webrtc/modules/audio_processing/logging/apm_data_dumper.cc
Per Åhgren 5dca3f1336 Add floating point support for writing and reading wav files
This CL adds support for reading and writing floating point
wav files in WebRTC. It also updates the former wav handling
code as well as adds some simplifications.

Beyond this, the CL also adds support in the APM data_dumper
and in the audioproc_f tool for using the floating point wav
format.

Bug: webrtc:11307
Change-Id: I2ea33fd12f590b6031ac85f75708f6cc88a266b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162902
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30423}
2020-01-30 13:38:19 +00:00

92 lines
2.8 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
#if defined(WEBRTC_WIN)
constexpr char kPathDelimiter = '\\';
#else
constexpr char kPathDelimiter = '/';
#endif
std::string FormFileName(const char* output_dir,
const char* name,
int instance_index,
int reinit_index,
const std::string& suffix) {
char buf[1024];
rtc::SimpleStringBuilder ss(buf);
const size_t output_dir_size = strlen(output_dir);
if (output_dir_size > 0) {
ss << output_dir;
if (output_dir[output_dir_size - 1] != kPathDelimiter) {
ss << kPathDelimiter;
}
}
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() = default;
#if WEBRTC_APM_DEBUG_DUMP == 1
bool ApmDataDumper::recording_activated_ = false;
char ApmDataDumper::output_dir_[] = "";
FILE* ApmDataDumper::GetRawFile(const char* name) {
std::string filename = FormFileName(output_dir_, name, instance_index_,
recording_set_index_, ".dat");
auto& f = raw_files_[filename];
if (!f) {
f.reset(fopen(filename.c_str(), "wb"));
RTC_CHECK(f.get()) << "Cannot write to " << filename << ".";
}
return f.get();
}
WavWriter* ApmDataDumper::GetWavFile(const char* name,
int sample_rate_hz,
int num_channels,
WavFile::SampleFormat format) {
std::string filename = FormFileName(output_dir_, 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, format));
}
return f.get();
}
#endif
} // namespace webrtc