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Cleaning up NetEQ statistics

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Removed struct MCUStats_t and all references to it.
Removed totalDiscardedPackets and totalFlushedPackets
from the PacketBuf_t struct.

Review URL: http://webrtc-codereview.appspot.com/293001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@999 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org 2011-11-23 11:06:05 +00:00
parent df10de4b27
commit 89ab652250
8 changed files with 2 additions and 189 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
src/modules/audio_coding/neteq/mcu.h
View File

@ -82,8 +82,6 @@ typedef struct
WebRtc_Word16 TSscalingInitialized;
enum TsScaling scalingFactor;
MCUStats_t statInst;
#ifdef NETEQ_STEREO
int usingStereo;
#endif

20
src/modules/audio_coding/neteq/mcu_reset.c
View File

@ -88,26 +88,6 @@ int WebRtcNetEQ_ResetMcuInCallStats(MCUInst_t *inst)
int WebRtcNetEQ_ResetMcuJitterStat(MCUInst_t *inst)
{
inst->statInst.jbAvgCount = 0;
inst->statInst.jbAvgSizeQ16 = 0;
inst->statInst.jbMaxSize = 0;
inst->statInst.jbMinSize = 0xFFFFFFFF;
inst->statInst.avgPacketCount = 0;
inst->statInst.avgPacketDelayMs = 0;
inst->statInst.minPacketDelayMs = 0xFFFFFFFF;
inst->statInst.maxPacketDelayMs = 0;
inst->statInst.jbChangeCount = 0;
inst->statInst.generatedSilentMs = 0;
inst->statInst.countExpandMoreThan120ms = 0;
inst->statInst.countExpandMoreThan250ms = 0;
inst->statInst.countExpandMoreThan500ms = 0;
inst->statInst.countExpandMoreThan2000ms = 0;
inst->statInst.longestExpandDurationMs = 0;
inst->statInst.accelerateMs = 0;
inst->PacketBuffer_inst.totalDiscardedPackets = 0;
inst->PacketBuffer_inst.totalFlushedPackets = 0;
inst->BufferStat_inst.Automode_inst.countIAT500ms = 0;
inst->BufferStat_inst.Automode_inst.countIAT1000ms = 0;
inst->BufferStat_inst.Automode_inst.countIAT2000ms = 0;

27
src/modules/audio_coding/neteq/neteq_statistics.h
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@ -36,32 +36,5 @@ typedef struct
} DSPStats_t;
/*
* Statistics struct on MCU side
* All variables are for post-call statistics; queried through WebRtcNetEQ_GetJitterStatistics.
*/
typedef struct
{
WebRtc_UWord32 jbMinSize; /* smallest Jitter Buffer size during call in ms */
WebRtc_UWord32 jbMaxSize; /* largest Jitter Buffer size during call in ms */
WebRtc_UWord32 jbAvgSizeQ16; /* the average JB size, measured over time in ms (Q16) */
WebRtc_UWord16 jbAvgCount; /* help counter for jbAveSize */
WebRtc_UWord32 minPacketDelayMs; /* min time incoming packet "waited" to be played in ms */
WebRtc_UWord32 maxPacketDelayMs; /* max time incoming packet "waited" to be played in ms */
WebRtc_UWord16 avgPacketDelayMs; /* avg time incoming packet "waited" to be played in ms */
WebRtc_UWord16 avgPacketCount; /* help counter for avgPacketDelayMs */
WebRtc_UWord32 jbChangeCount; /* count number of successful accelerate and pre-emptive
expand operations */
WebRtc_UWord32 generatedSilentMs; /* generated silence in ms */
WebRtc_UWord32 countExpandMoreThan120ms; /* count of tiny expansions */
WebRtc_UWord32 countExpandMoreThan250ms; /* count of small expansions */
WebRtc_UWord32 countExpandMoreThan500ms; /* count of medium expansions */
WebRtc_UWord32 countExpandMoreThan2000ms; /* count of large expansions */
WebRtc_UWord32 longestExpandDurationMs; /* duration of longest expansions in ms */
WebRtc_UWord32 accelerateMs; /* audio data removed through accelerate in ms */
} MCUStats_t;
#endif

8
src/modules/audio_coding/neteq/packet_buffer.c
View File

@ -96,8 +96,6 @@ int WebRtcNetEQ_PacketBufferInit(PacketBuf_t *bufferInst, int maxNoOfPackets,
/* Reset buffer statistics */
bufferInst->discardedPackets = 0;
bufferInst->totalDiscardedPackets = 0;
bufferInst->totalFlushedPackets = 0;
return (0);
}
@ -116,9 +114,6 @@ int WebRtcNetEQ_PacketBufferFlush(PacketBuf_t *bufferInst)
return (0);
}
/* Increase flush counter */
bufferInst->totalFlushedPackets += bufferInst->numPacketsInBuffer;
/* Set all payload lengths to zero */
WebRtcSpl_MemSetW16(bufferInst->payloadLengthBytes, 0, bufferInst->maxInsertPositions);
@ -437,9 +432,8 @@ int WebRtcNetEQ_PacketBufferFindLowestTimestamp(PacketBuf_t *bufferInst,
/* Reduce packet counter by one */
bufferInst->numPacketsInBuffer--;
/* Increase discard counter for in-call and post-call statistics */
/* Increase discard counter for in-call statistics */
bufferInst->discardedPackets++;
bufferInst->totalDiscardedPackets++;
}
else if (((newDiff < timeStampDiff) || ((newDiff == timeStampDiff)
&& (bufferInst->rcuPlCntr[i] < rcuPlCntr))) && (bufferInst->payloadLengthBytes[i]

4
src/modules/audio_coding/neteq/packet_buffer.h
View File

@ -50,10 +50,8 @@ typedef struct
WebRtc_Word16 *rcuPlCntr; /* zero for non-RCU payload, 1 for main payload
2 for redundant payload */
/* Statistics counters */
/* Statistics counter */
WebRtc_UWord16 discardedPackets; /* Number of discarded packets */
WebRtc_UWord32 totalDiscardedPackets; /* Total number of discarded packets */
WebRtc_UWord32 totalFlushedPackets; /* Total number of flushed packets */
} PacketBuf_t;

45
src/modules/audio_coding/neteq/recin.c
View File

@ -33,7 +33,6 @@ int WebRtcNetEQ_RecInInternal(MCUInst_t *MCU_inst, RTPPacket_t *RTPpacketInput,
int i_ok = 0, i_No_Of_Payloads = 1;
WebRtc_Word16 flushed = 0;
WebRtc_Word16 codecPos;
WebRtc_UWord32 diffTS, uw32_tmp;
int curr_Codec;
WebRtc_Word16 isREDPayload = 0;
WebRtc_Word32 temp_bufsize = MCU_inst->PacketBuffer_inst.numPacketsInBuffer;
@ -249,50 +248,6 @@ int WebRtcNetEQ_RecInInternal(MCUInst_t *MCU_inst, RTPPacket_t *RTPpacketInput,
MCU_inst->new_codec = 1;
}
/* update post-call statistics */
if (MCU_inst->new_codec != 1) /* not if in codec change */
{
diffTS = RTPpacket[i_k].timeStamp - MCU_inst->timeStamp; /* waiting time */
if (diffTS < 0x0FFFFFFF) /* guard against re-ordering */
{
/* waiting time in ms */
diffTS = WEBRTC_SPL_UDIV(diffTS,
WEBRTC_SPL_UDIV((WebRtc_UWord32) MCU_inst->fs, 1000) );
if (diffTS < MCU_inst->statInst.minPacketDelayMs)
{
/* new all-time low */
MCU_inst->statInst.minPacketDelayMs = diffTS;
}
if (diffTS > MCU_inst->statInst.maxPacketDelayMs)
{
/* new all-time high */
MCU_inst->statInst.maxPacketDelayMs = diffTS;
}
/* Update avg waiting time:
* avgPacketDelayMs =
* (avgPacketCount * avgPacketDelayMs + diffTS)/(avgPacketCount+1)
* with proper rounding.
*/
uw32_tmp
= WEBRTC_SPL_UMUL((WebRtc_UWord32) MCU_inst->statInst.avgPacketCount,
(WebRtc_UWord32) MCU_inst->statInst.avgPacketDelayMs);
uw32_tmp
= WEBRTC_SPL_ADD_SAT_W32(uw32_tmp,
(diffTS + (MCU_inst->statInst.avgPacketCount>>1)));
uw32_tmp = WebRtcSpl_DivU32U16(uw32_tmp,
(WebRtc_UWord16) (MCU_inst->statInst.avgPacketCount + 1));
MCU_inst->statInst.avgPacketDelayMs
= (WebRtc_UWord16) WEBRTC_SPL_MIN(uw32_tmp, (WebRtc_UWord32) 65535);
/* increase counter, but not to more than 65534 */
if (MCU_inst->statInst.avgPacketCount < (0xFFFF - 1))
{
MCU_inst->statInst.avgPacketCount++;
}
}
}
/* Parse the payload and insert it into the buffer */
i_ok = WebRtcNetEQ_SplitAndInsertPayload(&RTPpacket[i_k],
&MCU_inst->PacketBuffer_inst, &MCU_inst->PayloadSplit_inst, &flushed);

79
src/modules/audio_coding/neteq/signal_mcu.c
View File

@ -227,48 +227,11 @@ int WebRtcNetEQ_SignalMcu(MCUInst_t *inst)
/* Subtract (dspInfo.samplesLeft + inst->timestampsPerCall) from sampleMemory */
inst->BufferStat_inst.Automode_inst.sampleMemory -= dspInfo.samplesLeft
+ inst->timestampsPerCall;
/* Update post-call statistics */
inst->statInst.jbChangeCount++;
}
/* calculate total current buffer size (in ms*8), including sync buffer */
w32_bufsize = WebRtcSpl_DivW32W16((w32_bufsize + dspInfo.samplesLeft), fs_mult);
if (((WebRtc_UWord32) w32_bufsize >> 3) < inst->statInst.jbMinSize)
{
/* new all-time low */
inst->statInst.jbMinSize = ((WebRtc_UWord32) w32_bufsize >> 3); /* shift to ms */
}
if (((WebRtc_UWord32) w32_bufsize >> 3) > inst->statInst.jbMaxSize)
{
/* new all-time high */
inst->statInst.jbMaxSize = ((WebRtc_UWord32) w32_bufsize >> 3); /* shift to ms */
}
/* Update avg bufsize:
* jbAvgSize = (jbAvgCount * jbAvgSize + w32_bufsize/8)/(jbAvgCount+1)
* with proper rounding
*/
{
WebRtc_Word32 avgTmp;
/* Simplify the above formula to:
* jbAvgSizeQ16 =
* jbAvgSizeQ16 + ( (w32_bufsize/8 << 16) - jbAvgSizeQ16 + d ) / (jbAvgCount+1)
* where d = jbAvgCount/2 for proper rounding.
*/
avgTmp = (((WebRtc_UWord32) w32_bufsize >> 3) << 16) - inst->statInst.jbAvgSizeQ16;
avgTmp = WEBRTC_SPL_DIV( avgTmp + (inst->statInst.jbAvgCount>>1),
inst->statInst.jbAvgCount + 1 );
inst->statInst.jbAvgSizeQ16 += avgTmp;
if (inst->statInst.jbAvgCount < (0xFFFF - 1))
{
inst->statInst.jbAvgCount++;
}
}
#ifdef NETEQ_ATEVENT_DECODE
/* DTMF data will affect the decision */
if (WebRtcNetEQ_DtmfDecode(&inst->DTMF_inst, blockPtr + 1, blockPtr + 2,
@ -317,36 +280,6 @@ int WebRtcNetEQ_SignalMcu(MCUInst_t *inst)
}
else
{
/* update post-call statistics */
WebRtc_UWord32
expandTime =
WEBRTC_SPL_UDIV(WEBRTC_SPL_UMUL_32_16(
WEBRTC_SPL_UMUL_32_16((WebRtc_UWord32) inst->NoOfExpandCalls,
(WebRtc_UWord16) 1000),
inst->timestampsPerCall), inst->fs); /* expand time in ms */
if (expandTime > 2000)
{
inst->statInst.countExpandMoreThan2000ms++;
}
else if (expandTime > 500)
{
inst->statInst.countExpandMoreThan500ms++;
}
else if (expandTime > 250)
{
inst->statInst.countExpandMoreThan250ms++;
}
else if (expandTime > 120)
{
inst->statInst.countExpandMoreThan120ms++;
}
if (expandTime > inst->statInst.longestExpandDurationMs)
{
inst->statInst.longestExpandDurationMs = expandTime;
}
/* reset counter */
inst->NoOfExpandCalls = 0;
}
@ -542,12 +475,6 @@ int WebRtcNetEQ_SignalMcu(MCUInst_t *inst)
inst->timeStamp = dspInfo.playedOutTS + timeStampJump;
/* update post-call statistics (since we will reset the CNG counter) */
inst->statInst.generatedSilentMs
+= WEBRTC_SPL_UDIV(
WEBRTC_SPL_UMUL_32_16(inst->BufferStat_inst.uw32_CNGplayedTS, (WebRtc_UWord16) 1000),
inst->fs);
inst->BufferStat_inst.uw32_CNGplayedTS = 0;
inst->NoOfExpandCalls = 0;
@ -685,12 +612,6 @@ int WebRtcNetEQ_SignalMcu(MCUInst_t *inst)
inst->BufferStat_inst.w16_cngOn = CNG_OFF;
}
/* update post-call statistics */
inst->statInst.generatedSilentMs
+= WEBRTC_SPL_UDIV(
WEBRTC_SPL_UMUL_32_16(inst->BufferStat_inst.uw32_CNGplayedTS, (WebRtc_UWord16) 1000),
inst->fs);
/*
* Reset CNG timestamp as a new packet will be delivered.
* (Also if CNG packet, since playedOutTS is updated.)

6
src/modules/audio_coding/neteq/webrtc_neteq.c
View File

@ -1398,12 +1398,6 @@ int WebRtcNetEQ_GetNetworkStatistics(void *inst, WebRtcNetEQ_NetworkStatistics *
stats->currentAccelerateRate = 1 << 14; /* 1 in Q14 */
}
/* also transfer measure to post-call statistics */
NetEqMainInst->MCUinst.statInst.accelerateMs
+= WebRtcSpl_DivU32U16(
WEBRTC_SPL_UMUL_32_16( NetEqMainInst->DSPinst.statInst.accelerateLength, (WebRtc_UWord16) 1000),
NetEqMainInst->MCUinst.fs);
/* timestamps elapsed since last report */
tempU32 = NetEqMainInst->MCUinst.lastReportTS;