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SoftFM/AudioOutput.cc

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/*
* 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
*/
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#define _FILE_OFFSET_BITS 64
#include <string.h>
#include <errno.h>
#include <unistd.h>
#include <fcntl.h>
#include <cstdio>
#include <algorithm>
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#include <alsa/asoundlib.h>
#include "SoftFM.h"
#include "AudioOutput.h"
using namespace std;
/* **************** class AudioOutput **************** */
// Encode a list of samples as signed 16-bit little-endian integers.
void AudioOutput::samplesToInt16(const SampleVector& samples,
vector<uint8_t>& bytes)
{
bytes.resize(2 * samples.size());
SampleVector::const_iterator i = samples.begin();
SampleVector::const_iterator n = samples.end();
vector<uint8_t>::iterator k = bytes.begin();
while (i != n) {
Sample s = *(i++);
s = max(Sample(-1.0), min(Sample(1.0), s));
long v = lrint(s * 32767);
unsigned long u = v;
*(k++) = u & 0xff;
*(k++) = (u >> 8) & 0xff;
}
}
/* **************** class RawAudioOutput **************** */
// Construct raw audio writer.
RawAudioOutput::RawAudioOutput(const string& filename)
{
if (filename == "-") {
m_fd = STDOUT_FILENO;
} else {
m_fd = open(filename.c_str(), O_WRONLY | O_CREAT | O_TRUNC, 0666);
if (m_fd < 0) {
m_error = "can not open '" + filename + "' (" +
strerror(errno) + ")";
m_zombie = true;
return;
}
}
}
// Destructor.
RawAudioOutput::~RawAudioOutput()
{
// Close file descriptor.
if (m_fd >= 0 && m_fd != STDOUT_FILENO) {
close(m_fd);
}
}
// Write audio data.
bool RawAudioOutput::write(const SampleVector& samples)
{
if (m_fd < 0)
return false;
// Convert samples to bytes.
samplesToInt16(samples, m_bytebuf);
// Write data.
size_t p = 0;
size_t n = m_bytebuf.size();
while (p < n) {
ssize_t k = ::write(m_fd, m_bytebuf.data() + p, n - p);
if (k <= 0) {
if (k == 0 || errno != EINTR) {
m_error = "write failed (";
m_error += strerror(errno);
m_error += ")";
return false;
}
} else {
p += k;
}
}
return true;
}
/* **************** class WavAudioOutput **************** */
// Construct .WAV writer.
WavAudioOutput::WavAudioOutput(const std::string& filename,
unsigned int samplerate,
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;
}
// Write initial header with a dummy sample count.
// This will be replaced with the actual header once the WavFile is closed.
if (!write_header(0x7fff0000)) {
m_error = "can not write to '" + filename + "' (" +
strerror(errno) + ")";
m_zombie = true;
}
}
// Destructor.
WavAudioOutput::~WavAudioOutput()
{
// We need to go back and fill in the header ...
if (!m_zombie) {
const unsigned bytesPerSample = 2;
const long currentPosition = ftell(m_stream);
assert((currentPosition - 44) % bytesPerSample == 0);
const unsigned totalNumberOfSamples = (currentPosition - 44) / bytesPerSample;
assert(totalNumberOfSamples % numberOfChannels == 0);
// Put header in front
if (fseek(m_stream, 0, SEEK_SET) == 0) {
write_header(totalNumberOfSamples);
}
}
// 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;
}
// (Re)write .WAV header.
bool WavAudioOutput::write_header(unsigned int nsamples)
{
const unsigned bytesPerSample = 2;
const unsigned bitsPerSample = 16;
enum wFormatTagId
{
WAVE_FORMAT_PCM = 0x0001,
WAVE_FORMAT_IEEE_FLOAT = 0x0003
};
assert(nsamples % numberOfChannels == 0);
// synthesize header
uint8_t wavHeader[44];
encode_chunk_id (wavHeader + 0, "RIFF");
set_value<uint32_t>(wavHeader + 4, 36 + nsamples * 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, nsamples * bytesPerSample);
return fwrite(wavHeader, 1, 44, m_stream) == 44;
}
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;
}
}
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/* **************** class AlsaAudioOutput **************** */
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// Construct ALSA output stream.
AlsaAudioOutput::AlsaAudioOutput(const std::string& devname,
unsigned int samplerate,
bool stereo)
{
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m_pcm = NULL;
m_nchannels = stereo ? 2 : 1;
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int r = snd_pcm_open(&m_pcm, devname.c_str(),
SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
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if (r < 0) {
m_error = "can not open PCM device '" + devname + "' (" +
strerror(-r) + ")";
m_zombie = true;
return;
}
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snd_pcm_nonblock(m_pcm, 0);
r = snd_pcm_set_params(m_pcm,
SND_PCM_FORMAT_S16_LE,
SND_PCM_ACCESS_RW_INTERLEAVED,
m_nchannels,
samplerate,
1, // allow soft resampling
500000); // latency in us
if (r < 0) {
m_error = "can not set PCM parameters (";
m_error += strerror(-r);
m_error += ")";
m_zombie = true;
}
}
// Destructor.
AlsaAudioOutput::~AlsaAudioOutput()
{
// Close device.
if (m_pcm != NULL) {
snd_pcm_close(m_pcm);
}
}
// Write audio data.
bool AlsaAudioOutput::write(const SampleVector& samples)
{
if (m_zombie)
return false;
// Convert samples to bytes.
samplesToInt16(samples, m_bytebuf);
// Write data.
unsigned int p = 0;
unsigned int n = samples.size() / m_nchannels;
unsigned int framesize = 2 * m_nchannels;
while (p < n) {
int k = snd_pcm_writei(m_pcm,
m_bytebuf.data() + p * framesize, n - p);
if (k < 0) {
m_error = "write failed (";
m_error += strerror(errno);
m_error += ")";
// After an underrun, ALSA keeps returning error codes until we
// explicitly fix the stream.
snd_pcm_recover(m_pcm, k, 0);
return false;
} else {
p += k;
}
}
return true;
}
/* end */