Index: webrtc/modules/audio_processing/logging/apm_data_dumper.h |
diff --git a/webrtc/modules/audio_processing/logging/apm_data_dumper.h b/webrtc/modules/audio_processing/logging/apm_data_dumper.h |
index 18f9e5e18153282110c046a5cbf449a876107144..93232b7c9dc44b6add54730b9c59181cc6d96279 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 @@ class ApmDataDumper { |
#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, |