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Adds event traces and counters for WebRTC receive side.

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Review URL: https://webrtc-codereview.appspot.com/1279005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3766 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
edjee@google.com 2013-04-04 19:43:34 +00:00
parent 835dbf4516
commit 79b0289bfc
10 changed files with 345 additions and 4 deletions
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3
webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
View File

@ -22,6 +22,7 @@
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
#include "webrtc/system_wrappers/interface/trace.h"
#include "webrtc/system_wrappers/interface/trace_event.h"
namespace webrtc {
@ -2231,6 +2232,8 @@ AudioPlayoutMode AudioCodingModuleImpl::PlayoutMode() const {
// Automatic resample to the requested frequency.
WebRtc_Word32 AudioCodingModuleImpl::PlayoutData10Ms(
WebRtc_Word32 desired_freq_hz, AudioFrame* audio_frame) {
TRACE_EVENT0("webrtc_voe", "ACM::PlayoutData10Ms");
bool stereo_mode;
if (GetSilence(desired_freq_hz, audio_frame))

72
webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
View File

@ -18,11 +18,49 @@
#include "modules/rtp_rtcp/source/rtp_rtcp_impl.h"
#include "system_wrappers/interface/critical_section_wrapper.h"
#include "system_wrappers/interface/trace.h"
#include "system_wrappers/interface/trace_event.h"
namespace webrtc {
using RTCPUtility::RTCPCnameInformation;
NACKStringBuilder::NACKStringBuilder() :
_stream(""), _count(0), _consecutive(false)
{
// Empty.
}
void NACKStringBuilder::PushNACK(WebRtc_UWord16 nack)
{
if (_count == 0)
{
_stream << nack;
} else if (nack == _prevNack + 1)
{
_consecutive = true;
} else
{
if (_consecutive)
{
_stream << "-" << _prevNack;
_consecutive = false;
}
_stream << "," << nack;
}
_count++;
_prevNack = nack;
}
std::string NACKStringBuilder::GetResult()
{
if (_consecutive)
{
_stream << "-" << _prevNack;
_consecutive = false;
}
return _stream.str();
}
RTCPSender::RTCPSender(const WebRtc_Word32 id,
const bool audio,
Clock* clock,
@ -79,7 +117,10 @@ RTCPSender::RTCPSender(const WebRtc_Word32 id,
_appData(NULL),
_appLength(0),
_xrSendVoIPMetric(false),
_xrVoIPMetric()
_xrVoIPMetric(),
_nackCount(0),
_pliCount(0),
_fullIntraRequestCount(0)
{
memset(_CNAME, 0, sizeof(_CNAME));
memset(_lastSendReport, 0, sizeof(_lastSendReport));
@ -151,6 +192,11 @@ RTCPSender::Init()
memset(_CNAME, 0, sizeof(_CNAME));
memset(_lastSendReport, 0, sizeof(_lastSendReport));
memset(_lastRTCPTime, 0, sizeof(_lastRTCPTime));
_nackCount = 0;
_pliCount = 0;
_fullIntraRequestCount = 0;
return 0;
}
@ -1065,6 +1111,7 @@ RTCPSender::BuildREMB(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos)
ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _rembSSRC[i]);
pos += 4;
}
TRACE_COUNTER1("webrtc_rtcp", "Remb", _rembBitrate);
return 0;
}
@ -1294,7 +1341,8 @@ WebRtc_Word32
RTCPSender::BuildNACK(WebRtc_UWord8* rtcpbuffer,
WebRtc_UWord32& pos,
const WebRtc_Word32 nackSize,
const WebRtc_UWord16* nackList)
const WebRtc_UWord16* nackList,
std::string* nackString)
{
// sanity
if(pos + 16 >= IP_PACKET_SIZE)
@ -1324,12 +1372,14 @@ RTCPSender::BuildNACK(WebRtc_UWord8* rtcpbuffer,
// add the list
int i = 0;
int numOfNackFields = 0;
NACKStringBuilder stringBuilder;
while (nackSize > i && numOfNackFields < kRtcpMaxNackFields)
{
WebRtc_UWord16 nack = nackList[i];
// put dow our sequence number
ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer+pos, nack);
pos += 2;
stringBuilder.PushNACK(nack);
i++;
numOfNackFields++;
@ -1372,6 +1422,7 @@ RTCPSender::BuildNACK(WebRtc_UWord8* rtcpbuffer,
assert(!(shift > 15) && !(shift < 0));
bitmask += (1<< shift);
stringBuilder.PushNACK(nackList[i]);
i++;
if(nackSize > i)
{
@ -1403,6 +1454,7 @@ RTCPSender::BuildNACK(WebRtc_UWord8* rtcpbuffer,
}
}
rtcpbuffer[nackSizePos]=(WebRtc_UWord8)(2+numOfNackFields);
*nackString = stringBuilder.GetResult();
return 0;
}
@ -1784,6 +1836,9 @@ RTCPSender::SendRTCP(const WebRtc_UWord32 packetTypeFlags,
{
break; // out of buffer
}
TRACE_EVENT_INSTANT1("webrtc_rtcp", "SendRTCP", "type", "pli");
_pliCount++;
TRACE_COUNTER1("webrtc_rtcp", "PLI Count", _pliCount);
}
if(rtcpPacketTypeFlags & kRtcpFir)
{
@ -1796,6 +1851,9 @@ RTCPSender::SendRTCP(const WebRtc_UWord32 packetTypeFlags,
{
break; // out of buffer
}
TRACE_EVENT_INSTANT1("webrtc_rtcp", "SendRTCP", "type", "fir");
_fullIntraRequestCount++;
TRACE_COUNTER1("webrtc_rtcp", "FIR Count", _fullIntraRequestCount);
}
if(rtcpPacketTypeFlags & kRtcpSli)
{
@ -1837,6 +1895,8 @@ RTCPSender::SendRTCP(const WebRtc_UWord32 packetTypeFlags,
{
break; // out of buffer
}
TRACE_EVENT_INSTANT2("webrtc_rtcp", "SendRTCP", "type", "remb",
"bitrate", _rembBitrate);
}
if(rtcpPacketTypeFlags & kRtcpBye)
{
@ -1888,7 +1948,9 @@ RTCPSender::SendRTCP(const WebRtc_UWord32 packetTypeFlags,
}
if(rtcpPacketTypeFlags & kRtcpNack)
{
buildVal = BuildNACK(rtcpbuffer, pos, nackSize, nackList);
std::string nackString;
buildVal = BuildNACK(rtcpbuffer, pos, nackSize, nackList,
&nackString);
if(buildVal == -1)
{
return -1; // error
@ -1897,6 +1959,10 @@ RTCPSender::SendRTCP(const WebRtc_UWord32 packetTypeFlags,
{
break; // out of buffer
}
TRACE_EVENT_INSTANT2("webrtc_rtcp", "SendRTCP", "type", "nack",
"list", TRACE_STR_COPY(nackString.c_str()));
_nackCount++;
TRACE_COUNTER1("webrtc_rtcp", "Nacks", _nackCount);
}
if(rtcpPacketTypeFlags & kRtcpXrVoipMetric)
{

24
webrtc/modules/rtp_rtcp/source/rtcp_sender.h
View File

@ -12,6 +12,8 @@
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
#include <map>
#include <sstream>
#include <string>
#include "typedefs.h"
#include "rtcp_utility.h"
@ -26,6 +28,20 @@ namespace webrtc {
class ModuleRtpRtcpImpl;
class NACKStringBuilder
{
public:
NACKStringBuilder();
void PushNACK(WebRtc_UWord16 nack);
std::string GetResult();
private:
std::ostringstream _stream;
int _count;
WebRtc_UWord16 _prevNack;
bool _consecutive;
};
class RTCPSender
{
public:
@ -181,7 +197,8 @@ private:
WebRtc_Word32 BuildNACK(WebRtc_UWord8* rtcpbuffer,
WebRtc_UWord32& pos,
const WebRtc_Word32 nackSize,
const WebRtc_UWord16* nackList);
const WebRtc_UWord16* nackList,
std::string* nackString);
private:
WebRtc_Word32 _id;
@ -249,6 +266,11 @@ private:
// XR VoIP metric
bool _xrSendVoIPMetric;
RTCPVoIPMetric _xrVoIPMetric;
// Counters
WebRtc_UWord32 _nackCount;
WebRtc_UWord32 _pliCount;
WebRtc_UWord32 _fullIntraRequestCount;
};
} // namespace webrtc

155
webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
View File

@ -26,6 +26,161 @@
namespace webrtc {
TEST(NACKStringBuilderTest, TestCase1) {
NACKStringBuilder builder;
builder.PushNACK(5);
builder.PushNACK(7);
builder.PushNACK(9);
builder.PushNACK(10);
builder.PushNACK(11);
builder.PushNACK(12);
builder.PushNACK(15);
builder.PushNACK(18);
builder.PushNACK(19);
EXPECT_EQ(std::string("5,7,9-12,15,18-19"), builder.GetResult());
}
TEST(NACKStringBuilderTest, TestCase2) {
NACKStringBuilder builder;
builder.PushNACK(5);
builder.PushNACK(6);
builder.PushNACK(7);
builder.PushNACK(9);
builder.PushNACK(10);
builder.PushNACK(11);
builder.PushNACK(12);
builder.PushNACK(15);
builder.PushNACK(18);
builder.PushNACK(19);
EXPECT_EQ(std::string("5-7,9-12,15,18-19"), builder.GetResult());
}
TEST(NACKStringBuilderTest, TestCase3) {
NACKStringBuilder builder;
builder.PushNACK(5);
builder.PushNACK(7);
builder.PushNACK(9);
builder.PushNACK(10);
builder.PushNACK(11);
builder.PushNACK(12);
builder.PushNACK(15);
builder.PushNACK(18);
builder.PushNACK(19);
builder.PushNACK(21);
EXPECT_EQ(std::string("5,7,9-12,15,18-19,21"), builder.GetResult());
}
TEST(NACKStringBuilderTest, TestCase4) {
NACKStringBuilder builder;
builder.PushNACK(5);
builder.PushNACK(7);
builder.PushNACK(8);
builder.PushNACK(9);
builder.PushNACK(10);
builder.PushNACK(11);
builder.PushNACK(12);
builder.PushNACK(15);
builder.PushNACK(18);
builder.PushNACK(19);
EXPECT_EQ(std::string("5,7-12,15,18-19"), builder.GetResult());
}
TEST(NACKStringBuilderTest, TestCase5) {
NACKStringBuilder builder;
builder.PushNACK(5);
builder.PushNACK(7);
builder.PushNACK(9);
builder.PushNACK(10);
builder.PushNACK(11);
builder.PushNACK(12);
builder.PushNACK(15);
builder.PushNACK(16);
builder.PushNACK(18);
builder.PushNACK(19);
EXPECT_EQ(std::string("5,7,9-12,15-16,18-19"), builder.GetResult());
}
TEST(NACKStringBuilderTest, TestCase6) {
NACKStringBuilder builder;
builder.PushNACK(5);
builder.PushNACK(7);
builder.PushNACK(9);
builder.PushNACK(10);
builder.PushNACK(11);
builder.PushNACK(12);
builder.PushNACK(15);
builder.PushNACK(16);
builder.PushNACK(17);
builder.PushNACK(18);
builder.PushNACK(19);
EXPECT_EQ(std::string("5,7,9-12,15-19"), builder.GetResult());
}
TEST(NACKStringBuilderTest, TestCase7) {
NACKStringBuilder builder;
builder.PushNACK(5);
builder.PushNACK(6);
builder.PushNACK(7);
builder.PushNACK(8);
builder.PushNACK(11);
builder.PushNACK(12);
builder.PushNACK(13);
builder.PushNACK(14);
builder.PushNACK(15);
EXPECT_EQ(std::string("5-8,11-15"), builder.GetResult());
}
TEST(NACKStringBuilderTest, TestCase8) {
NACKStringBuilder builder;
builder.PushNACK(5);
builder.PushNACK(7);
builder.PushNACK(9);
builder.PushNACK(11);
builder.PushNACK(15);
builder.PushNACK(17);
builder.PushNACK(19);
EXPECT_EQ(std::string("5,7,9,11,15,17,19"), builder.GetResult());
}
TEST(NACKStringBuilderTest, TestCase9) {
NACKStringBuilder builder;
builder.PushNACK(5);
builder.PushNACK(6);
builder.PushNACK(7);
builder.PushNACK(8);
builder.PushNACK(9);
builder.PushNACK(10);
builder.PushNACK(11);
builder.PushNACK(12);
EXPECT_EQ(std::string("5-12"), builder.GetResult());
}
TEST(NACKStringBuilderTest, TestCase10) {
NACKStringBuilder builder;
builder.PushNACK(5);
EXPECT_EQ(std::string("5"), builder.GetResult());
}
TEST(NACKStringBuilderTest, TestCase11) {
NACKStringBuilder builder;
EXPECT_EQ(std::string(""), builder.GetResult());
}
TEST(NACKStringBuilderTest, TestCase12) {
NACKStringBuilder builder;
builder.PushNACK(5);
builder.PushNACK(6);
EXPECT_EQ(std::string("5-6"), builder.GetResult());
}
TEST(NACKStringBuilderTest, TestCase13) {
NACKStringBuilder builder;
builder.PushNACK(5);
builder.PushNACK(6);
builder.PushNACK(9);
EXPECT_EQ(std::string("5-6,9"), builder.GetResult());
}
void CreateRtpPacket(const bool marker_bit, const WebRtc_UWord8 payload,
const WebRtc_UWord16 seq_num, const WebRtc_UWord32 timestamp,
const WebRtc_UWord32 ssrc, WebRtc_UWord8* array,

3
webrtc/modules/rtp_rtcp/source/rtp_receiver.cc
View File

@ -21,6 +21,7 @@
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/trace.h"
#include "webrtc/system_wrappers/interface/trace_event.h"
namespace webrtc {
@ -1140,6 +1141,8 @@ void RTPReceiver::ProcessBitrate() {
CriticalSectionScoped cs(critical_section_rtp_receiver_);
Bitrate::Process();
TRACE_COUNTER1("webrtc_rtp", "Received Bitrate", BitrateLast());
TRACE_COUNTER1("webrtc_rtp", "Received Packet Rate", PacketRate());
}
} // namespace webrtc

4
webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
View File

@ -17,6 +17,7 @@
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/trace.h"
#include "webrtc/system_wrappers/interface/trace_event.h"
namespace webrtc {
RTPReceiverAudio::RTPReceiverAudio(const WebRtc_Word32 id,
@ -190,6 +191,9 @@ WebRtc_Word32 RTPReceiverAudio::ParseRtpPacket(
const WebRtc_UWord16 packet_length,
const WebRtc_Word64 timestamp_ms,
const bool is_first_packet) {
TRACE_EVENT_INSTANT2("webrtc_rtp", "RTPReceiverAudio::ParseRtpPacket",
"seqnum", rtp_header->header.sequenceNumber,
"timestamp", rtp_header->header.timestamp);
const WebRtc_UWord8* payload_data =
ModuleRTPUtility::GetPayloadData(rtp_header, packet);

4
webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
View File

@ -22,6 +22,7 @@
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/trace.h"
#include "webrtc/system_wrappers/interface/trace_event.h"
namespace webrtc {
WebRtc_UWord32 BitRateBPS(WebRtc_UWord16 x) {
@ -74,6 +75,9 @@ WebRtc_Word32 RTPReceiverVideo::ParseRtpPacket(
const WebRtc_UWord16 packet_length,
const WebRtc_Word64 timestamp_ms,
const bool is_first_packet) {
TRACE_EVENT_INSTANT2("webrtc_rtp", "RTPReceiverVideo::ParseRtpPacket",
"seqnum", rtp_header->header.sequenceNumber,
"timestamp", rtp_header->header.timestamp);
const WebRtc_UWord8* payload_data =
ModuleRTPUtility::GetPayloadData(rtp_header, packet);
const WebRtc_UWord16 payload_data_length =

7
webrtc/modules/video_coding/main/source/generic_decoder.cc
View File

@ -10,6 +10,7 @@
#include "video_coding.h"
#include "trace.h"
#include "trace_event.h"
#include "generic_decoder.h"
#include "internal_defines.h"
#include "webrtc/system_wrappers/interface/clock.h"
@ -63,6 +64,9 @@ int32_t VCMDecodedFrameCallback::Decoded(I420VideoFrame& decodedImage)
_frame.SwapFrame(&decodedImage);
_frame.set_render_time_ms(frameInfo->renderTimeMs);
int32_t callbackReturn = _receiveCallback->FrameToRender(_frame);
TRACE_EVENT_INSTANT2("webrtc_vie", "VCMDecodedFrameCallback::Decoded",
"timestamp", decodedImage.timestamp(),
"render_time_ms", decodedImage.render_time_ms());
if (callbackReturn < 0)
{
WEBRTC_TRACE(webrtc::kTraceDebug,
@ -166,6 +170,9 @@ int32_t VCMGenericDecoder::Decode(const VCMEncodedFrame& frame,
_nextFrameInfoIdx = (_nextFrameInfoIdx + 1) % kDecoderFrameMemoryLength;
TRACE_EVENT2("webrtc_vie", "VCMGenericDecoder::Decode",
"timestamp", frame.TimeStamp(),
"render_time_ms", frame.RenderTimeMs());
int32_t ret = _decoder.Decode(frame.EncodedImage(),
frame.MissingFrame(),
frame.FragmentationHeader(),

68
webrtc/modules/video_coding/main/source/jitter_buffer.cc
View File

@ -24,6 +24,7 @@
#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/logging.h"
#include "webrtc/system_wrappers/interface/trace.h"
#include "webrtc/system_wrappers/interface/trace_event.h"
namespace webrtc {
@ -214,6 +215,7 @@ void VCMJitterBuffer::Stop() {
running_ = false;
last_decoded_state_.Reset();
frame_list_.clear();
TRACE_EVENT_INSTANT1("webrtc_vie", "FrameListEmptied", "type", "Stop");
for (int i = 0; i < kMaxNumberOfFrames; i++) {
if (frame_buffers_[i] != NULL) {
static_cast<VCMFrameBuffer*>(frame_buffers_[i])->SetState(kStateFree);
@ -238,6 +240,7 @@ void VCMJitterBuffer::Flush() {
CriticalSectionScoped cs(crit_sect_);
// Erase all frames from the sorted list and set their state to free.
frame_list_.clear();
TRACE_EVENT_INSTANT1("webrtc_vie", "FrameListEmptied", "type", "Flush");
for (int i = 0; i < max_number_of_frames_; i++) {
ReleaseFrameIfNotDecoding(frame_buffers_[i]);
}
@ -258,6 +261,7 @@ void VCMJitterBuffer::Flush() {
WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
VCMId(vcm_id_, receiver_id_), "JB(0x%x): Jitter buffer: flush",
this);
TRACE_EVENT_INSTANT0("webrtc_vie", "VCMJitterBuffer::Flush");
}
// Get received key and delta frames
@ -336,6 +340,8 @@ void VCMJitterBuffer::IncomingRateStatistics(unsigned int* framerate,
bitrate = 0;
incoming_bit_rate_ = 0;
}
TRACE_COUNTER1("webrtc_vie", "IncomingFrameRate", incoming_frame_rate_);
TRACE_COUNTER1("webrtc_vie", "IncomingBitRate", incoming_bit_rate_);
}
// Wait for the first packet in the next frame to arrive.
@ -488,6 +494,10 @@ VCMEncodedFrame* VCMJitterBuffer::GetCompleteFrameForDecoding(
VCMFrameBuffer* oldest_frame = *it;
it = frame_list_.erase(it);
if (frame_list_.empty()) {
TRACE_EVENT_INSTANT1("webrtc_vie", "FrameListEmptied",
"type", "GetCompleteFrameForDecoding");
}
// Update jitter estimate.
const bool retransmitted = (oldest_frame->GetNackCount() > 0);
@ -512,6 +522,10 @@ VCMEncodedFrame* VCMJitterBuffer::GetCompleteFrameForDecoding(
crit_sect_->Leave();
TRACE_EVENT_INSTANT2("webrtc_vie",
"VCMJitterBuffer::GetCompleteFrameForDecoding",
"timestamp", oldest_frame->TimeStamp(),
"render_time_ms", oldest_frame->RenderTimeMs());
return oldest_frame;
}
@ -557,6 +571,10 @@ VCMEncodedFrame* VCMJitterBuffer::GetFrameForDecoding() {
waiting_for_completion_.timestamp = oldest_frame->TimeStamp();
}
frame_list_.erase(frame_list_.begin());
if (frame_list_.empty()) {
TRACE_EVENT_INSTANT1("webrtc_vie", "FrameListEmptied",
"type", "GetFrameForDecoding");
}
// Look for previous frame loss
VerifyAndSetPreviousFrameLost(oldest_frame);
@ -579,6 +597,10 @@ VCMEncodedFrame* VCMJitterBuffer::GetFrameForDecoding() {
last_decoded_state_.SetState(oldest_frame);
DropPacketsFromNackList(last_decoded_state_.sequence_num());
TRACE_EVENT_INSTANT2("webrtc_vie",
"VCMJitterBuffer::GetFrameForDecoding",
"timestamp", oldest_frame->TimeStamp(),
"render_time_ms", oldest_frame->RenderTimeMs());
return oldest_frame;
}
@ -605,6 +627,10 @@ int VCMJitterBuffer::GetFrame(const VCMPacket& packet,
if (packet.sizeBytes > 0) {
num_discarded_packets_++;
num_consecutive_old_packets_++;
TRACE_EVENT_INSTANT2("webrtc_vie", "OldPacketDropped",
"seqnum", packet.seqNum,
"timestamp", packet.timestamp);
TRACE_COUNTER1("webrtc_vie", "DroppedOldPackets", num_discarded_packets_);
}
// Update last decoded sequence number if the packet arrived late and
// belongs to a frame with a timestamp equal to the last decoded
@ -999,6 +1025,10 @@ VCMEncodedFrame* VCMJitterBuffer::GetFrameForDecodingNACK() {
UpdateJitterEstimate(*oldest_frame, false);
}
it = frame_list_.erase(it);
if (frame_list_.empty()) {
TRACE_EVENT_INSTANT1("webrtc_vie", "FrameListEmptied",
"type", "GetFrameForDecodingNACK");
}
// Look for previous frame loss.
VerifyAndSetPreviousFrameLost(oldest_frame);
@ -1019,6 +1049,10 @@ VCMEncodedFrame* VCMJitterBuffer::GetFrameForDecodingNACK() {
last_decoded_state_.SetState(oldest_frame);
DropPacketsFromNackList(last_decoded_state_.sequence_num());
TRACE_EVENT_INSTANT2("webrtc_vie",
"VCMJitterBuffer::GetFrameForDecodingNACK",
"timestamp", oldest_frame->TimeStamp(),
"render_time_ms", oldest_frame->RenderTimeMs());
return oldest_frame;
}
@ -1035,6 +1069,8 @@ VCMFrameBuffer* VCMJitterBuffer::GetEmptyFrame() {
return NULL;
}
TRACE_EVENT_INSTANT0("webrtc_vie", "VCMJitterBuffer::GetEmptyFrame");
crit_sect_->Enter();
for (int i = 0; i < max_number_of_frames_; ++i) {
@ -1058,6 +1094,8 @@ VCMFrameBuffer* VCMJitterBuffer::GetEmptyFrame() {
VCMId(vcm_id_, receiver_id_),
"JB(0x%x) FB(0x%x): Jitter buffer increased to:%d frames",
this, ptr_new_buffer, max_number_of_frames_);
TRACE_EVENT_INSTANT1("webrtc_vie", "JitterBufferIncreased",
"NewSize", max_number_of_frames_);
return ptr_new_buffer;
}
crit_sect_->Leave();
@ -1078,6 +1116,8 @@ bool VCMJitterBuffer::RecycleFramesUntilKeyFrame() {
VCMId(vcm_id_, receiver_id_),
"Jitter buffer drop count:%d, low_seq %d", drop_count_,
(*it)->GetLowSeqNum());
TRACE_EVENT_INSTANT0("webrtc_vie",
"VCMJitterBuffer::RecycleFramesUntilKeyFrame");
ReleaseFrameIfNotDecoding(*it);
it = frame_list_.erase(it);
if (it != frame_list_.end() && (*it)->FrameType() == kVideoFrameKey) {
@ -1087,6 +1127,10 @@ bool VCMJitterBuffer::RecycleFramesUntilKeyFrame() {
return true;
}
}
if (frame_list_.empty()) {
TRACE_EVENT_INSTANT1("webrtc_vie", "FrameListEmptied",
"type", "RecycleFramesUntilKeyFrame");
}
waiting_for_key_frame_ = true;
last_decoded_state_.Reset(); // TODO(mikhal): No sync.
missing_sequence_numbers_.clear();
@ -1116,13 +1160,30 @@ VCMFrameBufferEnum VCMJitterBuffer::UpdateFrameState(VCMFrameBuffer* frame) {
if (length != 0 && !frame->GetCountedFrame()) {
// Ignore ACK frames.
incoming_frame_count_++;
TRACE_EVENT_INSTANT1("webrtc_vie", "AddFrameToJitterBuffer",
"timestamp", frame->TimeStamp());
if (frame->FrameType() == kVideoFrameKey) {
TRACE_EVENT_INSTANT1("webrtc_vie", "AddKeyFrameToJitterBuffer",
"timestamp", frame->TimeStamp());
}
frame->SetCountedFrame(true);
} else {
TRACE_EVENT_INSTANT1("webrtc_vie",
"AddRetransmittedFrameToJitterBuffer",
"timestamp", frame->TimeStamp());
if (frame->FrameType() == kVideoFrameKey) {
TRACE_EVENT_INSTANT1("webrtc_vie",
"AddRetransmittedKeyFrameToJitterBuffer",
"timestamp", frame->TimeStamp());
}
}
// Check if we should drop the frame. A complete frame can arrive too late.
if (last_decoded_state_.IsOldFrame(frame)) {
// Frame is older than the latest decoded frame, drop it. Will be
// released by CleanUpOldFrames later.
TRACE_EVENT_INSTANT1("webrtc_vie", "DropLateFrame",
"timestamp", frame->TimeStamp());
frame->Reset();
frame->SetState(kStateEmpty);
WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
@ -1236,11 +1297,18 @@ void VCMJitterBuffer::CleanUpOldOrEmptyFrames() {
}
if (last_decoded_state_.IsOldFrame(oldest_frame)) {
ReleaseFrameIfNotDecoding(frame_list_.front());
TRACE_EVENT_INSTANT1("webrtc_vie", "OldFrameDropped",
"timestamp", oldest_frame->TimeStamp());
TRACE_COUNTER1("webrtc_vie", "DroppedLateFrames", drop_count_);
frame_list_.erase(frame_list_.begin());
} else {
break;
}
}
if (frame_list_.empty()) {
TRACE_EVENT_INSTANT1("webrtc_vie", "FrameListEmptied",
"type", "CleanUpOldOrEmptyFrames");
}
if (!last_decoded_state_.in_initial_state()) {
DropPacketsFromNackList(last_decoded_state_.sequence_num());
}

9
webrtc/video_engine/vie_sync_module.cc
View File

@ -14,6 +14,7 @@
#include "modules/video_coding/main/interface/video_coding.h"
#include "system_wrappers/interface/critical_section_wrapper.h"
#include "system_wrappers/interface/trace.h"
#include "system_wrappers/interface/trace_event.h"
#include "video_engine/stream_synchronization.h"
#include "video_engine/vie_channel.h"
#include "voice_engine/include/voe_video_sync.h"
@ -152,6 +153,10 @@ WebRtc_Word32 ViESyncModule::Process() {
return 0;
}
TRACE_COUNTER1("webrtc_sync", "CurrentVideoDelay",
total_video_delay_target_ms);
TRACE_COUNTER1("webrtc_sync", "CurrentAudioDelay", current_audio_delay_ms);
TRACE_COUNTER1("webrtc_sync", "RelativeDelay", relative_delay_ms);
int extra_audio_delay_ms = 0;
// Calculate the necessary extra audio delay and desired total video
// delay to get the streams in sync.
@ -161,6 +166,10 @@ WebRtc_Word32 ViESyncModule::Process() {
&total_video_delay_target_ms)) {
return 0;
}
TRACE_COUNTER1("webrtc_sync", "ExtraAudioDelayTarget", extra_audio_delay_ms);
TRACE_COUNTER1("webrtc_sync", "TotalVideoDelayTarget",
total_video_delay_target_ms);
if (voe_sync_interface_->SetMinimumPlayoutDelay(
voe_channel_id_, extra_audio_delay_ms) == -1) {
WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideo, vie_channel_->Id(),