1 files changed, 86 insertions, 0 deletions
diff --git a/third_party/webrtc/src/webrtc/modules/audio_processing/logging/aec_logging.h b/third_party/webrtc/src/webrtc/modules/audio_processing/logging/aec_logging.h
new file mode 100644
index 00000000..3cf9ff89
--- /dev/null
+++ b/third_party/webrtc/src/webrtc/modules/audio_processing/logging/aec_logging.h
@@ -0,0 +1,86 @@
+/*
+ *  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_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_LOGGING_
+
+#include <stdio.h>
+
+#include "webrtc/modules/audio_processing/logging/aec_logging_file_handling.h"
+
+// To enable AEC logging, invoke GYP with -Daec_debug_dump=1.
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+// Dumps a wav data to file.
+#define RTC_AEC_DEBUG_WAV_WRITE(file, data, num_samples) \
+  do {                                                   \
+    rtc_WavWriteSamples(file, data, num_samples);        \
+  } while (0)
+
+// (Re)opens a wav file for writing using the specified sample rate.
+#define RTC_AEC_DEBUG_WAV_REOPEN(name, instance_index, process_rate,     \
+                                 sample_rate, wav_file)                  \
+  do {                                                                   \
+    WebRtcAec_ReopenWav(name, instance_index, process_rate, sample_rate, \
+                        wav_file);                                       \
+  } while (0)
+
+// Closes a wav file.
+#define RTC_AEC_DEBUG_WAV_CLOSE(wav_file) \
+  do {                                    \
+    rtc_WavClose(wav_file);               \
+  } while (0)
+
+// Dumps a raw data to file.
+#define RTC_AEC_DEBUG_RAW_WRITE(file, data, data_size) \
+  do {                                                 \
+    (void) fwrite(data, data_size, 1, file);           \
+  } while (0)
+
+// Opens a raw data file for writing using the specified sample rate.
+#define RTC_AEC_DEBUG_RAW_OPEN(name, instance_counter, file) \
+  do {                                                       \
+    WebRtcAec_RawFileOpen(name, instance_counter, file);     \
+  } while (0)
+
+// Closes a raw data file.
+#define RTC_AEC_DEBUG_RAW_CLOSE(file) \
+  do {                                \
+    fclose(file);                     \
+  } while (0)
+
+#else  // RTC_AEC_DEBUG_DUMP
+#define RTC_AEC_DEBUG_WAV_WRITE(file, data, num_samples) \
+  do {                                                   \
+  } while (0)
+
+#define RTC_AEC_DEBUG_WAV_REOPEN(wav_file, name, instance_index, process_rate, \
+                                 sample_rate)                                  \
+  do {                                                                         \
+  } while (0)
+
+#define RTC_AEC_DEBUG_WAV_CLOSE(wav_file) \
+  do {                                    \
+  } while (0)
+
+#define RTC_AEC_DEBUG_RAW_WRITE(file, data, data_size) \
+  do {                                                 \
+  } while (0)
+
+#define RTC_AEC_DEBUG_RAW_OPEN(file, name, instance_counter) \
+  do {                                                       \
+  } while (0)
+
+#define RTC_AEC_DEBUG_RAW_CLOSE(file) \
+  do {                                \
+  } while (0)
+
+#endif  // WEBRTC_AEC_DEBUG_DUMP
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_LOGGING_