Introduced the new APM data logging functionality into the AEC echo_cancellation.* API layer.
BUG=webrtc:5298
Review-Url: https://codereview.webrtc.org/1952593002
Cr-Commit-Position: refs/heads/master@{#12635}
diff --git a/webrtc/modules/audio_processing/aec/echo_cancellation.cc b/webrtc/modules/audio_processing/aec/echo_cancellation.cc
index 6b1758a1..e1dba6e 100644
--- a/webrtc/modules/audio_processing/aec/echo_cancellation.cc
+++ b/webrtc/modules/audio_processing/aec/echo_cancellation.cc
@@ -14,9 +14,6 @@
#include "webrtc/modules/audio_processing/aec/echo_cancellation.h"
#include <math.h>
-#if WEBRTC_AEC_DEBUG_DUMP == 1
-#include <stdio.h>
-#endif
#include <stdlib.h>
#include <string.h>
@@ -27,14 +24,9 @@
#include "webrtc/modules/audio_processing/aec/aec_core.h"
#include "webrtc/modules/audio_processing/aec/aec_resampler.h"
#include "webrtc/modules/audio_processing/aec/echo_cancellation_internal.h"
+#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
#include "webrtc/typedefs.h"
-// Check to verify that the define is properly set.
-#if !defined(WEBRTC_AEC_DEBUG_DUMP) || \
- (WEBRTC_AEC_DEBUG_DUMP != 0 && WEBRTC_AEC_DEBUG_DUMP != 1)
-#error "Set WEBRTC_AEC_DEBUG_DUMP to either 0 or 1"
-#endif
-
namespace webrtc {
// Measured delays [ms]
@@ -127,11 +119,12 @@
int32_t skew);
void* WebRtcAec_Create() {
- Aec* aecpc = reinterpret_cast<Aec*>(malloc(sizeof(Aec)));
+ Aec* aecpc = new Aec();
if (!aecpc) {
return NULL;
}
+ aecpc->data_dumper.reset(new ApmDataDumper(aecpc->instance_count));
aecpc->aec = WebRtcAec_CreateAec(aecpc->instance_count);
if (!aecpc->aec) {
@@ -155,18 +148,7 @@
aecpc->initFlag = 0;
-#if WEBRTC_AEC_DEBUG_DUMP == 1
- char filename[64];
- snprintf(filename, sizeof(filename), "aec_buf%d.dat", aecpc->instance_count);
- aecpc->bufFile = fopen(filename, "wb");
- snprintf(filename, sizeof(filename), "aec_skew%d.dat", aecpc->instance_count);
- aecpc->skewFile = fopen(filename, "wb");
- snprintf(filename, sizeof(filename), "aec_delay%d.dat",
- aecpc->instance_count);
- aecpc->delayFile = fopen(filename, "wb");
-#endif
aecpc->instance_count++;
-
return aecpc;
}
@@ -179,19 +161,14 @@
WebRtc_FreeBuffer(aecpc->far_pre_buf);
-#if WEBRTC_AEC_DEBUG_DUMP == 1
- fclose(aecpc->bufFile);
- fclose(aecpc->skewFile);
- fclose(aecpc->delayFile);
-#endif
-
WebRtcAec_FreeAec(aecpc->aec);
WebRtcAec_FreeResampler(aecpc->resampler);
- free(aecpc);
+ delete aecpc;
}
int32_t WebRtcAec_Init(void* aecInst, int32_t sampFreq, int32_t scSampFreq) {
Aec* aecpc = reinterpret_cast<Aec*>(aecInst);
+ aecpc->data_dumper->InitiateNewSetOfRecordings();
AecConfig aecConfig;
if (sampFreq != 8000 && sampFreq != 16000 && sampFreq != 32000 &&
@@ -376,15 +353,9 @@
msInSndCardBuf, skew);
}
-#if WEBRTC_AEC_DEBUG_DUMP == 1
- {
- int16_t far_buf_size_ms = (int16_t)(WebRtcAec_system_delay(aecpc->aec) /
- (sampMsNb * aecpc->rate_factor));
- (void)fwrite(&far_buf_size_ms, 2, 1, aecpc->bufFile);
- (void)fwrite(&aecpc->knownDelay, sizeof(aecpc->knownDelay), 1,
- aecpc->delayFile);
- }
-#endif
+ int far_buf_size_samples = WebRtcAec_system_delay(aecpc->aec);
+ aecpc->data_dumper->DumpRaw("aec_system_delay", 1, &far_buf_size_samples);
+ aecpc->data_dumper->DumpRaw("aec_known_delay", 1, &aecpc->knownDelay);
return retVal;
}
@@ -603,9 +574,7 @@
aecpc->skew = maxSkewEst;
}
-#if WEBRTC_AEC_DEBUG_DUMP == 1
- (void)fwrite(&aecpc->skew, sizeof(aecpc->skew), 1, aecpc->skewFile);
-#endif
+ aecpc->data_dumper->DumpRaw("aec_skew", 1, &aecpc->skew);
}
}
diff --git a/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h b/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h
index 188fb57..5e79626 100644
--- a/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h
+++ b/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h
@@ -11,6 +11,8 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_INTERNAL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_INTERNAL_H_
+#include <memory>
+
extern "C" {
#include "webrtc/common_audio/ring_buffer.h"
}
@@ -18,7 +20,11 @@
namespace webrtc {
+class ApmDataDumper;
+
typedef struct Aec {
+ std::unique_ptr<ApmDataDumper> data_dumper;
+
int delayCtr;
int sampFreq;
int splitSampFreq;
@@ -45,12 +51,6 @@
int checkBuffSize;
short lastDelayDiff;
-#if WEBRTC_AEC_DEBUG_DUMP
- FILE* bufFile;
- FILE* delayFile;
- FILE* skewFile;
-#endif
-
// Structures
void* resampler;
diff --git a/webrtc/modules/audio_processing/logging/apm_data_dumper.h b/webrtc/modules/audio_processing/logging/apm_data_dumper.h
index 18f9e5e..93232b7 100644
--- a/webrtc/modules/audio_processing/logging/apm_data_dumper.h
+++ b/webrtc/modules/audio_processing/logging/apm_data_dumper.h
@@ -73,6 +73,32 @@
#endif
}
+ void DumpRaw(const char* name, int v_length, const int16_t* v) {
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+ FILE* file = GetRawFile(name);
+ fwrite(v, sizeof(v[0]), v_length, file);
+#endif
+ }
+
+ void DumpRaw(const char* name, rtc::ArrayView<const int16_t> v) {
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+ DumpRaw(name, v.size(), v.data());
+#endif
+ }
+
+ void DumpRaw(const char* name, int v_length, const int32_t* v) {
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+ FILE* file = GetRawFile(name);
+ fwrite(v, sizeof(v[0]), v_length, file);
+#endif
+ }
+
+ void DumpRaw(const char* name, rtc::ArrayView<const int32_t> v) {
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+ DumpRaw(name, v.size(), v.data());
+#endif
+ }
+
void DumpWav(const char* name,
int v_length,
const float* v,
