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;
}
/* public methods (static but exported through the sysdep_dsp or plugin
struct) */
static void *arts_dsp_create(const void *flags)
{
struct sysdep_dsp_struct *dsp = NULL;
struct arts_dsp_priv_data *priv = 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;
}
if(!(priv = calloc(1, sizeof(struct arts_dsp_priv_data))))
{
fprintf(stderr,
"error malloc failed for struct arts_dsp_priv_data\n");
return NULL;
}
/* fill in the functions and some data */
priv->stream=0;
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;
dsp->hw_info.bufsize = 1024;
/* open the sound device */
arts_init();
priv->stream=arts_play_stream(dsp->hw_info.samplerate,
(dsp->hw_info.type&SYSDEP_DSP_16BIT)?16:8,
(dsp->hw_info.type&SYSDEP_DSP_STEREO)?2:1,
"xmame arts");
/* Set the buffering time */
arts_stream_set(priv->stream,ARTS_P_BUFFER_TIME,btime);
/* set non-blocking mode if selected */
if(params->flags & SYSDEP_DSP_O_NONBLOCK)
{
arts_stream_set(priv->stream,ARTS_P_BLOCKING,0);
}
return dsp;
}
/* 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 acntl(int request, void *arg)
{
int tmp, tmp1, samples;
switch(request)
{
case PM_REQ_DISCARD: /* Discard stream */
arts_close_stream(stream);
stream=NULL;
return 0;
case PM_REQ_RATE: /* Change sample rate */
arts_close_stream(stream);
tmp = (dpm.encoding & PE_16BIT) ? 16 : 8;
tmp1 = (dpm.encoding & PE_MONO) ? 1 : 2;
stream = arts_play_stream(*(int*)arg,
LE_LONG(tmp),
tmp1,
"timidity");
server_buffer = arts_stream_get(stream, ARTS_P_SERVER_LATENCY) * dpm.rate * (tmp/8) * tmp1 / 1000;
return 0;
case PM_REQ_GETQSIZ: /* Get maximum queue size */
*(int*)arg = arts_stream_get(stream, ARTS_P_BUFFER_SIZE);
return 0;
case PM_REQ_SETQSIZ: /* Set queue size */
*(int*)arg = arts_stream_set(stream, ARTS_P_BUFFER_SIZE, *(int*)arg);
return 0;
case PM_REQ_GETFRAGSIZ: /* Get device fragment size */
*(int*)arg = arts_stream_get(stream, ARTS_P_PACKET_SIZE);
return 0;
case PM_REQ_GETSAMPLES: /* Get current play sample */
tmp = arts_stream_get(stream, ARTS_P_BUFFER_SIZE) -
arts_stream_get(stream, ARTS_P_BUFFER_SPACE) +
server_buffer;
samples = output_count - tmp;
if(samples < 0)
samples = 0;
if(!(dpm.encoding & PE_MONO)) samples >>= 1;
if(dpm.encoding & PE_16BIT) samples >>= 1;
*(int*)arg = samples;
return 0;
case PM_REQ_GETFILLABLE: /* Get fillable device queue size */
*(int*)arg = arts_stream_get(stream, ARTS_P_BUFFER_SPACE);
return 0;
case PM_REQ_GETFILLED: /* Get filled device queue size */
*(int*)arg = arts_stream_get(stream, ARTS_P_BUFFER_SIZE) - arts_stream_get(stream, ARTS_P_BUFFER_SPACE);
return 0;
/* The following are not (yet) implemented: */
case PM_REQ_FLUSH: /* Wait until playback is complete */
case PM_REQ_MIDI: /* Send MIDI event */
case PM_REQ_INST_NAME: /* Get instrument name */
return -1;
case PM_REQ_OUTPUT_FINISH: /* Sent after last output_data */
case PM_REQ_PLAY_START: /* Called just before playing */
case PM_REQ_PLAY_END: /* Called just after playing */
return 0;
}
return -1;
}
/* configure the audio buffer sizes. The latency parameter is in millisec. */
void audio_set_latency(audio_dev_handle* handle, int latency)
{
handle->latency = latency;
#ifdef HAS_ARTS
if(handle->use_arts)
{
arts_close_stream(handle->arts_handle);
handle->arts_handle = arts_play_stream(handle->rate, 8, handle->channels+1, "arts-whitenoise");
arts_stream_set(handle->arts_handle, ARTS_P_BUFFER_TIME, latency);
}
else
#endif
{
snd_pcm_close(handle->alsa_handle);
alsa_init(handle);
}
}
static int op_arts_open(sample_format_t sf, const channel_position_t *channel_map)
{
int buffer_time, server_latency, total_latency;
int blocking;
arts_sf = sf;
arts_stream = arts_play_stream(sf_get_rate(arts_sf), sf_get_bits(arts_sf),
sf_get_channels(arts_sf), "cmus");
blocking = arts_stream_set(arts_stream, ARTS_P_BLOCKING, 0);
if (blocking) {
}
arts_buffer_size = arts_stream_get(arts_stream, ARTS_P_BUFFER_SIZE);
if (arts_buffer_size < 0) {
}
buffer_time = arts_stream_get(arts_stream, ARTS_P_BUFFER_TIME);
server_latency = arts_stream_get(arts_stream, ARTS_P_SERVER_LATENCY);
total_latency = arts_stream_get(arts_stream, ARTS_P_TOTAL_LATENCY);
d_print("buffer_time: %d\n", buffer_time);
d_print("server_latency: %d\n", server_latency);
d_print("total_latency: %d\n", total_latency);
return 0;
}
static int ARTSC_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
int bits, frag_spec;
Uint16 test_format, format;
/* Reset the timer synchronization flag */
frame_ticks = 0.0;
mixbuf = NULL;
/* Try for a closest match on audio format */
format = 0;
bits = 0;
for ( test_format = SDL_FirstAudioFormat(spec->format);
! format && test_format; ) {
#ifdef DEBUG_AUDIO
fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
#endif
switch ( test_format ) {
case AUDIO_U8:
bits = 8;
format = 1;
break;
case AUDIO_S16LSB:
bits = 16;
format = 1;
break;
default:
format = 0;
break;
}
if ( ! format ) {
test_format = SDL_NextAudioFormat();
}
}
if ( format == 0 ) {
SDL_SetError("Couldn't find any hardware audio formats");
return(-1);
}
spec->format = test_format;
stream = arts_play_stream(spec->freq, bits, spec->channels, "SDL");
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(spec);
/* Determine the power of two of the fragment size */
for ( frag_spec = 0; (0x01<<frag_spec) < spec->size; ++frag_spec );
if ( (0x01<<frag_spec) != spec->size ) {
SDL_SetError("Fragment size must be a power of two");
return(-1);
}
frag_spec |= 0x00020000; /* two fragments, for low latency */
#ifdef ARTS_P_PACKET_SETTINGS
arts_stream_set(stream, ARTS_P_PACKET_SETTINGS, frag_spec);
#else
arts_stream_set(stream, ARTS_P_PACKET_SIZE, frag_spec&0xffff);
arts_stream_set(stream, ARTS_P_PACKET_COUNT, frag_spec>>16);
#endif
spec->size = arts_stream_get(stream, ARTS_P_PACKET_SIZE);
/* Allocate mixing buffer */
mixlen = spec->size;
mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
if ( mixbuf == NULL ) {
return(-1);
}
memset(mixbuf, spec->silence, spec->size);
/* Get the parent process id (we're the parent of the audio thread) */
parent = getpid();
/* We're ready to rock and roll. :-) */
return(0);
}
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;
}