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libtgvoip/webrtc_dsp/webrtc/common_audio/wav_file.cc
Grishka 333c4a1101 Added working audio i/o for OS X
Added simple audio resampler
Replaced prebuilt static libs with their sources & added that to all project files (closes #5)
2017-04-09 19:14:33 +03:00

206 lines
6.3 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 "webrtc/common_audio/wav_file.h"
#include <algorithm>
#include <cstdio>
#include <limits>
#include <sstream>
#include "webrtc/base/checks.h"
#include "webrtc/base/safe_conversions.h"
#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/common_audio/wav_header.h"
namespace webrtc {
// We write 16-bit PCM WAV files.
static const WavFormat kWavFormat = kWavFormatPcm;
static const 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) {}
virtual size_t Read(void* buf, size_t num_bytes) {
return fread(buf, 1, num_bytes, file_);
}
private:
FILE* file_;
};
std::string WavFile::FormatAsString() const {
std::ostringstream s;
s << "Sample rate: " << sample_rate() << " Hz, Channels: " << num_channels()
<< ", Duration: "
<< (1.f * num_samples()) / (num_channels() * sample_rate()) << " s";
return s.str();
}
WavReader::WavReader(const std::string& filename)
: file_handle_(fopen(filename.c_str(), "rb")) {
RTC_CHECK(file_handle_) << "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_ = NULL;
}
WavWriter::WavWriter(const std::string& filename, int sample_rate,
size_t num_channels)
: sample_rate_(sample_rate),
num_channels_(num_channels),
num_samples_(0),
file_handle_(fopen(filename.c_str(), "wb")) {
RTC_CHECK(file_handle_) << "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_ = NULL;
}
} // 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();
}