Replaced the data logging functionality in the AEC with a generic logging functionality

webrtc/modules/audio_processing/aec/aec_core.cc

Index: webrtc/modules/audio_processing/aec/aec_core.cc
diff --git a/webrtc/modules/audio_processing/aec/aec_core.cc b/webrtc/modules/audio_processing/aec/aec_core.cc
index 8849ad787b6c0faefb6f4d1e1693f4e6a5be175b..1c8441d0ba923492cc42cf44ded685c0b7050e00 100644
--- a/webrtc/modules/audio_processing/aec/aec_core.cc
+++ b/webrtc/modules/audio_processing/aec/aec_core.cc
@@ -14,10 +14,6 @@
 
 #include "webrtc/modules/audio_processing/aec/aec_core.h"
 
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-#include <stdio.h>
-#endif
-
 #include <algorithm>
 #include <assert.h>
 #include <math.h>
@@ -33,7 +29,6 @@ extern "C" {
 #include "webrtc/modules/audio_processing/aec/aec_common.h"
 #include "webrtc/modules/audio_processing/aec/aec_core_internal.h"
 #include "webrtc/modules/audio_processing/aec/aec_rdft.h"
-#include "webrtc/modules/audio_processing/logging/aec_logging.h"
 #include "webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h"
 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
 #include "webrtc/typedefs.h"
@@ -132,10 +127,6 @@ const float WebRtcAec_kNormalSmoothingCoefficients[2][2] = {{0.9f, 0.1f},
 // Number of partitions forming the NLP's "preferred" bands.
 enum { kPrefBandSize = 24 };
 
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-extern int webrtc_aec_instance_count;
-#endif
-
 WebRtcAecFilterFar WebRtcAec_FilterFar;
 WebRtcAecScaleErrorSignal WebRtcAec_ScaleErrorSignal;
 WebRtcAecFilterAdaptation WebRtcAec_FilterAdaptation;
@@ -206,7 +197,10 @@ void DivergentFilterFraction::Clear() {
 }
 
 // TODO(minyue): Moving some initialization from WebRtcAec_CreateAec() to ctor.
-AecCore::AecCore() = default;
+AecCore::AecCore(int instance_index)
+    : data_dumper(new ApmDataDumper(instance_index)) {}
+
+AecCore::~AecCore() {}
 
 static int CmpFloat(const void* a, const void* b) {
   const float* da = (const float*)a;
@@ -1309,15 +1303,10 @@ static void ProcessBlock(AecCore* aec) {
   WebRtc_ReadBuffer(aec->far_time_buf, reinterpret_cast<void**>(&farend_ptr),
                     farend, 1);
 
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-  {
-    // TODO(minyue): |farend_ptr| starts from buffered samples. This will be
-    // modified when |aec->far_time_buf| is revised.
-    RTC_AEC_DEBUG_WAV_WRITE(aec->farFile, &farend_ptr[PART_LEN], PART_LEN);
-
-    RTC_AEC_DEBUG_WAV_WRITE(aec->nearFile, nearend_ptr, PART_LEN);
-  }
-#endif
+  aec->data_dumper->DumpWav("aec_far", PART_LEN, &farend_ptr[PART_LEN],

[kwiberg-webrtc, 2016/05/03 00:53:00]

At each of these call sites, the compiler is converting the const char* to a
temporary std::string for you, even if the debug dumping is switched off. I
suspect the compiler may not even be allowed to optimize that away, since the
heap allocations can have the side effect of exhausting memory.

Make the dump methods take const char* instead, and convert to std::string in
the #if-protected method bodies if necessary.

[peah-webrtc, 2016/05/03 06:29:22]

Great point!!!
Done.

+                            aec->sampFreq > 16000 ? 16000 : aec->sampFreq, 1);
+  aec->data_dumper->DumpWav("aec_near", PART_LEN, nearend_ptr,
+                            aec->sampFreq > 16000 ? 16000 : aec->sampFreq, 1);

[kwiberg-webrtc, 2016/05/03 00:53:01]

std::min(aec->sampFreq, 16000)

(Also in several places below.)

[peah-webrtc, 2016/05/03 06:29:22]

Done.

 
   if (aec->metricsMode == 1) {
     // Update power levels
@@ -1417,7 +1406,8 @@ static void ProcessBlock(AecCore* aec) {
                   aec->xfBuf, nearend_ptr, aec->xPow, aec->wfBuf,
                   echo_subtractor_output);
 
-  RTC_AEC_DEBUG_WAV_WRITE(aec->outLinearFile, echo_subtractor_output, PART_LEN);
+  aec->data_dumper->DumpWav("aec_out_linear", PART_LEN, echo_subtractor_output,
+                            aec->sampFreq > 16000 ? 16000 : aec->sampFreq, 1);
 
   if (aec->metricsMode == 1) {
     UpdateLevel(&aec->linoutlevel,
@@ -1439,12 +1429,14 @@ static void ProcessBlock(AecCore* aec) {
     WebRtc_WriteBuffer(aec->outFrBufH[i], outputH[i], PART_LEN);
   }
 
-  RTC_AEC_DEBUG_WAV_WRITE(aec->outFile, output, PART_LEN);
+  aec->data_dumper->DumpWav("aec_out", PART_LEN, output,
+                            aec->sampFreq > 16000 ? 16000 : aec->sampFreq, 1);
 }
 
-AecCore* WebRtcAec_CreateAec() {
+AecCore* WebRtcAec_CreateAec(int instance_count) {
   int i;
-  AecCore* aec = new AecCore;
+  AecCore* aec = new AecCore(instance_count);
+
   if (!aec) {
     return NULL;
   }
@@ -1488,12 +1480,6 @@ AecCore* WebRtcAec_CreateAec() {
     return NULL;
   }
 
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-  aec->instance_index = webrtc_aec_instance_count;
-
-  aec->farFile = aec->nearFile = aec->outFile = aec->outLinearFile = NULL;
-  aec->debug_dump_count = 0;
-#endif
   aec->delay_estimator_farend =
       WebRtc_CreateDelayEstimatorFarend(PART_LEN1, kHistorySizeBlocks);
   if (aec->delay_estimator_farend == NULL) {
@@ -1571,12 +1557,6 @@ void WebRtcAec_FreeAec(AecCore* aec) {
 
   WebRtc_FreeBuffer(aec->far_time_buf);
 
-  RTC_AEC_DEBUG_WAV_CLOSE(aec->farFile);
-  RTC_AEC_DEBUG_WAV_CLOSE(aec->nearFile);
-  RTC_AEC_DEBUG_WAV_CLOSE(aec->outFile);
-  RTC_AEC_DEBUG_WAV_CLOSE(aec->outLinearFile);
-  RTC_AEC_DEBUG_RAW_CLOSE(aec->e_fft_file);
-
   WebRtc_FreeDelayEstimator(aec->delay_estimator);
   WebRtc_FreeDelayEstimatorFarend(aec->delay_estimator_farend);
 
@@ -1621,6 +1601,7 @@ static void SetErrorThreshold(AecCore* aec) {
 
 int WebRtcAec_InitAec(AecCore* aec, int sampFreq) {
   int i;
+  aec->data_dumper->InitiateNewSetOfRecordings();
 
   aec->sampFreq = sampFreq;
 
@@ -1643,27 +1624,6 @@ int WebRtcAec_InitAec(AecCore* aec, int sampFreq) {
   // Initialize far-end buffers.
   WebRtc_InitBuffer(aec->far_time_buf);
 
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-  {
-    int process_rate = sampFreq > 16000 ? 16000 : sampFreq;
-    RTC_AEC_DEBUG_WAV_REOPEN("aec_far", aec->instance_index,
-                             aec->debug_dump_count, process_rate,
-                             &aec->farFile);
-    RTC_AEC_DEBUG_WAV_REOPEN("aec_near", aec->instance_index,
-                             aec->debug_dump_count, process_rate,
-                             &aec->nearFile);
-    RTC_AEC_DEBUG_WAV_REOPEN("aec_out", aec->instance_index,
-                             aec->debug_dump_count, process_rate,
-                             &aec->outFile);
-    RTC_AEC_DEBUG_WAV_REOPEN("aec_out_linear", aec->instance_index,
-                             aec->debug_dump_count, process_rate,
-                             &aec->outLinearFile);
-  }
-
-  RTC_AEC_DEBUG_RAW_OPEN("aec_e_fft", aec->debug_dump_count, &aec->e_fft_file);
-
-  ++aec->debug_dump_count;
-#endif
   aec->system_delay = 0;
 
   if (WebRtc_InitDelayEstimatorFarend(aec->delay_estimator_farend) != 0) {