You cannot select more than 25 topics Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
akode/akode/plugins/oss_sink/oss_sink.cpp

219 lines
4.9 KiB
C++

/* aKode: OSS-Sink
Copyright (C) 2004 Allan Sandfeld Jensen <kde@carewolf.com>
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Library General Public
License as published by the Free Software Foundation; either
version 2 of the License, or (at your option) any later version.
This library 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
Library General Public License for more details.
You should have received a copy of the GNU Library General Public License
along with this library; see the file COPYING.LIB. If not, write to
the Free Software Foundation, Inc., 51 Franklin Steet, Fifth Floor,
Boston, MA 02110-1301, USA.
*/
#include <config.h>
#if defined(HAVE_SYS_SOUNDCARD_H)
#include <sys/soundcard.h>
#elif defined(HAVE_SOUNDCARD_H)
#include <soundcard.h>
#endif
#include <sys/ioctl.h>
#include <errno.h>
#include <fcntl.h>
#include <unistd.h>
#include <audioframe.h>
#include "oss_sink.h"
#include <iostream>
namespace aKode {
extern "C" { OSSSinkPlugin oss_sink; }
struct OSSSink::private_data
{
private_data() : audio_fd(-1), device(0), valid(false), buffer(0), buffer_length(0) {};
int audio_fd;
const char *device;
AudioConfiguration config;
bool valid;
char* buffer;
int buffer_length;
};
static const char *_devices[] = {
"/dev/dsp",
"/dev/sound/dsp0",
"/dev/audio",
0
};
OSSSink::OSSSink()
{
d = new private_data;
}
OSSSink::~OSSSink()
{
close();
delete d;
}
bool OSSSink::open()
{
const char** device = _devices;
while (*device) {
if(::access(*device, F_OK) == 0) {
break;
}
device++;
}
if (!*device) {
std::cerr << "akode: No OSS device found\n";
d->valid = false;
return false;
}
return openDevice(*device);
}
bool OSSSink::openDevice(const char *device)
{
d->device = device;
// Set non-blocking to not block on open
d->audio_fd = ::open(d->device, O_WRONLY | O_NONBLOCK, 0);
if (d->audio_fd == -1) {
std::cerr << "akode: Could not open " << d->device << " for writing\n";
goto failed;
}
// set blocking again to block on write
fcntl(d->audio_fd, F_SETFL, O_WRONLY);
d->valid = true;
return true;
failed:
d->valid = false;
return false;
}
void OSSSink::close() {
if (d->audio_fd != -1) ::close(d->audio_fd);
d->audio_fd = -1;
delete d->buffer;
d->buffer = 0;
d->buffer_length = 0;
d->valid = false;
}
int OSSSink::setAudioConfiguration(const AudioConfiguration* config)
{
d->config = *config;
int format = 0;
if (config->sample_width > 0 && config->sample_width <= 8)
format = AFMT_S8;
else
format = AFMT_S16_NE; // 16bit native endian
ioctl(d->audio_fd, SNDCTL_DSP_SETFMT, &format);
if (format == AFMT_S16_NE)
d->config.sample_width = 16;
else
if (format == AFMT_S8)
d->config.sample_width = 8;
else
return -1;
int stereo;
if (config->channels == 1)
stereo = 0;
else
stereo = 1;
ioctl(d->audio_fd, SNDCTL_DSP_STEREO, &stereo);
d->config.channel_config = MonoStereo;
if (stereo == 0)
d->config.channels = 1;
else
d->config.channels = 2;
ioctl(d->audio_fd, SNDCTL_DSP_SPEED, &d->config.sample_rate);
return 1;
}
const AudioConfiguration* OSSSink::audioConfiguration() const
{
return &d->config;
}
bool OSSSink::writeFrame(AudioFrame* frame)
{
if (!d->valid) return false;
if ( frame->sample_width != d->config.sample_width
|| frame->channels != d->config.channels )
{
if (setAudioConfiguration(frame) < 0)
return false;
}
int channels = d->config.channels;
int length = frame->length;
int bytelen = length*channels*((d->config.sample_width+7)/8);
if (bytelen > d->buffer_length) {
delete d->buffer;
d->buffer = new char[bytelen];
d->buffer_length = bytelen;
}
if (d->config.sample_width == 8) {
int8_t *buffer = (int8_t*)d->buffer;
int8_t** data = (int8_t**)frame->data;
for(int i=0; i<length; i++)
for(int j=0; j<channels; j++)
buffer[i*channels+j] = data[j][i];
} else {
int16_t *buffer = (int16_t*)d->buffer;
int16_t** data = (int16_t**)frame->data;
for(int i=0; i<length; i++)
for(int j=0; j<channels; j++)
buffer[i*channels+j] = data[j][i];
}
int status = 0;
do {
status = ::write(d->audio_fd, d->buffer, bytelen);
if (status == -1) {
// if (errno == EAGAIN) continue;
if (errno == EINTR) continue;
return false;
}
} while(false);
return true;
}
} // namespace