static int init(int rate_hz, int channels, int format, int flags)
{
int err;
int frag_spec;
if( (err=arts_init()) ) {
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ARTS_CantInit, arts_error_text(err));
return 0;
}
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ARTS_ServerConnect);
/*
* arts supports 8bit unsigned and 16bit signed sample formats
* (16bit apparently in little endian format, even in the case
* when artsd runs on a big endian cpu).
*
* Unsupported formats are translated to one of these two formats
* using mplayer's audio filters.
*/
switch (format) {
case AF_FORMAT_U8:
case AF_FORMAT_S8:
format = AF_FORMAT_U8;
break;
default:
format = AF_FORMAT_S16_LE; /* artsd always expects little endian?*/
break;
}
ao_data.format = format;
ao_data.channels = channels;
ao_data.samplerate = rate_hz;
ao_data.bps = (rate_hz*channels);
if(format != AF_FORMAT_U8 && format != AF_FORMAT_S8)
ao_data.bps*=2;
stream=arts_play_stream(rate_hz, OBTAIN_BITRATE(format), channels, "MPlayer");
if(stream == NULL) {
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ARTS_CantOpenStream);
arts_free();
return 0;
}
/* Set the stream to blocking: it will not block anyway, but it seems */
/* to be working better */
arts_stream_set(stream, ARTS_P_BLOCKING, 1);
frag_spec = ARTS_PACKET_SIZE_LOG2 | ARTS_PACKETS << 16;
arts_stream_set(stream, ARTS_P_PACKET_SETTINGS, frag_spec);
ao_data.buffersize = arts_stream_get(stream, ARTS_P_BUFFER_SIZE);
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ARTS_StreamOpen);
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ARTS_BufferSize,
ao_data.buffersize);
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ARTS_BufferSize,
arts_stream_get(stream, ARTS_P_PACKET_SIZE));
return 1;
}
static int output_data(char *buf, int32 nbytes)
{
int n;
if (stream == 0) return -1;
while(nbytes > 0)
{
if((n = arts_write(stream, buf, nbytes)) < 0)
{
ctl->cmsg(CMSG_WARNING, VERB_VERBOSE,
"%s: %s", dpm.name, arts_error_text(n));
if(errno == EWOULDBLOCK)
{
/* It is possible to come here because of bug of the
* sound driver.
*/
continue;
}
return -1;
}
output_count += n;
buf += n;
nbytes -= n;
}
return 0;
}
static gboolean gst_artsdsink_open_audio (GstArtsdsink * sink)
{
const char connname[] = "gstreamer";
int errcode;
/* Name used by aRtsd for this connection. */
if (sink->connect_name != NULL)
connname = sink->connect_name;
/* FIXME: this should only ever happen once per process. */
/* Really, artsc needs to be made thread safe to fix this (and other related */
/* problems). */
errcode = arts_init ();
if (errcode < 0) {
fprintf (stderr, "arts_init error: %s\n", arts_error_text (errcode));
return FALSE;
}
GST_DEBUG ("artsdsink: attempting to open connection to aRtsd server");
sink->stream = arts_play_stream (sink->frequency, sink->depth,
sink->channels, connname);
/* FIXME: check connection */
/* GST_DEBUG ("artsdsink: can't open connection to aRtsd server"); */
GST_OBJECT_FLAG_SET (sink, GST_ARTSDSINK_OPEN);
sink->connected = TRUE;
return TRUE;
}
/* initialize the sound card or connect to aRts server */
void audio_init(audio_dev_handle* handle, int rate, int latency, int try_arts)
{
#ifdef HAS_ARTS
int artserr = 0;
handle->arts_handle = NULL;
handle->use_arts = 0;
#endif
handle->alsa_handle = NULL;
handle->channels = 1; /* mono */
handle->format = SND_PCM_FORMAT_U8;
handle->rate = rate;
handle->latency = latency; /* in ms */
#ifdef HAS_ARTS
if( try_arts )
{
artserr = arts_init();
if( artserr < 0 )
{
fprintf(stderr, "Error initializing aRts: %s\n", arts_error_text(artserr));
exit(-1);
}
handle->arts_handle = arts_play_stream( *rate, 8, handle->channels, "arts-whitenoise" );
arts_stream_set(handle->arts_handle, ARTS_P_BUFFER_TIME, *latency);
handle->use_arts = 1;
}
else
#endif
{
alsa_init(handle);
}
}
static int arts_dsp_write(struct sysdep_dsp_struct *dsp, unsigned char *data,
int count)
{
int result;
struct arts_dsp_priv_data *priv = dsp->_priv;
result = arts_write(priv->stream, data, count *
arts_dsp_bytes_per_sample[dsp->hw_info.type]);
if (result < 0)
{
fprintf(stderr,
"arts_write error: %s\n", arts_error_text(result));
return -1;
}
return result / arts_dsp_bytes_per_sample[dsp->hw_info.type];
}
static void
gst_artsdsink_chain (GstPad * pad, GstData * _data)
{
GstBuffer *buf = GST_BUFFER (_data);
GstArtsdsink *artsdsink;
g_return_if_fail (pad != NULL);
g_return_if_fail (GST_IS_PAD (pad));
g_return_if_fail (buf != NULL);
artsdsink = GST_ARTSDSINK (gst_pad_get_parent (pad));
if (GST_BUFFER_DATA (buf) != NULL) {
gst_trace_add_entry (NULL, 0, GPOINTER_TO_INT (buf),
"artsdsink: writing to server");
if (!artsdsink->mute && artsdsink->connected) {
int bytes;
void *bufptr = GST_BUFFER_DATA (buf);
int bufsize = GST_BUFFER_SIZE (buf);
GST_DEBUG ("artsdsink: stream=%p data=%p size=%d",
artsdsink->stream, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
do {
bytes = arts_write (artsdsink->stream, bufptr, bufsize);
if (bytes < 0) {
fprintf (stderr, "arts_write error: %s\n", arts_error_text (bytes));
gst_buffer_unref (buf);
return;
}
bufptr += bytes;
bufsize -= bytes;
} while (bufsize > 0);
}
}
gst_buffer_unref (buf);
}
static int open_output(void)
{
int i, include_enc, exclude_enc;
int sample_width, channels;
include_enc = 0;
exclude_enc = PE_ULAW|PE_ALAW|PE_BYTESWAP; /* They can't mean these */
if(dpm.encoding & PE_16BIT)
include_enc |= PE_SIGNED;
else
exclude_enc |= PE_SIGNED;
dpm.encoding = validate_encoding(dpm.encoding, include_enc, exclude_enc);
sample_width = (dpm.encoding & PE_16BIT) ? 16 : 8;
channels = (dpm.encoding & PE_MONO) ? 1 : 2;
/* Open the audio device */
switch (arts_init_state) {
case 0:
if((i = arts_init()) != 0)
{
ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "%s: %s",
dpm.name, arts_error_text(i));
return -1;
}
arts_init_state = 1;
break;
case 2:
ctl->cmsg(CMSG_ERROR, VERB_NORMAL,
"TiMidity aRts bug: open_output() after close_output() not supported");
return -1;
}
stream = arts_play_stream(dpm.rate,
LE_LONG(sample_width),
channels,
"timidity");
if(stream == NULL)
{
ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "%s: %s",
dpm.name, strerror(errno));
return -1;
}
arts_stream_set(stream, ARTS_P_BLOCKING, 1);
server_buffer = arts_stream_get(stream, ARTS_P_SERVER_LATENCY) * dpm.rate * (sample_width/8) * channels / 1000;
output_count = 0;
return 0;
/* "this aRts function isnot yet implemented"
*
if (dpm.extra_param[0]) {
i = arts_stream_set(stream,
ARTS_P_PACKET_COUNT,
dpm.extra_param[0]);
if (i < 0) {
ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "%s: %s",
dpm.name, arts_error_text(i));
return 1;
}
}
return 0;
*
*/
}
/* public methods (static but exported through the sysdep_dsp or plugin
struct) */
static void *arts_dsp_create(const void *flags)
{
int i, j, result, bits, channels, block;
struct arts_dsp_priv_data *priv = NULL;
struct sysdep_dsp_struct *dsp = NULL;
const struct sysdep_dsp_create_params *params = flags;
/* allocate the dsp struct */
if (!(dsp = calloc(1, sizeof(struct sysdep_dsp_struct))))
{
fprintf(stderr,
"error malloc failed for struct sysdep_dsp_struct\n");
return NULL;
}
/* alloc private data */
if(!(priv = calloc(1, sizeof(struct arts_dsp_priv_data))))
{
fprintf(stderr,
"error malloc failed for struct arts_dsp_priv_data\n");
arts_dsp_destroy(dsp);
return NULL;
}
/* fill in the functions and some data */
priv->stream = NULL;
dsp->_priv = priv;
dsp->write = arts_dsp_write;
dsp->destroy = arts_dsp_destroy;
dsp->hw_info.type = params->type;
dsp->hw_info.samplerate = params->samplerate;
result = arts_init();
if (result < 0)
{
fprintf(stderr,
"arts_init error: %s\n", arts_error_text(result));
arts_dsp_destroy(dsp);
return NULL;
}
bits = (dsp->hw_info.type & SYSDEP_DSP_16BIT) ? 16 : 8;
channels = (dsp->hw_info.type & SYSDEP_DSP_STEREO) ? 2 : 1;
priv->stream = arts_play_stream(dsp->hw_info.samplerate, bits, channels,
"xmame arts");
block = (params->flags & SYSDEP_DSP_O_NONBLOCK) ? 0 : 1;
result = arts_stream_set(priv->stream, ARTS_P_BLOCKING, block);
if (result < 0)
{
fprintf(stderr,
"arts_stream_set error: %s\n", arts_error_text(result));
arts_dsp_destroy(dsp);
return NULL;
}
else
{
if (result != block)
{
fprintf(stderr,
"arts_stream_set ARTS_P_BLOCKING to %d failed\n", block);
arts_dsp_destroy(dsp);
return NULL;
}
}
/* calculate fragsize & number of frags */
/* fragsize (as power of 2) */
i = 7;
if (dsp->hw_info.type & SYSDEP_DSP_16BIT) i++;
if (dsp->hw_info.type & SYSDEP_DSP_STEREO) i++;
i += dsp->hw_info.samplerate / 22050;
/* number of frags */
j = ((dsp->hw_info.samplerate * arts_dsp_bytes_per_sample[dsp->hw_info.type] * params->bufsize) / (0x01 << i)) + 1;
arts_stream_set(priv->stream, ARTS_P_BUFFER_SIZE, (0x01<<i)*j);
#ifdef ARTS_DEBUG
/* print some info messages ;) */
fprintf(stderr, "info: aRts buffer size : %d\n",
arts_stream_get(priv->stream, ARTS_P_BUFFER_SIZE));
fprintf(stderr, "info: aRts buffer time : %d\n",
arts_stream_get(priv->stream, ARTS_P_BUFFER_TIME));
fprintf(stderr, "info: aRts server latency : %d\n",
arts_stream_get(priv->stream, ARTS_P_SERVER_LATENCY));
fprintf(stderr, "info: aRts total latency : %d\n",
arts_stream_get(priv->stream, ARTS_P_TOTAL_LATENCY));
fprintf(stderr, "info: aRts blocking : %s\n",
arts_stream_get(priv->stream, ARTS_P_BLOCKING) ? "yes" : "no");
#endif /* ifdef ARTS_DEBUG */
return dsp;
}
