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-rw-r--r--akode/plugins/oss_sink/oss_sink.cpp218
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diff --git a/akode/plugins/oss_sink/oss_sink.cpp b/akode/plugins/oss_sink/oss_sink.cpp
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+/* 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