mirror of
https://github.com/danog/libtgvoip.git
synced 2024-12-02 17:51:06 +01:00
5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
76 lines
2.1 KiB
C++
76 lines
2.1 KiB
C++
/*
|
|
* Copyright (c) 2016 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.
|
|
*/
|
|
|
|
#include "modules/audio_processing/logging/apm_data_dumper.h"
|
|
|
|
#include "rtc_base/strings/string_builder.h"
|
|
|
|
// Check to verify that the define is properly set.
|
|
#if !defined(WEBRTC_APM_DEBUG_DUMP) || \
|
|
(WEBRTC_APM_DEBUG_DUMP != 0 && WEBRTC_APM_DEBUG_DUMP != 1)
|
|
#error "Set WEBRTC_APM_DEBUG_DUMP to either 0 or 1"
|
|
#endif
|
|
|
|
namespace webrtc {
|
|
|
|
namespace {
|
|
|
|
#if WEBRTC_APM_DEBUG_DUMP == 1
|
|
std::string FormFileName(const char* name,
|
|
int instance_index,
|
|
int reinit_index,
|
|
const std::string& suffix) {
|
|
char buf[1024];
|
|
rtc::SimpleStringBuilder ss(buf);
|
|
ss << name << "_" << instance_index << "-" << reinit_index << suffix;
|
|
return ss.str();
|
|
}
|
|
#endif
|
|
|
|
} // namespace
|
|
|
|
#if WEBRTC_APM_DEBUG_DUMP == 1
|
|
ApmDataDumper::ApmDataDumper(int instance_index)
|
|
: instance_index_(instance_index) {}
|
|
#else
|
|
ApmDataDumper::ApmDataDumper(int instance_index) {}
|
|
#endif
|
|
|
|
ApmDataDumper::~ApmDataDumper() {}
|
|
|
|
#if WEBRTC_APM_DEBUG_DUMP == 1
|
|
bool ApmDataDumper::recording_activated_ = false;
|
|
;
|
|
FILE* ApmDataDumper::GetRawFile(const char* name) {
|
|
std::string filename =
|
|
FormFileName(name, instance_index_, recording_set_index_, ".dat");
|
|
auto& f = raw_files_[filename];
|
|
if (!f) {
|
|
f.reset(fopen(filename.c_str(), "wb"));
|
|
}
|
|
return f.get();
|
|
}
|
|
|
|
WavWriter* ApmDataDumper::GetWavFile(const char* name,
|
|
int sample_rate_hz,
|
|
int num_channels) {
|
|
std::string filename =
|
|
FormFileName(name, instance_index_, recording_set_index_, ".wav");
|
|
auto& f = wav_files_[filename];
|
|
if (!f) {
|
|
f.reset(new WavWriter(filename.c_str(), sample_rate_hz, num_channels));
|
|
}
|
|
return f.get();
|
|
}
|
|
|
|
#endif
|
|
|
|
} // namespace webrtc
|