• Home
  • Line#
  • Scopes#
  • Navigate#
  • Raw
  • Download
1 /*
2  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3  *
4  *  Use of this source code is governed by a BSD-style license
5  *  that can be found in the LICENSE file in the root of the source
6  *  tree. An additional intellectual property rights grant can be found
7  *  in the file PATENTS.  All contributing project authors may
8  *  be found in the AUTHORS file in the root of the source tree.
9  */
10 
11 #include "webrtc/modules/audio_processing/logging/aec_logging_file_handling.h"
12 
13 #include <stdint.h>
14 #include <stdio.h>
15 
16 #include "webrtc/base/checks.h"
17 #include "webrtc/base/stringutils.h"
18 #include "webrtc/common_audio/wav_file.h"
19 #include "webrtc/typedefs.h"
20 
21 #ifdef WEBRTC_AEC_DEBUG_DUMP
WebRtcAec_ReopenWav(const char * name,int instance_index,int process_rate,int sample_rate,rtc_WavWriter ** wav_file)22 void WebRtcAec_ReopenWav(const char* name,
23                          int instance_index,
24                          int process_rate,
25                          int sample_rate,
26                          rtc_WavWriter** wav_file) {
27   if (*wav_file) {
28     if (rtc_WavSampleRate(*wav_file) == sample_rate)
29       return;
30     rtc_WavClose(*wav_file);
31   }
32   char filename[64];
33   int written = rtc::sprintfn(filename, sizeof(filename), "%s%d-%d.wav", name,
34                               instance_index, process_rate);
35 
36   // Ensure there was no buffer output error.
37   RTC_DCHECK_GE(written, 0);
38   // Ensure that the buffer size was sufficient.
39   RTC_DCHECK_LT(static_cast<size_t>(written), sizeof(filename));
40 
41   *wav_file = rtc_WavOpen(filename, sample_rate, 1);
42 }
43 
WebRtcAec_RawFileOpen(const char * name,int instance_index,FILE ** file)44 void WebRtcAec_RawFileOpen(const char* name, int instance_index, FILE** file) {
45   char filename[64];
46   int written = rtc::sprintfn(filename, sizeof(filename), "%s_%d.dat", name,
47                               instance_index);
48 
49   // Ensure there was no buffer output error.
50   RTC_DCHECK_GE(written, 0);
51   // Ensure that the buffer size was sufficient.
52   RTC_DCHECK_LT(static_cast<size_t>(written), sizeof(filename));
53 
54   *file = fopen(filename, "wb");
55 }
56 
57 #endif  // WEBRTC_AEC_DEBUG_DUMP
58