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Inject TickTimeInterface into VCM and tests

Browse Source 
The purpose of this change is to introduce dependency injection
of the timer into the video coding module.

Review URL: http://webrtc-codereview.appspot.com/332003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1220 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org 2011-12-16 14:40:05 +00:00
parent 5249cc8f77
commit a70f945086
45 changed files with 329 additions and 295 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

4
src/modules/video_coding/main/interface/video_coding.h
View File

@ -18,6 +18,7 @@
namespace webrtc
{
class TickTimeInterface;
class VideoEncoder;
class VideoDecoder;
struct CodecSpecificInfo;
@ -27,6 +28,9 @@ class VideoCodingModule : public Module
public:
static VideoCodingModule* Create(const WebRtc_Word32 id);
static VideoCodingModule* Create(const WebRtc_Word32 id,
TickTimeInterface* clock);
static void Destroy(VideoCodingModule* module);
// Get number of supported codecs

1
src/modules/video_coding/main/source/content_metrics_processing.cc
View File

@ -9,7 +9,6 @@
*/
#include "content_metrics_processing.h"
#include "tick_time.h"
#include "module_common_types.h"
#include "video_coding_defines.h"

13
src/modules/video_coding/main/source/generic_decoder.cc
View File

@ -12,13 +12,15 @@
#include "trace.h"
#include "generic_decoder.h"
#include "internal_defines.h"
#include "tick_time.h"
#include "tick_time_interface.h"
namespace webrtc {
VCMDecodedFrameCallback::VCMDecodedFrameCallback(VCMTiming& timing)
VCMDecodedFrameCallback::VCMDecodedFrameCallback(VCMTiming& timing,
TickTimeInterface* clock)
:
_critSect(CriticalSectionWrapper::CreateCriticalSection()),
_clock(clock),
_receiveCallback(NULL),
_timing(timing),
_timestampMap(kDecoderFrameMemoryLength)
@ -53,7 +55,7 @@ WebRtc_Word32 VCMDecodedFrameCallback::Decoded(RawImage& decodedImage)
_timing.StopDecodeTimer(
decodedImage._timeStamp,
frameInfo->decodeStartTimeMs,
VCMTickTime::MillisecondTimestamp());
_clock->MillisecondTimestamp());
if (_receiveCallback != NULL)
{
@ -146,7 +148,8 @@ WebRtc_Word32 VCMGenericDecoder::InitDecode(const VideoCodec* settings,
return _decoder.InitDecode(settings, numberOfCores);
}
WebRtc_Word32 VCMGenericDecoder::Decode(const VCMEncodedFrame& frame)
WebRtc_Word32 VCMGenericDecoder::Decode(const VCMEncodedFrame& frame,
int64_t nowMs)
{
if (_requireKeyFrame &&
!_keyFrameDecoded &&
@ -157,7 +160,7 @@ WebRtc_Word32 VCMGenericDecoder::Decode(const VCMEncodedFrame& frame)
// before we can decode delta frames.
return VCM_CODEC_ERROR;
}
_frameInfos[_nextFrameInfoIdx].decodeStartTimeMs = VCMTickTime::MillisecondTimestamp();
_frameInfos[_nextFrameInfoIdx].decodeStartTimeMs = nowMs;
_frameInfos[_nextFrameInfoIdx].renderTimeMs = frame.RenderTimeMs();
_callback->Map(frame.TimeStamp(), &_frameInfos[_nextFrameInfoIdx]);

5
src/modules/video_coding/main/source/generic_decoder.h
View File

@ -34,7 +34,7 @@ struct VCMFrameInformation
class VCMDecodedFrameCallback : public DecodedImageCallback
{
public:
VCMDecodedFrameCallback(VCMTiming& timing);
VCMDecodedFrameCallback(VCMTiming& timing, TickTimeInterface* clock);
virtual ~VCMDecodedFrameCallback();
void SetUserReceiveCallback(VCMReceiveCallback* receiveCallback);
@ -49,6 +49,7 @@ public:
private:
CriticalSectionWrapper* _critSect;
TickTimeInterface* _clock;
VideoFrame _frame;
VCMReceiveCallback* _receiveCallback;
VCMTiming& _timing;
@ -76,7 +77,7 @@ public:
*
* inputVideoBuffer reference to encoded video frame
*/
WebRtc_Word32 Decode(const VCMEncodedFrame& inputFrame);
WebRtc_Word32 Decode(const VCMEncodedFrame& inputFrame, int64_t nowMs);
/**
* Free the decoder memory

16
src/modules/video_coding/main/source/inter_frame_delay.cc
View File

@ -9,20 +9,19 @@
*/
#include "inter_frame_delay.h"
#include "tick_time.h"
namespace webrtc {
VCMInterFrameDelay::VCMInterFrameDelay()
VCMInterFrameDelay::VCMInterFrameDelay(int64_t currentWallClock)
{
Reset();
Reset(currentWallClock);
}
// Resets the delay estimate
void
VCMInterFrameDelay::Reset()
VCMInterFrameDelay::Reset(int64_t currentWallClock)
{
_zeroWallClock = VCMTickTime::MillisecondTimestamp();
_zeroWallClock = currentWallClock;
_wrapArounds = 0;
_prevWallClock = 0;
_prevTimestamp = 0;
@ -34,13 +33,8 @@ VCMInterFrameDelay::Reset()
bool
VCMInterFrameDelay::CalculateDelay(WebRtc_UWord32 timestamp,
WebRtc_Word64 *delay,
WebRtc_Word64 currentWallClock /* = -1 */)
int64_t currentWallClock)
{
if (currentWallClock <= -1)
{
currentWallClock = VCMTickTime::MillisecondTimestamp();
}
if (_prevWallClock == 0)
{
// First set of data, initialization, wait for next frame

6
src/modules/video_coding/main/source/inter_frame_delay.h
View File

@ -19,10 +19,10 @@ namespace webrtc
class VCMInterFrameDelay
{
public:
VCMInterFrameDelay();
VCMInterFrameDelay(int64_t currentWallClock);
// Resets the estimate. Zeros are given as parameters.
void Reset();
void Reset(int64_t currentWallClock);
// Calculates the delay of a frame with the given timestamp.
// This method is called when the frame is complete.
@ -35,7 +35,7 @@ public:
// Return value : true if OK, false when reordered timestamps
bool CalculateDelay(WebRtc_UWord32 timestamp,
WebRtc_Word64 *delay,
WebRtc_Word64 currentWallClock = -1);
int64_t currentWallClock);
// Returns the current difference between incoming timestamps
//

24
src/modules/video_coding/main/source/jitter_buffer.cc
View File

@ -20,7 +20,7 @@
#include "event.h"
#include "trace.h"
#include "tick_time.h"
#include "modules/video_coding/main/source/tick_time_interface.h"
#include "list_wrapper.h"
#include <cassert>
@ -57,10 +57,13 @@ VCMJitterBuffer::CompleteDecodableKeyFrameCriteria(VCMFrameBuffer* frame,
}
// Constructor
VCMJitterBuffer::VCMJitterBuffer(WebRtc_Word32 vcmId, WebRtc_Word32 receiverId,
VCMJitterBuffer::VCMJitterBuffer(TickTimeInterface* clock,
WebRtc_Word32 vcmId,
WebRtc_Word32 receiverId,
bool master) :
_vcmId(vcmId),
_receiverId(receiverId),
_clock(clock),
_running(false),
_critSect(CriticalSectionWrapper::CreateCriticalSection()),
_master(master),
@ -81,6 +84,7 @@ VCMJitterBuffer::VCMJitterBuffer(WebRtc_Word32 vcmId, WebRtc_Word32 receiverId,
_numConsecutiveOldPackets(0),
_discardedPackets(0),
_jitterEstimate(vcmId, receiverId),
_delayEstimate(_clock->MillisecondTimestamp()),
_rttMs(0),
_nackMode(kNoNack),
_lowRttNackThresholdMs(-1),
@ -180,7 +184,7 @@ VCMJitterBuffer::Start()
_incomingFrameCount = 0;
_incomingFrameRate = 0;
_incomingBitCount = 0;
_timeLastIncomingFrameCount = VCMTickTime::MillisecondTimestamp();
_timeLastIncomingFrameCount = _clock->MillisecondTimestamp();
memset(_receiveStatistics, 0, sizeof(_receiveStatistics));
_numConsecutiveOldFrames = 0;
@ -262,7 +266,7 @@ VCMJitterBuffer::FlushInternal()
// Also reset the jitter and delay estimates
_jitterEstimate.Reset();
_delayEstimate.Reset();
_delayEstimate.Reset(_clock->MillisecondTimestamp());
_waitingForCompletion.frameSize = 0;
_waitingForCompletion.timestamp = 0;
@ -602,7 +606,7 @@ WebRtc_Word32
VCMJitterBuffer::GetUpdate(WebRtc_UWord32& frameRate, WebRtc_UWord32& bitRate)
{
CriticalSectionScoped cs(_critSect);
const WebRtc_Word64 now = VCMTickTime::MillisecondTimestamp();
const WebRtc_Word64 now = _clock->MillisecondTimestamp();
WebRtc_Word64 diff = now - _timeLastIncomingFrameCount;
if (diff < 1000 && _incomingFrameRate > 0 && _incomingBitRate > 0)
{
@ -657,7 +661,7 @@ VCMJitterBuffer::GetUpdate(WebRtc_UWord32& frameRate, WebRtc_UWord32& bitRate)
else
{
// No frames since last call
_timeLastIncomingFrameCount = VCMTickTime::MillisecondTimestamp();
_timeLastIncomingFrameCount = _clock->MillisecondTimestamp();
frameRate = 0;
bitRate = 0;
_incomingBitRate = 0;
@ -698,7 +702,7 @@ VCMJitterBuffer::GetCompleteFrameForDecoding(WebRtc_UWord32 maxWaitTimeMS)
_critSect->Leave();
return NULL;
}
const WebRtc_Word64 endWaitTimeMs = VCMTickTime::MillisecondTimestamp()
const WebRtc_Word64 endWaitTimeMs = _clock->MillisecondTimestamp()
+ maxWaitTimeMS;
WebRtc_Word64 waitTimeMs = maxWaitTimeMS;
while (waitTimeMs > 0)
@ -727,7 +731,7 @@ VCMJitterBuffer::GetCompleteFrameForDecoding(WebRtc_UWord32 maxWaitTimeMS)
if (oldestFrame == NULL)
{
waitTimeMs = endWaitTimeMs -
VCMTickTime::MillisecondTimestamp();
_clock->MillisecondTimestamp();
}
else
{
@ -1514,7 +1518,7 @@ VCMFrameBufferEnum
VCMJitterBuffer::InsertPacket(VCMEncodedFrame* buffer, const VCMPacket& packet)
{
CriticalSectionScoped cs(_critSect);
WebRtc_Word64 nowMs = VCMTickTime::MillisecondTimestamp();
WebRtc_Word64 nowMs = _clock->MillisecondTimestamp();
VCMFrameBufferEnum bufferReturn = kSizeError;
VCMFrameBufferEnum ret = kSizeError;
VCMFrameBuffer* frame = static_cast<VCMFrameBuffer*>(buffer);
@ -1525,7 +1529,7 @@ VCMJitterBuffer::InsertPacket(VCMEncodedFrame* buffer, const VCMPacket& packet)
{
// Now it's time to start estimating jitter
// reset the delay estimate.
_delayEstimate.Reset();
_delayEstimate.Reset(_clock->MillisecondTimestamp());
_firstPacket = false;
}

5
src/modules/video_coding/main/source/jitter_buffer.h
View File

@ -33,6 +33,7 @@ enum VCMNackMode
};
// forward declarations
class TickTimeInterface;
class VCMFrameBuffer;
class VCMPacket;
class VCMEncodedFrame;
@ -49,7 +50,8 @@ public:
class VCMJitterBuffer
{
public:
VCMJitterBuffer(WebRtc_Word32 vcmId = -1,
VCMJitterBuffer(TickTimeInterface* clock,
WebRtc_Word32 vcmId = -1,
WebRtc_Word32 receiverId = -1,
bool master = true);
virtual ~VCMJitterBuffer();
@ -191,6 +193,7 @@ private:
WebRtc_Word32 _vcmId;
WebRtc_Word32 _receiverId;
TickTimeInterface* _clock;
// If we are running (have started) or not
bool _running;
CriticalSectionWrapper* _critSect;

2
src/modules/video_coding/main/source/jitter_estimator.cc
View File

@ -12,8 +12,8 @@
#include "internal_defines.h"
#include "jitter_estimator.h"
#include "rtt_filter.h"
#include "tick_time.h"
#include <assert.h>
#include <math.h>
#include <stdlib.h>
#include <string.h>

41
src/modules/video_coding/main/source/media_opt_util.cc
View File

@ -551,7 +551,7 @@ VCMFecMethod::UpdateParameters(const VCMProtectionParameters* parameters)
return true;
}
VCMLossProtectionLogic::VCMLossProtectionLogic():
VCMLossProtectionLogic::VCMLossProtectionLogic(int64_t nowMs):
_selectedMethod(NULL),
_currentParameters(),
_rtt(0),
@ -572,7 +572,7 @@ _boostRateKey(2),
_codecWidth(0),
_codecHeight(0)
{
Reset();
Reset(nowMs);
}
VCMLossProtectionLogic::~VCMLossProtectionLogic()
@ -661,13 +661,13 @@ VCMLossProtectionLogic::UpdateResidualPacketLoss(float residualPacketLoss)
}
void
VCMLossProtectionLogic::UpdateLossPr(WebRtc_UWord8 lossPr255)
VCMLossProtectionLogic::UpdateLossPr(WebRtc_UWord8 lossPr255,
int64_t nowMs)
{
const WebRtc_Word64 now = VCMTickTime::MillisecondTimestamp();
UpdateMaxLossHistory(lossPr255, now);
_lossPr255.Apply(static_cast<float> (now - _lastPrUpdateT),
UpdateMaxLossHistory(lossPr255, nowMs);
_lossPr255.Apply(static_cast<float> (nowMs - _lastPrUpdateT),
static_cast<float> (lossPr255));
_lastPrUpdateT = now;
_lastPrUpdateT = nowMs;
_lossPr = _lossPr255.Value() / 255.0f;
}
@ -741,14 +741,14 @@ VCMLossProtectionLogic::MaxFilteredLossPr(WebRtc_Word64 nowMs) const
}
WebRtc_UWord8
VCMLossProtectionLogic::FilteredLoss() const
VCMLossProtectionLogic::FilteredLoss(int64_t nowMs) const
{
if (_selectedMethod != NULL &&
(_selectedMethod->Type() == kFec ||
_selectedMethod->Type() == kNackFec))
{
// Take the windowed max of the received loss.
return MaxFilteredLossPr(VCMTickTime::MillisecondTimestamp());
return MaxFilteredLossPr(nowMs);
}
else
{
@ -770,21 +770,19 @@ VCMLossProtectionLogic::UpdateBitRate(float bitRate)
}
void
VCMLossProtectionLogic::UpdatePacketsPerFrame(float nPackets)
VCMLossProtectionLogic::UpdatePacketsPerFrame(float nPackets, int64_t nowMs)
{
const WebRtc_Word64 now = VCMTickTime::MillisecondTimestamp();
_packetsPerFrame.Apply(static_cast<float>(now - _lastPacketPerFrameUpdateT),
_packetsPerFrame.Apply(static_cast<float>(nowMs - _lastPacketPerFrameUpdateT),
nPackets);
_lastPacketPerFrameUpdateT = now;
_lastPacketPerFrameUpdateT = nowMs;
}
void
VCMLossProtectionLogic::UpdatePacketsPerFrameKey(float nPackets)
VCMLossProtectionLogic::UpdatePacketsPerFrameKey(float nPackets, int64_t nowMs)
{
const WebRtc_Word64 now = VCMTickTime::MillisecondTimestamp();
_packetsPerFrameKey.Apply(static_cast<float>(now -
_packetsPerFrameKey.Apply(static_cast<float>(nowMs -
_lastPacketPerFrameUpdateTKey), nPackets);
_lastPacketPerFrameUpdateTKey = now;
_lastPacketPerFrameUpdateTKey = nowMs;
}
void
@ -836,12 +834,11 @@ VCMLossProtectionLogic::SelectedType() const
}
void
VCMLossProtectionLogic::Reset()
VCMLossProtectionLogic::Reset(int64_t nowMs)
{
const WebRtc_Word64 now = VCMTickTime::MillisecondTimestamp();
_lastPrUpdateT = now;
_lastPacketPerFrameUpdateT = now;
_lastPacketPerFrameUpdateTKey = now;
_lastPrUpdateT = nowMs;
_lastPacketPerFrameUpdateT = nowMs;
_lastPacketPerFrameUpdateTKey = nowMs;
_lossPr255.Reset(0.9999f);
_packetsPerFrame.Reset(0.9999f);
_fecRateDelta = _fecRateKey = 0;

13
src/modules/video_coding/main/source/media_opt_util.h
View File

@ -15,7 +15,6 @@
#include "trace.h"
#include "exp_filter.h"
#include "internal_defines.h"
#include "tick_time.h"
#include "qm_select.h"
#include <cmath>
@ -212,7 +211,7 @@ private:
class VCMLossProtectionLogic
{
public:
VCMLossProtectionLogic();
VCMLossProtectionLogic(int64_t nowMs);
~VCMLossProtectionLogic();
// Set the protection method to be used
@ -251,7 +250,7 @@ public:
// Input:
// - lossPr255 : The packet loss probability [0, 255],
// reported by RTCP.
void UpdateLossPr(WebRtc_UWord8 lossPr255);
void UpdateLossPr(WebRtc_UWord8 lossPr255, int64_t nowMs);
// Update the filtered packet loss.
//
@ -270,13 +269,13 @@ public:
//
// Input:
// - nPackets : Number of packets in the latest sent frame.
void UpdatePacketsPerFrame(float nPackets);
void UpdatePacketsPerFrame(float nPackets, int64_t nowMs);
// Update the number of packets per frame estimate, for key frames
//
// Input:
// - nPackets : umber of packets in the latest sent frame.
void UpdatePacketsPerFrameKey(float nPackets);
void UpdatePacketsPerFrameKey(float nPackets, int64_t nowMs);
// Update the keyFrameSize estimate
//
@ -324,9 +323,9 @@ public:
// Returns the filtered loss probability in the interval [0, 255].
//
// Return value : The filtered loss probability
WebRtc_UWord8 FilteredLoss() const;
WebRtc_UWord8 FilteredLoss(int64_t nowMs) const;
void Reset();
void Reset(int64_t nowMs);
void Release();

36
src/modules/video_coding/main/source/media_optimization.cc
View File

@ -12,11 +12,14 @@
#include "content_metrics_processing.h"
#include "frame_dropper.h"
#include "qm_select.h"
#include "modules/video_coding/main/source/tick_time_interface.h"
namespace webrtc {
VCMMediaOptimization::VCMMediaOptimization(WebRtc_Word32 id):
VCMMediaOptimization::VCMMediaOptimization(WebRtc_Word32 id,
TickTimeInterface* clock):
_id(id),
_clock(clock),
_maxBitRate(0),
_sendCodecType(kVideoCodecUnknown),
_codecWidth(0),
@ -42,7 +45,7 @@ _lastChangeTime(0)
memset(_incomingFrameTimes, -1, sizeof(_incomingFrameTimes));
_frameDropper = new VCMFrameDropper(_id);
_lossProtLogic = new VCMLossProtectionLogic();
_lossProtLogic = new VCMLossProtectionLogic(_clock->MillisecondTimestamp());
_content = new VCMContentMetricsProcessing();
_qmResolution = new VCMQmResolution();
}
@ -62,12 +65,12 @@ VCMMediaOptimization::Reset()
memset(_incomingFrameTimes, -1, sizeof(_incomingFrameTimes));
InputFrameRate(); // Resets _incomingFrameRate
_frameDropper->Reset();
_lossProtLogic->Reset();
_lossProtLogic->Reset(_clock->MillisecondTimestamp());
_frameDropper->SetRates(0, 0);
_content->Reset();
_qmResolution->Reset();
_lossProtLogic->UpdateFrameRate(_incomingFrameRate);
_lossProtLogic->Reset();
_lossProtLogic->Reset(_clock->MillisecondTimestamp());
_sendStatisticsZeroEncode = 0;
_targetBitRate = 0;
_codecWidth = 0;
@ -93,7 +96,7 @@ VCMMediaOptimization::SetTargetRates(WebRtc_UWord32 bitRate,
{
VCMProtectionMethod *selectedMethod = _lossProtLogic->SelectedMethod();
_lossProtLogic->UpdateBitRate(static_cast<float>(bitRate));
_lossProtLogic->UpdateLossPr(fractionLost);
_lossProtLogic->UpdateLossPr(fractionLost, _clock->MillisecondTimestamp());
_lossProtLogic->UpdateRtt(roundTripTimeMs);
_lossProtLogic->UpdateResidualPacketLoss(static_cast<float>(fractionLost));
@ -116,7 +119,8 @@ VCMMediaOptimization::SetTargetRates(WebRtc_UWord32 bitRate,
// average or max filter may be used.
// We should think about which filter is appropriate for low/high bit rates,
// low/high loss rates, etc.
WebRtc_UWord8 packetLossEnc = _lossProtLogic->FilteredLoss();
WebRtc_UWord8 packetLossEnc = _lossProtLogic->FilteredLoss(
_clock->MillisecondTimestamp());
// For now use the filtered loss for computing the robustness settings
_lossProtLogic->UpdateFilteredLossPr(packetLossEnc);
@ -253,7 +257,7 @@ VCMMediaOptimization::SetEncodingData(VideoCodecType sendCodecType, WebRtc_Word3
// has changed. If native dimension values have changed, then either user
// initiated change, or QM initiated change. Will be able to determine only
// after the processing of the first frame.
_lastChangeTime = VCMTickTime::MillisecondTimestamp();
_lastChangeTime = _clock->MillisecondTimestamp();
_content->Reset();
_content->UpdateFrameRate(frameRate);
@ -336,7 +340,7 @@ VCMMediaOptimization::SentFrameRate()
float
VCMMediaOptimization::SentBitRate()
{
UpdateBitRateEstimate(-1, VCMTickTime::MillisecondTimestamp());
UpdateBitRateEstimate(-1, _clock->MillisecondTimestamp());
return _avgSentBitRateBps / 1000.0f;
}
@ -351,7 +355,7 @@ VCMMediaOptimization::UpdateWithEncodedData(WebRtc_Word32 encodedLength,
FrameType encodedFrameType)
{
// look into the ViE version - debug mode - needs also number of layers.
UpdateBitRateEstimate(encodedLength, VCMTickTime::MillisecondTimestamp());
UpdateBitRateEstimate(encodedLength, _clock->MillisecondTimestamp());
if(encodedLength > 0)
{
const bool deltaFrame = (encodedFrameType != kVideoFrameKey &&
@ -364,11 +368,13 @@ VCMMediaOptimization::UpdateWithEncodedData(WebRtc_Word32 encodedLength,
static_cast<float>(_maxPayloadSize);
if (deltaFrame)
{
_lossProtLogic->UpdatePacketsPerFrame(minPacketsPerFrame);
_lossProtLogic->UpdatePacketsPerFrame(
minPacketsPerFrame, _clock->MillisecondTimestamp());
}
else
{
_lossProtLogic->UpdatePacketsPerFrameKey(minPacketsPerFrame);
_lossProtLogic->UpdatePacketsPerFrameKey(
minPacketsPerFrame, _clock->MillisecondTimestamp());
}
if (_enableQm)
@ -519,7 +525,7 @@ VCMMediaOptimization::SelectQuality()
_qmResolution->ResetRates();
// Reset counters
_lastQMUpdateTime = VCMTickTime::MillisecondTimestamp();
_lastQMUpdateTime = _clock->MillisecondTimestamp();
// Reset content metrics
_content->Reset();
@ -542,7 +548,7 @@ VCMMediaOptimization::checkStatusForQMchange()
// (to sample the metrics) from the event lastChangeTime
// lastChangeTime is the time where user changed the size/rate/frame rate
// (via SetEncodingData)
WebRtc_Word64 now = VCMTickTime::MillisecondTimestamp();
WebRtc_Word64 now = _clock->MillisecondTimestamp();
if ((now - _lastQMUpdateTime) < kQmMinIntervalMs ||
(now - _lastChangeTime) < kQmMinIntervalMs)
{
@ -612,7 +618,7 @@ VCMMediaOptimization::QMUpdate(VCMResolutionScale* qm)
void
VCMMediaOptimization::UpdateIncomingFrameRate()
{
WebRtc_Word64 now = VCMTickTime::MillisecondTimestamp();
WebRtc_Word64 now = _clock->MillisecondTimestamp();
if (_incomingFrameTimes[0] == 0)
{
// first no shift
@ -664,7 +670,7 @@ VCMMediaOptimization::ProcessIncomingFrameRate(WebRtc_Word64 now)
WebRtc_UWord32
VCMMediaOptimization::InputFrameRate()
{
ProcessIncomingFrameRate(VCMTickTime::MillisecondTimestamp());
ProcessIncomingFrameRate(_clock->MillisecondTimestamp());
return WebRtc_UWord32 (_incomingFrameRate + 0.5f);
}

5
src/modules/video_coding/main/source/media_optimization.h
View File

@ -24,6 +24,7 @@ namespace webrtc
enum { kBitrateMaxFrameSamples = 60 };
enum { kBitrateAverageWinMs = 1000 };
class TickTimeInterface;
class VCMContentMetricsProcessing;
class VCMFrameDropper;
@ -38,7 +39,7 @@ struct VCMEncodedFrameSample
class VCMMediaOptimization
{
public:
VCMMediaOptimization(WebRtc_Word32 id);
VCMMediaOptimization(WebRtc_Word32 id, TickTimeInterface* clock);
~VCMMediaOptimization(void);
/*
* Reset the Media Optimization module
@ -162,7 +163,7 @@ private:
enum { kFrameHistoryWinMs = 2000};
WebRtc_Word32 _id;
TickTimeInterface* _clock;
WebRtc_Word32 _maxBitRate;
VideoCodecType _sendCodecType;
WebRtc_UWord16 _codecWidth;

28
src/modules/video_coding/main/source/receiver.cc
View File

@ -13,7 +13,7 @@
#include "encoded_frame.h"
#include "internal_defines.h"
#include "media_opt_util.h"
#include "tick_time.h"
#include "tick_time_interface.h"
#include "trace.h"
#include "video_coding.h"
@ -22,15 +22,17 @@
namespace webrtc {
VCMReceiver::VCMReceiver(VCMTiming& timing,
TickTimeInterface* clock,
WebRtc_Word32 vcmId,
WebRtc_Word32 receiverId,
bool master)
:
_critSect(CriticalSectionWrapper::CreateCriticalSection()),
_vcmId(vcmId),
_clock(clock),
_receiverId(receiverId),
_master(master),
_jitterBuffer(vcmId, receiverId, master),
_jitterBuffer(_clock, vcmId, receiverId, master),
_timing(timing),
_renderWaitEvent(*new VCMEvent()),
_state(kPassive)
@ -118,10 +120,10 @@ VCMReceiver::InsertPacket(const VCMPacket& packet,
VCMId(_vcmId, _receiverId),
"Packet seqNo %u of frame %u at %u",
packet.seqNum, packet.timestamp,
MaskWord64ToUWord32(VCMTickTime::MillisecondTimestamp()));
MaskWord64ToUWord32(_clock->MillisecondTimestamp()));
}
const WebRtc_Word64 nowMs = VCMTickTime::MillisecondTimestamp();
const WebRtc_Word64 nowMs = _clock->MillisecondTimestamp();
WebRtc_Word64 renderTimeMs = _timing.RenderTimeMs(packet.timestamp, nowMs);
@ -130,7 +132,7 @@ VCMReceiver::InsertPacket(const VCMPacket& packet,
// Render time error. Assume that this is due to some change in
// the incoming video stream and reset the JB and the timing.
_jitterBuffer.Flush();
_timing.Reset();
_timing.Reset(_clock->MillisecondTimestamp());
return VCM_FLUSH_INDICATOR;
}
else if (renderTimeMs < nowMs - kMaxVideoDelayMs)
@ -139,7 +141,7 @@ VCMReceiver::InsertPacket(const VCMPacket& packet,
"This frame should have been rendered more than %u ms ago."
"Flushing jitter buffer and resetting timing.", kMaxVideoDelayMs);
_jitterBuffer.Flush();
_timing.Reset();
_timing.Reset(_clock->MillisecondTimestamp());
return VCM_FLUSH_INDICATOR;
}
else if (_timing.TargetVideoDelay() > kMaxVideoDelayMs)
@ -148,14 +150,14 @@ VCMReceiver::InsertPacket(const VCMPacket& packet,
"More than %u ms target delay. Flushing jitter buffer and resetting timing.",
kMaxVideoDelayMs);
_jitterBuffer.Flush();
_timing.Reset();
_timing.Reset(_clock->MillisecondTimestamp());
return VCM_FLUSH_INDICATOR;
}
// First packet received belonging to this frame.
if (buffer->Length() == 0)
{
const WebRtc_Word64 nowMs = VCMTickTime::MillisecondTimestamp();
const WebRtc_Word64 nowMs = _clock->MillisecondTimestamp();
if (_master)
{
// Only trace the primary receiver to make it possible to parse and plot the trace file.
@ -199,7 +201,7 @@ VCMReceiver::FrameForDecoding(WebRtc_UWord16 maxWaitTimeMs, WebRtc_Word64& nextR
// is thread-safe.
FrameType incomingFrameType = kVideoFrameDelta;
nextRenderTimeMs = -1;
const WebRtc_Word64 startTimeMs = VCMTickTime::MillisecondTimestamp();
const WebRtc_Word64 startTimeMs = _clock->MillisecondTimestamp();
WebRtc_Word64 ret = _jitterBuffer.GetNextTimeStamp(maxWaitTimeMs,
incomingFrameType,
nextRenderTimeMs);
@ -215,7 +217,7 @@ VCMReceiver::FrameForDecoding(WebRtc_UWord16 maxWaitTimeMs, WebRtc_Word64& nextR
_timing.UpdateCurrentDelay(timeStamp);
const WebRtc_Word32 tempWaitTime = maxWaitTimeMs -
static_cast<WebRtc_Word32>(VCMTickTime::MillisecondTimestamp() - startTimeMs);
static_cast<WebRtc_Word32>(_clock->MillisecondTimestamp() - startTimeMs);
WebRtc_UWord16 newMaxWaitTime = static_cast<WebRtc_UWord16>(VCM_MAX(tempWaitTime, 0));
VCMEncodedFrame* frame = NULL;
@ -256,7 +258,7 @@ VCMReceiver::FrameForDecoding(WebRtc_UWord16 maxWaitTimeMs,
{
// How long can we wait until we must decode the next frame
WebRtc_UWord32 waitTimeMs = _timing.MaxWaitingTime(nextRenderTimeMs,
VCMTickTime::MillisecondTimestamp());
_clock->MillisecondTimestamp());
// Try to get a complete frame from the jitter buffer
VCMEncodedFrame* frame = _jitterBuffer.GetCompleteFrameForDecoding(0);
@ -284,7 +286,7 @@ VCMReceiver::FrameForDecoding(WebRtc_UWord16 maxWaitTimeMs,
{
// Get an incomplete frame
if (_timing.MaxWaitingTime(nextRenderTimeMs,
VCMTickTime::MillisecondTimestamp()) > 0)
_clock->MillisecondTimestamp()) > 0)
{
// Still time to wait for a complete frame
return NULL;
@ -316,7 +318,7 @@ VCMReceiver::FrameForRendering(WebRtc_UWord16 maxWaitTimeMs,
// as possible before giving the frame to the decoder, which will render the frame as soon
// as it has been decoded.
WebRtc_UWord32 waitTimeMs = _timing.MaxWaitingTime(nextRenderTimeMs,
VCMTickTime::MillisecondTimestamp());
_clock->MillisecondTimestamp());
if (maxWaitTimeMs < waitTimeMs)
{
// If we're not allowed to wait until the frame is supposed to be rendered

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

@ -13,6 +13,7 @@
#include "critical_section_wrapper.h"
#include "jitter_buffer.h"
#include "modules/video_coding/main/source/tick_time_interface.h"
#include "timing.h"
#include "packet.h"
@ -40,6 +41,7 @@ class VCMReceiver
{
public:
VCMReceiver(VCMTiming& timing,
TickTimeInterface* clock,
WebRtc_Word32 vcmId = -1,
WebRtc_Word32 receiverId = -1,
bool master = true);
@ -83,6 +85,7 @@ private:
CriticalSectionWrapper* _critSect;
WebRtc_Word32 _vcmId;
TickTimeInterface* _clock;
WebRtc_Word32 _receiverId;
bool _master;
VCMJitterBuffer _jitterBuffer;

55
src/modules/video_coding/main/source/tick_time.h
View File

@ -1,55 +0,0 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_TICK_TIME_H_
#define WEBRTC_MODULES_VIDEO_CODING_TICK_TIME_H_
#include "tick_util.h"
#include <assert.h>
namespace webrtc
{
//#define TICK_TIME_DEBUG
class VCMTickTime : public TickTime
{
#ifdef TICK_TIME_DEBUG
public:
/*
* Get current time
*/
static TickTime Now() { assert(false); };
/*
* Get time in milli seconds
*/
static WebRtc_Word64 MillisecondTimestamp() { return _timeNowDebug; };
/*
* Get time in micro seconds
*/
static WebRtc_Word64 MicrosecondTimestamp() { return _timeNowDebug * 1000LL; };
static void IncrementDebugClock() { _timeNowDebug++; };
private:
static WebRtc_Word64 _timeNowDebug;
#else
public:
static void IncrementDebugClock() { assert(false); };
#endif
};
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_TICK_TIME_H_

11
src/modules/video_coding/main/source/timestamp_extrapolator.cc
View File

@ -9,17 +9,20 @@
*/
#include "internal_defines.h"
#include "modules/video_coding/main/source/tick_time_interface.h"
#include "timestamp_extrapolator.h"
#include "tick_time.h"
#include "trace.h"
namespace webrtc {
VCMTimestampExtrapolator::VCMTimestampExtrapolator(WebRtc_Word32 vcmId, WebRtc_Word32 id)
VCMTimestampExtrapolator::VCMTimestampExtrapolator(TickTimeInterface* clock,
WebRtc_Word32 vcmId,
WebRtc_Word32 id)
:
_rwLock(RWLockWrapper::CreateRWLock()),
_vcmId(vcmId),
_id(id),
_clock(clock),
_startMs(0),
_firstTimestamp(0),
_wrapArounds(0),
@ -35,7 +38,7 @@ _accDrift(6600), // in timestamp ticks, i.e. 15 ms
_accMaxError(7000),
_P11(1e10)
{
Reset(VCMTickTime::MillisecondTimestamp());
Reset(_clock->MillisecondTimestamp());
}
VCMTimestampExtrapolator::~VCMTimestampExtrapolator()
@ -53,7 +56,7 @@ VCMTimestampExtrapolator::Reset(const WebRtc_Word64 nowMs /* = -1 */)
}
else
{
_startMs = VCMTickTime::MillisecondTimestamp();
_startMs = _clock->MillisecondTimestamp();
}
_prevMs = _startMs;
_firstTimestamp = 0;

7
src/modules/video_coding/main/source/timestamp_extrapolator.h
View File

@ -17,10 +17,14 @@
namespace webrtc
{
class TickTimeInterface;
class VCMTimestampExtrapolator
{
public:
VCMTimestampExtrapolator(WebRtc_Word32 vcmId = 0, WebRtc_Word32 receiverId = 0);
VCMTimestampExtrapolator(TickTimeInterface* clock,
WebRtc_Word32 vcmId = 0,
WebRtc_Word32 receiverId = 0);
~VCMTimestampExtrapolator();
void Update(WebRtc_Word64 tMs, WebRtc_UWord32 ts90khz, bool trace = true);
WebRtc_UWord32 ExtrapolateTimestamp(WebRtc_Word64 tMs) const;
@ -33,6 +37,7 @@ private:
RWLockWrapper* _rwLock;
WebRtc_Word32 _vcmId;
WebRtc_Word32 _id;
TickTimeInterface* _clock;
bool _trace;
double _w[2];
double _P[2][2];

8
src/modules/video_coding/main/source/timing.cc
View File

@ -16,10 +16,14 @@
namespace webrtc {
VCMTiming::VCMTiming(WebRtc_Word32 vcmId, WebRtc_Word32 timingId, VCMTiming* masterTiming)
VCMTiming::VCMTiming(TickTimeInterface* clock,
WebRtc_Word32 vcmId,
WebRtc_Word32 timingId,
VCMTiming* masterTiming)
:
_critSect(CriticalSectionWrapper::CreateCriticalSection()),
_vcmId(vcmId),
_clock(clock),
_timingId(timingId),
_master(false),
_tsExtrapolator(),
@ -33,7 +37,7 @@ _prevFrameTimestamp(0)
if (masterTiming == NULL)
{
_master = true;
_tsExtrapolator = new VCMTimestampExtrapolator(vcmId, timingId);
_tsExtrapolator = new VCMTimestampExtrapolator(_clock, vcmId, timingId);
}
else
{

5
src/modules/video_coding/main/source/timing.h
View File

@ -18,6 +18,7 @@
namespace webrtc
{
class TickTimeInterface;
class VCMTimestampExtrapolator;
class VCMTiming
@ -25,7 +26,8 @@ class VCMTiming
public:
// The primary timing component should be passed
// if this is the dual timing component.
VCMTiming(WebRtc_Word32 vcmId = 0,
VCMTiming(TickTimeInterface* clock,
WebRtc_Word32 vcmId = 0,
WebRtc_Word32 timingId = 0,
VCMTiming* masterTiming = NULL);
~VCMTiming();
@ -92,6 +94,7 @@ protected:
private:
CriticalSectionWrapper* _critSect;
WebRtc_Word32 _vcmId;
TickTimeInterface* _clock;
WebRtc_Word32 _timingId;
bool _master;
VCMTimestampExtrapolator* _tsExtrapolator;

1
src/modules/video_coding/main/source/video_coding.gypi
View File

@ -63,7 +63,6 @@
'receiver.h',
'rtt_filter.h',
'session_info.h',
'tick_time.h',
'tick_time_interface.h',
'timestamp_extrapolator.h',
'timestamp_map.h',

55
src/modules/video_coding/main/source/video_coding_impl.cc
View File

@ -15,6 +15,7 @@
#include "packet.h"
#include "trace.h"
#include "video_codec_interface.h"
#include "modules/video_coding/main/source/tick_time_interface.h"
namespace webrtc
{
@ -33,26 +34,30 @@ VCMProcessTimer::TimeUntilProcess() const
{
return static_cast<WebRtc_UWord32>(
VCM_MAX(static_cast<WebRtc_Word64>(_periodMs) -
(VCMTickTime::MillisecondTimestamp() - _latestMs), 0));
(_clock->MillisecondTimestamp() - _latestMs), 0));
}
void
VCMProcessTimer::Processed()
{
_latestMs = VCMTickTime::MillisecondTimestamp();
_latestMs = _clock->MillisecondTimestamp();
}
VideoCodingModuleImpl::VideoCodingModuleImpl(const WebRtc_Word32 id)
VideoCodingModuleImpl::VideoCodingModuleImpl(const WebRtc_Word32 id,
TickTimeInterface* clock,
bool delete_clock_on_destroy)
:
_id(id),
clock_(clock),
delete_clock_on_destroy_(delete_clock_on_destroy),
_receiveCritSect(CriticalSectionWrapper::CreateCriticalSection()),
_receiverInited(false),
_timing(id, 1),
_dualTiming(id, 2, &_timing),
_receiver(_timing, id, 1),
_dualReceiver(_dualTiming, id, 2, false),
_decodedFrameCallback(_timing),
_dualDecodedFrameCallback(_dualTiming),
_timing(clock_, id, 1),
_dualTiming(clock_, id, 2, &_timing),
_receiver(_timing, clock_, id, 1),
_dualReceiver(_dualTiming, clock_, id, 2, false),
_decodedFrameCallback(_timing, clock_),
_dualDecodedFrameCallback(_dualTiming, clock_),
_frameTypeCallback(NULL),
_frameStorageCallback(NULL),
_receiveStatsCallback(NULL),
@ -67,17 +72,18 @@ _scheduleKeyRequest(false),
_sendCritSect(CriticalSectionWrapper::CreateCriticalSection()),
_encoder(),
_encodedFrameCallback(),
_mediaOpt(id),
_mediaOpt(id, clock_),
_sendCodecType(kVideoCodecUnknown),
_sendStatsCallback(NULL),
_encoderInputFile(NULL),
_codecDataBase(id),
_receiveStatsTimer(1000),
_sendStatsTimer(1000),
_retransmissionTimer(10),
_keyRequestTimer(500)
_receiveStatsTimer(1000, clock_),
_sendStatsTimer(1000, clock_),
_retransmissionTimer(10, clock_),
_keyRequestTimer(500, clock_)
{
assert(clock_);
for (int i = 0; i < kMaxSimulcastStreams; i++)
{
_nextFrameType[i] = kVideoFrameDelta;
@ -98,6 +104,7 @@ VideoCodingModuleImpl::~VideoCodingModuleImpl()
}
delete _receiveCritSect;
delete _sendCritSect;
if (delete_clock_on_destroy_) delete clock_;
#ifdef DEBUG_DECODER_BIT_STREAM
fclose(_bitStreamBeforeDecoder);
#endif
@ -113,7 +120,18 @@ VideoCodingModule::Create(const WebRtc_Word32 id)
webrtc::kTraceVideoCoding,
VCMId(id),
"VideoCodingModule::Create()");
return new VideoCodingModuleImpl(id);
return new VideoCodingModuleImpl(id, new TickTimeInterface(), true);
}
VideoCodingModule*
VideoCodingModule::Create(const WebRtc_Word32 id, TickTimeInterface* clock)
{
WEBRTC_TRACE(webrtc::kTraceModuleCall,
webrtc::kTraceVideoCoding,
VCMId(id),
"VideoCodingModule::Create()");
assert(clock);
return new VideoCodingModuleImpl(id, clock, false);
}
void
@ -1085,7 +1103,7 @@ VideoCodingModuleImpl::Decode(WebRtc_UWord16 maxWaitTimeMs)
// If this frame was too late, we should adjust the delay accordingly
_timing.UpdateCurrentDelay(frame->RenderTimeMs(),
VCMTickTime::MillisecondTimestamp());
clock_->MillisecondTimestamp());
#ifdef DEBUG_DECODER_BIT_STREAM
if (_bitStreamBeforeDecoder != NULL)
@ -1202,7 +1220,8 @@ VideoCodingModuleImpl::DecodeDualFrame(WebRtc_UWord16 maxWaitTimeMs)
"Decoding frame %u with dual decoder",
dualFrame->TimeStamp());
// Decode dualFrame and try to catch up
WebRtc_Word32 ret = _dualDecoder->Decode(*dualFrame);
WebRtc_Word32 ret = _dualDecoder->Decode(*dualFrame,
clock_->MillisecondTimestamp());
if (ret != WEBRTC_VIDEO_CODEC_OK)
{
WEBRTC_TRACE(webrtc::kTraceWarning,
@ -1250,7 +1269,7 @@ VideoCodingModuleImpl::Decode(const VCMEncodedFrame& frame)
return VCM_NO_CODEC_REGISTERED;
}
// Decode a frame
WebRtc_Word32 ret = _decoder->Decode(frame);
WebRtc_Word32 ret = _decoder->Decode(frame, clock_->MillisecondTimestamp());
// Check for failed decoding, run frame type request callback if needed.
if (ret < 0)

14
src/modules/video_coding/main/source/video_coding_impl.h
View File

@ -21,6 +21,7 @@
#include "generic_decoder.h"
#include "generic_encoder.h"
#include "media_optimization.h"
#include "modules/video_coding/main/source/tick_time_interface.h"
#include <stdio.h>
@ -30,13 +31,16 @@ namespace webrtc
class VCMProcessTimer
{
public:
VCMProcessTimer(WebRtc_UWord32 periodMs) :
_periodMs(periodMs), _latestMs(VCMTickTime::MillisecondTimestamp()) {}
VCMProcessTimer(WebRtc_UWord32 periodMs, TickTimeInterface* clock)
: _clock(clock),
_periodMs(periodMs),
_latestMs(_clock->MillisecondTimestamp()) {}
WebRtc_UWord32 Period() const;
WebRtc_UWord32 TimeUntilProcess() const;
void Processed();
private:
TickTimeInterface* _clock;
WebRtc_UWord32 _periodMs;
WebRtc_Word64 _latestMs;
};
@ -53,7 +57,9 @@ enum VCMKeyRequestMode
class VideoCodingModuleImpl : public VideoCodingModule
{
public:
VideoCodingModuleImpl(const WebRtc_Word32 id);
VideoCodingModuleImpl(const WebRtc_Word32 id,
TickTimeInterface* clock,
bool delete_clock_on_destroy);
virtual ~VideoCodingModuleImpl();
@ -259,6 +265,8 @@ protected:
private:
WebRtc_Word32 _id;
TickTimeInterface* clock_;
bool delete_clock_on_destroy_;
CriticalSectionWrapper* _receiveCritSect;
bool _receiverInited;
VCMTiming _timing;

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

@ -12,9 +12,9 @@
#include "video_coding.h"
#include "rtp_rtcp.h"
#include "trace.h"
#include "tick_time.h"
#include "../source/event.h"
#include "rtp_player.h"
#include "modules/video_coding/main/source/mock/fake_tick_time_interface.h"
using namespace webrtc;
@ -35,8 +35,8 @@ private:
int DecodeFromStorageTest(CmdArgs& args)
{
// Make sure this test isn't executed without simulated clocks
#if !defined(TICK_TIME_DEBUG) || !defined(EVENT_DEBUG)
// Make sure this test isn't executed without simulated events.
#if !defined(EVENT_DEBUG)
return -1;
#endif
// BEGIN Settings
@ -64,8 +64,10 @@ int DecodeFromStorageTest(CmdArgs& args)
Trace::SetLevelFilter(webrtc::kTraceAll);
VideoCodingModule* vcm = VideoCodingModule::Create(1);
VideoCodingModule* vcmPlayback = VideoCodingModule::Create(2);
FakeTickTime clock(0);
// TODO(hlundin): This test was not verified after changing to FakeTickTime.
VideoCodingModule* vcm = VideoCodingModule::Create(1, &clock);
VideoCodingModule* vcmPlayback = VideoCodingModule::Create(2, &clock);
FrameStorageCallback storageCallback(vcmPlayback);
RtpDataCallback dataCallback(vcm);
WebRtc_Word32 ret = vcm->InitializeReceiver();
@ -80,7 +82,7 @@ int DecodeFromStorageTest(CmdArgs& args)
}
vcm->RegisterFrameStorageCallback(&storageCallback);
vcmPlayback->RegisterReceiveCallback(&receiveCallback);
RTPPlayer rtpStream(rtpFilename.c_str(), &dataCallback);
RTPPlayer rtpStream(rtpFilename.c_str(), &dataCallback, &clock);
ListWrapper payloadTypes;
payloadTypes.PushFront(new PayloadCodecTuple(VCM_VP8_PAYLOAD_TYPE, "VP8", kVideoCodecVP8));
@ -124,9 +126,9 @@ int DecodeFromStorageTest(CmdArgs& args)
ret = 0;
// RTP stream main loop
while ((ret = rtpStream.NextPacket(VCMTickTime::MillisecondTimestamp())) == 0)
while ((ret = rtpStream.NextPacket(clock.MillisecondTimestamp())) == 0)
{
if (VCMTickTime::MillisecondTimestamp() % 5 == 0)
if (clock.MillisecondTimestamp() % 5 == 0)
{
ret = vcm->Decode();
if (ret < 0)
@ -138,11 +140,11 @@ int DecodeFromStorageTest(CmdArgs& args)
{
vcm->Process();
}
if (MAX_RUNTIME_MS > -1 && VCMTickTime::MillisecondTimestamp() >= MAX_RUNTIME_MS)
if (MAX_RUNTIME_MS > -1 && clock.MillisecondTimestamp() >= MAX_RUNTIME_MS)
{
break;
}
VCMTickTime::IncrementDebugClock();
clock.IncrementDebugClock(1);
}
switch (ret)

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

@ -11,11 +11,11 @@
#include "generic_codec_test.h"
#include <cmath>
#include <stdio.h>
#include "tick_time.h"
#include "../source/event.h"
#include "rtp_rtcp.h"
#include "module_common_types.h"
#include "test_macros.h"
#include "modules/video_coding/main/source/mock/fake_tick_time_interface.h"
using namespace webrtc;
@ -23,12 +23,13 @@ enum { kMaxWaitEncTimeMs = 100 };
int GenericCodecTest::RunTest(CmdArgs& args)
{
#if !defined(TICK_TIME_DEBUG) || !defined(EVENT_DEBUG)
printf("\n\nEnable debug time to run this test!\n\n");
#if !defined(EVENT_DEBUG)
printf("\n\nEnable debug events to run this test!\n\n");
return -1;
#endif
VideoCodingModule* vcm = VideoCodingModule::Create(1);
GenericCodecTest* get = new GenericCodecTest(vcm);
FakeTickTime clock(0);
VideoCodingModule* vcm = VideoCodingModule::Create(1, &clock);
GenericCodecTest* get = new GenericCodecTest(vcm, &clock);
Trace::CreateTrace();
Trace::SetTraceFile(
(test::OutputPath() + "genericCodecTestTrace.txt").c_str());
@ -40,7 +41,8 @@ int GenericCodecTest::RunTest(CmdArgs& args)
return 0;
}
GenericCodecTest::GenericCodecTest(VideoCodingModule* vcm):
GenericCodecTest::GenericCodecTest(VideoCodingModule* vcm, FakeTickTime* clock):
_clock(clock),
_vcm(vcm),
_width(0),
_height(0),
@ -307,7 +309,7 @@ GenericCodecTest::Perform(CmdArgs& args)
_vcm->SetChannelParameters((WebRtc_UWord32)_bitRate, 0, 20);
_frameCnt = 0;
totalBytes = 0;
startTime = VCMTickTime::MicrosecondTimestamp();
startTime = _clock->MicrosecondTimestamp();
_encodeCompleteCallback->Initialize();
sendStats.SetTargetFrameRate(static_cast<WebRtc_UWord32>(_frameRate));
_vcm->RegisterSendStatisticsCallback(&sendStats);
@ -331,7 +333,7 @@ GenericCodecTest::Perform(CmdArgs& args)
//currentTime = VCMTickTime::MillisecondTimestamp();//clock()/(double)CLOCKS_PER_SEC;
if (_frameCnt == _frameRate)// @ 1sec
{
oneSecTime = VCMTickTime::MicrosecondTimestamp();
oneSecTime = _clock->MicrosecondTimestamp();
totalBytesOneSec = _encodeCompleteCallback->EncodedBytes();//totalBytes;
}
TEST(_vcm->TimeUntilNextProcess() >= 0);
@ -341,7 +343,7 @@ GenericCodecTest::Perform(CmdArgs& args)
// estimating rates
// complete sequence
// bit rate assumes input frame rate is as specified
currentTime = VCMTickTime::MicrosecondTimestamp();
currentTime = _clock->MicrosecondTimestamp();
totalBytes = _encodeCompleteCallback->EncodedBytes();
actualBitrate = (float)(8.0/1000)*(totalBytes / (_frameCnt / _frameRate));
@ -514,8 +516,8 @@ GenericCodecTest::Print()
float
GenericCodecTest::WaitForEncodedFrame() const
{
WebRtc_Word64 startTime = TickTime::MillisecondTimestamp();
while (TickTime::MillisecondTimestamp() - startTime < kMaxWaitEncTimeMs*10)
WebRtc_Word64 startTime = _clock->MillisecondTimestamp();
while (_clock->MillisecondTimestamp() - startTime < kMaxWaitEncTimeMs*10)
{
if (_encodeCompleteCallback->EncodeComplete())
{
@ -528,11 +530,7 @@ GenericCodecTest::WaitForEncodedFrame() const
void
GenericCodecTest::IncrementDebugClock(float frameRate)
{
for (int t= 0; t < 1000/frameRate; t++)
{
VCMTickTime::IncrementDebugClock();
}
return;
_clock->IncrementDebugClock(1000/frameRate);
}
int

6
src/modules/video_coding/main/test/generic_codec_test.h
View File

@ -31,10 +31,13 @@ namespace webrtc {
int VCMGenericCodecTest(CmdArgs& args);
class FakeTickTime;
class GenericCodecTest
{
public:
GenericCodecTest(webrtc::VideoCodingModule* vcm);
GenericCodecTest(webrtc::VideoCodingModule* vcm,
webrtc::FakeTickTime* clock);
~GenericCodecTest();
static int RunTest(CmdArgs& args);
WebRtc_Word32 Perform(CmdArgs& args);
@ -46,6 +49,7 @@ private:
WebRtc_Word32 TearDown();
void IncrementDebugClock(float frameRate);
webrtc::FakeTickTime* _clock;
webrtc::VideoCodingModule* _vcm;
webrtc::VideoCodec _sendCodec;
webrtc::VideoCodec _receiveCodec;

11
src/modules/video_coding/main/test/jitter_buffer_test.cc
View File

@ -19,10 +19,10 @@
#include "jitter_estimate_test.h"
#include "jitter_estimator.h"
#include "media_opt_util.h"
#include "modules/video_coding/main/source/tick_time_interface.h"
#include "packet.h"
#include "test_util.h"
#include "test_macros.h"
#include "tick_time.h"
using namespace webrtc;
@ -92,10 +92,11 @@ int CheckOutFrame(VCMEncodedFrame* frameOut, unsigned int size, bool startCode)
int JitterBufferTest(CmdArgs& args)
{
// Don't run these tests with debug time
#if defined(TICK_TIME_DEBUG) || defined(EVENT_DEBUG)
// Don't run these tests with debug event.
#if defined(EVENT_DEBUG)
return -1;
#endif
TickTimeInterface clock;
// Start test
WebRtc_UWord16 seqNum = 1234;
@ -114,7 +115,7 @@ int JitterBufferTest(CmdArgs& args)
packet.seqNum = seqNum;
packet.payloadType = 126;
seqNum++;
fb->InsertPacket(packet, VCMTickTime::MillisecondTimestamp(), false, 0);
fb->InsertPacket(packet, clock.MillisecondTimestamp(), false, 0);
TEST(frameList.Insert(fb) == 0);
}
VCMFrameListItem* item = NULL;
@ -135,7 +136,7 @@ int JitterBufferTest(CmdArgs& args)
//printf("DONE timestamp ordered frame list\n");
VCMJitterBuffer jb;
VCMJitterBuffer jb(&clock);
seqNum = 1234;
timeStamp = 123*90;

1
src/modules/video_coding/main/test/jitter_estimate_test.cc
View File

@ -11,7 +11,6 @@
#include <stdio.h>
#include <ctime>
#include "JitterEstimateTest.h"
#include "tick_time.h"
using namespace webrtc;

11
src/modules/video_coding/main/test/media_opt_test.cc
View File

@ -34,8 +34,9 @@ int MediaOptTest::RunTest(int testNum, CmdArgs& args)
Trace::CreateTrace();
Trace::SetTraceFile((test::OutputPath() + "mediaOptTestTrace.txt").c_str());
Trace::SetLevelFilter(webrtc::kTraceAll);
VideoCodingModule* vcm = VideoCodingModule::Create(1);
MediaOptTest* mot = new MediaOptTest(vcm);
TickTimeInterface clock;
VideoCodingModule* vcm = VideoCodingModule::Create(1, &clock);
MediaOptTest* mot = new MediaOptTest(vcm, &clock);
if (testNum == 0)
{ // regular
mot->Setup(0, args);
@ -66,8 +67,9 @@ int MediaOptTest::RunTest(int testNum, CmdArgs& args)
}
MediaOptTest::MediaOptTest(VideoCodingModule* vcm):
MediaOptTest::MediaOptTest(VideoCodingModule* vcm, TickTimeInterface* clock):
_vcm(vcm),
_clock(clock),
_width(0),
_height(0),
_lengthSourceFrame(0),
@ -279,7 +281,8 @@ MediaOptTest::Perform()
encodeCompleteCallback->SetCodecType(ConvertCodecType(_codecName.c_str()));
encodeCompleteCallback->SetFrameDimensions(_width, _height);
// frame ready to be sent to network
RTPSendCompleteCallback* outgoingTransport = new RTPSendCompleteCallback(_rtp);
RTPSendCompleteCallback* outgoingTransport =
new RTPSendCompleteCallback(_rtp, _clock);
_rtp->RegisterSendTransport(outgoingTransport);
//FrameReceiveCallback
VCMDecodeCompleteCallback receiveCallback(_decodedFile);

4
src/modules/video_coding/main/test/media_opt_test.h
View File

@ -31,7 +31,8 @@
class MediaOptTest
{
public:
MediaOptTest(webrtc::VideoCodingModule* vcm);
MediaOptTest(webrtc::VideoCodingModule* vcm,
webrtc::TickTimeInterface* clock);
~MediaOptTest();
static int RunTest(int testNum, CmdArgs& args);
@ -51,6 +52,7 @@ private:
webrtc::VideoCodingModule* _vcm;
webrtc::RtpRtcp* _rtp;
webrtc::TickTimeInterface* _clock;
std::string _inname;
std::string _outname;
std::string _actualSourcename;

6
src/modules/video_coding/main/test/mt_rx_tx_test.cc
View File

@ -170,7 +170,8 @@ int MTRxTxTest(CmdArgs& args)
TEST(rtp->SetGenericFECStatus(fecEnabled, VCM_RED_PAYLOAD_TYPE, VCM_ULPFEC_PAYLOAD_TYPE) == 0);
//VCM
VideoCodingModule* vcm = VideoCodingModule::Create(1);
TickTimeInterface clock;
VideoCodingModule* vcm = VideoCodingModule::Create(1, &clock);
if (vcm->InitializeReceiver() < 0)
{
return -1;
@ -215,7 +216,8 @@ int MTRxTxTest(CmdArgs& args)
encodeCompleteCallback->SetCodecType(ConvertCodecType(args.codecName.c_str()));
encodeCompleteCallback->SetFrameDimensions(width, height);
// frame ready to be sent to network
RTPSendCompleteCallback* outgoingTransport = new RTPSendCompleteCallback(rtp, "dump.rtp");
RTPSendCompleteCallback* outgoingTransport =
new RTPSendCompleteCallback(rtp, &clock, "dump.rtp");
rtp->RegisterSendTransport(outgoingTransport);
// FrameReceiveCallback
VCMDecodeCompleteCallback receiveCallback(decodedFile);

10
src/modules/video_coding/main/test/mt_test_common.cc
View File

@ -12,13 +12,15 @@
#include <cmath>
#include "modules/video_coding/main/source/tick_time_interface.h"
#include "rtp_dump.h"
namespace webrtc {
TransportCallback::TransportCallback(webrtc::RtpRtcp* rtp,
TickTimeInterface* clock,
const char* filename):
RTPSendCompleteCallback(rtp, filename)
RTPSendCompleteCallback(rtp, clock, filename)
{
//
}
@ -49,7 +51,8 @@ TransportCallback::SendPacket(int channel, const void *data, int len)
transmitPacket = PacketLoss();
}
WebRtc_UWord64 now = VCMTickTime::MillisecondTimestamp();
TickTimeInterface clock;
int64_t now = clock.MillisecondTimestamp();
// Insert outgoing packet into list
if (transmitPacket)
{
@ -73,7 +76,8 @@ TransportCallback::TransportPackets()
{
// Are we ready to send packets to the receiver?
rtpPacket* packet = NULL;
WebRtc_UWord64 now = VCMTickTime::MillisecondTimestamp();
TickTimeInterface clock;
int64_t now = clock.MillisecondTimestamp();
while (!_rtpPackets.Empty())
{

2
src/modules/video_coding/main/test/mt_test_common.h
View File

@ -47,7 +47,7 @@ class TransportCallback:public RTPSendCompleteCallback
{
public:
// constructor input: (receive side) rtp module to send encoded data to
TransportCallback(webrtc::RtpRtcp* rtp,
TransportCallback(webrtc::RtpRtcp* rtp, TickTimeInterface* clock,
const char* filename = NULL);
virtual ~TransportCallback();
// Add packets to list

28
src/modules/video_coding/main/test/normal_test.cc
View File

@ -17,10 +17,10 @@
#include "../source/event.h"
#include "common_types.h"
#include "modules/video_coding/main/source/mock/fake_tick_time_interface.h"
#include "test_callbacks.h"
#include "test_macros.h"
#include "test_util.h"
#include "tick_time.h"
#include "trace.h"
#include "testsupport/metrics/video_metrics.h"
@ -28,20 +28,22 @@ using namespace webrtc;
int NormalTest::RunTest(CmdArgs& args)
{
// Don't run this test with debug time
#if defined(TICK_TIME_DEBUG) || defined(EVENT_DEBUG)
#if defined(EVENT_DEBUG)
printf("SIMULATION TIME\n");
TickTimeInterface* clock = new FakeTickTime(0);
#else
printf("REAL-TIME\n");
TickTimeInterface* clock = new TickTimeInterface;
#endif
Trace::CreateTrace();
Trace::SetTraceFile(
(test::OutputPath() + "VCMNormalTestTrace.txt").c_str());
Trace::SetLevelFilter(webrtc::kTraceAll);
VideoCodingModule* vcm = VideoCodingModule::Create(1);
NormalTest VCMNTest(vcm);
VideoCodingModule* vcm = VideoCodingModule::Create(1, clock);
NormalTest VCMNTest(vcm, clock);
VCMNTest.Perform(args);
VideoCodingModule::Destroy(vcm);
delete clock;
Trace::ReturnTrace();
return 0;
}
@ -182,8 +184,9 @@ VCMNTDecodeCompleCallback::DecodedBytes()
//VCM Normal Test Class implementation
NormalTest::NormalTest(VideoCodingModule* vcm)
NormalTest::NormalTest(VideoCodingModule* vcm, TickTimeInterface* clock)
:
_clock(clock),
_vcm(vcm),
_sumEncBytes(0),
_timeStamp(0),
@ -281,8 +284,8 @@ NormalTest::Perform(CmdArgs& args)
while (feof(_sourceFile) == 0)
{
#if !(defined(TICK_TIME_DEBUG) || defined(EVENT_DEBUG))
WebRtc_Word64 processStartTime = VCMTickTime::MillisecondTimestamp();
#if !defined(EVENT_DEBUG)
WebRtc_Word64 processStartTime = _clock->MillisecondTimestamp();
#endif
TEST(fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) > 0 ||
feof(_sourceFile));
@ -314,13 +317,10 @@ NormalTest::Perform(CmdArgs& args)
_vcm->Process();
}
WebRtc_UWord32 framePeriod = static_cast<WebRtc_UWord32>(1000.0f/static_cast<float>(_sendCodec.maxFramerate) + 0.5f);
#if defined(TICK_TIME_DEBUG) || defined(EVENT_DEBUG)
for (unsigned int i=0; i < framePeriod; i++)
{
VCMTickTime::IncrementDebugClock();
}
#if defined(EVENT_DEBUG)
static_cast<FakeTickTime*>(_clock)->IncrementDebugClock(framePeriod);
#else
WebRtc_Word64 timeSpent = VCMTickTime::MillisecondTimestamp() - processStartTime;
WebRtc_Word64 timeSpent = _clock->MillisecondTimestamp() - processStartTime;
if (timeSpent < framePeriod)
{
waitEvent->Wait(framePeriod - timeSpent);

4
src/modules/video_coding/main/test/normal_test.h
View File

@ -83,7 +83,8 @@ private:
class NormalTest
{
public:
NormalTest(webrtc::VideoCodingModule* vcm);
NormalTest(webrtc::VideoCodingModule* vcm,
webrtc::TickTimeInterface* clock);
~NormalTest();
static int RunTest(CmdArgs& args);
WebRtc_Word32 Perform(CmdArgs& args);
@ -105,6 +106,7 @@ protected:
// calculating pipeline delay, and decoding time
void FrameDecoded(WebRtc_UWord32 timeStamp);
webrtc::TickTimeInterface* _clock;
webrtc::VideoCodingModule* _vcm;
webrtc::VideoCodec _sendCodec;
webrtc::VideoCodec _receiveCodec;

15
src/modules/video_coding/main/test/quality_modes_test.cc
View File

@ -15,6 +15,7 @@
#include <time.h>
#include "../source/event.h"
#include "modules/video_coding/main/source/tick_time_interface.h"
#include "test_callbacks.h"
#include "test_macros.h"
#include "testsupport/metrics/video_metrics.h"
@ -24,20 +25,22 @@ using namespace webrtc;
int qualityModeTest()
{
// Don't run this test with debug time
#if defined(TICK_TIME_DEBUG) || defined(EVENT_DEBUG)
// Don't run this test with debug events.
#if defined(EVENT_DEBUG)
return -1;
#endif
VideoCodingModule* vcm = VideoCodingModule::Create(1);
QualityModesTest QMTest(vcm);
TickTimeInterface clock;
VideoCodingModule* vcm = VideoCodingModule::Create(1, &clock);
QualityModesTest QMTest(vcm, &clock);
QMTest.Perform();
VideoCodingModule::Destroy(vcm);
return 0;
}
QualityModesTest::QualityModesTest(VideoCodingModule *vcm):
NormalTest(vcm),
QualityModesTest::QualityModesTest(VideoCodingModule* vcm,
TickTimeInterface* clock):
NormalTest(vcm, clock),
_vpm()
{
//

3
src/modules/video_coding/main/test/quality_modes_test.h
View File

@ -20,7 +20,8 @@ int qualityModeTest();
class QualityModesTest : public NormalTest
{
public:
QualityModesTest(webrtc::VideoCodingModule* vcm);
QualityModesTest(webrtc::VideoCodingModule* vcm,
webrtc::TickTimeInterface* clock);
virtual ~QualityModesTest();
WebRtc_Word32 Perform();

8
src/modules/video_coding/main/test/receiver_timing_tests.cc
View File

@ -11,7 +11,6 @@
#include "receiver_tests.h"
#include "video_coding.h"
#include "trace.h"
#include "tick_time.h"
#include "../source/event.h"
#include "../source/internal_defines.h"
#include "timing.h"
@ -49,8 +48,8 @@ public:
int ReceiverTimingTests(CmdArgs& args)
{
// Make sure this test is never executed with simulated clocks
#if defined(TICK_TIME_DEBUG) || defined(EVENT_DEBUG)
// Make sure this test is never executed with simulated events.
#if defined(EVENT_DEBUG)
return -1;
#endif
@ -62,7 +61,8 @@ int ReceiverTimingTests(CmdArgs& args)
// A static random seed
srand(0);
VCMTiming timing;
TickTimeInterface clock;
VCMTiming timing(&clock);
float clockInMs = 0.0;
WebRtc_UWord32 waitTime = 0;
WebRtc_UWord32 jitterDelayMs = 0;

27
src/modules/video_coding/main/test/rtp_player.cc
View File

@ -20,8 +20,8 @@
#include "../source/internal_defines.h"
#include "gtest/gtest.h"
#include "modules/video_coding/main/source/tick_time_interface.h"
#include "rtp_rtcp.h"
#include "tick_time.h"
using namespace webrtc;
@ -82,7 +82,9 @@ WebRtc_UWord32 LostPackets::AddPacket(WebRtc_UWord8* rtpData, WebRtc_UWord16 rtp
return 0;
}
WebRtc_UWord32 LostPackets::SetResendTime(WebRtc_UWord16 sequenceNumber, WebRtc_Word64 resendTime)
WebRtc_UWord32 LostPackets::SetResendTime(WebRtc_UWord16 sequenceNumber,
WebRtc_Word64 resendTime,
WebRtc_Word64 nowMs)
{
CriticalSectionScoped cs(_critSect);
ListItem* item = First();
@ -90,7 +92,6 @@ WebRtc_UWord32 LostPackets::SetResendTime(WebRtc_UWord16 sequenceNumber, WebRtc_
{
RawRtpPacket* packet = static_cast<RawRtpPacket*>(item->GetItem());
const WebRtc_UWord16 seqNo = (packet->rtpData[2] << 8) + packet->rtpData[3];
const WebRtc_Word64 nowMs = VCMTickTime::MillisecondTimestamp();
if (sequenceNumber == seqNo && packet->resendTimeMs + 10 < nowMs)
{
if (_debugFile != NULL)
@ -123,18 +124,21 @@ WebRtc_UWord32 LostPackets::NumberOfPacketsToResend() const
return count;
}
void LostPackets::ResentPacket(WebRtc_UWord16 seqNo)
void LostPackets::ResentPacket(WebRtc_UWord16 seqNo, WebRtc_Word64 nowMs)
{
CriticalSectionScoped cs(_critSect);
if (_debugFile != NULL)
{
fprintf(_debugFile, "Resent %u at %u\n", seqNo,
MaskWord64ToUWord32(VCMTickTime::MillisecondTimestamp()));
MaskWord64ToUWord32(nowMs));
}
}
RTPPlayer::RTPPlayer(const char* filename, RtpData* callback)
RTPPlayer::RTPPlayer(const char* filename,
RtpData* callback,
TickTimeInterface* clock)
:
_clock(clock),
_rtpModule(*RtpRtcp::CreateRtpRtcp(1, false)),
_nextRtpTime(0),
_dataCallback(callback),
@ -272,7 +276,7 @@ WebRtc_Word32 RTPPlayer::ReadHeader()
WebRtc_UWord32 RTPPlayer::TimeUntilNextPacket() const
{
WebRtc_Word64 timeLeft = (_nextRtpTime - _firstPacketRtpTime) - (VCMTickTime::MillisecondTimestamp() - _firstPacketTimeMs);
WebRtc_Word64 timeLeft = (_nextRtpTime - _firstPacketRtpTime) - (_clock->MillisecondTimestamp() - _firstPacketTimeMs);
if (timeLeft < 0)
{
return 0;
@ -305,7 +309,8 @@ WebRtc_Word32 RTPPlayer::NextPacket(const WebRtc_Word64 timeNow)
_resendPacketCount++;
if (ret > 0)
{
_lostPackets.ResentPacket(seqNo);
_lostPackets.ResentPacket(seqNo,
_clock->MillisecondTimestamp());
}
else if (ret < 0)
{
@ -327,7 +332,7 @@ WebRtc_Word32 RTPPlayer::NextPacket(const WebRtc_Word64 timeNow)
if (_firstPacket)
{
_firstPacketRtpTime = static_cast<WebRtc_Word64>(_nextRtpTime);
_firstPacketTimeMs = VCMTickTime::MillisecondTimestamp();
_firstPacketTimeMs = _clock->MillisecondTimestamp();
}
if (_reordering && _reorderBuffer == NULL)
{
@ -447,7 +452,9 @@ WebRtc_Word32 RTPPlayer::ResendPackets(const WebRtc_UWord16* sequenceNumbers, We
}
for (int i=0; i < length; i++)
{
_lostPackets.SetResendTime(sequenceNumbers[i], VCMTickTime::MillisecondTimestamp() + _rttMs);
_lostPackets.SetResendTime(sequenceNumbers[i],
_clock->MillisecondTimestamp() + _rttMs,
_clock->MillisecondTimestamp());
}
return 0;
}

12
src/modules/video_coding/main/test/rtp_player.h
View File

@ -16,6 +16,7 @@
#include "list_wrapper.h"
#include "critical_section_wrapper.h"
#include "video_coding_defines.h"
#include "modules/video_coding/main/source/tick_time_interface.h"
#include <stdio.h>
#include <string>
@ -42,10 +43,12 @@ public:
~LostPackets();
WebRtc_UWord32 AddPacket(WebRtc_UWord8* rtpData, WebRtc_UWord16 rtpLen);
WebRtc_UWord32 SetResendTime(WebRtc_UWord16 sequenceNumber, WebRtc_Word64 resendTime);
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);
void ResentPacket(WebRtc_UWord16 seqNo, WebRtc_Word64 nowMs);
void Lock() {_critSect->Enter();};
void Unlock() {_critSect->Leave();};
private:
@ -66,7 +69,9 @@ struct PayloadCodecTuple
class RTPPlayer : public webrtc::VCMPacketRequestCallback
{
public:
RTPPlayer(const char* filename, webrtc::RtpData* callback);
RTPPlayer(const char* filename,
webrtc::RtpData* callback,
webrtc::TickTimeInterface* clock);
virtual ~RTPPlayer();
WebRtc_Word32 Initialize(const webrtc::ListWrapper& payloadList);
@ -81,6 +86,7 @@ private:
WebRtc_Word32 SendPacket(WebRtc_UWord8* rtpData, WebRtc_UWord16 rtpLen);
WebRtc_Word32 ReadPacket(WebRtc_Word16* rtpdata, WebRtc_UWord32* offset);
WebRtc_Word32 ReadHeader();
webrtc::TickTimeInterface* _clock;
FILE* _rtpFile;
webrtc::RtpRtcp& _rtpModule;
WebRtc_UWord32 _nextRtpTime;

5
src/modules/video_coding/main/test/test_callbacks.cc
View File

@ -12,6 +12,7 @@
#include <cmath>
#include "modules/video_coding/main/source/tick_time_interface.h"
#include "rtp_dump.h"
#include "test_macros.h"
@ -199,7 +200,9 @@ VCMDecodeCompleteCallback::DecodedBytes()
}
RTPSendCompleteCallback::RTPSendCompleteCallback(RtpRtcp* rtp,
TickTimeInterface* clock,
const char* filename):
_clock(clock),
_sendCount(0),
_rtp(rtp),
_lossPct(0),
@ -251,7 +254,7 @@ RTPSendCompleteCallback::SendPacket(int channel, const void *data, int len)
bool transmitPacket = true;
transmitPacket = PacketLoss();
WebRtc_UWord64 now = VCMTickTime::MillisecondTimestamp();
WebRtc_UWord64 now = _clock->MillisecondTimestamp();
// Insert outgoing packet into list
if (transmitPacket)
{

4
src/modules/video_coding/main/test/test_callbacks.h
View File

@ -24,7 +24,6 @@
#include "module_common_types.h"
#include "rtp_rtcp.h"
#include "test_util.h"
#include "tick_time.h"
#include "trace.h"
#include "video_coding.h"
@ -157,7 +156,7 @@ class RTPSendCompleteCallback: public Transport
{
public:
// Constructor input: (receive side) rtp module to send encoded data to
RTPSendCompleteCallback(RtpRtcp* rtp,
RTPSendCompleteCallback(RtpRtcp* rtp, TickTimeInterface* clock,
const char* filename = NULL);
virtual ~RTPSendCompleteCallback();
// Send Packet to receive side RTP module
@ -184,6 +183,7 @@ protected:
// Random uniform loss model
bool UnifomLoss(double lossPct);
TickTimeInterface* _clock;
WebRtc_UWord32 _sendCount;
RtpRtcp* _rtp;
double _lossPct;

10
src/modules/video_coding/main/test/tester_main.cc
View File

@ -27,16 +27,10 @@
using namespace webrtc;
/*
* Build with TICK_TIME_DEBUG and EVENT_DEBUG defined
* to build the tests with simulated clock.
* Build with EVENT_DEBUG defined
* to build the tests with simulated events.
*/
// TODO(holmer): How do we get debug time into the cmd line interface?
/* Debug time */
#if defined(TICK_TIME_DEBUG) && defined(EVENT_DEBUG)
WebRtc_Word64 VCMTickTime::_timeNowDebug = 0; // current time in ms
#endif
int vcmMacrosTests = 0;
int vcmMacrosErrors = 0;

19
src/modules/video_coding/main/test/video_rtp_play.cc
View File

@ -12,11 +12,11 @@
#include "video_coding.h"
#include "rtp_rtcp.h"
#include "trace.h"
#include "tick_time.h"
#include "../source/event.h"
#include "../source/internal_defines.h"
#include "test_macros.h"
#include "rtp_player.h"
#include "modules/video_coding/main/source/mock/fake_tick_time_interface.h"
#include <stdio.h>
#include <string.h>
@ -72,8 +72,8 @@ FrameReceiveCallback::FrameToRender(VideoFrame& videoFrame)
int RtpPlay(CmdArgs& args)
{
// Make sure this test isn't executed without simulated clocks
#if !defined(TICK_TIME_DEBUG) || !defined(EVENT_DEBUG)
// Make sure this test isn't executed without simulated events.
#if !defined(EVENT_DEBUG)
return -1;
#endif
// BEGIN Settings
@ -90,9 +90,10 @@ int RtpPlay(CmdArgs& args)
if (outFile == "")
outFile = test::OutputPath() + "RtpPlay_decoded.yuv";
FrameReceiveCallback receiveCallback(outFile);
VideoCodingModule* vcm = VideoCodingModule::Create(1);
FakeTickTime clock(0);
VideoCodingModule* vcm = VideoCodingModule::Create(1, &clock);
RtpDataCallback dataCallback(vcm);
RTPPlayer rtpStream(args.inputFile.c_str(), &dataCallback);
RTPPlayer rtpStream(args.inputFile.c_str(), &dataCallback, &clock);
ListWrapper payloadTypes;
@ -150,9 +151,9 @@ int RtpPlay(CmdArgs& args)
ret = 0;
// RTP stream main loop
while ((ret = rtpStream.NextPacket(VCMTickTime::MillisecondTimestamp())) == 0)
while ((ret = rtpStream.NextPacket(clock.MillisecondTimestamp())) == 0)
{
if (VCMTickTime::MillisecondTimestamp() % 5 == 0)
if (clock.MillisecondTimestamp() % 5 == 0)
{
ret = vcm->Decode();
if (ret < 0)
@ -165,11 +166,11 @@ int RtpPlay(CmdArgs& args)
{
vcm->Process();
}
if (MAX_RUNTIME_MS > -1 && VCMTickTime::MillisecondTimestamp() >= MAX_RUNTIME_MS)
if (MAX_RUNTIME_MS > -1 && clock.MillisecondTimestamp() >= MAX_RUNTIME_MS)
{
break;
}
VCMTickTime::IncrementDebugClock();
clock.IncrementDebugClock(1);
}
switch (ret)

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

@ -14,7 +14,6 @@
#include "trace.h"
#include "thread_wrapper.h"
#include "../source/event.h"
#include "tick_time.h"
#include "test_macros.h"
#include "rtp_player.h"
@ -40,8 +39,8 @@ bool RtpReaderThread(void* obj)
SharedState* state = static_cast<SharedState*>(obj);
EventWrapper& waitEvent = *EventWrapper::Create();
// RTP stream main loop
WebRtc_Word64 nowMs = VCMTickTime::MillisecondTimestamp();
if (state->_rtpPlayer.NextPacket(nowMs) < 0)
TickTimeInterface clock;
if (state->_rtpPlayer.NextPacket(clock.MillisecondTimestamp()) < 0)
{
return false;
}
@ -60,8 +59,8 @@ bool DecodeThread(void* obj)
int RtpPlayMT(CmdArgs& args, int releaseTestNo, webrtc::VideoCodecType releaseTestVideoType)
{
// Don't run these tests with debug time
#if defined(TICK_TIME_DEBUG) || defined(EVENT_DEBUG)
// Don't run these tests with debug events.
#if defined(EVENT_DEBUG)
return -1;
#endif
@ -82,8 +81,9 @@ int RtpPlayMT(CmdArgs& args, int releaseTestNo, webrtc::VideoCodecType releaseTe
(protection == kProtectionDualDecoder ||
protection == kProtectionNack ||
kProtectionNackFEC));
TickTimeInterface clock;
VideoCodingModule* vcm =
VideoCodingModule::Create(1);
VideoCodingModule::Create(1, &clock);
RtpDataCallback dataCallback(vcm);
std::string rtpFilename;
rtpFilename = args.inputFile;
@ -136,7 +136,7 @@ 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);
RTPPlayer rtpStream(rtpFilename.c_str(), &dataCallback, &clock);
ListWrapper payloadTypes;
payloadTypes.PushFront(new PayloadCodecTuple(VCM_VP8_PAYLOAD_TYPE,
"VP8", kVideoCodecVP8));