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libtgvoip/webrtc_dsp/common_audio/wav_file.cc
Grishka 5caaaafa42 Updated WebRTC APM
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.
2018-11-23 04:02:53 +03:00

240 lines
7.7 KiB
C++

/*
* Copyright (c) 2014 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 "common_audio/wav_file.h"
#include <errno.h>
#include <algorithm>
#include <cstdio>
#include <type_traits>
#include "common_audio/include/audio_util.h"
#include "common_audio/wav_header.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/system/arch.h"
namespace webrtc {
namespace {
// We write 16-bit PCM WAV files.
constexpr WavFormat kWavFormat = kWavFormatPcm;
static_assert(std::is_trivially_destructible<WavFormat>::value, "");
constexpr size_t kBytesPerSample = 2;
// Doesn't take ownership of the file handle and won't close it.
class ReadableWavFile : public ReadableWav {
public:
explicit ReadableWavFile(FILE* file) : file_(file) {}
ReadableWavFile(const ReadableWavFile&) = delete;
ReadableWavFile& operator=(const ReadableWavFile&) = delete;
size_t Read(void* buf, size_t num_bytes) override {
return fread(buf, 1, num_bytes, file_);
}
bool Eof() const override { return feof(file_) != 0; }
bool SeekForward(uint32_t num_bytes) override {
return fseek(file_, num_bytes, SEEK_CUR) == 0;
}
private:
FILE* file_;
};
} // namespace
WavReader::WavReader(const std::string& filename)
: WavReader(rtc::OpenPlatformFileReadOnly(filename)) {}
WavReader::WavReader(rtc::PlatformFile file) {
RTC_CHECK_NE(file, rtc::kInvalidPlatformFileValue)
<< "Invalid file. Could not create file handle for wav file.";
file_handle_ = rtc::FdopenPlatformFile(file, "rb");
if (!file_handle_) {
RTC_LOG(LS_ERROR) << "Could not open wav file for reading: " << errno;
// Even though we failed to open a FILE*, the file is still open
// and needs to be closed.
if (!rtc::ClosePlatformFile(file)) {
RTC_LOG(LS_ERROR) << "Can't close file.";
}
FATAL() << "Could not open wav file for reading.";
}
ReadableWavFile readable(file_handle_);
WavFormat format;
size_t bytes_per_sample;
RTC_CHECK(ReadWavHeader(&readable, &num_channels_, &sample_rate_, &format,
&bytes_per_sample, &num_samples_));
num_samples_remaining_ = num_samples_;
RTC_CHECK_EQ(kWavFormat, format);
RTC_CHECK_EQ(kBytesPerSample, bytes_per_sample);
}
WavReader::~WavReader() {
Close();
}
int WavReader::sample_rate() const {
return sample_rate_;
}
size_t WavReader::num_channels() const {
return num_channels_;
}
size_t WavReader::num_samples() const {
return num_samples_;
}
size_t WavReader::ReadSamples(size_t num_samples, int16_t* samples) {
#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
#error "Need to convert samples to big-endian when reading from WAV file"
#endif
// There could be metadata after the audio; ensure we don't read it.
num_samples = std::min(num_samples, num_samples_remaining_);
const size_t read =
fread(samples, sizeof(*samples), num_samples, file_handle_);
// If we didn't read what was requested, ensure we've reached the EOF.
RTC_CHECK(read == num_samples || feof(file_handle_));
RTC_CHECK_LE(read, num_samples_remaining_);
num_samples_remaining_ -= read;
return read;
}
size_t WavReader::ReadSamples(size_t num_samples, float* samples) {
static const size_t kChunksize = 4096 / sizeof(uint16_t);
size_t read = 0;
for (size_t i = 0; i < num_samples; i += kChunksize) {
int16_t isamples[kChunksize];
size_t chunk = std::min(kChunksize, num_samples - i);
chunk = ReadSamples(chunk, isamples);
for (size_t j = 0; j < chunk; ++j)
samples[i + j] = isamples[j];
read += chunk;
}
return read;
}
void WavReader::Close() {
RTC_CHECK_EQ(0, fclose(file_handle_));
file_handle_ = nullptr;
}
WavWriter::WavWriter(const std::string& filename,
int sample_rate,
size_t num_channels)
// Unlike plain fopen, CreatePlatformFile takes care of filename utf8 ->
// wchar conversion on windows.
: WavWriter(rtc::CreatePlatformFile(filename), sample_rate, num_channels) {}
WavWriter::WavWriter(rtc::PlatformFile file,
int sample_rate,
size_t num_channels)
: sample_rate_(sample_rate), num_channels_(num_channels), num_samples_(0) {
// Handle errors from the CreatePlatformFile call in above constructor.
RTC_CHECK_NE(file, rtc::kInvalidPlatformFileValue)
<< "Invalid file. Could not create wav file.";
file_handle_ = rtc::FdopenPlatformFile(file, "wb");
if (!file_handle_) {
RTC_LOG(LS_ERROR) << "Could not open wav file for writing.";
// Even though we failed to open a FILE*, the file is still open
// and needs to be closed.
if (!rtc::ClosePlatformFile(file)) {
RTC_LOG(LS_ERROR) << "Can't close file.";
}
FATAL() << "Could not open wav file for writing.";
}
RTC_CHECK(CheckWavParameters(num_channels_, sample_rate_, kWavFormat,
kBytesPerSample, num_samples_));
// Write a blank placeholder header, since we need to know the total number
// of samples before we can fill in the real data.
static const uint8_t blank_header[kWavHeaderSize] = {0};
RTC_CHECK_EQ(1, fwrite(blank_header, kWavHeaderSize, 1, file_handle_));
}
WavWriter::~WavWriter() {
Close();
}
int WavWriter::sample_rate() const {
return sample_rate_;
}
size_t WavWriter::num_channels() const {
return num_channels_;
}
size_t WavWriter::num_samples() const {
return num_samples_;
}
void WavWriter::WriteSamples(const int16_t* samples, size_t num_samples) {
#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
#error "Need to convert samples to little-endian when writing to WAV file"
#endif
const size_t written =
fwrite(samples, sizeof(*samples), num_samples, file_handle_);
RTC_CHECK_EQ(num_samples, written);
num_samples_ += written;
RTC_CHECK(num_samples_ >= written); // detect size_t overflow
}
void WavWriter::WriteSamples(const float* samples, size_t num_samples) {
static const size_t kChunksize = 4096 / sizeof(uint16_t);
for (size_t i = 0; i < num_samples; i += kChunksize) {
int16_t isamples[kChunksize];
const size_t chunk = std::min(kChunksize, num_samples - i);
FloatS16ToS16(samples + i, chunk, isamples);
WriteSamples(isamples, chunk);
}
}
void WavWriter::Close() {
RTC_CHECK_EQ(0, fseek(file_handle_, 0, SEEK_SET));
uint8_t header[kWavHeaderSize];
WriteWavHeader(header, num_channels_, sample_rate_, kWavFormat,
kBytesPerSample, num_samples_);
RTC_CHECK_EQ(1, fwrite(header, kWavHeaderSize, 1, file_handle_));
RTC_CHECK_EQ(0, fclose(file_handle_));
file_handle_ = nullptr;
}
} // namespace webrtc
rtc_WavWriter* rtc_WavOpen(const char* filename,
int sample_rate,
size_t num_channels) {
return reinterpret_cast<rtc_WavWriter*>(
new webrtc::WavWriter(filename, sample_rate, num_channels));
}
void rtc_WavClose(rtc_WavWriter* wf) {
delete reinterpret_cast<webrtc::WavWriter*>(wf);
}
void rtc_WavWriteSamples(rtc_WavWriter* wf,
const float* samples,
size_t num_samples) {
reinterpret_cast<webrtc::WavWriter*>(wf)->WriteSamples(samples, num_samples);
}
int rtc_WavSampleRate(const rtc_WavWriter* wf) {
return reinterpret_cast<const webrtc::WavWriter*>(wf)->sample_rate();
}
size_t rtc_WavNumChannels(const rtc_WavWriter* wf) {
return reinterpret_cast<const webrtc::WavWriter*>(wf)->num_channels();
}
size_t rtc_WavNumSamples(const rtc_WavWriter* wf) {
return reinterpret_cast<const webrtc::WavWriter*>(wf)->num_samples();
}