
void Update(int64_t tMs, uint32_t ts90khz);
int64_t ExtrapolateLocalTime(uint32_t timestamp90khz);
void Reset(int64_t start_ms);
@@ -53,4 +53,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_EXTRAPOLATOR_H_
+#endif // SYSTEM_WRAPPERS_INTERFACE_TIMESTAMP_EXTRAPOLATOR_H_
diff --git a/webrtc/system_wrappers/source/Android.mk b/webrtc/system_wrappers/source/Android.mk
index f48019d..4767e72 100644
--- a/webrtc/system_wrappers/source/Android.mk
+++ b/webrtc/system_wrappers/source/Android.mk
@@ -35,11 +35,13 @@
condition_variable_posix.cc \
critical_section_posix.cc \
event_posix.cc \
+ rtp_to_ntp.cc \
sleep.cc \
thread_posix.cc \
tick_util.cc \
+ timestamp_extrapolator.cc \
trace_posix.cc \
- rw_lock_posix.cc
+ rw_lock_posix.cc
LOCAL_CFLAGS := \
$(MY_WEBRTC_COMMON_DEFS)
diff --git a/webrtc/modules/remote_bitrate_estimator/rtp_to_ntp.cc b/webrtc/system_wrappers/source/rtp_to_ntp.cc
similarity index 93%
rename from webrtc/modules/remote_bitrate_estimator/rtp_to_ntp.cc
rename to webrtc/system_wrappers/source/rtp_to_ntp.cc
index 775cd0d..d6b7b14 100644
--- a/webrtc/modules/remote_bitrate_estimator/rtp_to_ntp.cc
+++ b/webrtc/system_wrappers/source/rtp_to_ntp.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/remote_bitrate_estimator/include/rtp_to_ntp.h"
+#include "webrtc/system_wrappers/interface/rtp_to_ntp.h"
#include "webrtc/system_wrappers/interface/clock.h"
@@ -16,8 +16,6 @@
namespace webrtc {
-namespace synchronization {
-
RtcpMeasurement::RtcpMeasurement()
: ntp_secs(0), ntp_frac(0), rtp_timestamp(0) {}
@@ -47,8 +45,7 @@
uint32_t old_timestamp,
int64_t* compensated_timestamp) {
assert(compensated_timestamp);
- int64_t wraps = synchronization::CheckForWrapArounds(new_timestamp,
- old_timestamp);
+ int64_t wraps = CheckForWrapArounds(new_timestamp, old_timestamp);
if (wraps < 0) {
// Reordering, don't use this packet.
return false;
@@ -96,7 +93,7 @@
// |rtp_timestamp_in_ms|. This function compensates for wrap arounds in RTP
// timestamps and returns false if it can't do the conversion due to reordering.
bool RtpToNtpMs(int64_t rtp_timestamp,
- const synchronization::RtcpList& rtcp,
+ const RtcpList& rtcp,
int64_t* rtp_timestamp_in_ms) {
assert(rtcp.size() == 2);
int64_t rtcp_ntp_ms_new = Clock::NtpToMs(rtcp.front().ntp_secs,
@@ -149,5 +146,5 @@
}
return 0;
}
-} // namespace synchronization
+
} // namespace webrtc
diff --git a/webrtc/system_wrappers/source/rtp_to_ntp_unittest.cc b/webrtc/system_wrappers/source/rtp_to_ntp_unittest.cc
new file mode 100644
index 0000000..a4d75ae
--- /dev/null
+++ b/webrtc/system_wrappers/source/rtp_to_ntp_unittest.cc
@@ -0,0 +1,146 @@
+/*
+ * Copyright (c) 2012 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 "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/system_wrappers/interface/rtp_to_ntp.h"
+
+namespace webrtc {
+
+TEST(WrapAroundTests, NoWrap) {
+ EXPECT_EQ(0, CheckForWrapArounds(0xFFFFFFFF, 0xFFFFFFFE));
+ EXPECT_EQ(0, CheckForWrapArounds(1, 0));
+ EXPECT_EQ(0, CheckForWrapArounds(0x00010000, 0x0000FFFF));
+}
+
+TEST(WrapAroundTests, ForwardWrap) {
+ EXPECT_EQ(1, CheckForWrapArounds(0, 0xFFFFFFFF));
+ EXPECT_EQ(1, CheckForWrapArounds(0, 0xFFFF0000));
+ EXPECT_EQ(1, CheckForWrapArounds(0x0000FFFF, 0xFFFFFFFF));
+ EXPECT_EQ(1, CheckForWrapArounds(0x0000FFFF, 0xFFFF0000));
+}
+
+TEST(WrapAroundTests, BackwardWrap) {
+ EXPECT_EQ(-1, CheckForWrapArounds(0xFFFFFFFF, 0));
+ EXPECT_EQ(-1, CheckForWrapArounds(0xFFFF0000, 0));
+ EXPECT_EQ(-1, CheckForWrapArounds(0xFFFFFFFF, 0x0000FFFF));
+ EXPECT_EQ(-1, CheckForWrapArounds(0xFFFF0000, 0x0000FFFF));
+}
+
+TEST(WrapAroundTests, OldRtcpWrapped) {
+ RtcpList rtcp;
+ uint32_t ntp_sec = 0;
+ uint32_t ntp_frac = 0;
+ uint32_t timestamp = 0;
+ const uint32_t kOneMsInNtpFrac = 4294967;
+ const uint32_t kTimestampTicksPerMs = 90;
+ rtcp.push_front(RtcpMeasurement(ntp_sec, ntp_frac, timestamp));
+ ntp_frac += kOneMsInNtpFrac;
+ timestamp -= kTimestampTicksPerMs;
+ rtcp.push_front(RtcpMeasurement(ntp_sec, ntp_frac, timestamp));
+ ntp_frac += kOneMsInNtpFrac;
+ timestamp -= kTimestampTicksPerMs;
+ int64_t timestamp_in_ms = -1;
+ // This expected to fail since it's highly unlikely that the older RTCP
+ // has a much smaller RTP timestamp than the newer.
+ EXPECT_FALSE(RtpToNtpMs(timestamp, rtcp, ×tamp_in_ms));
+}
+
+TEST(WrapAroundTests, NewRtcpWrapped) {
+ RtcpList rtcp;
+ uint32_t ntp_sec = 0;
+ uint32_t ntp_frac = 0;
+ uint32_t timestamp = 0xFFFFFFFF;
+ const uint32_t kOneMsInNtpFrac = 4294967;
+ const uint32_t kTimestampTicksPerMs = 90;
+ rtcp.push_front(RtcpMeasurement(ntp_sec, ntp_frac, timestamp));
+ ntp_frac += kOneMsInNtpFrac;
+ timestamp += kTimestampTicksPerMs;
+ rtcp.push_front(RtcpMeasurement(ntp_sec, ntp_frac, timestamp));
+ int64_t timestamp_in_ms = -1;
+ EXPECT_TRUE(RtpToNtpMs(rtcp.back().rtp_timestamp, rtcp, ×tamp_in_ms));
+ // Since this RTP packet has the same timestamp as the RTCP packet constructed
+ // at time 0 it should be mapped to 0 as well.
+ EXPECT_EQ(0, timestamp_in_ms);
+}
+
+TEST(WrapAroundTests, RtpWrapped) {
+ const uint32_t kOneMsInNtpFrac = 4294967;
+ const uint32_t kTimestampTicksPerMs = 90;
+ RtcpList rtcp;
+ uint32_t ntp_sec = 0;
+ uint32_t ntp_frac = 0;
+ uint32_t timestamp = 0xFFFFFFFF - 2 * kTimestampTicksPerMs;
+ rtcp.push_front(RtcpMeasurement(ntp_sec, ntp_frac, timestamp));
+ ntp_frac += kOneMsInNtpFrac;
+ timestamp += kTimestampTicksPerMs;
+ rtcp.push_front(RtcpMeasurement(ntp_sec, ntp_frac, timestamp));
+ ntp_frac += kOneMsInNtpFrac;
+ timestamp += kTimestampTicksPerMs;
+ int64_t timestamp_in_ms = -1;
+ EXPECT_TRUE(RtpToNtpMs(timestamp, rtcp, ×tamp_in_ms));
+ // Since this RTP packet has the same timestamp as the RTCP packet constructed
+ // at time 0 it should be mapped to 0 as well.
+ EXPECT_EQ(2, timestamp_in_ms);
+}
+
+TEST(WrapAroundTests, OldRtp_RtcpsWrapped) {
+ const uint32_t kOneMsInNtpFrac = 4294967;
+ const uint32_t kTimestampTicksPerMs = 90;
+ RtcpList rtcp;
+ uint32_t ntp_sec = 0;
+ uint32_t ntp_frac = 0;
+ uint32_t timestamp = 0;
+ rtcp.push_front(RtcpMeasurement(ntp_sec, ntp_frac, timestamp));
+ ntp_frac += kOneMsInNtpFrac;
+ timestamp += kTimestampTicksPerMs;
+ rtcp.push_front(RtcpMeasurement(ntp_sec, ntp_frac, timestamp));
+ ntp_frac += kOneMsInNtpFrac;
+ timestamp -= 2*kTimestampTicksPerMs;
+ int64_t timestamp_in_ms = -1;
+ EXPECT_FALSE(RtpToNtpMs(timestamp, rtcp, ×tamp_in_ms));
+}
+
+TEST(WrapAroundTests, OldRtp_NewRtcpWrapped) {
+ const uint32_t kOneMsInNtpFrac = 4294967;
+ const uint32_t kTimestampTicksPerMs = 90;
+ RtcpList rtcp;
+ uint32_t ntp_sec = 0;
+ uint32_t ntp_frac = 0;
+ uint32_t timestamp = 0xFFFFFFFF;
+ rtcp.push_front(RtcpMeasurement(ntp_sec, ntp_frac, timestamp));
+ ntp_frac += kOneMsInNtpFrac;
+ timestamp += kTimestampTicksPerMs;
+ rtcp.push_front(RtcpMeasurement(ntp_sec, ntp_frac, timestamp));
+ ntp_frac += kOneMsInNtpFrac;
+ timestamp -= kTimestampTicksPerMs;
+ int64_t timestamp_in_ms = -1;
+ EXPECT_TRUE(RtpToNtpMs(timestamp, rtcp, ×tamp_in_ms));
+ // Constructed at the same time as the first RTCP and should therefore be
+ // mapped to zero.
+ EXPECT_EQ(0, timestamp_in_ms);
+}
+
+TEST(WrapAroundTests, OldRtp_OldRtcpWrapped) {
+ const uint32_t kOneMsInNtpFrac = 4294967;
+ const uint32_t kTimestampTicksPerMs = 90;
+ RtcpList rtcp;
+ uint32_t ntp_sec = 0;
+ uint32_t ntp_frac = 0;
+ uint32_t timestamp = 0;
+ rtcp.push_front(RtcpMeasurement(ntp_sec, ntp_frac, timestamp));
+ ntp_frac += kOneMsInNtpFrac;
+ timestamp -= kTimestampTicksPerMs;
+ rtcp.push_front(RtcpMeasurement(ntp_sec, ntp_frac, timestamp));
+ ntp_frac += kOneMsInNtpFrac;
+ timestamp += 2*kTimestampTicksPerMs;
+ int64_t timestamp_in_ms = -1;
+ EXPECT_FALSE(RtpToNtpMs(timestamp, rtcp, ×tamp_in_ms));
+}
+}; // namespace webrtc
diff --git a/webrtc/system_wrappers/source/system_wrappers.gyp b/webrtc/system_wrappers/source/system_wrappers.gyp
index 2deefaa..36d4d9c 100644
--- a/webrtc/system_wrappers/source/system_wrappers.gyp
+++ b/webrtc/system_wrappers/source/system_wrappers.gyp
@@ -40,6 +40,7 @@
'../interface/logcat_trace_context.h',
'../interface/logging.h',
'../interface/ref_count.h',
+ '../interface/rtp_to_ntp.h',
'../interface/rw_lock_wrapper.h',
'../interface/scoped_ptr.h',
'../interface/scoped_refptr.h',
@@ -52,6 +53,7 @@
'../interface/thread_annotations.h',
'../interface/thread_wrapper.h',
'../interface/tick_util.h',
+ '../interface/timestamp_extrapolator.h',
'../interface/trace.h',
'../interface/trace_event.h',
'../interface/utf_util_win.h',
@@ -87,6 +89,7 @@
'file_impl.h',
'logcat_trace_context.cc',
'logging.cc',
+ 'rtp_to_ntp.cc',
'rw_lock.cc',
'rw_lock_generic.cc',
'rw_lock_generic.h',
@@ -103,6 +106,7 @@
'thread_posix.h',
'thread_win.cc',
'thread_win.h',
+ 'timestamp_extrapolator.cc',
'trace_impl.cc',
'trace_impl.h',
'trace_posix.cc',
diff --git a/webrtc/system_wrappers/source/system_wrappers_tests.gyp b/webrtc/system_wrappers/source/system_wrappers_tests.gyp
index 91ff028..59eef36 100644
--- a/webrtc/system_wrappers/source/system_wrappers_tests.gyp
+++ b/webrtc/system_wrappers/source/system_wrappers_tests.gyp
@@ -29,6 +29,7 @@
'data_log_helpers_unittest.cc',
'data_log_c_helpers_unittest.c',
'data_log_c_helpers_unittest.h',
+ 'rtp_to_ntp_unittest.cc',
'scoped_vector_unittest.cc',
'stringize_macros_unittest.cc',
'stl_util_unittest.cc',
diff --git a/webrtc/modules/video_coding/main/source/timestamp_extrapolator.cc b/webrtc/system_wrappers/source/timestamp_extrapolator.cc
similarity index 92%
rename from webrtc/modules/video_coding/main/source/timestamp_extrapolator.cc
rename to webrtc/system_wrappers/source/timestamp_extrapolator.cc
index bde2f0a..afd212b 100644
--- a/webrtc/modules/video_coding/main/source/timestamp_extrapolator.cc
+++ b/webrtc/system_wrappers/source/timestamp_extrapolator.cc
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/video_coding/main/source/timestamp_extrapolator.h"
+#include "webrtc/system_wrappers/interface/timestamp_extrapolator.h"
#include <algorithm>
namespace webrtc {
-VCMTimestampExtrapolator::VCMTimestampExtrapolator(int64_t start_ms)
+TimestampExtrapolator::TimestampExtrapolator(int64_t start_ms)
: _rwLock(RWLockWrapper::CreateRWLock()),
_startMs(0),
_firstTimestamp(0),
@@ -34,13 +34,12 @@
Reset(start_ms);
}
-VCMTimestampExtrapolator::~VCMTimestampExtrapolator()
+TimestampExtrapolator::~TimestampExtrapolator()
{
delete _rwLock;
}
-void
-VCMTimestampExtrapolator::Reset(int64_t start_ms)
+void TimestampExtrapolator::Reset(int64_t start_ms)
{
WriteLockScoped wl(*_rwLock);
_startMs = start_ms;
@@ -61,7 +60,7 @@
}
void
-VCMTimestampExtrapolator::Update(int64_t tMs, uint32_t ts90khz)
+TimestampExtrapolator::Update(int64_t tMs, uint32_t ts90khz)
{
_rwLock->AcquireLockExclusive();
@@ -143,7 +142,7 @@
}
int64_t
-VCMTimestampExtrapolator::ExtrapolateLocalTime(uint32_t timestamp90khz)
+TimestampExtrapolator::ExtrapolateLocalTime(uint32_t timestamp90khz)
{
ReadLockScoped rl(*_rwLock);
int64_t localTimeMs = 0;
@@ -181,7 +180,7 @@
// Investigates if the timestamp clock has overflowed since the last timestamp and
// keeps track of the number of wrap arounds since reset.
void
-VCMTimestampExtrapolator::CheckForWrapArounds(uint32_t ts90khz)
+TimestampExtrapolator::CheckForWrapArounds(uint32_t ts90khz)
{
if (_prevWrapTimestamp == -1)
{
@@ -210,7 +209,7 @@
}
bool
-VCMTimestampExtrapolator::DelayChangeDetection(double error)
+TimestampExtrapolator::DelayChangeDetection(double error)
{
// CUSUM detection of sudden delay changes
error = (error > 0) ? std::min(error, _accMaxError) :
diff --git a/webrtc/video/call_perf_tests.cc b/webrtc/video/call_perf_tests.cc
index 85f21da..9830dd1 100644
--- a/webrtc/video/call_perf_tests.cc
+++ b/webrtc/video/call_perf_tests.cc
@@ -17,10 +17,10 @@
#include "webrtc/call.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/remote_bitrate_estimator/include/rtp_to_ntp.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/rtp_to_ntp.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/thread_annotations.h"
#include "webrtc/test/direct_transport.h"
@@ -105,7 +105,7 @@
packet_type = parser.Iterate()) {
if (packet_type == RTCPUtility::kRtcpSrCode) {
const RTCPUtility::RTCPPacket& packet = parser.Packet();
- synchronization::RtcpMeasurement ntp_rtp_pair(
+ RtcpMeasurement ntp_rtp_pair(
packet.SR.NTPMostSignificant,
packet.SR.NTPLeastSignificant,
packet.SR.RTPTimestamp);
@@ -122,16 +122,16 @@
// TODO(stefan): We can't EXPECT_TRUE on this call due to a bug in the
// RTCP sender where it sends RTCP SR before any RTP packets, which leads
// to a bogus NTP/RTP mapping.
- synchronization::RtpToNtpMs(timestamp, ntp_rtp_pairs_, ×tamp_in_ms);
+ RtpToNtpMs(timestamp, ntp_rtp_pairs_, ×tamp_in_ms);
return timestamp_in_ms;
}
return -1;
}
private:
- void StoreNtpRtpPair(synchronization::RtcpMeasurement ntp_rtp_pair) {
+ void StoreNtpRtpPair(RtcpMeasurement ntp_rtp_pair) {
CriticalSectionScoped lock(crit_.get());
- for (synchronization::RtcpList::iterator it = ntp_rtp_pairs_.begin();
+ for (RtcpList::iterator it = ntp_rtp_pairs_.begin();
it != ntp_rtp_pairs_.end();
++it) {
if (ntp_rtp_pair.ntp_secs == it->ntp_secs &&
@@ -149,7 +149,7 @@
}
const scoped_ptr<CriticalSectionWrapper> crit_;
- synchronization::RtcpList ntp_rtp_pairs_ GUARDED_BY(crit_);
+ RtcpList ntp_rtp_pairs_ GUARDED_BY(crit_);
};
class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer {
diff --git a/webrtc/video_engine/stream_synchronization.cc b/webrtc/video_engine/stream_synchronization.cc
index a877c7f..8f72fa9 100644
--- a/webrtc/video_engine/stream_synchronization.cc
+++ b/webrtc/video_engine/stream_synchronization.cc
@@ -64,15 +64,15 @@
return false;
}
int64_t audio_last_capture_time_ms;
- if (!synchronization::RtpToNtpMs(audio_measurement.latest_timestamp,
- audio_measurement.rtcp,
- &audio_last_capture_time_ms)) {
+ if (!RtpToNtpMs(audio_measurement.latest_timestamp,
+ audio_measurement.rtcp,
+ &audio_last_capture_time_ms)) {
return false;
}
int64_t video_last_capture_time_ms;
- if (!synchronization::RtpToNtpMs(video_measurement.latest_timestamp,
- video_measurement.rtcp,
- &video_last_capture_time_ms)) {
+ if (!RtpToNtpMs(video_measurement.latest_timestamp,
+ video_measurement.rtcp,
+ &video_last_capture_time_ms)) {
return false;
}
if (video_last_capture_time_ms < 0) {
diff --git a/webrtc/video_engine/stream_synchronization.h b/webrtc/video_engine/stream_synchronization.h
index 078b938..5fa9536d17 100644
--- a/webrtc/video_engine/stream_synchronization.h
+++ b/webrtc/video_engine/stream_synchronization.h
@@ -13,7 +13,7 @@
#include <list>
-#include "webrtc/modules/remote_bitrate_estimator/include/rtp_to_ntp.h"
+#include "webrtc/system_wrappers/interface/rtp_to_ntp.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -24,7 +24,7 @@
public:
struct Measurements {
Measurements() : rtcp(), latest_receive_time_ms(0), latest_timestamp(0) {}
- synchronization::RtcpList rtcp;
+ RtcpList rtcp;
int64_t latest_receive_time_ms;
uint32_t latest_timestamp;
};
diff --git a/webrtc/video_engine/stream_synchronization_unittest.cc b/webrtc/video_engine/stream_synchronization_unittest.cc
index 8876485..7136f1e 100644
--- a/webrtc/video_engine/stream_synchronization_unittest.cc
+++ b/webrtc/video_engine/stream_synchronization_unittest.cc
@@ -33,9 +33,8 @@
: kNtpJan1970(2208988800UL),
time_now_ms_(offset) {}
- synchronization::RtcpMeasurement GenerateRtcp(int frequency,
- uint32_t offset) const {
- synchronization::RtcpMeasurement rtcp;
+ RtcpMeasurement GenerateRtcp(int frequency, uint32_t offset) const {
+ RtcpMeasurement rtcp;
NowNtp(&rtcp.ntp_secs, &rtcp.ntp_frac);
rtcp.rtp_timestamp = NowRtp(frequency, offset);
return rtcp;
diff --git a/webrtc/video_engine/vie_receiver.cc b/webrtc/video_engine/vie_receiver.cc
index 2775ac9..ce6bdf2 100644
--- a/webrtc/video_engine/vie_receiver.cc
+++ b/webrtc/video_engine/vie_receiver.cc
@@ -21,10 +21,10 @@
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/utility/interface/rtp_dump.h"
#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/source/timestamp_extrapolator.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/logging.h"
#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/interface/timestamp_extrapolator.h"
#include "webrtc/system_wrappers/interface/trace.h"
namespace webrtc {
@@ -49,7 +49,7 @@
remote_bitrate_estimator_(remote_bitrate_estimator),
clock_(Clock::GetRealTimeClock()),
ts_extrapolator_(
- new VCMTimestampExtrapolator(clock_->TimeInMilliseconds())),
+ new TimestampExtrapolator(clock_->TimeInMilliseconds())),
rtp_dump_(NULL),
receiving_(false),
restored_packet_in_use_(false),
@@ -194,9 +194,9 @@
}
int64_t sender_capture_ntp_ms = 0;
- if (!synchronization::RtpToNtpMs(rtp_header->header.timestamp,
- rtcp_list_,
- &sender_capture_ntp_ms)) {
+ if (!RtpToNtpMs(rtp_header->header.timestamp,
+ rtcp_list_,
+ &sender_capture_ntp_ms)) {
return;
}
uint32_t timestamp = sender_capture_ntp_ms * 90;
@@ -390,8 +390,11 @@
}
bool new_rtcp_sr = false;
- if (!synchronization::UpdateRtcpList(
- ntp_secs, ntp_frac, rtp_timestamp, &rtcp_list_, &new_rtcp_sr)) {
+ if (!UpdateRtcpList(ntp_secs,
+ ntp_frac,
+ rtp_timestamp,
+ &rtcp_list_,
+ &new_rtcp_sr)) {
return false;
}
diff --git a/webrtc/video_engine/vie_receiver.h b/webrtc/video_engine/vie_receiver.h
index c85975c..bd753ae 100644
--- a/webrtc/video_engine/vie_receiver.h
+++ b/webrtc/video_engine/vie_receiver.h
@@ -14,17 +14,14 @@
#include <list>
#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/remote_bitrate_estimator/include/rtp_to_ntp.h"
#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/system_wrappers/interface/rtp_to_ntp.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/typedefs.h"
#include "webrtc/video_engine/include/vie_network.h"
#include "webrtc/video_engine/vie_defines.h"
-// TODO(wu): Move rtp_to_ntp.h and timestamp_extrapolator.h to somewhere that
-// can be shared between audio and video.
-
namespace webrtc {
class CriticalSectionWrapper;
@@ -36,7 +33,7 @@
class RTPPayloadRegistry;
class RtpReceiver;
class RtpRtcp;
-class VCMTimestampExtrapolator;
+class TimestampExtrapolator;
class VideoCodingModule;
struct ReceiveBandwidthEstimatorStats;
@@ -124,8 +121,8 @@
RemoteBitrateEstimator* remote_bitrate_estimator_;
Clock* clock_;
- scoped_ptr<VCMTimestampExtrapolator> ts_extrapolator_;
- synchronization::RtcpList rtcp_list_;
+ scoped_ptr<TimestampExtrapolator> ts_extrapolator_;
+ RtcpList rtcp_list_;
RtpDump* rtp_dump_;
bool receiving_;
diff --git a/webrtc/video_engine/vie_sync_module.cc b/webrtc/video_engine/vie_sync_module.cc
index b7c74a7..31a7a6d 100644
--- a/webrtc/video_engine/vie_sync_module.cc
+++ b/webrtc/video_engine/vie_sync_module.cc
@@ -43,7 +43,7 @@
}
bool new_rtcp_sr = false;
- if (!synchronization::UpdateRtcpList(
+ if (!UpdateRtcpList(
ntp_secs, ntp_frac, rtp_timestamp, &stream->rtcp, &new_rtcp_sr)) {
return -1;
}
