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Add number of inserted samples to NetEq statistics.

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BUG=

Review URL: https://webrtc-codereview.appspot.com/964030

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3289 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
roosa@google.com
2012-12-14 00:06:18 +00:00
parent c454fab03b
commit b8ba4d8109
10 changed files with 17 additions and 4 deletions
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2
DEPS
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@ -14,7 +14,7 @@ vars = {
# External resources like video and audio files used for testing purposes.
# Downloaded on demand when needed.
"webrtc_resources_revision": "11",
"webrtc_resources_revision": "12",
}
# NOTE: Prefer revision numbers to tags for svn deps. Use http rather than

2
webrtc/common_types.h
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@ -314,6 +314,8 @@ struct NetworkStatistics // NETEQ statistics
int minWaitingTimeMs;
// max packet waiting time in the jitter buffer (ms)
int maxWaitingTimeMs;
// added samples in off mode due to packet loss
int addedSamples;
};
typedef struct

2
webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h
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@ -165,6 +165,7 @@ enum ACMAMRPackingFormat {
// -medianWaitingTimeMs : median packet waiting time in the buffer
// -minWaitingTimeMs : min packet waiting time in the buffer
// -maxWaitingTimeMs : max packet waiting time in the buffer
// -addedSamples : samples inserted because of packet loss in off mode
typedef struct {
WebRtc_UWord16 currentBufferSize;
WebRtc_UWord16 preferredBufferSize;
@ -179,6 +180,7 @@ typedef struct {
int medianWaitingTimeMs;
int minWaitingTimeMs;
int maxWaitingTimeMs;
int addedSamples;
} ACMNetworkStatistics;
///////////////////////////////////////////////////////////////////////////

1
webrtc/modules/audio_coding/main/source/acm_neteq.cc
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@ -366,6 +366,7 @@ WebRtc_Word32 ACMNetEQ::NetworkStatistics(
statistics->currentPreemptiveRate = stats.currentPreemptiveRate;
statistics->preferredBufferSize = stats.preferredBufferSize;
statistics->clockDriftPPM = stats.clockDriftPPM;
statistics->addedSamples = stats.addedSamples;
} else {
LogError("getNetworkStatistics", 0);
return -1;

4
webrtc/modules/audio_coding/neteq/dsp.c
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@ -365,12 +365,11 @@ int WebRtcNetEQ_AddressInit(DSPInst_t *inst, const void *data2McuAddress,
int WebRtcNetEQ_ClearInCallStats(DSPInst_t *inst)
{
/* Reset statistics counters */
inst->statInst.accelerateLength = 0;
inst->statInst.expandLength = 0;
inst->statInst.preemptiveLength = 0;
inst->statInst.addedSamples = 0;
return (0);
}
@ -394,7 +393,6 @@ int WebRtcNetEQ_ClearPostCallStats(DSPInst_t *inst)
/* Reset statistics counters */
inst->statInst.expandedVoiceSamples = 0;
inst->statInst.expandedNoiseSamples = 0;
return (0);
}

2
webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_internal.h
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@ -107,6 +107,8 @@ typedef struct
* acceleration (in Q14). */
int32_t clockDriftPPM; /* Average clock-drift in parts-per-
* million (positive or negative). */
int addedSamples; /* Number of zero samples added in off
* mode */
} WebRtcNetEQ_NetworkStatistics;
/*

1
webrtc/modules/audio_coding/neteq/neteq_statistics.h
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@ -28,6 +28,7 @@ typedef struct
WebRtc_UWord32 preemptiveLength; /* number of samples produced through pre-emptive
expand */
WebRtc_UWord32 accelerateLength; /* number of samples removed through accelerate */
int addedSamples; /* number of samples inserted in off mode */
/* variables for post-call statistics; queried through WebRtcNetEQ_GetJitterStatistics */
WebRtc_UWord32 expandedVoiceSamples; /* number of voice samples produced through expand */

2
webrtc/modules/audio_coding/neteq/recout.c
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@ -1090,6 +1090,8 @@ int WebRtcNetEQ_RecOutInternal(DSPInst_t *inst, WebRtc_Word16 *pw16_outData,
len = inst->timestampsPerCall;
/* ZeroStuffing... */
WebRtcSpl_MemSetW16(pw16_NetEqAlgorithm_buffer, 0, len);
/* By not advancing the timestamp, NetEq inserts samples. */
inst->statInst.addedSamples += len;
}
inst->ExpandInst.w16_consecExp = 0;
break;

2
webrtc/modules/audio_coding/neteq/webrtc_neteq.c
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@ -1171,6 +1171,8 @@ int WebRtcNetEQ_GetNetworkStatistics(void *inst, WebRtcNetEQ_NetworkStatistics *
/* Instance sanity */
if (NetEqMainInst == NULL) return (-1);
stats->addedSamples = NetEqMainInst->DSPinst.statInst.addedSamples;
/*******************/
/* Get buffer size */
/*******************/

3
webrtc/voice_engine/test/auto_test/standard/neteq_stats_test.cc
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@ -49,4 +49,7 @@ TEST_F(NetEQStatsTest, ManualPrintStatisticsAfterRunningAWhile) {
network_statistics.minWaitingTimeMs);
TEST_LOG(" maxWaitingTimeMs = %i \n",
network_statistics.maxWaitingTimeMs);
// This is only set to a non-zero value in off-mode.
EXPECT_EQ(0, network_statistics.addedSamples);
}