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Implement .WAV file writing based on Sidney's WavFile.h.

This commit is contained in:
Joris van Rantwijk 2014-01-03 18:07:28 +01:00
parent 3e8c2232d2
commit 8a2cef5e35
4 changed files with 158 additions and 25 deletions

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@ -1,3 +1,24 @@
/*
* Audio output handling for SoftFM
*
* Copyright (C) 2013, Joris van Rantwijk.
*
* .WAV file writing by Sidney Cadot,
* adapted for SoftFM by Joris van Rantwijk.
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, see http://www.gnu.org/licenses/gpl-2.0.html
*/
#define _FILE_OFFSET_BITS 64 #define _FILE_OFFSET_BITS 64
@ -5,6 +26,7 @@
#include <errno.h> #include <errno.h>
#include <unistd.h> #include <unistd.h>
#include <fcntl.h> #include <fcntl.h>
#include <cstdio>
#include <algorithm> #include <algorithm>
#include <alsa/asoundlib.h> #include <alsa/asoundlib.h>
@ -75,7 +97,7 @@ RawAudioOutput::~RawAudioOutput()
bool RawAudioOutput::write(const SampleVector& samples) bool RawAudioOutput::write(const SampleVector& samples)
{ {
if (m_fd < 0) if (m_fd < 0)
return -1; return false;
// Convert samples to bytes. // Convert samples to bytes.
samplesToInt16(samples, m_bytebuf); samplesToInt16(samples, m_bytebuf);
@ -102,33 +124,134 @@ bool RawAudioOutput::write(const SampleVector& samples)
} }
#if 0 /* **************** class WavAudioOutput **************** */
/** Write audio data as .WAV file. */ // Construct .WAV writer.
class WavAudioOutput WavAudioOutput::WavAudioOutput(const std::string& filename,
{
public:
/**
* Construct .WAV writer.
*
* filename :: file name (including path) or "-" to write to stdout
* samplerate :: audio sample rate in Hz
* stereo :: true if the output stream contains stereo data
*/
WavAudioOutput(const std::string& filename,
unsigned int samplerate, unsigned int samplerate,
bool stereo); bool stereo)
: numberOfChannels(stereo ? 2 : 1)
, sampleRate(samplerate)
{
m_stream = fopen(filename.c_str(), "wb");
if (m_stream == NULL) {
m_error = "can not open '" + filename + "' (" +
strerror(errno) + ")";
m_zombie = true;
return;
}
~WavAudioOutput(); // Write a 44-byte placeholder for the header.
bool write(const SampleVector& samples); // This will be replaced with the actual header once the WavFile is closed.
std::string error(); size_t k = fwrite("[44-byte WAV-file header -- to be filled in]", 1, 44, m_stream);
if (k != 44) {
m_error = "can not write to '" + filename + "' (" +
strerror(errno) + ")";
m_zombie = true;
}
}
private:
// TODO
};
#endif // Destructor.
WavAudioOutput::~WavAudioOutput()
{
// We need to go back and fill in the header ...
const unsigned bytesPerSample = 2;
const unsigned bitsPerSample = 16;
enum wFormatTagId
{
WAVE_FORMAT_PCM = 0x0001,
WAVE_FORMAT_IEEE_FLOAT = 0x0003
};
if (!m_zombie) {
const long currentPosition = ftell(m_stream);
assert((currentPosition - 44) % bytesPerSample == 0);
const unsigned totalNumberOfSamples = (currentPosition - 44) / bytesPerSample;
assert(totalNumberOfSamples % numberOfChannels == 0);
// synthesize header
uint8_t wavHeader[44];
encode_chunk_id (wavHeader + 0, "RIFF");
set_value<uint32_t>(wavHeader + 4, 36 + totalNumberOfSamples * bytesPerSample);
encode_chunk_id (wavHeader + 8, "WAVE");
encode_chunk_id (wavHeader + 12, "fmt ");
set_value<uint32_t>(wavHeader + 16, 16);
set_value<uint16_t>(wavHeader + 20, WAVE_FORMAT_PCM);
set_value<uint16_t>(wavHeader + 22, numberOfChannels);
set_value<uint32_t>(wavHeader + 24, sampleRate ); // sample rate
set_value<uint32_t>(wavHeader + 28, sampleRate * numberOfChannels * bytesPerSample); // byte rate
set_value<uint16_t>(wavHeader + 32, numberOfChannels * bytesPerSample); // block size
set_value<uint16_t>(wavHeader + 34, bitsPerSample);
encode_chunk_id (wavHeader + 36, "data");
set_value<uint32_t>(wavHeader + 40, totalNumberOfSamples * bytesPerSample);
// Put header in front
if (fseek(m_stream, 0, SEEK_SET) == 0) {
fwrite(wavHeader, 1, 44, m_stream);
}
}
// Done writing the file
if (m_stream) {
fclose(m_stream);
}
}
// Write audio data.
bool WavAudioOutput::write(const SampleVector& samples)
{
if (m_zombie)
return false;
// Convert samples to bytes.
samplesToInt16(samples, m_bytebuf);
// Write samples to file.
size_t k = fwrite(m_bytebuf.data(), 1, m_bytebuf.size(), m_stream);
if (k != m_bytebuf.size()) {
m_error = "write failed (";
m_error += strerror(errno);
m_error += ")";
return false;
}
return true;
}
void WavAudioOutput::encode_chunk_id(uint8_t * ptr, const char * chunkname)
{
for (unsigned i = 0; i < 4; ++i)
{
assert(chunkname[i] != '\0');
ptr[i] = chunkname[i];
}
assert(chunkname[4] == '\0');
}
template <typename T>
void WavAudioOutput::set_value(uint8_t * ptr, T value)
{
for (size_t i = 0; i < sizeof(T); ++i)
{
ptr[i] = value & 0xff;
value >>= 8;
}
}
/* **************** class AlsaAudioOutput **************** */ /* **************** class AlsaAudioOutput **************** */

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@ -2,6 +2,7 @@
#define SOFTFM_AUDIOOUTPUT_H #define SOFTFM_AUDIOOUTPUT_H
#include <cstdint> #include <cstdint>
#include <cstdio>
#include <string> #include <string>
#include <vector> #include <vector>
#include "SoftFM.h" #include "SoftFM.h"
@ -95,7 +96,16 @@ public:
bool write(const SampleVector& samples); bool write(const SampleVector& samples);
private: private:
// TODO
static void encode_chunk_id(uint8_t * ptr, const char * chunkname);
template <typename T>
static void set_value(uint8_t * ptr, T value);
const unsigned numberOfChannels;
const unsigned sampleRate;
FILE *m_stream;
std::vector<uint8_t> m_bytebuf;
}; };

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@ -1,4 +1,3 @@
* (feature) implement WAV file writing
* (feature) implement stereo pilot pulse-per-second * (feature) implement stereo pilot pulse-per-second
* (quality) consider DC offset calibration * (quality) consider DC offset calibration
* (speedup) maybe replace high-order FIR downsampling filter with 2nd order butterworth followed by lower order FIR filter * (speedup) maybe replace high-order FIR downsampling filter with 2nd order butterworth followed by lower order FIR filter

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@ -438,7 +438,8 @@ int main(int argc, char **argv)
audio_output.reset(new RawAudioOutput(filename)); audio_output.reset(new RawAudioOutput(filename));
break; break;
case MODE_WAV: case MODE_WAV:
abort(); audio_output.reset(new WavAudioOutput(filename, pcmrate, stereo));
break;
case MODE_ALSA: case MODE_ALSA:
audio_output.reset(new AlsaAudioOutput(devname, pcmrate, stereo)); audio_output.reset(new AlsaAudioOutput(devname, pcmrate, stereo));
break; break;