Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(182)

Unified Diff: webrtc/modules/audio_processing/logging/aec_logging_file_handling.h

Issue 1272403003: Replaced the wav file dumping functionality in aec_core.c with the newly added corresponding macros (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Added logging using the raw variant of the new aec logging macros Created 5 years, 4 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
Index: webrtc/modules/audio_processing/logging/aec_logging_file_handling.h
diff --git a/webrtc/modules/audio_processing/logging/aec_logging_file_handling.h b/webrtc/modules/audio_processing/logging/aec_logging_file_handling.h
new file mode 100644
index 0000000000000000000000000000000000000000..d21029217c9a8f6f7660fdd829277c6696436855
--- /dev/null
+++ b/webrtc/modules/audio_processing/logging/aec_logging_file_handling.h
@@ -0,0 +1,35 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_LOGGING_FILE_HANDLING_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_LOGGING_FILE_HANDLING_
+
+// To enable AEC logging, run this command from trunk/ :
+// python webrtc/build/gyp_webrtc -Daec_debug_dump=1
hlundin-webrtc 2015/08/10 13:04:40 I would rephrase this as: To enable AEC logging, i
Andrew MacDonald 2015/08/10 15:28:44 I like this better than my suggestion.
peah-webrtc 2015/08/12 20:25:51 Done.
peah-webrtc 2015/08/12 20:25:51 Done.
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+#include <stdio.h>
+#include "webrtc/common_audio/wav_file.h"
+#include "webrtc/typedefs.h"
+
+
+// (Re)open wav file using the specified sample rate
hlundin-webrtc 2015/08/10 13:04:40 (Re)opens... http://google-styleguide.googlecode.
hlundin-webrtc 2015/08/10 13:04:41 What do the seq1 and seq2 parameters do?
hlundin-webrtc 2015/08/10 13:04:41 End comment with period (.) http://google-stylegui
peah-webrtc 2015/08/12 20:25:51 Done. (In response to another comment I changed th
peah-webrtc 2015/08/12 20:25:51 Done.
peah-webrtc 2015/08/12 20:25:51 Done.
+void WebRtcAec_ReopenWav(rtc_WavWriter** wav_file,
+ const char* const name,
+ const int seq1,
hlundin-webrtc 2015/08/10 13:04:41 We typically don't qualify parameters passed by va
peah-webrtc 2015/08/12 20:25:51 Done.
+ const int seq2,
+ const int sample_rate);
+
+// Open dumpfile with instance-specific filename
hlundin-webrtc 2015/08/10 13:04:41 Opens... End with .
peah-webrtc 2015/08/12 20:25:51 Done.
+void WebRtcAec_RawFileOpen(FILE** file,
+ const char * const name,
hlundin-webrtc 2015/08/10 13:04:41 Connect the type with the *. const char* const nam
peah-webrtc 2015/08/12 20:25:51 Done.
+ const int16_t instanceCtr);
hlundin-webrtc 2015/08/10 13:04:40 Variable names should be all lowercase, with under
hlundin-webrtc 2015/08/10 13:04:41 Drop const.
hlundin-webrtc 2015/08/10 13:04:41 Why int16_t? Almost always use int. http://google-
peah-webrtc 2015/08/12 20:25:51 Done.
peah-webrtc 2015/08/12 20:25:51 Done.
peah-webrtc 2015/08/12 20:25:51 Done.
+
+#endif // WEBRTC_AEC_DEBUG_DUMP
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_LOGGING_FILE_HANDLING_

Powered by Google App Engine
This is Rietveld 408576698