Add API to get the number of packets discarded by the video jitter buffer due to being too late.
BUG= TEST= Review URL: http://webrtc-codereview.appspot.com/200001 git-svn-id: http://webrtc.googlecode.com/svn/trunk@723 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
parent
06887aebae
commit
791eec7424
@ -490,6 +490,12 @@ public:
|
||||
// Return value : VCM_OK, on success.
|
||||
// <0, on error.
|
||||
virtual WebRtc_Word32 ReceivedFrameCount(VCMFrameCount& frameCount) const = 0;
|
||||
|
||||
// Returns the number of packets discarded by the jitter buffer due to being
|
||||
// too late. This can include duplicated packets which arrived after the
|
||||
// frame was sent to the decoder. Therefore packets which were prematurely
|
||||
// NACKed will be counted.
|
||||
virtual WebRtc_UWord32 DiscardedPackets() const = 0;
|
||||
};
|
||||
|
||||
} // namespace webrtc
|
||||
|
@ -81,6 +81,7 @@ VCMJitterBuffer::VCMJitterBuffer(WebRtc_Word32 vcmId, WebRtc_Word32 receiverId,
|
||||
_dropCount(0),
|
||||
_numConsecutiveOldFrames(0),
|
||||
_numConsecutiveOldPackets(0),
|
||||
_discardedPackets(0),
|
||||
_jitterEstimate(vcmId, receiverId),
|
||||
_rttMs(0),
|
||||
_nackMode(kNoNack),
|
||||
@ -134,6 +135,7 @@ VCMJitterBuffer::operator=(const VCMJitterBuffer& rhs)
|
||||
_dropCount = rhs._dropCount;
|
||||
_numConsecutiveOldFrames = rhs._numConsecutiveOldFrames;
|
||||
_numConsecutiveOldPackets = rhs._numConsecutiveOldPackets;
|
||||
_discardedPackets = rhs._discardedPackets;
|
||||
_jitterEstimate = rhs._jitterEstimate;
|
||||
_delayEstimate = rhs._delayEstimate;
|
||||
_waitingForCompletion = rhs._waitingForCompletion;
|
||||
@ -210,6 +212,7 @@ VCMJitterBuffer::Start()
|
||||
|
||||
_numConsecutiveOldFrames = 0;
|
||||
_numConsecutiveOldPackets = 0;
|
||||
_discardedPackets = 0;
|
||||
|
||||
_frameEvent.Reset(); // start in a non-signaled state
|
||||
_packetEvent.Reset(); // start in a non-signaled state
|
||||
@ -438,6 +441,11 @@ VCMJitterBuffer::GetFrameStatistics(WebRtc_UWord32& receivedDeltaFrames,
|
||||
return 0;
|
||||
}
|
||||
|
||||
WebRtc_UWord32 VCMJitterBuffer::DiscardedPackets() const {
|
||||
CriticalSectionScoped cs(_critSect);
|
||||
return _discardedPackets;
|
||||
}
|
||||
|
||||
// Gets frame to use for this timestamp. If no match, get empty frame.
|
||||
WebRtc_Word32
|
||||
VCMJitterBuffer::GetFrame(const VCMPacket& packet, VCMEncodedFrame*& frame)
|
||||
@ -448,11 +456,12 @@ VCMJitterBuffer::GetFrame(const VCMPacket& packet, VCMEncodedFrame*& frame)
|
||||
}
|
||||
|
||||
_critSect.Enter();
|
||||
// Make sure that old empty packets are inserted.
|
||||
if (LatestTimestamp(static_cast<WebRtc_UWord32>(_lastDecodedTimeStamp),
|
||||
packet.timestamp) == _lastDecodedTimeStamp
|
||||
&& packet.sizeBytes > 0)
|
||||
// Make sure that old Empty packets are inserted.
|
||||
{
|
||||
_discardedPackets++; // Only counts discarded media packets
|
||||
// Trying to get an old frame.
|
||||
_numConsecutiveOldPackets++;
|
||||
if (_numConsecutiveOldPackets > kMaxConsecutiveOldPackets)
|
||||
|
@ -68,6 +68,9 @@ public:
|
||||
WebRtc_Word32 GetFrameStatistics(WebRtc_UWord32& receivedDeltaFrames,
|
||||
WebRtc_UWord32& receivedKeyFrames) const;
|
||||
|
||||
// Get number of packets discarded by the jitter buffer
|
||||
WebRtc_UWord32 DiscardedPackets() const;
|
||||
|
||||
// Statistics, Calculate frame and bit rates
|
||||
WebRtc_Word32 GetUpdate(WebRtc_UWord32& frameRate, WebRtc_UWord32& bitRate);
|
||||
|
||||
@ -223,6 +226,9 @@ private:
|
||||
WebRtc_UWord32 _numConsecutiveOldFrames;
|
||||
// Number of packets in a row that have been too old
|
||||
WebRtc_UWord32 _numConsecutiveOldPackets;
|
||||
// Number of packets discarded by the jitter buffer
|
||||
WebRtc_UWord32 _discardedPackets;
|
||||
|
||||
// Filters for estimating jitter
|
||||
VCMJitterEstimator _jitterEstimate;
|
||||
// Calculates network delays used for jitter calculations
|
||||
|
@ -344,6 +344,10 @@ VCMReceiver::ReceivedFrameCount(VCMFrameCount& frameCount) const
|
||||
frameCount.numKeyFrames);
|
||||
}
|
||||
|
||||
WebRtc_UWord32 VCMReceiver::DiscardedPackets() const {
|
||||
return _jitterBuffer.DiscardedPackets();
|
||||
}
|
||||
|
||||
void
|
||||
VCMReceiver::SetNackMode(VCMNackMode nackMode)
|
||||
{
|
||||
|
@ -57,6 +57,7 @@ public:
|
||||
void ReleaseFrame(VCMEncodedFrame* frame);
|
||||
WebRtc_Word32 ReceiveStatistics(WebRtc_UWord32& bitRate, WebRtc_UWord32& frameRate);
|
||||
WebRtc_Word32 ReceivedFrameCount(VCMFrameCount& frameCount) const;
|
||||
WebRtc_UWord32 DiscardedPackets() const;
|
||||
|
||||
// NACK
|
||||
void SetNackMode(VCMNackMode nackMode);
|
||||
|
File diff suppressed because it is too large
Load Diff
@ -43,9 +43,11 @@ private:
|
||||
|
||||
enum VCMKeyRequestMode
|
||||
{
|
||||
kKeyOnError, // Normal mode, request key frames on decoder error
|
||||
kKeyOnKeyLoss, // Request key frames on decoder error and on packet loss in key frames
|
||||
kKeyOnLoss, // Request key frames on decoder error and on packet loss in any frame
|
||||
kKeyOnError, // Normal mode, request key frames on decoder error
|
||||
kKeyOnKeyLoss, // Request key frames on decoder error and on packet loss
|
||||
// in key frames.
|
||||
kKeyOnLoss, // Request key frames on decoder error and on packet loss
|
||||
// in any frame
|
||||
};
|
||||
|
||||
class VideoCodingModuleImpl : public VideoCodingModule
|
||||
@ -65,7 +67,8 @@ public:
|
||||
// Change the unique identifier of this object
|
||||
virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
|
||||
|
||||
// Returns the number of milliseconds until the module want a worker thread to call Process
|
||||
// Returns the number of milliseconds until the module want a worker thread
|
||||
// to call Process
|
||||
virtual WebRtc_Word32 TimeUntilNextProcess();
|
||||
|
||||
virtual WebRtc_Word32 Process();
|
||||
@ -97,7 +100,8 @@ public:
|
||||
bool internalSource = false);
|
||||
|
||||
// Get codec config parameters
|
||||
virtual WebRtc_Word32 CodecConfigParameters(WebRtc_UWord8* buffer, WebRtc_Word32 size);
|
||||
virtual WebRtc_Word32 CodecConfigParameters(WebRtc_UWord8* buffer,
|
||||
WebRtc_Word32 size);
|
||||
|
||||
// Get encode bitrate
|
||||
virtual WebRtc_UWord32 Bitrate() const;
|
||||
@ -106,30 +110,38 @@ public:
|
||||
virtual WebRtc_UWord32 FrameRate() const;
|
||||
|
||||
// Set channel parameters
|
||||
virtual WebRtc_Word32 SetChannelParameters(WebRtc_UWord32 availableBandWidth,
|
||||
WebRtc_UWord8 lossRate,
|
||||
WebRtc_UWord32 RTT);
|
||||
virtual WebRtc_Word32 SetChannelParameters(
|
||||
WebRtc_UWord32 availableBandWidth,
|
||||
WebRtc_UWord8 lossRate,
|
||||
WebRtc_UWord32 RTT);
|
||||
|
||||
// Set recieve channel parameters
|
||||
virtual WebRtc_Word32 SetReceiveChannelParameters(WebRtc_UWord32 RTT);
|
||||
|
||||
// Register a transport callback which will be called to deliver the encoded buffers
|
||||
virtual WebRtc_Word32 RegisterTransportCallback(VCMPacketizationCallback* transport);
|
||||
// Register a transport callback which will be called to deliver the
|
||||
// encoded buffers
|
||||
virtual WebRtc_Word32 RegisterTransportCallback(
|
||||
VCMPacketizationCallback* transport);
|
||||
|
||||
// Register a send statistics callback which will be called to deliver information
|
||||
// about the video stream produced by the encoder,
|
||||
// Register a send statistics callback which will be called to deliver
|
||||
// information about the video stream produced by the encoder,
|
||||
// for instance the average frame rate and bit rate.
|
||||
virtual WebRtc_Word32 RegisterSendStatisticsCallback(VCMSendStatisticsCallback* sendStats);
|
||||
virtual WebRtc_Word32 RegisterSendStatisticsCallback(
|
||||
VCMSendStatisticsCallback* sendStats);
|
||||
|
||||
// Register a video quality settings callback which will be called when
|
||||
// frame rate/dimensions need to be updated for video quality optimization
|
||||
virtual WebRtc_Word32 RegisterVideoQMCallback(VCMQMSettingsCallback* videoQMSettings);
|
||||
virtual WebRtc_Word32 RegisterVideoQMCallback(
|
||||
VCMQMSettingsCallback* videoQMSettings);
|
||||
|
||||
// Register a video protection callback which will be called to deliver
|
||||
// the requested FEC rate and NACK status (on/off).
|
||||
virtual WebRtc_Word32 RegisterProtectionCallback(VCMProtectionCallback* protection);
|
||||
virtual WebRtc_Word32 RegisterProtectionCallback(
|
||||
VCMProtectionCallback* protection);
|
||||
|
||||
// Enable or disable a video protection method.
|
||||
virtual WebRtc_Word32 SetVideoProtection(VCMVideoProtection videoProtection, bool enable);
|
||||
virtual WebRtc_Word32 SetVideoProtection(VCMVideoProtection videoProtection,
|
||||
bool enable);
|
||||
|
||||
// Add one raw video frame to the encoder, blocking.
|
||||
virtual WebRtc_Word32 AddVideoFrame(
|
||||
@ -164,31 +176,35 @@ public:
|
||||
WebRtc_UWord8 payloadType,
|
||||
bool internalRenderTiming);
|
||||
|
||||
// Register a receive callback. Will be called whenever there are a new frame ready
|
||||
// for rendering.
|
||||
virtual WebRtc_Word32 RegisterReceiveCallback(VCMReceiveCallback* receiveCallback);
|
||||
// Register a receive callback. Will be called whenever there are a new
|
||||
// frame ready for rendering.
|
||||
virtual WebRtc_Word32 RegisterReceiveCallback(
|
||||
VCMReceiveCallback* receiveCallback);
|
||||
|
||||
// Register a receive statistics callback which will be called to deliver information
|
||||
// about the video stream received by the receiving side of the VCM, for instance
|
||||
// the average frame rate and bit rate.
|
||||
// Register a receive statistics callback which will be called to deliver
|
||||
// information about the video stream received by the receiving side of the
|
||||
// VCM, for instance the average frame rate and bit rate.
|
||||
virtual WebRtc_Word32 RegisterReceiveStatisticsCallback(
|
||||
VCMReceiveStatisticsCallback* receiveStats);
|
||||
VCMReceiveStatisticsCallback* receiveStats);
|
||||
|
||||
// Register a frame type request callback.
|
||||
virtual WebRtc_Word32 RegisterFrameTypeCallback(VCMFrameTypeCallback* frameTypeCallback);
|
||||
virtual WebRtc_Word32 RegisterFrameTypeCallback(
|
||||
VCMFrameTypeCallback* frameTypeCallback);
|
||||
|
||||
// Register a frame storage callback.
|
||||
virtual WebRtc_Word32 RegisterFrameStorageCallback(
|
||||
VCMFrameStorageCallback* frameStorageCallback);
|
||||
VCMFrameStorageCallback* frameStorageCallback);
|
||||
|
||||
// Nack callback
|
||||
virtual WebRtc_Word32 RegisterPacketRequestCallback(VCMPacketRequestCallback* callback);
|
||||
virtual WebRtc_Word32 RegisterPacketRequestCallback(
|
||||
VCMPacketRequestCallback* callback);
|
||||
|
||||
// Decode next frame, blocks for a maximum of maxWaitTimeMs milliseconds.
|
||||
// Should be called as often as possible to get the most out of the decoder.
|
||||
virtual WebRtc_Word32 Decode(WebRtc_UWord16 maxWaitTimeMs = 200);
|
||||
|
||||
// Decode next dual frame, blocks for a maximum of maxWaitTimeMs milliseconds.
|
||||
// Decode next dual frame, blocks for a maximum of maxWaitTimeMs
|
||||
// milliseconds.
|
||||
virtual WebRtc_Word32 DecodeDualFrame(WebRtc_UWord16 maxWaitTimeMs = 200);
|
||||
|
||||
// Reset the decoder state
|
||||
@ -207,17 +223,19 @@ public:
|
||||
|
||||
// A part of an encoded frame to be decoded.
|
||||
// Used in conjunction with VCMFrameStorageCallback.
|
||||
virtual WebRtc_Word32 DecodeFromStorage(const EncodedVideoData& frameFromStorage);
|
||||
virtual WebRtc_Word32 DecodeFromStorage(
|
||||
const EncodedVideoData& frameFromStorage);
|
||||
|
||||
// Set codec config parameters
|
||||
virtual WebRtc_Word32 SetCodecConfigParameters(WebRtc_UWord8 payloadType,
|
||||
const WebRtc_UWord8* buffer,
|
||||
WebRtc_Word32 length);
|
||||
|
||||
// Minimum playout delay (Used for lip-sync). This is the minimum delay required
|
||||
// to sync with audio. Not included in VideoCodingModule::Delay()
|
||||
// Minimum playout delay (Used for lip-sync). This is the minimum delay
|
||||
// required to sync with audio. Not included in VideoCodingModule::Delay()
|
||||
// Defaults to 0 ms.
|
||||
virtual WebRtc_Word32 SetMinimumPlayoutDelay(WebRtc_UWord32 minPlayoutDelayMs);
|
||||
virtual WebRtc_Word32 SetMinimumPlayoutDelay(
|
||||
WebRtc_UWord32 minPlayoutDelayMs);
|
||||
|
||||
// The estimated delay caused by rendering
|
||||
virtual WebRtc_Word32 SetRenderDelay(WebRtc_UWord32 timeMS);
|
||||
@ -228,15 +246,19 @@ public:
|
||||
// Received frame counters
|
||||
virtual WebRtc_Word32 ReceivedFrameCount(VCMFrameCount& frameCount) const;
|
||||
|
||||
// Returns the number of packets discarded by the jitter buffer.
|
||||
virtual WebRtc_UWord32 DiscardedPackets() const;
|
||||
|
||||
protected:
|
||||
WebRtc_Word32 Decode(const webrtc::VCMEncodedFrame& frame);
|
||||
WebRtc_Word32 RequestKeyFrame();
|
||||
WebRtc_Word32 RequestSliceLossIndication(const WebRtc_UWord64 pictureID) const;
|
||||
WebRtc_Word32 RequestSliceLossIndication(
|
||||
const WebRtc_UWord64 pictureID) const;
|
||||
WebRtc_Word32 NackList(WebRtc_UWord16* nackList, WebRtc_UWord16& size);
|
||||
|
||||
private:
|
||||
WebRtc_Word32 _id;
|
||||
CriticalSectionWrapper& _receiveCritSect; // Critical section for receive side
|
||||
CriticalSectionWrapper& _receiveCritSect;
|
||||
bool _receiverInited;
|
||||
VCMTiming _timing;
|
||||
VCMTiming _dualTiming;
|
||||
@ -255,7 +277,7 @@ private:
|
||||
VCMKeyRequestMode _keyRequestMode;
|
||||
bool _scheduleKeyRequest;
|
||||
|
||||
CriticalSectionWrapper& _sendCritSect; // Critical section for send side
|
||||
CriticalSectionWrapper& _sendCritSect;
|
||||
VCMGenericEncoder* _encoder;
|
||||
VCMEncodedFrameCallback _encodedFrameCallback;
|
||||
FrameType _nextFrameType;
|
||||
|
@ -800,11 +800,6 @@ int JitterBufferTest(CmdArgs& args)
|
||||
packet.insertStartCode = false;
|
||||
//printf("DONE H.264 insert start code test 2 packets\n");
|
||||
|
||||
|
||||
// Temporarily do this to make the rest of the test work:
|
||||
timeStamp += 33*90;
|
||||
seqNum += 4;
|
||||
|
||||
//
|
||||
// TEST statistics
|
||||
//
|
||||
@ -823,8 +818,35 @@ int JitterBufferTest(CmdArgs& args)
|
||||
TEST(frameRate > 30);
|
||||
TEST(bitRate > 10000000);
|
||||
|
||||
|
||||
// Insert 3 old packets and verify that we have 3 discarded packets
|
||||
packet.timestamp = timeStamp - 1000;
|
||||
frameIn = jb.GetFrame(packet);
|
||||
TEST(frameIn == NULL);
|
||||
|
||||
packet.timestamp = timeStamp - 500;
|
||||
frameIn = jb.GetFrame(packet);
|
||||
TEST(frameIn == NULL);
|
||||
|
||||
packet.timestamp = timeStamp - 100;
|
||||
frameIn = jb.GetFrame(packet);
|
||||
TEST(frameIn == NULL);
|
||||
|
||||
TEST(jb.DiscardedPackets() == 3);
|
||||
|
||||
jb.Flush();
|
||||
|
||||
// This statistic shouldn't be reset by a flush.
|
||||
TEST(jb.DiscardedPackets() == 3);
|
||||
|
||||
//printf("DONE Statistics\n");
|
||||
|
||||
|
||||
// Temporarily do this to make the rest of the test work:
|
||||
timeStamp += 33*90;
|
||||
seqNum += 4;
|
||||
|
||||
|
||||
//
|
||||
// TEST delta frame 100 packets with seqNum wrap
|
||||
//
|
||||
@ -833,7 +855,6 @@ int JitterBufferTest(CmdArgs& args)
|
||||
// ---------------------------------------
|
||||
//
|
||||
|
||||
// test flush
|
||||
jb.Flush();
|
||||
|
||||
// insert first packet
|
||||
|
@ -140,6 +140,10 @@ public:
|
||||
unsigned int& keyFrames,
|
||||
unsigned int& deltaFrames) const = 0;
|
||||
|
||||
// Gets the number of packets discarded by the jitter buffer because they
|
||||
// arrived too late.
|
||||
virtual unsigned int GetDiscardedPackets(const int videoChannel) const = 0;
|
||||
|
||||
// Enables key frame request callback in ViEDecoderObserver.
|
||||
virtual int SetKeyFrameRequestCallbackStatus(const int videoChannel,
|
||||
const bool enable) = 0;
|
||||
|
@ -515,6 +515,14 @@ WebRtc_Word32 ViEChannel::ReceiveCodecStatistics(WebRtc_UWord32& numKeyFrames,
|
||||
return 0;
|
||||
}
|
||||
|
||||
WebRtc_UWord32 ViEChannel::DiscardedPackets() const {
|
||||
WEBRTC_TRACE(webrtc::kTraceInfo,
|
||||
webrtc::kTraceVideo,
|
||||
ViEId(_engineId, _channelId),
|
||||
"%s", __FUNCTION__);
|
||||
return _vcm.DiscardedPackets();
|
||||
}
|
||||
|
||||
// ----------------------------------------------------------------------------
|
||||
// WaitForKeyFrame
|
||||
//
|
||||
|
@ -95,6 +95,8 @@ public:
|
||||
WebRtc_Word32 ReceiveCodecStatistics(WebRtc_UWord32& numKeyFrames,
|
||||
WebRtc_UWord32& numDeltaFrames);
|
||||
|
||||
WebRtc_UWord32 DiscardedPackets() const;
|
||||
|
||||
WebRtc_Word32 WaitForKeyFrame(bool wait);
|
||||
|
||||
WebRtc_Word32 SetSignalPacketLossStatus(bool enable, bool onlyKeyFrames);
|
||||
|
@ -566,6 +566,25 @@ int ViECodecImpl::GetReceiveCodecStastistics(const int videoChannel,
|
||||
|
||||
}
|
||||
|
||||
unsigned int ViECodecImpl::GetDiscardedPackets(const int videoChannel) const {
|
||||
WEBRTC_TRACE(webrtc::kTraceApiCall, webrtc::kTraceVideo,
|
||||
ViEId(_instanceId, videoChannel),
|
||||
"%s(videoChannel: %d, codecType: %d)", __FUNCTION__,
|
||||
videoChannel);
|
||||
|
||||
ViEChannelManagerScoped cs(_channelManager);
|
||||
ViEChannel* vieChannel = cs.Channel(videoChannel);
|
||||
if (vieChannel == NULL)
|
||||
{
|
||||
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
|
||||
ViEId(_instanceId, videoChannel), "%s: No channel %d",
|
||||
__FUNCTION__, videoChannel);
|
||||
SetLastError(kViECodecInvalidChannelId);
|
||||
return -1;
|
||||
}
|
||||
return vieChannel->DiscardedPackets();
|
||||
}
|
||||
|
||||
// Callbacks
|
||||
// ----------------------------------------------------------------------------
|
||||
// SetKeyFrameRequestCallbackStatus
|
||||
|
@ -72,6 +72,8 @@ public:
|
||||
unsigned int& keyFrames,
|
||||
unsigned int& deltaFrames) const;
|
||||
|
||||
virtual unsigned int GetDiscardedPackets(const int videoChannel) const;
|
||||
|
||||
// Callbacks
|
||||
virtual int SetKeyFrameRequestCallbackStatus(const int videoChannel,
|
||||
const bool enable);
|
||||
|
Loading…
x
Reference in New Issue
Block a user