mirror of
https://github.com/danog/libtgvoip.git
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333c4a1101
Added simple audio resampler Replaced prebuilt static libs with their sources & added that to all project files (closes #5)
206 lines
6.3 KiB
C++
206 lines
6.3 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/common_audio/wav_file.h"
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#include <algorithm>
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#include <cstdio>
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#include <limits>
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#include <sstream>
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#include "webrtc/base/checks.h"
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#include "webrtc/base/safe_conversions.h"
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#include "webrtc/common_audio/include/audio_util.h"
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#include "webrtc/common_audio/wav_header.h"
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namespace webrtc {
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// We write 16-bit PCM WAV files.
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static const WavFormat kWavFormat = kWavFormatPcm;
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static const size_t kBytesPerSample = 2;
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// Doesn't take ownership of the file handle and won't close it.
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class ReadableWavFile : public ReadableWav {
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public:
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explicit ReadableWavFile(FILE* file) : file_(file) {}
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virtual size_t Read(void* buf, size_t num_bytes) {
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return fread(buf, 1, num_bytes, file_);
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}
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private:
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FILE* file_;
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};
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std::string WavFile::FormatAsString() const {
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std::ostringstream s;
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s << "Sample rate: " << sample_rate() << " Hz, Channels: " << num_channels()
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<< ", Duration: "
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<< (1.f * num_samples()) / (num_channels() * sample_rate()) << " s";
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return s.str();
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}
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WavReader::WavReader(const std::string& filename)
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: file_handle_(fopen(filename.c_str(), "rb")) {
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RTC_CHECK(file_handle_) << "Could not open wav file for reading.";
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ReadableWavFile readable(file_handle_);
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WavFormat format;
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size_t bytes_per_sample;
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RTC_CHECK(ReadWavHeader(&readable, &num_channels_, &sample_rate_, &format,
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&bytes_per_sample, &num_samples_));
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num_samples_remaining_ = num_samples_;
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RTC_CHECK_EQ(kWavFormat, format);
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RTC_CHECK_EQ(kBytesPerSample, bytes_per_sample);
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}
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WavReader::~WavReader() {
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Close();
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}
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int WavReader::sample_rate() const {
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return sample_rate_;
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}
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size_t WavReader::num_channels() const {
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return num_channels_;
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}
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size_t WavReader::num_samples() const {
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return num_samples_;
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}
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size_t WavReader::ReadSamples(size_t num_samples, int16_t* samples) {
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#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
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#error "Need to convert samples to big-endian when reading from WAV file"
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#endif
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// There could be metadata after the audio; ensure we don't read it.
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num_samples = std::min(num_samples, num_samples_remaining_);
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const size_t read =
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fread(samples, sizeof(*samples), num_samples, file_handle_);
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// If we didn't read what was requested, ensure we've reached the EOF.
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RTC_CHECK(read == num_samples || feof(file_handle_));
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RTC_CHECK_LE(read, num_samples_remaining_);
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num_samples_remaining_ -= read;
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return read;
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}
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size_t WavReader::ReadSamples(size_t num_samples, float* samples) {
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static const size_t kChunksize = 4096 / sizeof(uint16_t);
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size_t read = 0;
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for (size_t i = 0; i < num_samples; i += kChunksize) {
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int16_t isamples[kChunksize];
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size_t chunk = std::min(kChunksize, num_samples - i);
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chunk = ReadSamples(chunk, isamples);
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for (size_t j = 0; j < chunk; ++j)
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samples[i + j] = isamples[j];
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read += chunk;
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}
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return read;
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}
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void WavReader::Close() {
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RTC_CHECK_EQ(0, fclose(file_handle_));
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file_handle_ = NULL;
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}
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WavWriter::WavWriter(const std::string& filename, int sample_rate,
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size_t num_channels)
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: sample_rate_(sample_rate),
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num_channels_(num_channels),
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num_samples_(0),
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file_handle_(fopen(filename.c_str(), "wb")) {
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RTC_CHECK(file_handle_) << "Could not open wav file for writing.";
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RTC_CHECK(CheckWavParameters(num_channels_, sample_rate_, kWavFormat,
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kBytesPerSample, num_samples_));
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// Write a blank placeholder header, since we need to know the total number
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// of samples before we can fill in the real data.
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static const uint8_t blank_header[kWavHeaderSize] = {0};
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RTC_CHECK_EQ(1, fwrite(blank_header, kWavHeaderSize, 1, file_handle_));
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}
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WavWriter::~WavWriter() {
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Close();
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}
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int WavWriter::sample_rate() const {
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return sample_rate_;
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}
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size_t WavWriter::num_channels() const {
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return num_channels_;
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}
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size_t WavWriter::num_samples() const {
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return num_samples_;
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}
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void WavWriter::WriteSamples(const int16_t* samples, size_t num_samples) {
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#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
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#error "Need to convert samples to little-endian when writing to WAV file"
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#endif
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const size_t written =
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fwrite(samples, sizeof(*samples), num_samples, file_handle_);
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RTC_CHECK_EQ(num_samples, written);
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num_samples_ += written;
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RTC_CHECK(num_samples_ >= written); // detect size_t overflow
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}
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void WavWriter::WriteSamples(const float* samples, size_t num_samples) {
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static const size_t kChunksize = 4096 / sizeof(uint16_t);
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for (size_t i = 0; i < num_samples; i += kChunksize) {
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int16_t isamples[kChunksize];
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const size_t chunk = std::min(kChunksize, num_samples - i);
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FloatS16ToS16(samples + i, chunk, isamples);
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WriteSamples(isamples, chunk);
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}
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}
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void WavWriter::Close() {
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RTC_CHECK_EQ(0, fseek(file_handle_, 0, SEEK_SET));
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uint8_t header[kWavHeaderSize];
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WriteWavHeader(header, num_channels_, sample_rate_, kWavFormat,
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kBytesPerSample, num_samples_);
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RTC_CHECK_EQ(1, fwrite(header, kWavHeaderSize, 1, file_handle_));
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RTC_CHECK_EQ(0, fclose(file_handle_));
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file_handle_ = NULL;
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}
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} // namespace webrtc
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rtc_WavWriter* rtc_WavOpen(const char* filename,
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int sample_rate,
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size_t num_channels) {
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return reinterpret_cast<rtc_WavWriter*>(
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new webrtc::WavWriter(filename, sample_rate, num_channels));
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}
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void rtc_WavClose(rtc_WavWriter* wf) {
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delete reinterpret_cast<webrtc::WavWriter*>(wf);
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}
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void rtc_WavWriteSamples(rtc_WavWriter* wf,
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const float* samples,
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size_t num_samples) {
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reinterpret_cast<webrtc::WavWriter*>(wf)->WriteSamples(samples, num_samples);
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}
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int rtc_WavSampleRate(const rtc_WavWriter* wf) {
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return reinterpret_cast<const webrtc::WavWriter*>(wf)->sample_rate();
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}
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size_t rtc_WavNumChannels(const rtc_WavWriter* wf) {
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return reinterpret_cast<const webrtc::WavWriter*>(wf)->num_channels();
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}
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size_t rtc_WavNumSamples(const rtc_WavWriter* wf) {
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return reinterpret_cast<const webrtc::WavWriter*>(wf)->num_samples();
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}
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