753eada3aa
Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1 Update notes: * Pull in third party license file * Replace .gypi files with BUILD.gn to keep track of what changes upstream * Bunch of new filse pulled in as dependencies * Won't build yet due to changes needed on top of these
87 lines
3.1 KiB
C
87 lines
3.1 KiB
C
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_LOGGING_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_LOGGING_
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#include <stdio.h>
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#include "webrtc/modules/audio_processing/logging/aec_logging_file_handling.h"
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// To enable AEC logging, invoke GYP with -Daec_debug_dump=1.
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#ifdef WEBRTC_AEC_DEBUG_DUMP
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// Dumps a wav data to file.
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#define RTC_AEC_DEBUG_WAV_WRITE(file, data, num_samples) \
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do { \
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rtc_WavWriteSamples(file, data, num_samples); \
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} while (0)
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// (Re)opens a wav file for writing using the specified sample rate.
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#define RTC_AEC_DEBUG_WAV_REOPEN(name, instance_index, process_rate, \
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sample_rate, wav_file) \
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do { \
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WebRtcAec_ReopenWav(name, instance_index, process_rate, sample_rate, \
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wav_file); \
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} while (0)
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// Closes a wav file.
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#define RTC_AEC_DEBUG_WAV_CLOSE(wav_file) \
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do { \
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rtc_WavClose(wav_file); \
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} while (0)
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// Dumps a raw data to file.
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#define RTC_AEC_DEBUG_RAW_WRITE(file, data, data_size) \
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do { \
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(void) fwrite(data, data_size, 1, file); \
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} while (0)
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// Opens a raw data file for writing using the specified sample rate.
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#define RTC_AEC_DEBUG_RAW_OPEN(name, instance_counter, file) \
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do { \
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WebRtcAec_RawFileOpen(name, instance_counter, file); \
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} while (0)
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// Closes a raw data file.
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#define RTC_AEC_DEBUG_RAW_CLOSE(file) \
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do { \
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fclose(file); \
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} while (0)
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#else // RTC_AEC_DEBUG_DUMP
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#define RTC_AEC_DEBUG_WAV_WRITE(file, data, num_samples) \
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do { \
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} while (0)
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#define RTC_AEC_DEBUG_WAV_REOPEN(wav_file, name, instance_index, process_rate, \
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sample_rate) \
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do { \
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} while (0)
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#define RTC_AEC_DEBUG_WAV_CLOSE(wav_file) \
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do { \
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} while (0)
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#define RTC_AEC_DEBUG_RAW_WRITE(file, data, data_size) \
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do { \
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} while (0)
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#define RTC_AEC_DEBUG_RAW_OPEN(file, name, instance_counter) \
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do { \
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} while (0)
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#define RTC_AEC_DEBUG_RAW_CLOSE(file) \
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do { \
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} while (0)
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#endif // WEBRTC_AEC_DEBUG_DUMP
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_LOGGING_
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