Pass capture time (wallclock) to the RTP sender to compute transmission offset
- Change how the transmission offset is calculated, to
incorporate the time since the frame was captured.
- Break out RtpRtcpClock and move it to system_wrappers.
- Use RtpRtcpClock to set the capture time in ms in the capture module.
We must use the same clock as in the RTP module to be able to measure
the time from capture until transmission.
- Enables the RTP header extension for packet transmission time offsets.
BUG=
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/666006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2489 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/src/common_video/interface/video_image.h b/src/common_video/interface/video_image.h
index 82e7b15..a899bb3 100644
--- a/src/common_video/interface/video_image.h
+++ b/src/common_video/interface/video_image.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -29,17 +29,29 @@
class RawImage
{
public:
- RawImage() : _width(0), _height(0), _timeStamp(0), _buffer(NULL),
- _length(0), _size(0) {}
+ RawImage()
+ : _width(0),
+ _height(0),
+ _timeStamp(0),
+ capture_time_ms_(0),
+ _buffer(NULL),
+ _length(0),
+ _size(0) {}
RawImage(WebRtc_UWord8* buffer, WebRtc_UWord32 length,
- WebRtc_UWord32 size) :
- _width(0), _height(0), _timeStamp(0),
- _buffer(buffer), _length(length), _size(size) {}
+ WebRtc_UWord32 size)
+ : _width(0),
+ _height(0),
+ _timeStamp(0),
+ capture_time_ms_(0),
+ _buffer(buffer),
+ _length(length),
+ _size(size) {}
WebRtc_UWord32 _width;
WebRtc_UWord32 _height;
WebRtc_UWord32 _timeStamp;
+ int64_t capture_time_ms_;
WebRtc_UWord8* _buffer;
WebRtc_UWord32 _length;
WebRtc_UWord32 _size;
@@ -48,21 +60,34 @@
class EncodedImage
{
public:
- EncodedImage() :
- _encodedWidth(0), _encodedHeight(0), _timeStamp(0),
- _frameType(kDeltaFrame), _buffer(NULL), _length(0),
- _size(0), _completeFrame(false) {}
+ EncodedImage()
+ : _encodedWidth(0),
+ _encodedHeight(0),
+ _timeStamp(0),
+ capture_time_ms_(0),
+ _frameType(kDeltaFrame),
+ _buffer(NULL),
+ _length(0),
+ _size(0),
+ _completeFrame(false) {}
EncodedImage(WebRtc_UWord8* buffer,
WebRtc_UWord32 length,
- WebRtc_UWord32 size) :
- _encodedWidth(0), _encodedHeight(0), _timeStamp(0),
- _frameType(kDeltaFrame), _buffer(buffer), _length(length),
- _size(size), _completeFrame(false) {}
+ WebRtc_UWord32 size)
+ : _encodedWidth(0),
+ _encodedHeight(0),
+ _timeStamp(0),
+ capture_time_ms_(0),
+ _frameType(kDeltaFrame),
+ _buffer(buffer),
+ _length(length),
+ _size(size),
+ _completeFrame(false) {}
WebRtc_UWord32 _encodedWidth;
WebRtc_UWord32 _encodedHeight;
WebRtc_UWord32 _timeStamp;
+ int64_t capture_time_ms_;
VideoFrameType _frameType;
WebRtc_UWord8* _buffer;
WebRtc_UWord32 _length;
diff --git a/src/modules/rtp_rtcp/interface/rtp_rtcp.h b/src/modules/rtp_rtcp/interface/rtp_rtcp.h
index 7db9810..ab3a4bb 100644
--- a/src/modules/rtp_rtcp/interface/rtp_rtcp.h
+++ b/src/modules/rtp_rtcp/interface/rtp_rtcp.h
@@ -492,6 +492,7 @@
const FrameType frameType,
const WebRtc_Word8 payloadType,
const WebRtc_UWord32 timeStamp,
+ int64_t capture_time_ms,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord32 payloadSize,
const RTPFragmentationHeader* fragmentation = NULL,
diff --git a/src/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/src/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index 2ac9125..a6849a9 100644
--- a/src/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/src/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -145,10 +145,11 @@
void(WebRtc_UWord32* totalRate, WebRtc_UWord32* videoRate, WebRtc_UWord32* fecRate, WebRtc_UWord32* nackRate));
MOCK_CONST_METHOD1(EstimatedReceiveBandwidth,
int(WebRtc_UWord32* available_bandwidth));
- MOCK_METHOD7(SendOutgoingData,
+ MOCK_METHOD8(SendOutgoingData,
WebRtc_Word32(const FrameType frameType,
const WebRtc_Word8 payloadType,
const WebRtc_UWord32 timeStamp,
+ int64_t capture_time_ms,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord32 payloadSize,
const RTPFragmentationHeader* fragmentation,
diff --git a/src/modules/rtp_rtcp/source/rtp_header_extension.h b/src/modules/rtp_rtcp/source/rtp_header_extension.h
index d5740a6..bb6dd81 100644
--- a/src/modules/rtp_rtcp/source/rtp_header_extension.h
+++ b/src/modules/rtp_rtcp/source/rtp_header_extension.h
@@ -13,7 +13,7 @@
#include <map>
-#include "rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "typedefs.h"
namespace webrtc {
diff --git a/src/modules/rtp_rtcp/source/rtp_packet_history.cc b/src/modules/rtp_rtcp/source/rtp_packet_history.cc
index b3a93e5..3f09f6d 100644
--- a/src/modules/rtp_rtcp/source/rtp_packet_history.cc
+++ b/src/modules/rtp_rtcp/source/rtp_packet_history.cc
@@ -108,6 +108,7 @@
int32_t RTPPacketHistory::PutRTPPacket(const uint8_t* packet,
uint16_t packet_length,
uint16_t max_packet_length,
+ int64_t capture_time_ms,
StorageType type) {
if (type == kDontStore) {
return 0;
@@ -138,7 +139,7 @@
stored_seq_nums_[prev_index_] = seq_num;
stored_lengths_[prev_index_] = packet_length;
- stored_times_[prev_index_] = clock_.GetTimeInMS();
+ stored_times_[prev_index_] = capture_time_ms;
stored_resend_times_[prev_index_] = 0; // packet not resent
stored_types_[prev_index_] = type;
@@ -173,7 +174,7 @@
uint32_t min_elapsed_time_ms,
uint8_t* packet,
uint16_t* packet_length,
- uint32_t* stored_time_ms,
+ int64_t* stored_time_ms,
StorageType* type) const {
webrtc::CriticalSectionScoped cs(*critsect_);
if (!store_) {
diff --git a/src/modules/rtp_rtcp/source/rtp_packet_history.h b/src/modules/rtp_rtcp/source/rtp_packet_history.h
index 581bffb..5efadcd 100644
--- a/src/modules/rtp_rtcp/source/rtp_packet_history.h
+++ b/src/modules/rtp_rtcp/source/rtp_packet_history.h
@@ -37,6 +37,7 @@
int32_t PutRTPPacket(const uint8_t* packet,
uint16_t packet_length,
uint16_t max_packet_length,
+ int64_t capture_time_ms,
StorageType type);
// Gets stored RTP packet corresponding to the input sequence number.
@@ -54,7 +55,7 @@
uint32_t min_elapsed_time_ms,
uint8_t* packet,
uint16_t* packet_length,
- uint32_t* stored_time_ms,
+ int64_t* stored_time_ms,
StorageType* type) const;
bool HasRTPPacket(uint16_t sequence_number) const;
diff --git a/src/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc b/src/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
index 1bf55ed..47f3df5 100644
--- a/src/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
+++ b/src/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
@@ -87,12 +87,13 @@
TEST_F(RtpPacketHistoryTest, NoStoreStatus) {
EXPECT_FALSE(hist_->StorePackets());
uint16_t len = 0;
+ int64_t capture_time_ms = fake_clock_.GetTimeInMS();
CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
- kAllowRetransmission));
+ capture_time_ms, kAllowRetransmission));
// Packet should not be stored.
len = kMaxPacketLength;
- uint32_t time;
+ int64_t time;
StorageType type;
EXPECT_FALSE(hist_->GetRTPPacket(kSeqNum, 0, packet_, &len, &time, &type));
}
@@ -100,32 +101,37 @@
TEST_F(RtpPacketHistoryTest, DontStore) {
hist_->SetStorePacketsStatus(true, 10);
uint16_t len = 0;
+ int64_t capture_time_ms = fake_clock_.GetTimeInMS();
CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
- EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength, kDontStore));
+ EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
+ capture_time_ms, kDontStore));
// Packet should not be stored.
len = kMaxPacketLength;
- uint32_t time;
+ int64_t time;
StorageType type;
EXPECT_FALSE(hist_->GetRTPPacket(kSeqNum, 0, packet_, &len, &time, &type));
}
TEST_F(RtpPacketHistoryTest, PutRtpPacket_TooLargePacketLength) {
hist_->SetStorePacketsStatus(true, 10);
+ int64_t capture_time_ms = fake_clock_.GetTimeInMS();
EXPECT_EQ(-1, hist_->PutRTPPacket(packet_,
kMaxPacketLength + 1,
kMaxPacketLength,
+ capture_time_ms,
kAllowRetransmission));
}
TEST_F(RtpPacketHistoryTest, GetRtpPacket_TooSmallBuffer) {
hist_->SetStorePacketsStatus(true, 10);
uint16_t len = 0;
+ int64_t capture_time_ms = fake_clock_.GetTimeInMS();
CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
- kAllowRetransmission));
+ capture_time_ms, kAllowRetransmission));
uint16_t len_out = len - 1;
- uint32_t time;
+ int64_t time;
StorageType type;
EXPECT_FALSE(hist_->GetRTPPacket(kSeqNum, 0, packet_, &len_out, &time,
&type));
@@ -134,7 +140,7 @@
TEST_F(RtpPacketHistoryTest, GetRtpPacket_NotStored) {
hist_->SetStorePacketsStatus(true, 10);
uint16_t len = kMaxPacketLength;
- uint32_t time;
+ int64_t time;
StorageType type;
EXPECT_FALSE(hist_->GetRTPPacket(0, 0, packet_, &len, &time, &type));
}
@@ -145,25 +151,28 @@
CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
EXPECT_FALSE(hist_->HasRTPPacket(kSeqNum));
+ int64_t capture_time_ms = fake_clock_.GetTimeInMS();
EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
- kAllowRetransmission));
+ capture_time_ms, kAllowRetransmission));
EXPECT_TRUE(hist_->HasRTPPacket(kSeqNum));
}
TEST_F(RtpPacketHistoryTest, GetRtpPacket) {
hist_->SetStorePacketsStatus(true, 10);
uint16_t len = 0;
+ int64_t capture_time_ms = fake_clock_.GetTimeInMS();
CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
- kAllowRetransmission));
+ capture_time_ms, kAllowRetransmission));
uint16_t len_out = kMaxPacketLength;
- uint32_t time;
+ int64_t time;
StorageType type;
EXPECT_TRUE(hist_->GetRTPPacket(kSeqNum, 0, packet_out_, &len_out, &time,
&type));
EXPECT_EQ(len, len_out);
EXPECT_EQ(kAllowRetransmission, type);
+ EXPECT_EQ(capture_time_ms, time);
for (int i = 0; i < len; i++) {
EXPECT_EQ(packet_[i], packet_out_[i]);
}
@@ -172,26 +181,28 @@
TEST_F(RtpPacketHistoryTest, DontRetransmit) {
hist_->SetStorePacketsStatus(true, 10);
uint16_t len = 0;
+ int64_t capture_time_ms = fake_clock_.GetTimeInMS();
CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
- kDontRetransmit));
+ capture_time_ms, kDontRetransmit));
uint16_t len_out = kMaxPacketLength;
- uint32_t time;
+ int64_t time;
StorageType type;
EXPECT_TRUE(hist_->GetRTPPacket(kSeqNum, 0, packet_out_, &len_out, &time,
&type));
EXPECT_EQ(len, len_out);
EXPECT_EQ(kDontRetransmit, type);
+ EXPECT_EQ(capture_time_ms, time);
}
TEST_F(RtpPacketHistoryTest, MinResendTime) {
hist_->SetStorePacketsStatus(true, 10);
- WebRtc_Word64 store_time = fake_clock_.GetTimeInMS();
uint16_t len = 0;
+ int64_t capture_time_ms = fake_clock_.GetTimeInMS();
CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
- kAllowRetransmission));
+ capture_time_ms, kAllowRetransmission));
hist_->UpdateResendTime(kSeqNum);
fake_clock_.IncrementTime(100);
@@ -199,10 +210,10 @@
// Time has elapsed.
len = kMaxPacketLength;
StorageType type;
- uint32_t time;
+ int64_t time;
EXPECT_TRUE(hist_->GetRTPPacket(kSeqNum, 100, packet_, &len, &time, &type));
EXPECT_GT(len, 0);
- EXPECT_EQ(store_time, time);
+ EXPECT_EQ(capture_time_ms, time);
// Time has not elapsed. Packet should be found, but no bytes copied.
len = kMaxPacketLength;
diff --git a/src/modules/rtp_rtcp/source/rtp_receiver_video.cc b/src/modules/rtp_rtcp/source/rtp_receiver_video.cc
index 2fc3258b..0af375e 100644
--- a/src/modules/rtp_rtcp/source/rtp_receiver_video.cc
+++ b/src/modules/rtp_rtcp/source/rtp_receiver_video.cc
@@ -92,10 +92,12 @@
// Ethernet header here as well.
const WebRtc_UWord16 packetSize = payloadDataLength + _packetOverHead +
rtpHeader->header.headerLength + rtpHeader->header.paddingLength;
+ uint32_t compensated_timestamp = rtpHeader->header.timestamp +
+ rtpHeader->extension.transmissionTimeOffset;
remote_bitrate_->IncomingPacket(rtpHeader->header.ssrc,
packetSize,
nowMS,
- rtpHeader->header.timestamp,
+ compensated_timestamp,
-1);
if (isRED) {
diff --git a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index b2851a7..56a4236 100644
--- a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -798,6 +798,7 @@
FrameType frameType,
WebRtc_Word8 payloadType,
WebRtc_UWord32 timeStamp,
+ int64_t capture_time_ms,
const WebRtc_UWord8* payloadData,
WebRtc_UWord32 payloadSize,
const RTPFragmentationHeader* fragmentation,
@@ -818,6 +819,7 @@
return _rtpSender.SendOutgoingData(frameType,
payloadType,
timeStamp,
+ capture_time_ms,
payloadData,
payloadSize,
fragmentation,
@@ -847,6 +849,7 @@
return rtpSender.SendOutgoingData(frameType,
payloadType,
timeStamp,
+ capture_time_ms,
payloadData,
payloadSize,
fragmentation,
@@ -863,6 +866,7 @@
retVal = rtpSender.SendOutgoingData(frameType,
payloadType,
timeStamp,
+ capture_time_ms,
payloadData,
payloadSize,
fragmentation,
@@ -878,6 +882,7 @@
retVal = rtpSender.SendOutgoingData(frameType,
payloadType,
timeStamp,
+ capture_time_ms,
payloadData,
payloadSize,
fragmentation,
diff --git a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index ece25d1..046e4e1 100644
--- a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -176,6 +176,7 @@
const FrameType frameType,
const WebRtc_Word8 payloadType,
const WebRtc_UWord32 timeStamp,
+ int64_t capture_time_ms,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord32 payloadSize,
const RTPFragmentationHeader* fragmentation = NULL,
diff --git a/src/modules/rtp_rtcp/source/rtp_sender.cc b/src/modules/rtp_rtcp/source/rtp_sender.cc
index c122e52..5bfb76a 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/src/modules/rtp_rtcp/source/rtp_sender.cc
@@ -464,6 +464,7 @@
RTPSender::SendOutgoingData(const FrameType frame_type,
const WebRtc_Word8 payload_type,
const WebRtc_UWord32 capture_timestamp,
+ int64_t capture_time_ms,
const WebRtc_UWord8* payload_data,
const WebRtc_UWord32 payload_size,
const RTPFragmentationHeader* fragmentation,
@@ -500,12 +501,14 @@
frame_type != kAudioFrameCN);
if (frame_type == kFrameEmpty) {
- return SendPaddingAccordingToBitrate(payload_type, capture_timestamp);
+ return SendPaddingAccordingToBitrate(payload_type, capture_timestamp,
+ capture_time_ms);
}
return _video->SendVideo(video_type,
frame_type,
payload_type,
capture_timestamp,
+ capture_time_ms,
payload_data,
payload_size,
fragmentation,
@@ -516,7 +519,8 @@
WebRtc_Word32 RTPSender::SendPaddingAccordingToBitrate(
WebRtc_Word8 payload_type,
- WebRtc_UWord32 capture_timestamp) {
+ WebRtc_UWord32 capture_timestamp,
+ int64_t capture_time_ms) {
// Current bitrate since last estimate(1 second) averaged with the
// estimate since then, to get the most up to date bitrate.
uint32_t current_bitrate = BitrateNow();
@@ -535,13 +539,14 @@
}
}
// Send padding data.
- return SendPadData(payload_type, capture_timestamp, bytes);
+ return SendPadData(payload_type, capture_timestamp, capture_time_ms, bytes);
}
return 0;
}
WebRtc_Word32 RTPSender::SendPadData(WebRtc_Word8 payload_type,
WebRtc_UWord32 capture_timestamp,
+ int64_t capture_time_ms,
WebRtc_Word32 bytes) {
// Drop this packet if we're not sending media packets
if (!_sendingMedia) {
@@ -586,6 +591,7 @@
if (0 > SendToNetwork(data_buffer,
padding_bytes_in_packet,
header_length,
+ capture_time_ms,
kDontRetransmit)) {
// Error sending the packet.
break;
@@ -616,7 +622,7 @@
WebRtc_UWord8 data_buffer[IP_PACKET_SIZE];
WebRtc_UWord8* buffer_to_send_ptr = data_buffer;
- WebRtc_UWord32 stored_time_in_ms;
+ int64_t stored_time_in_ms;
StorageType type;
bool found = _packetHistory->GetRTPPacket(packet_id,
min_resend_time, data_buffer, &length, &stored_time_in_ms, &type);
@@ -848,7 +854,7 @@
WebRtc_UWord8 data_buffer[IP_PACKET_SIZE];
WebRtc_UWord16 length = IP_PACKET_SIZE;
- WebRtc_UWord32 stored_time_ms;
+ int64_t stored_time_ms;
StorageType type;
bool found = _packetHistory->GetRTPPacket(seq_num, 0, data_buffer, &length,
&stored_time_ms, &type);
@@ -888,14 +894,16 @@
}
WebRtc_Word32
-RTPSender::SendToNetwork(const WebRtc_UWord8* buffer,
+RTPSender::SendToNetwork(WebRtc_UWord8* buffer,
const WebRtc_UWord16 length,
const WebRtc_UWord16 rtpLength,
+ int64_t capture_time_ms,
const StorageType storage)
{
// Used for NACK or to spead out the transmission of packets.
if (_packetHistory->PutRTPPacket(
- buffer, rtpLength + length, _maxPayloadLength, storage) != 0) {
+ buffer, rtpLength + length, _maxPayloadLength, capture_time_ms, storage)
+ != 0) {
return -1;
}
@@ -906,6 +914,15 @@
return 0;
}
+ if (capture_time_ms >= 0) {
+ ModuleRTPUtility::RTPHeaderParser rtpParser(buffer, length);
+ WebRtcRTPHeader rtp_header;
+ rtpParser.Parse(rtp_header);
+ int64_t time_now = _clock.GetTimeInMS();
+ UpdateTransmissionTimeOffset(buffer, length, rtp_header,
+ time_now - capture_time_ms);
+ }
+
// Send packet
WebRtc_Word32 bytes_sent = -1;
{
diff --git a/src/modules/rtp_rtcp/source/rtp_sender.h b/src/modules/rtp_rtcp/source/rtp_sender.h
index 963f5cc..eadfca3 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender.h
+++ b/src/modules/rtp_rtcp/source/rtp_sender.h
@@ -56,9 +56,10 @@
virtual WebRtc_UWord16 PacketOverHead() const = 0;
virtual WebRtc_UWord16 ActualSendBitrateKbit() const = 0;
- virtual WebRtc_Word32 SendToNetwork(const WebRtc_UWord8* dataBuffer,
+ virtual WebRtc_Word32 SendToNetwork(WebRtc_UWord8* dataBuffer,
const WebRtc_UWord16 payloadLength,
const WebRtc_UWord16 rtpHeaderLength,
+ int64_t capture_time_ms,
const StorageType storage) = 0;
};
@@ -133,6 +134,7 @@
WebRtc_Word32 SendOutgoingData(const FrameType frameType,
const WebRtc_Word8 payloadType,
const WebRtc_UWord32 timeStamp,
+ int64_t capture_time_ms,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord32 payloadSize,
const RTPFragmentationHeader* fragmentation,
@@ -141,6 +143,7 @@
WebRtc_Word32 SendPadData(WebRtc_Word8 payload_type,
WebRtc_UWord32 capture_timestamp,
+ int64_t capture_time_ms,
WebRtc_Word32 bytes);
/*
* RTP header extension
@@ -219,9 +222,10 @@
virtual WebRtc_UWord32 Timestamp() const;
virtual WebRtc_UWord32 SSRC() const;
- virtual WebRtc_Word32 SendToNetwork(const WebRtc_UWord8* dataBuffer,
+ virtual WebRtc_Word32 SendToNetwork(WebRtc_UWord8* dataBuffer,
const WebRtc_UWord16 payloadLength,
const WebRtc_UWord16 rtpHeaderLength,
+ int64_t capture_time_ms,
const StorageType storage);
/*
@@ -290,7 +294,8 @@
WebRtc_Word32 SendPaddingAccordingToBitrate(
WebRtc_Word8 payload_type,
- WebRtc_UWord32 capture_timestamp);
+ WebRtc_UWord32 capture_timestamp,
+ int64_t capture_time_ms);
WebRtc_Word32 _id;
const bool _audioConfigured;
diff --git a/src/modules/rtp_rtcp/source/rtp_sender_audio.cc b/src/modules/rtp_rtcp/source/rtp_sender_audio.cc
index 76738ca..71aa72a 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/src/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -457,6 +457,7 @@
return _rtpSender->SendToNetwork(dataBuffer,
payloadSize,
static_cast<WebRtc_UWord16>(rtpHeaderLength),
+ -1,
kAllowRetransmission);
}
@@ -591,7 +592,7 @@
ModuleRTPUtility::AssignUWord16ToBuffer(dtmfbuffer+14, duration);
_sendAudioCritsect->Leave();
- retVal = _rtpSender->SendToNetwork(dtmfbuffer, 4, 12,
+ retVal = _rtpSender->SendToNetwork(dtmfbuffer, 4, 12, -1,
kAllowRetransmission);
sendCount--;
diff --git a/src/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/src/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 113e65b..7bbb190 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/src/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -207,7 +207,10 @@
kTimestamp);
// Packet should be sent immediately.
- EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
+ EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_,
+ 0,
+ rtp_length,
+ kTimestamp / 90,
kAllowRetransmission));
EXPECT_EQ(1, transport_.packets_sent_);
EXPECT_EQ(rtp_length, transport_.last_sent_packet_len_);
@@ -225,7 +228,10 @@
kTimestamp);
// Packet should be stored in a send bucket.
- EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
+ EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_,
+ 0,
+ rtp_length,
+ fake_clock_.GetTimeInMS(),
kAllowRetransmission));
EXPECT_EQ(0, transport_.packets_sent_);
diff --git a/src/modules/rtp_rtcp/source/rtp_sender_video.cc b/src/modules/rtp_rtcp/source/rtp_sender_video.cc
index a9c8be7..41f0230 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/src/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -105,9 +105,10 @@
}
WebRtc_Word32
-RTPSenderVideo::SendVideoPacket(const WebRtc_UWord8* data_buffer,
+RTPSenderVideo::SendVideoPacket(WebRtc_UWord8* data_buffer,
const WebRtc_UWord16 payload_length,
const WebRtc_UWord16 rtp_header_length,
+ int64_t capture_time_ms,
StorageType storage,
bool protect) {
if(_fecEnabled) {
@@ -124,6 +125,7 @@
red_packet->data(),
red_packet->length() - rtp_header_length,
rtp_header_length,
+ capture_time_ms,
storage);
ret |= packet_success;
@@ -156,6 +158,7 @@
red_packet->data(),
red_packet->length() - rtp_header_length,
rtp_header_length,
+ capture_time_ms,
storage);
ret |= packet_success;
@@ -173,6 +176,7 @@
int ret = _rtpSender.SendToNetwork(data_buffer,
payload_length,
rtp_header_length,
+ capture_time_ms,
storage);
if (ret == 0) {
_videoBitrate.Update(payload_length + rtp_header_length);
@@ -195,7 +199,7 @@
ModuleRTPUtility::AssignUWord32ToBuffer(data+4, _rtpSender.SSRC());
- return _rtpSender.SendToNetwork(data, 0, length, kAllowRetransmission);
+ return _rtpSender.SendToNetwork(data, 0, length, -1, kAllowRetransmission);
}
WebRtc_Word32
@@ -248,7 +252,8 @@
RTPSenderVideo::SendVideo(const RtpVideoCodecTypes videoType,
const FrameType frameType,
const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
+ const uint32_t captureTimeStamp,
+ int64_t capture_time_ms,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord32 payloadSize,
const RTPFragmentationHeader* fragmentation,
@@ -276,12 +281,18 @@
switch(videoType)
{
case kRtpNoVideo:
- retVal = SendGeneric(payloadType,captureTimeStamp, payloadData,
- payloadSize);
+ retVal = SendGeneric(payloadType, captureTimeStamp, capture_time_ms,
+ payloadData, payloadSize);
break;
case kRtpVp8Video:
- retVal = SendVP8(frameType, payloadType, captureTimeStamp,
- payloadData, payloadSize, fragmentation, rtpTypeHdr);
+ retVal = SendVP8(frameType,
+ payloadType,
+ captureTimeStamp,
+ capture_time_ms,
+ payloadData,
+ payloadSize,
+ fragmentation,
+ rtpTypeHdr);
break;
default:
assert(false);
@@ -298,7 +309,8 @@
WebRtc_Word32
RTPSenderVideo::SendGeneric(const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
+ const uint32_t captureTimeStamp,
+ int64_t capture_time_ms,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord32 payloadSize)
{
@@ -348,6 +360,7 @@
if(-1 == SendVideoPacket(dataBuffer,
payloadBytesInPacket,
rtpHeaderLength,
+ capture_time_ms,
kAllowRetransmission,
true))
{
@@ -378,7 +391,8 @@
WebRtc_Word32
RTPSenderVideo::SendVP8(const FrameType frameType,
const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
+ const uint32_t captureTimeStamp,
+ int64_t capture_time_ms,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord32 payloadSize,
const RTPFragmentationHeader* fragmentation,
@@ -438,7 +452,7 @@
_rtpSender.BuildRTPheader(dataBuffer, payloadType, last,
captureTimeStamp);
if (-1 == SendVideoPacket(dataBuffer, payloadBytesInPacket,
- rtpHeaderLength, storage, protect))
+ rtpHeaderLength, capture_time_ms, storage, protect))
{
WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
"RTPSenderVideo::SendVP8 failed to send packet number"
diff --git a/src/modules/rtp_rtcp/source/rtp_sender_video.h b/src/modules/rtp_rtcp/source/rtp_sender_video.h
index 5c3f70c..64ac058 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender_video.h
+++ b/src/modules/rtp_rtcp/source/rtp_sender_video.h
@@ -50,7 +50,8 @@
WebRtc_Word32 SendVideo(const RtpVideoCodecTypes videoType,
const FrameType frameType,
const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
+ const uint32_t captureTimeStamp,
+ int64_t capture_time_ms,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord32 payloadSize,
const RTPFragmentationHeader* fragmentation,
@@ -88,21 +89,24 @@
int SetSelectiveRetransmissions(uint8_t settings);
protected:
- virtual WebRtc_Word32 SendVideoPacket(const WebRtc_UWord8* dataBuffer,
+ virtual WebRtc_Word32 SendVideoPacket(WebRtc_UWord8* dataBuffer,
const WebRtc_UWord16 payloadLength,
const WebRtc_UWord16 rtpHeaderLength,
+ int64_t capture_time_ms,
StorageType storage,
bool protect);
private:
WebRtc_Word32 SendGeneric(const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
+ const uint32_t captureTimeStamp,
+ int64_t capture_time_ms,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord32 payloadSize);
WebRtc_Word32 SendVP8(const FrameType frameType,
const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
+ const uint32_t captureTimeStamp,
+ int64_t capture_time_ms,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord32 payloadSize,
const RTPFragmentationHeader* fragmentation,
diff --git a/src/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/src/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
index 25b91a0..bf88ace 100644
--- a/src/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
+++ b/src/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
@@ -203,7 +203,7 @@
// Send an empty RTP packet.
// Should fail since we have not registerd the payload type.
EXPECT_EQ(-1, module1->SendOutgoingData(webrtc::kAudioFrameSpeech,
- 96, 0, NULL, 0));
+ 96, 0, -1, NULL, 0));
CodecInst voiceCodec;
voiceCodec.pltype = 96;
@@ -219,7 +219,7 @@
const WebRtc_UWord8 test[5] = "test";
EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
- 0, test, 4));
+ 0, -1, test, 4));
EXPECT_EQ(test_ssrc, module2->RemoteSSRC());
EXPECT_EQ(test_timestamp, module2->RemoteTimestamp());
@@ -270,7 +270,7 @@
const WebRtc_UWord8 test[5] = "test";
// Send a RTP packet.
EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech,
- 96, 160, test, 4,
+ 96, 160, -1, test, 4,
&fragmentation));
EXPECT_EQ(0, module1->SetSendREDPayloadType(-1));
@@ -316,7 +316,7 @@
// pause between = 2560ms + 1600ms = 4160ms
for (;timeStamp <= 250 * 160; timeStamp += 160) {
EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
- timeStamp, test, 4));
+ timeStamp, -1, test, 4));
fake_clock.IncrementTime(20);
module1->Process();
}
@@ -324,7 +324,7 @@
for (;timeStamp <= 740 * 160; timeStamp += 160) {
EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
- timeStamp, test, 4));
+ timeStamp, -1, test, 4));
fake_clock.IncrementTime(20);
module1->Process();
}
diff --git a/src/modules/rtp_rtcp/test/testAPI/test_api_nack.cc b/src/modules/rtp_rtcp/test/testAPI/test_api_nack.cc
index ad4dd6a..6fa81e71 100644
--- a/src/modules/rtp_rtcp/test/testAPI/test_api_nack.cc
+++ b/src/modules/rtp_rtcp/test/testAPI/test_api_nack.cc
@@ -156,6 +156,7 @@
for (int frame = 0; frame < 10; ++frame) {
EXPECT_EQ(0, video_module_->SendOutgoingData(webrtc::kVideoFrameDelta, 123,
timestamp,
+ timestamp / 90,
payload_data,
payload_data_length));
@@ -209,10 +210,12 @@
WebRtc_UWord16 nack_list[kVideoNackListSize];
for (int frame = 0; frame < 10; ++frame) {
- EXPECT_EQ(0, video_module_->SendOutgoingData(webrtc::kVideoFrameDelta, 123,
- timestamp,
- payload_data,
- payload_data_length));
+ EXPECT_EQ(0, video_module_->SendOutgoingData(webrtc::kVideoFrameDelta,
+ 123,
+ timestamp,
+ timestamp / 90,
+ payload_data,
+ payload_data_length));
std::sort(nack_receiver_->sequence_numbers_.begin(),
nack_receiver_->sequence_numbers_.end());
diff --git a/src/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc b/src/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
index bdf3c6b..29596f9 100644
--- a/src/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
+++ b/src/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
@@ -140,7 +140,7 @@
// send RTP packet with the data "testtest"
const WebRtc_UWord8 test[9] = "testtest";
EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
- 0, test, 8));
+ 0, -1, test, 8));
}
virtual void TearDown() {
diff --git a/src/modules/rtp_rtcp/test/testAPI/test_api_video.cc b/src/modules/rtp_rtcp/test/testAPI/test_api_video.cc
index c5b134a..fdd3ed3 100644
--- a/src/modules/rtp_rtcp/test/testAPI/test_api_video.cc
+++ b/src/modules/rtp_rtcp/test/testAPI/test_api_video.cc
@@ -87,6 +87,7 @@
WebRtc_UWord32 timestamp = 3000;
EXPECT_EQ(0, video_module->SendOutgoingData(webrtc::kVideoFrameDelta, 123,
timestamp,
+ timestamp / 90,
payload_data,
payload_data_length));
diff --git a/src/modules/utility/source/video_coder.cc b/src/modules/utility/source/video_coder.cc
index 06c7ffd..7276723 100644
--- a/src/modules/utility/source/video_coder.cc
+++ b/src/modules/utility/source/video_coder.cc
@@ -127,9 +127,10 @@
}
WebRtc_Word32 VideoCoder::SendData(
- FrameType frameType,
- WebRtc_UWord8 payloadType,
- WebRtc_UWord32 timeStamp,
+ const FrameType frameType,
+ const WebRtc_UWord8 payloadType,
+ const WebRtc_UWord32 timeStamp,
+ int64_t capture_time_ms,
const WebRtc_UWord8* payloadData,
WebRtc_UWord32 payloadSize,
const RTPFragmentationHeader& fragmentationHeader,
diff --git a/src/modules/utility/source/video_coder.h b/src/modules/utility/source/video_coder.h
index 33174dd..dce6f1a 100644
--- a/src/modules/utility/source/video_coder.h
+++ b/src/modules/utility/source/video_coder.h
@@ -54,6 +54,7 @@
const FrameType /*frameType*/,
const WebRtc_UWord8 /*payloadType*/,
const WebRtc_UWord32 /*timeStamp*/,
+ int64_t capture_time_ms,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord32 payloadSize,
const RTPFragmentationHeader& /* fragmentationHeader*/,
diff --git a/src/modules/video_coding/codecs/vp8/main/source/vp8.cc b/src/modules/video_coding/codecs/vp8/main/source/vp8.cc
index a112718..0255dd3 100644
--- a/src/modules/video_coding/codecs/vp8/main/source/vp8.cc
+++ b/src/modules/video_coding/codecs/vp8/main/source/vp8.cc
@@ -469,6 +469,7 @@
if (encoded_image_._length > 0) {
encoded_image_._timeStamp = input_image._timeStamp;
+ encoded_image_.capture_time_ms_ = input_image.capture_time_ms_;
// Figure out where partition boundaries are located.
RTPFragmentationHeader fragInfo;
@@ -542,6 +543,7 @@
}
if (encoded_image_._length > 0) {
encoded_image_._timeStamp = input_image._timeStamp;
+ encoded_image_.capture_time_ms_ = input_image.capture_time_ms_;
encoded_image_._encodedHeight = raw_->h;
encoded_image_._encodedWidth = raw_->w;
encoded_complete_callback_->Encoded(encoded_image_, &codec_specific,
diff --git a/src/modules/video_coding/main/interface/video_coding_defines.h b/src/modules/video_coding/main/interface/video_coding_defines.h
index bd2a98c..324b24b 100644
--- a/src/modules/video_coding/main/interface/video_coding_defines.h
+++ b/src/modules/video_coding/main/interface/video_coding_defines.h
@@ -69,9 +69,12 @@
class VCMPacketizationCallback {
public:
virtual WebRtc_Word32 SendData(
- const FrameType frameType, const WebRtc_UWord8 payloadType,
- const WebRtc_UWord32 timeStamp, const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize,
+ FrameType frameType,
+ WebRtc_UWord8 payloadType,
+ WebRtc_UWord32 timeStamp,
+ int64_t capture_time_ms,
+ const WebRtc_UWord8* payloadData,
+ WebRtc_UWord32 payloadSize,
const RTPFragmentationHeader& fragmentationHeader,
const RTPVideoHeader* rtpVideoHdr) = 0;
protected:
diff --git a/src/modules/video_coding/main/source/generic_encoder.cc b/src/modules/video_coding/main/source/generic_encoder.cc
index 613c3ec..fb02c99 100644
--- a/src/modules/video_coding/main/source/generic_encoder.cc
+++ b/src/modules/video_coding/main/source/generic_encoder.cc
@@ -67,6 +67,7 @@
rawImage._width = inputFrame.Width();
rawImage._height = inputFrame.Height();
rawImage._timeStamp = inputFrame.TimeStamp();
+ rawImage.capture_time_ms_ = inputFrame.RenderTimeMs();
VideoFrameType videoFrameType =
VCMEncodedFrame::ConvertFrameType(frameType);
@@ -207,6 +208,7 @@
frameType,
_payloadType,
encodedImage._timeStamp,
+ encodedImage.capture_time_ms_,
encodedImage._buffer,
encodedBytes,
*fragmentationHeader,
diff --git a/src/modules/video_coding/main/test/generic_codec_test.cc b/src/modules/video_coding/main/test/generic_codec_test.cc
index 8d12325..773f7ab 100644
--- a/src/modules/video_coding/main/test/generic_codec_test.cc
+++ b/src/modules/video_coding/main/test/generic_codec_test.cc
@@ -570,6 +570,7 @@
const FrameType frameType,
const WebRtc_UWord8 payloadType,
const WebRtc_UWord32 timeStamp,
+ int64_t capture_time_ms,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord32 payloadSize,
const RTPFragmentationHeader& /*fragmentationHeader*/,
diff --git a/src/modules/video_coding/main/test/generic_codec_test.h b/src/modules/video_coding/main/test/generic_codec_test.h
index c88280f..f60893b 100644
--- a/src/modules/video_coding/main/test/generic_codec_test.h
+++ b/src/modules/video_coding/main/test/generic_codec_test.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -94,6 +94,7 @@
const webrtc::FrameType frameType,
const WebRtc_UWord8 payloadType,
WebRtc_UWord32 timeStamp,
+ int64_t capture_time_ms,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord32 payloadSize,
const webrtc::RTPFragmentationHeader& fragmentationHeader,
diff --git a/src/modules/video_coding/main/test/normal_test.cc b/src/modules/video_coding/main/test/normal_test.cc
index f31cd22..bd37766 100644
--- a/src/modules/video_coding/main/test/normal_test.cc
+++ b/src/modules/video_coding/main/test/normal_test.cc
@@ -76,6 +76,7 @@
const FrameType frameType,
const WebRtc_UWord8 payloadType,
const WebRtc_UWord32 timeStamp,
+ int64_t capture_time_ms,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord32 payloadSize,
const RTPFragmentationHeader& /*fragmentationHeader*/,
diff --git a/src/modules/video_coding/main/test/normal_test.h b/src/modules/video_coding/main/test/normal_test.h
index 6f75dfb..6165d65 100644
--- a/src/modules/video_coding/main/test/normal_test.h
+++ b/src/modules/video_coding/main/test/normal_test.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -31,6 +31,7 @@
WebRtc_Word32 SendData(const webrtc::FrameType frameType,
const WebRtc_UWord8 payloadType,
const WebRtc_UWord32 timeStamp,
+ int64_t capture_time_ms,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord32 payloadSize,
const webrtc::RTPFragmentationHeader& fragmentationHeader,
diff --git a/src/modules/video_coding/main/test/test_callbacks.cc b/src/modules/video_coding/main/test/test_callbacks.cc
index 34c7927..2f18dd1 100644
--- a/src/modules/video_coding/main/test/test_callbacks.cc
+++ b/src/modules/video_coding/main/test/test_callbacks.cc
@@ -51,6 +51,7 @@
const FrameType frameType,
const WebRtc_UWord8 payloadType,
const WebRtc_UWord32 timeStamp,
+ int64_t capture_time_ms,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord32 payloadSize,
const RTPFragmentationHeader& fragmentationHeader,
@@ -144,6 +145,7 @@
const FrameType frameType,
const WebRtc_UWord8 payloadType,
const WebRtc_UWord32 timeStamp,
+ int64_t capture_time_ms,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord32 payloadSize,
const RTPFragmentationHeader& fragmentationHeader,
@@ -155,6 +157,7 @@
return _RTPModule->SendOutgoingData(frameType,
payloadType,
timeStamp,
+ capture_time_ms,
payloadData,
payloadSize,
&fragmentationHeader,
diff --git a/src/modules/video_coding/main/test/test_callbacks.h b/src/modules/video_coding/main/test/test_callbacks.h
index 9f179e2..eae963b 100644
--- a/src/modules/video_coding/main/test/test_callbacks.h
+++ b/src/modules/video_coding/main/test/test_callbacks.h
@@ -43,10 +43,13 @@
// Process encoded data received from the encoder, pass stream to the
// VCMReceiver module
WebRtc_Word32 SendData(const FrameType frameType,
- const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData, const WebRtc_UWord32 payloadSize,
- const RTPFragmentationHeader& fragmentationHeader,
- const RTPVideoHeader* videoHdr);
+ const WebRtc_UWord8 payloadType,
+ const WebRtc_UWord32 timeStamp,
+ int64_t capture_time_ms,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ const RTPFragmentationHeader& fragmentationHeader,
+ const RTPVideoHeader* videoHdr);
// Register exisitng VCM. Currently - encode and decode under same module.
void RegisterReceiverVCM(VideoCodingModule *vcm) {_VCMReceiver = vcm;}
// Return size of last encoded frame data (all frames in the sequence)
@@ -99,10 +102,13 @@
// Process encoded data received from the encoder, pass stream to the
// RTP module
WebRtc_Word32 SendData(const FrameType frameType,
- const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData, const WebRtc_UWord32 payloadSize,
- const RTPFragmentationHeader& fragmentationHeader,
- const RTPVideoHeader* videoHdr);
+ const WebRtc_UWord8 payloadType,
+ const WebRtc_UWord32 timeStamp,
+ int64_t capture_time_ms,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ const RTPFragmentationHeader& fragmentationHeader,
+ const RTPVideoHeader* videoHdr);
// Return size of last encoded frame. Value good for one call
// (resets to zero after call to inform test of frame drop)
float EncodedBytes();
diff --git a/src/modules/video_processing/main/source/spatial_resampler.cc b/src/modules/video_processing/main/source/spatial_resampler.cc
index dd018cf..8837370 100644
--- a/src/modules/video_processing/main/source/spatial_resampler.cc
+++ b/src/modules/video_processing/main/source/spatial_resampler.cc
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -89,6 +89,7 @@
(target_half_width * target_half_height));
outFrame.VerifyAndAllocate(requiredSize);
outFrame.SetTimeStamp(inFrame.TimeStamp());
+ outFrame.SetRenderTime(inFrame.RenderTimeMs());
outFrame.SetWidth(_targetWidth);
outFrame.SetHeight(_targetHeight);
diff --git a/src/video_engine/vie_channel.cc b/src/video_engine/vie_channel.cc
index 42101e0..20d32c0 100644
--- a/src/video_engine/vie_channel.cc
+++ b/src/video_engine/vie_channel.cc
@@ -121,6 +121,21 @@
WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
"%s: RTP::SetRTCPStatus failure", __FUNCTION__);
}
+ if (rtp_rtcp_->RegisterSendRtpHeaderExtension(
+ kRtpExtensionTransmissionTimeOffset, 1) != 0) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: RTP::RegisterSendRtpHeaderExtension failure",
+ __FUNCTION__);
+ return -1;
+ }
+ if (rtp_rtcp_->RegisterReceiveRtpHeaderExtension(
+ kRtpExtensionTransmissionTimeOffset, 1) != 0) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: RTP::RegisterReceiveRtpHeaderExtension failure",
+ __FUNCTION__);
+ return -1;
+ }
+
// VCM initialization
if (vcm_.InitializeReceiver() != 0) {
WEBRTC_TRACE(kTraceError, kTraceVideo,
@@ -284,6 +299,13 @@
if (restart_rtp) {
rtp_rtcp->SetSendingStatus(true);
}
+ if (rtp_rtcp_->RegisterReceiveRtpHeaderExtension(
+ kRtpExtensionTransmissionTimeOffset, 1) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: could not register transmission time offset extension",
+ __FUNCTION__);
+ return -1;
+ }
}
// |RegisterSimulcastRtpRtcpModules| resets all old weak pointers and old
// modules can be deleted after this step.
diff --git a/src/video_engine/vie_encoder.cc b/src/video_engine/vie_encoder.cc
index 72610cc..25e4dca 100644
--- a/src/video_engine/vie_encoder.cc
+++ b/src/video_engine/vie_encoder.cc
@@ -150,6 +150,13 @@
"%s RegisterSendPayload failure", __FUNCTION__);
return false;
}
+ if (default_rtp_rtcp_->RegisterSendRtpHeaderExtension(
+ kRtpExtensionTransmissionTimeOffset, 1) != 0) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "%s RegisterSendRtpHeaderExtension failure", __FUNCTION__);
+ return false;
+ }
#else
VideoCodec video_codec;
if (vcm_.Codec(webrtc::kVideoCodecI420, &video_codec) == VCM_OK) {
@@ -682,6 +689,7 @@
const FrameType frame_type,
const WebRtc_UWord8 payload_type,
const WebRtc_UWord32 time_stamp,
+ int64_t capture_time_ms,
const WebRtc_UWord8* payload_data,
const WebRtc_UWord32 payload_size,
const webrtc::RTPFragmentationHeader& fragmentation_header,
@@ -702,8 +710,11 @@
}
// New encoded data, hand over to the rtp module.
- return default_rtp_rtcp_->SendOutgoingData(frame_type, payload_type,
- time_stamp, payload_data,
+ return default_rtp_rtcp_->SendOutgoingData(frame_type,
+ payload_type,
+ time_stamp,
+ capture_time_ms,
+ payload_data,
payload_size,
&fragmentation_header,
rtp_video_hdr);
diff --git a/src/video_engine/vie_encoder.h b/src/video_engine/vie_encoder.h
index 52410bf..b6c3321 100644
--- a/src/video_engine/vie_encoder.h
+++ b/src/video_engine/vie_encoder.h
@@ -109,6 +109,7 @@
const FrameType frame_type,
const WebRtc_UWord8 payload_type,
const WebRtc_UWord32 time_stamp,
+ int64_t capture_time_ms,
const WebRtc_UWord8* payload_data,
const WebRtc_UWord32 payload_size,
const RTPFragmentationHeader& fragmentation_header,
diff --git a/src/voice_engine/main/source/channel.cc b/src/voice_engine/main/source/channel.cc
index a4c1dcb..d46a8d5 100644
--- a/src/voice_engine/main/source/channel.cc
+++ b/src/voice_engine/main/source/channel.cc
@@ -63,6 +63,10 @@
if (_rtpRtcpModule->SendOutgoingData((FrameType&)frameType,
payloadType,
timeStamp,
+ // Leaving the time when this frame was
+ // received from the capture device as
+ // undefined for voice for now.
+ -1,
payloadData,
payloadSize,
fragmentation) == -1)
@@ -5728,6 +5732,10 @@
if (_rtpRtcpModule->SendOutgoingData(kAudioFrameSpeech,
_lastPayloadType,
_lastLocalTimeStamp,
+ // Leaving the time when this frame was
+ // received from the capture device as
+ // undefined for voice for now.
+ -1,
(const WebRtc_UWord8*) payloadData,
payloadSize) != 0)
{
