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Refactor to use std::list in the video rtp play tools.

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BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/349013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1504 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org 2012-01-23 14:56:14 +00:00
parent 152c34cf11
commit 8fe03af674
11 changed files with 216 additions and 272 deletions
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4
src/modules/video_coding/main/source/media_opt_util.h
View File

@ -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
@ -23,8 +23,6 @@
namespace webrtc
{
class ListWrapper;
// Number of time periods used for (max) window filter for packet loss
// TODO (marpan): set reasonable window size for filtered packet loss,
// adjustment should be based on logged/real data of loss stats/correlation.

3
src/modules/video_coding/main/source/media_optimization.h
View File

@ -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
@ -11,7 +11,6 @@
#ifndef WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPTIMIZATION_H_
#define WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPTIMIZATION_H_
#include "list_wrapper.h"
#include "module_common_types.h"
#include "video_coding.h"
#include "trace.h"

30
src/modules/video_coding/main/test/decode_from_storage_test.cc
View File

@ -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
@ -83,14 +83,14 @@ int DecodeFromStorageTest(CmdArgs& args)
vcm->RegisterFrameStorageCallback(&storageCallback);
vcmPlayback->RegisterReceiveCallback(&receiveCallback);
RTPPlayer rtpStream(rtpFilename.c_str(), &dataCallback, &clock);
ListWrapper payloadTypes;
payloadTypes.PushFront(new PayloadCodecTuple(VCM_VP8_PAYLOAD_TYPE, "VP8", kVideoCodecVP8));
PayloadTypeList payloadTypes;
payloadTypes.push_front(new PayloadCodecTuple(VCM_VP8_PAYLOAD_TYPE, "VP8",
kVideoCodecVP8));
// Register receive codecs in VCM
ListItem* item = payloadTypes.First();
while (item != NULL)
{
PayloadCodecTuple* payloadType = static_cast<PayloadCodecTuple*>(item->GetItem());
for (PayloadTypeList::iterator it = payloadTypes.begin();
it != payloadTypes.end(); ++it) {
PayloadCodecTuple* payloadType = *it;
if (payloadType != NULL)
{
VideoCodec codec;
@ -108,9 +108,8 @@ int DecodeFromStorageTest(CmdArgs& args)
return -1;
}
}
item = payloadTypes.Next(item);
}
if (rtpStream.Initialize(payloadTypes) < 0)
if (rtpStream.Initialize(&payloadTypes) < 0)
{
return -1;
}
@ -163,17 +162,10 @@ int DecodeFromStorageTest(CmdArgs& args)
rtpStream.Print();
// Tear down
item = payloadTypes.First();
while (item != NULL)
while (!payloadTypes.empty())
{
PayloadCodecTuple* payloadType = static_cast<PayloadCodecTuple*>(item->GetItem());
if (payloadType != NULL)
{
delete payloadType;
}
ListItem* itemToRemove = item;
item = payloadTypes.Next(item);
payloadTypes.Erase(itemToRemove);
delete payloadTypes.front();
payloadTypes.pop_front();
}
VideoCodingModule::Destroy(vcm);
vcm = NULL;

14
src/modules/video_coding/main/test/mt_test_common.cc
View File

@ -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
@ -56,7 +56,7 @@ TransportCallback::SendPacket(int channel, const void *data, int len)
// Insert outgoing packet into list
if (transmitPacket)
{
rtpPacket* newPacket = new rtpPacket();
RtpPacket* newPacket = new RtpPacket();
memcpy(newPacket->data, data, len);
newPacket->length = len;
// Simulate receive time = network delay + packet jitter
@ -66,7 +66,7 @@ TransportCallback::SendPacket(int channel, const void *data, int len)
simulatedDelay = (WebRtc_Word32)NormalDist(_networkDelayMs,
sqrt(_jitterVar));
newPacket->receiveTime = now + simulatedDelay;
_rtpPackets.PushBack(newPacket);
_rtpPackets.push_back(newPacket);
}
return 0;
}
@ -75,14 +75,14 @@ int
TransportCallback::TransportPackets()
{
// Are we ready to send packets to the receiver?
rtpPacket* packet = NULL;
RtpPacket* packet = NULL;
TickTimeBase clock;
int64_t now = clock.MillisecondTimestamp();
while (!_rtpPackets.Empty())
while (!_rtpPackets.empty())
{
// Take first packet in list
packet = static_cast<rtpPacket*>((_rtpPackets.First())->GetItem());
packet = _rtpPackets.front();
WebRtc_Word64 timeToReceive = packet->receiveTime - now;
if (timeToReceive > 0)
{
@ -90,7 +90,7 @@ TransportCallback::TransportPackets()
break;
}
_rtpPackets.PopFront();
_rtpPackets.pop_front();
// Send to receive side
if (_rtp->IncomingPacket((const WebRtc_UWord8*)packet->data,
packet->length) < 0)

262
src/modules/video_coding/main/test/rtp_player.cc
View File

@ -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
@ -25,113 +25,114 @@
using namespace webrtc;
RawRtpPacket::RawRtpPacket(WebRtc_UWord8* data, WebRtc_UWord16 len)
:
rtpData(), rtpLen(len), resendTimeMs(-1)
{
rtpData = new WebRtc_UWord8[rtpLen];
memcpy(rtpData, data, rtpLen);
RawRtpPacket::RawRtpPacket(uint8_t* rtp_data, uint16_t rtp_length)
: data(rtp_data),
length(rtp_length),
resend_time_ms(-1) {
data = new uint8_t[length];
memcpy(data, rtp_data, length);
}
RawRtpPacket::~RawRtpPacket()
{
delete [] rtpData;
RawRtpPacket::~RawRtpPacket() {
delete [] data;
}
LostPackets::LostPackets()
:
_critSect(CriticalSectionWrapper::CreateCriticalSection()),
_lossCount(0),
_debugFile(NULL)
{
_debugFile = fopen("PacketLossDebug.txt", "w");
: crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
loss_count_(0),
debug_file_(NULL),
packets_() {
debug_file_ = fopen("PacketLossDebug.txt", "w");
}
LostPackets::~LostPackets()
{
if (_debugFile)
{
fclose(_debugFile);
}
ListItem* item = First();
while (item != NULL)
{
RawRtpPacket* packet = static_cast<RawRtpPacket*>(item->GetItem());
if (packet != NULL)
{
delete packet;
LostPackets::~LostPackets() {
if (debug_file_) {
fclose(debug_file_);
}
while (!packets_.empty()) {
delete packets_.front();
packets_.pop_front();
}
delete crit_sect_;
}
void LostPackets::AddPacket(RawRtpPacket* packet) {
CriticalSectionScoped cs(crit_sect_);
packets_.push_back(packet);
uint16_t seq_num = (packet->data[2] << 8) + packet->data[3];
if (debug_file_ != NULL) {
fprintf(debug_file_, "%u Lost packet: %u\n", loss_count_, seq_num);
}
++loss_count_;
}
void LostPackets::SetResendTime(uint16_t resend_seq_num,
int64_t resend_time_ms,
int64_t now_ms) {
CriticalSectionScoped cs(crit_sect_);
for (RtpPacketIterator it = packets_.begin(); it != packets_.end(); ++it) {
const uint16_t seq_num = ((*it)->data[2] << 8) +
(*it)->data[3];
if (resend_seq_num == seq_num) {
if ((*it)->resend_time_ms + 10 < now_ms) {
if (debug_file_ != NULL) {
fprintf(debug_file_, "Resend %u at %u\n", seq_num,
MaskWord64ToUWord32(resend_time_ms));
}
Erase(item);
item = First();
(*it)->resend_time_ms = resend_time_ms;
}
return;
}
delete _critSect;
}
assert(false);
}
WebRtc_UWord32 LostPackets::AddPacket(WebRtc_UWord8* rtpData, WebRtc_UWord16 rtpLen)
{
CriticalSectionScoped cs(_critSect);
RawRtpPacket* packet = new RawRtpPacket(rtpData, rtpLen);
ListItem* newItem = new ListItem(packet);
Insert(Last(), newItem);
const WebRtc_UWord16 seqNo = (rtpData[2] << 8) + rtpData[3];
if (_debugFile != NULL)
{
fprintf(_debugFile, "%u Lost packet: %u\n", _lossCount, seqNo);
RawRtpPacket* LostPackets::NextPacketToResend(int64_t timeNow) {
CriticalSectionScoped cs(crit_sect_);
for (RtpPacketIterator it = packets_.begin(); it != packets_.end(); ++it) {
if (timeNow >= (*it)->resend_time_ms && (*it)->resend_time_ms != -1) {
RawRtpPacket* packet = *it;
it = packets_.erase(it);
return packet;
}
_lossCount++;
return 0;
}
return NULL;
}
WebRtc_UWord32 LostPackets::SetResendTime(WebRtc_UWord16 sequenceNumber,
WebRtc_Word64 resendTime,
WebRtc_Word64 nowMs)
{
CriticalSectionScoped cs(_critSect);
ListItem* item = First();
while (item != NULL)
{
RawRtpPacket* packet = static_cast<RawRtpPacket*>(item->GetItem());
const WebRtc_UWord16 seqNo = (packet->rtpData[2] << 8) + packet->rtpData[3];
if (sequenceNumber == seqNo && packet->resendTimeMs + 10 < nowMs)
{
if (_debugFile != NULL)
{
fprintf(_debugFile, "Resend %u at %u\n", seqNo, MaskWord64ToUWord32(resendTime));
}
packet->resendTimeMs = resendTime;
return 0;
}
item = Next(item);
int LostPackets::NumberOfPacketsToResend() const {
CriticalSectionScoped cs(crit_sect_);
int count = 0;
for (ConstRtpPacketIterator it = packets_.begin(); it != packets_.end();
++it) {
if ((*it)->resend_time_ms >= 0) {
count++;
}
fprintf(_debugFile, "Packet not lost %u\n", sequenceNumber);
return -1;
}
return count;
}
WebRtc_UWord32 LostPackets::NumberOfPacketsToResend() const
{
CriticalSectionScoped cs(_critSect);
WebRtc_UWord32 count = 0;
ListItem* item = First();
while (item != NULL)
{
RawRtpPacket* packet = static_cast<RawRtpPacket*>(item->GetItem());
if (packet->resendTimeMs >= 0)
{
count++;
}
item = Next(item);
}
return count;
void LostPackets::SetPacketResent(uint16_t seq_num, int64_t now_ms) {
CriticalSectionScoped cs(crit_sect_);
if (debug_file_ != NULL) {
fprintf(debug_file_, "Resent %u at %u\n", seq_num,
MaskWord64ToUWord32(now_ms));
}
}
void LostPackets::ResentPacket(WebRtc_UWord16 seqNo, WebRtc_Word64 nowMs)
{
CriticalSectionScoped cs(_critSect);
if (_debugFile != NULL)
{
fprintf(_debugFile, "Resent %u at %u\n", seqNo,
MaskWord64ToUWord32(nowMs));
}
void LostPackets::Print() const {
CriticalSectionScoped cs(crit_sect_);
printf("Lost packets: %u\n", loss_count_);
printf("Packets waiting to be resent: %u\n",
NumberOfPacketsToResend());
printf("Packets still lost: %u\n",
static_cast<unsigned int>(packets_.size()));
printf("Sequence numbers:\n");
for (ConstRtpPacketIterator it = packets_.begin(); it != packets_.end();
++it) {
uint16_t seq_num = ((*it)->data[2] << 8) + (*it)->data[3];
printf("%u, ", seq_num);
}
printf("\n");
}
RTPPlayer::RTPPlayer(const char* filename,
@ -176,7 +177,7 @@ RTPPlayer::~RTPPlayer()
}
}
WebRtc_Word32 RTPPlayer::Initialize(const ListWrapper& payloadList)
WebRtc_Word32 RTPPlayer::Initialize(const PayloadTypeList* payloadList)
{
std::srand(321);
for (int i=0; i < RAND_VEC_LENGTH; i++)
@ -205,10 +206,9 @@ WebRtc_Word32 RTPPlayer::Initialize(const ListWrapper& payloadList)
return -1;
}
// Register payload types
ListItem* item = payloadList.First();
while (item != NULL)
{
PayloadCodecTuple* payloadType = static_cast<PayloadCodecTuple*>(item->GetItem());
for (PayloadTypeList::const_iterator it = payloadList->begin();
it != payloadList->end(); ++it) {
PayloadCodecTuple* payloadType = *it;
if (payloadType != NULL)
{
VideoCodec videoCodec;
@ -219,7 +219,6 @@ WebRtc_Word32 RTPPlayer::Initialize(const ListWrapper& payloadList)
return -1;
}
}
item = payloadList.Next(item);
}
if (ReadHeader() < 0)
{
@ -286,43 +285,21 @@ WebRtc_UWord32 RTPPlayer::TimeUntilNextPacket() const
WebRtc_Word32 RTPPlayer::NextPacket(const WebRtc_Word64 timeNow)
{
// Send any packets ready to be resent
_lostPackets.Lock();
ListItem* item = _lostPackets.First();
_lostPackets.Unlock();
while (item != NULL)
{
_lostPackets.Lock();
RawRtpPacket* packet = static_cast<RawRtpPacket*>(item->GetItem());
_lostPackets.Unlock();
if (timeNow >= packet->resendTimeMs && packet->resendTimeMs != -1)
{
const WebRtc_UWord16 seqNo = (packet->rtpData[2] << 8) + packet->rtpData[3];
printf("Resend: %u\n", seqNo);
WebRtc_Word32 ret = SendPacket(packet->rtpData, packet->rtpLen);
ListItem* itemToRemove = item;
_lostPackets.Lock();
item = _lostPackets.Next(item);
_lostPackets.Erase(itemToRemove);
delete packet;
_lostPackets.Unlock();
_resendPacketCount++;
if (ret > 0)
{
_lostPackets.ResentPacket(seqNo,
_clock->MillisecondTimestamp());
}
else if (ret < 0)
{
return ret;
}
}
else
{
_lostPackets.Lock();
item = _lostPackets.Next(item);
_lostPackets.Unlock();
}
// Send any packets ready to be resent,
RawRtpPacket* resend_packet = _lostPackets.NextPacketToResend(timeNow);
while (resend_packet != NULL) {
const uint16_t seqNo = (resend_packet->data[2] << 8) +
resend_packet->data[3];
printf("Resend: %u\n", seqNo);
int ret = SendPacket(resend_packet->data, resend_packet->length);
delete resend_packet;
_resendPacketCount++;
if (ret > 0) {
_lostPackets.SetPacketResent(seqNo, _clock->MillisecondTimestamp());
} else if (ret < 0) {
return ret;
}
resend_packet = _lostPackets.NextPacketToResend(timeNow);
}
// Send any packets from rtp file
@ -344,7 +321,7 @@ WebRtc_Word32 RTPPlayer::NextPacket(const WebRtc_Word64 timeNow)
{
RawRtpPacket* rtpPacket = _reorderBuffer;
_reorderBuffer = NULL;
SendPacket(rtpPacket->rtpData, rtpPacket->rtpLen);
SendPacket(rtpPacket->data, rtpPacket->length);
delete rtpPacket;
}
_firstPacket = false;
@ -379,11 +356,11 @@ WebRtc_Word32 RTPPlayer::SendPacket(WebRtc_UWord8* rtpData, WebRtc_UWord16 rtpLe
{
const WebRtc_UWord16 seqNo = (rtpData[2] << 8) + rtpData[3];
printf("Throw: %u\n", seqNo);
_lostPackets.AddPacket(rtpData, rtpLen);
_lostPackets.AddPacket(new RawRtpPacket(rtpData, rtpLen));
return 0;
}
}
else
else if (rtpLen > 0)
{
WebRtc_Word32 ret = _rtpModule.IncomingPacket(rtpData, rtpLen);
if (ret < 0)
@ -461,17 +438,6 @@ WebRtc_Word32 RTPPlayer::ResendPackets(const WebRtc_UWord16* sequenceNumbers, We
void RTPPlayer::Print() const
{
printf("Lost packets: %u, resent packets: %u\n", _lostPackets.TotalNumberOfLosses(), _resendPacketCount);
printf("Packets still lost: %u\n", _lostPackets.GetSize());
printf("Packets waiting to be resent: %u\n", _lostPackets.NumberOfPacketsToResend());
printf("Sequence numbers:\n");
ListItem* item = _lostPackets.First();
while (item != NULL)
{
RawRtpPacket* packet = static_cast<RawRtpPacket*>(item->GetItem());
const WebRtc_UWord16 seqNo = (packet->rtpData[2] << 8) + packet->rtpData[3];
printf("%u, ", seqNo);
item = _lostPackets.Next(item);
}
printf("\n");
printf("Resent packets: %u\n", _resendPacketCount);
_lostPackets.Print();
}

57
src/modules/video_coding/main/test/rtp_player.h
View File

@ -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
@ -13,48 +13,55 @@
#include "typedefs.h"
#include "rtp_rtcp.h"
#include "list_wrapper.h"
#include "critical_section_wrapper.h"
#include "video_coding_defines.h"
#include "modules/video_coding/main/source/tick_time_base.h"
#include <stdio.h>
#include <list>
#include <string>
#define HDR_SIZE 8 // rtpplay packet header size in bytes
#define FIRSTLINELEN 40
#define RAND_VEC_LENGTH 4096
class PayloadCodecTuple;
struct RawRtpPacket
{
public:
RawRtpPacket(WebRtc_UWord8* data, WebRtc_UWord16 len);
RawRtpPacket(WebRtc_UWord8* rtp_data, WebRtc_UWord16 rtp_length);
~RawRtpPacket();
WebRtc_UWord8* rtpData;
WebRtc_UWord16 rtpLen;
WebRtc_Word64 resendTimeMs;
uint8_t* data;
uint16_t length;
int64_t resend_time_ms;
};
class LostPackets : public webrtc::ListWrapper
{
public:
LostPackets();
~LostPackets();
typedef std::list<PayloadCodecTuple*> PayloadTypeList;
typedef std::list<RawRtpPacket*> RtpPacketList;
typedef RtpPacketList::iterator RtpPacketIterator;
typedef RtpPacketList::const_iterator ConstRtpPacketIterator;
WebRtc_UWord32 AddPacket(WebRtc_UWord8* rtpData, WebRtc_UWord16 rtpLen);
WebRtc_UWord32 SetResendTime(WebRtc_UWord16 sequenceNumber,
WebRtc_Word64 resendTime,
WebRtc_Word64 nowMs);
WebRtc_UWord32 TotalNumberOfLosses() const { return _lossCount; };
WebRtc_UWord32 NumberOfPacketsToResend() const;
void ResentPacket(WebRtc_UWord16 seqNo, WebRtc_Word64 nowMs);
void Lock() {_critSect->Enter();};
void Unlock() {_critSect->Leave();};
private:
webrtc::CriticalSectionWrapper* _critSect;
WebRtc_UWord32 _lossCount;
FILE* _debugFile;
class LostPackets {
public:
LostPackets();
~LostPackets();
void AddPacket(RawRtpPacket* packet);
void SetResendTime(uint16_t sequenceNumber,
int64_t resendTime,
int64_t nowMs);
RawRtpPacket* NextPacketToResend(int64_t timeNow);
int NumberOfPacketsToResend() const;
void SetPacketResent(uint16_t seqNo, int64_t nowMs);
void Print() const;
private:
webrtc::CriticalSectionWrapper* crit_sect_;
int loss_count_;
FILE* debug_file_;
RtpPacketList packets_;
};
struct PayloadCodecTuple
@ -74,7 +81,7 @@ public:
webrtc::TickTimeBase* clock);
virtual ~RTPPlayer();
WebRtc_Word32 Initialize(const webrtc::ListWrapper& payloadList);
WebRtc_Word32 Initialize(const PayloadTypeList* payloadList);
WebRtc_Word32 NextPacket(const WebRtc_Word64 timeNow);
WebRtc_UWord32 TimeUntilNextPacket() const;
WebRtc_Word32 SimulatePacketLoss(float lossRate, bool enableNack = false, WebRtc_UWord32 rttMs = 0);

22
src/modules/video_coding/main/test/test_callbacks.cc
View File

@ -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
@ -225,11 +225,11 @@ RTPSendCompleteCallback::~RTPSendCompleteCallback()
RtpDump::DestroyRtpDump(_rtpDump);
}
// Delete remaining packets
while (!_rtpPackets.Empty())
while (!_rtpPackets.empty())
{
// Take first packet in list
delete static_cast<rtpPacket*>((_rtpPackets.First())->GetItem());
_rtpPackets.PopFront();
// Take first packet in list
delete _rtpPackets.front();
_rtpPackets.pop_front();
}
}
@ -254,7 +254,7 @@ RTPSendCompleteCallback::SendPacket(int channel, const void *data, int len)
// Insert outgoing packet into list
if (transmitPacket)
{
rtpPacket* newPacket = new rtpPacket();
RtpPacket* newPacket = new RtpPacket();
memcpy(newPacket->data, data, len);
newPacket->length = len;
// Simulate receive time = network delay + packet jitter
@ -264,16 +264,16 @@ RTPSendCompleteCallback::SendPacket(int channel, const void *data, int len)
simulatedDelay = (WebRtc_Word32)NormalDist(_networkDelayMs,
sqrt(_jitterVar));
newPacket->receiveTime = now + simulatedDelay;
_rtpPackets.PushBack(newPacket);
_rtpPackets.push_back(newPacket);
}
// Are we ready to send packets to the receiver?
rtpPacket* packet = NULL;
RtpPacket* packet = NULL;
while (!_rtpPackets.Empty())
while (!_rtpPackets.empty())
{
// Take first packet in list
packet = static_cast<rtpPacket*>((_rtpPackets.First())->GetItem());
packet = _rtpPackets.front();
WebRtc_Word64 timeToReceive = packet->receiveTime - now;
if (timeToReceive > 0)
{
@ -281,7 +281,7 @@ RTPSendCompleteCallback::SendPacket(int channel, const void *data, int len)
break;
}
_rtpPackets.PopFront();
_rtpPackets.pop_front();
// Send to receive side
if (_rtp->IncomingPacket((const WebRtc_UWord8*)packet->data,
packet->length) < 0)

11
src/modules/video_coding/main/test/test_callbacks.h
View File

@ -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
@ -16,11 +16,11 @@
*/
#include <string.h>
#include <cstdlib>
#include <fstream>
#include <string.h>
#include <list>
#include "list_wrapper.h"
#include "module_common_types.h"
#include "rtp_rtcp.h"
#include "test_util.h"
@ -29,7 +29,7 @@
namespace webrtc
{
class RtpDump;
class RtpDump;
// Send Side - Packetization callback - send an encoded frame to the VCMReceiver
class VCMEncodeCompleteCallback: public VCMPacketizationCallback
@ -147,7 +147,6 @@ private:
WebRtc_UWord32 _decodedBytes;
}; // end of VCMDecodeCompleCallback class
// Transport callback
// Called by the RTP Sender - simulates sending packets through a network to the
// RTP receiver. User can set network conditions as: RTT, packet loss,
@ -192,7 +191,7 @@ protected:
double _jitterVar;
bool _prevLossState;
WebRtc_UWord32 _totalSentLength;
ListWrapper _rtpPackets;
std::list<RtpPacket*> _rtpPackets;
RtpDump* _rtpDump;
};

13
src/modules/video_coding/main/test/test_util.h
View File

@ -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
@ -61,12 +61,11 @@ class CmdArgs
int MTRxTxTest(CmdArgs& args);
double NormalDist(double mean, double stdDev);
typedef struct
{
WebRtc_Word8 data[1650]; // max packet size
WebRtc_Word32 length;
WebRtc_Word64 receiveTime;
} rtpPacket;
struct RtpPacket {
WebRtc_Word8 data[1650]; // max packet size
WebRtc_Word32 length;
WebRtc_Word64 receiveTime;
};
// Codec type conversion

40
src/modules/video_coding/main/test/video_rtp_play.cc
View File

@ -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
@ -80,7 +80,7 @@ int RtpPlay(CmdArgs& args)
bool protectionEnabled = false;
VCMVideoProtection protectionMethod = kProtectionNack;
WebRtc_UWord32 rttMS = 10;
WebRtc_UWord32 rttMS = 0;
float lossRate = 0.0f;
bool reordering = false;
WebRtc_UWord32 renderDelayMs = 0;
@ -96,8 +96,9 @@ int RtpPlay(CmdArgs& args)
RTPPlayer rtpStream(args.inputFile.c_str(), &dataCallback, &clock);
ListWrapper payloadTypes;
payloadTypes.PushFront(new PayloadCodecTuple(VCM_VP8_PAYLOAD_TYPE, "VP8", kVideoCodecVP8));
PayloadTypeList payloadTypes;
payloadTypes.push_front(new PayloadCodecTuple(VCM_VP8_PAYLOAD_TYPE, "VP8",
kVideoCodecVP8));
Trace::CreateTrace();
Trace::SetTraceFile((test::OutputPath() + "receiverTestTrace.txt").c_str());
@ -115,10 +116,9 @@ int RtpPlay(CmdArgs& args)
vcm->RegisterPacketRequestCallback(&rtpStream);
// Register receive codecs in VCM
ListItem* item = payloadTypes.First();
while (item != NULL)
{
PayloadCodecTuple* payloadType = static_cast<PayloadCodecTuple*>(item->GetItem());
for (PayloadTypeList::iterator it = payloadTypes.begin();
it != payloadTypes.end(); ++it) {
PayloadCodecTuple* payloadType = *it;
if (payloadType != NULL)
{
VideoCodec codec;
@ -132,15 +132,15 @@ int RtpPlay(CmdArgs& args)
return -1;
}
}
item = payloadTypes.Next(item);
}
if (rtpStream.Initialize(payloadTypes) < 0)
if (rtpStream.Initialize(&payloadTypes) < 0)
{
return -1;
}
bool nackEnabled = protectionEnabled && (protectionMethod == kProtectionNack ||
protectionMethod == kProtectionDualDecoder);
bool nackEnabled = protectionEnabled &&
(protectionMethod == kProtectionNack ||
protectionMethod == kProtectionDualDecoder);
rtpStream.SimulatePacketLoss(lossRate, nackEnabled, rttMS);
rtpStream.SetReordering(reordering);
vcm->SetChannelParameters(0, 0, rttMS);
@ -166,7 +166,8 @@ int RtpPlay(CmdArgs& args)
{
vcm->Process();
}
if (MAX_RUNTIME_MS > -1 && clock.MillisecondTimestamp() >= MAX_RUNTIME_MS)
if (MAX_RUNTIME_MS > -1 && clock.MillisecondTimestamp() >=
MAX_RUNTIME_MS)
{
break;
}
@ -189,17 +190,10 @@ int RtpPlay(CmdArgs& args)
rtpStream.Print();
// Tear down
item = payloadTypes.First();
while (item != NULL)
while (!payloadTypes.empty())
{
PayloadCodecTuple* payloadType = static_cast<PayloadCodecTuple*>(item->GetItem());
if (payloadType != NULL)
{
delete payloadType;
}
ListItem* itemToRemove = item;
item = payloadTypes.Next(item);
payloadTypes.Erase(itemToRemove);
delete payloadTypes.front();
payloadTypes.pop_front();
}
delete vcm;
vcm = NULL;

32
src/modules/video_coding/main/test/video_rtp_play_mt.cc
View File

@ -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
@ -137,10 +137,9 @@ int RtpPlayMT(CmdArgs& args, int releaseTestNo, webrtc::VideoCodecType releaseTe
printf("Watch %s to verify that the output is reasonable\n", outFilename.c_str());
}
RTPPlayer rtpStream(rtpFilename.c_str(), &dataCallback, &clock);
ListWrapper payloadTypes;
payloadTypes.PushFront(new PayloadCodecTuple(VCM_VP8_PAYLOAD_TYPE,
"VP8", kVideoCodecVP8));
PayloadTypeList payloadTypes;
payloadTypes.push_front(new PayloadCodecTuple(VCM_VP8_PAYLOAD_TYPE, "VP8",
kVideoCodecVP8));
Trace::CreateTrace();
Trace::SetTraceFile("receiverTestTrace.txt");
Trace::SetLevelFilter(webrtc::kTraceAll);
@ -151,7 +150,7 @@ int RtpPlayMT(CmdArgs& args, int releaseTestNo, webrtc::VideoCodecType releaseTe
SharedState mtState(*vcm, rtpStream);
if (rtpStream.Initialize(payloadTypes) < 0)
if (rtpStream.Initialize(&payloadTypes) < 0)
{
return -1;
}
@ -172,10 +171,9 @@ int RtpPlayMT(CmdArgs& args, int releaseTestNo, webrtc::VideoCodecType releaseTe
&mtState, kNormalPriority, "DecodeThread");
// Register receive codecs in VCM
ListItem* item = payloadTypes.First();
while (item != NULL)
{
PayloadCodecTuple* payloadType = static_cast<PayloadCodecTuple*>(item->GetItem());
for (PayloadTypeList::iterator it = payloadTypes.begin();
it != payloadTypes.end(); ++it) {
PayloadCodecTuple* payloadType = *it;
if (payloadType != NULL)
{
VideoCodec codec;
@ -186,7 +184,6 @@ int RtpPlayMT(CmdArgs& args, int releaseTestNo, webrtc::VideoCodecType releaseTe
return -1;
}
}
item = payloadTypes.Next(item);
}
if (processingThread != NULL)
@ -235,17 +232,10 @@ int RtpPlayMT(CmdArgs& args, int releaseTestNo, webrtc::VideoCodecType releaseTe
waitEvent.Wait(MAX_RUNTIME_MS);
// Tear down
item = payloadTypes.First();
while (item != NULL)
while (!payloadTypes.empty())
{
PayloadCodecTuple* payloadType = static_cast<PayloadCodecTuple*>(item->GetItem());
if (payloadType != NULL)
{
delete payloadType;
}
ListItem* itemToRemove = item;
item = payloadTypes.Next(item);
payloadTypes.Erase(itemToRemove);
delete payloadTypes.front();
payloadTypes.pop_front();
}
while (!processingThread->Stop())
{