Add number of inserted samples to NetEq statistics.

BUG=

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3289 4adac7df-926f-26a2-2b94-8c16560cd09d

DEPS

@@ -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

webrtc/common_types.h

@@ -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

webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h

@@ -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;
 
 ///////////////////////////////////////////////////////////////////////////

webrtc/modules/audio_coding/main/source/acm_neteq.cc

@@ -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;

webrtc/modules/audio_coding/neteq/dsp.c

@@ -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);
 }
 

webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_internal.h

@@ -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;
 
 /*

webrtc/modules/audio_coding/neteq/neteq_statistics.h

@@ -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 */

webrtc/modules/audio_coding/neteq/recout.c

@@ -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;

webrtc/modules/audio_coding/neteq/webrtc_neteq.c

@@ -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 */
     /*******************/

webrtc/voice_engine/test/auto_test/standard/neteq_stats_test.cc

@@ -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);
 }