static int init(struct ao *ao)
{
struct priv *p = ao->priv;
if (AF_FORMAT_IS_IEC61937(ao->format)) {
MP_WARN(ao, "detected IEC61937, redirecting to coreaudio_exclusive\n");
ao->redirect = "coreaudio_exclusive";
return CONTROL_ERROR;
}
if (!reinit_device(ao))
goto coreaudio_error;
if (p->change_physical_format)
init_physical_format(ao);
if (!ca_init_chmap(ao, p->device))
goto coreaudio_error;
ao->format = af_fmt_from_planar(ao->format);
AudioStreamBasicDescription asbd;
ca_fill_asbd(ao, &asbd);
if (!init_audiounit(ao, asbd))
goto coreaudio_error;
return CONTROL_OK;
coreaudio_error:
return CONTROL_ERROR;
}
void ca_fill_asbd(struct ao *ao, AudioStreamBasicDescription *asbd)
{
asbd->mSampleRate = ao->samplerate;
// Set "AC3" for other spdif formats too - unknown if that works.
asbd->mFormatID = AF_FORMAT_IS_IEC61937(ao->format) ?
kAudioFormat60958AC3 :
kAudioFormatLinearPCM;
asbd->mChannelsPerFrame = ao->channels.num;
asbd->mBitsPerChannel = af_fmt2bits(ao->format);
asbd->mFormatFlags = kAudioFormatFlagIsPacked;
if ((ao->format & AF_FORMAT_TYPE_MASK) == AF_FORMAT_F)
asbd->mFormatFlags |= kAudioFormatFlagIsFloat;
if ((ao->format & AF_FORMAT_SIGN_MASK) == AF_FORMAT_SI)
asbd->mFormatFlags |= kAudioFormatFlagIsSignedInteger;
if (BYTE_ORDER == BIG_ENDIAN)
asbd->mFormatFlags |= kAudioFormatFlagIsBigEndian;
asbd->mFramesPerPacket = 1;
asbd->mBytesPerPacket = asbd->mBytesPerFrame =
asbd->mFramesPerPacket * asbd->mChannelsPerFrame *
(asbd->mBitsPerChannel / 8);
}
/* to set/get/query special features/parameters */
static int control(int cmd, void *arg)
{
switch(cmd) {
case AOCONTROL_QUERY_FORMAT:
return CONTROL_TRUE;
case AOCONTROL_GET_VOLUME:
case AOCONTROL_SET_VOLUME:
{
ao_control_vol_t *vol = (ao_control_vol_t *)arg;
int err;
snd_mixer_t *handle;
snd_mixer_elem_t *elem;
snd_mixer_selem_id_t *sid;
char *mix_name = "PCM";
char *card = "default";
int mix_index = 0;
long pmin, pmax;
long get_vol, set_vol;
float f_multi;
if(AF_FORMAT_IS_IEC61937(ao_data.format))
return CONTROL_TRUE;
if(mixer_channel) {
char *test_mix_index;
mix_name = strdup(mixer_channel);
if ((test_mix_index = strchr(mix_name, ','))){
*test_mix_index = 0;
test_mix_index++;
mix_index = strtol(test_mix_index, &test_mix_index, 0);
if (*test_mix_index){
mp_msg(MSGT_AO,MSGL_ERR,
MSGTR_AO_ALSA_InvalidMixerIndexDefaultingToZero);
mix_index = 0 ;
}
}
}
if(mixer_device) card = mixer_device;
//allocate simple id
snd_mixer_selem_id_alloca(&sid);
//sets simple-mixer index and name
snd_mixer_selem_id_set_index(sid, mix_index);
snd_mixer_selem_id_set_name(sid, mix_name);
if (mixer_channel) {
free(mix_name);
mix_name = NULL;
}
if ((err = snd_mixer_open(&handle, 0)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerOpenError, snd_strerror(err));
return CONTROL_ERROR;
}
if ((err = snd_mixer_attach(handle, card)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerAttachError,
card, snd_strerror(err));
snd_mixer_close(handle);
return CONTROL_ERROR;
}
if ((err = snd_mixer_selem_register(handle, NULL, NULL)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerRegisterError, snd_strerror(err));
snd_mixer_close(handle);
return CONTROL_ERROR;
}
err = snd_mixer_load(handle);
if (err < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerLoadError, snd_strerror(err));
snd_mixer_close(handle);
return CONTROL_ERROR;
}
elem = snd_mixer_find_selem(handle, sid);
if (!elem) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToFindSimpleControl,
snd_mixer_selem_id_get_name(sid), snd_mixer_selem_id_get_index(sid));
snd_mixer_close(handle);
return CONTROL_ERROR;
}
snd_mixer_selem_get_playback_volume_range(elem,&pmin,&pmax);
f_multi = (100 / (float)(pmax - pmin));
if (cmd == AOCONTROL_SET_VOLUME) {
set_vol = vol->left / f_multi + pmin + 0.5;
//setting channels
if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSettingLeftChannel,
snd_strerror(err));
snd_mixer_close(handle);
return CONTROL_ERROR;
}
mp_msg(MSGT_AO,MSGL_DBG2,"left=%li, ", set_vol);
set_vol = vol->right / f_multi + pmin + 0.5;
if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSettingRightChannel,
snd_strerror(err));
snd_mixer_close(handle);
return CONTROL_ERROR;
}
mp_msg(MSGT_AO,MSGL_DBG2,"right=%li, pmin=%li, pmax=%li, mult=%f\n",
set_vol, pmin, pmax, f_multi);
if (snd_mixer_selem_has_playback_switch(elem)) {
int lmute = (vol->left == 0.0);
int rmute = (vol->right == 0.0);
if (snd_mixer_selem_has_playback_switch_joined(elem)) {
lmute = rmute = lmute && rmute;
} else {
snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_RIGHT, !rmute);
}
snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT, !lmute);
}
}
else {
snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, &get_vol);
vol->left = (get_vol - pmin) * f_multi;
snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, &get_vol);
vol->right = (get_vol - pmin) * f_multi;
mp_msg(MSGT_AO,MSGL_DBG2,"left=%f, right=%f\n",vol->left,vol->right);
}
snd_mixer_close(handle);
return CONTROL_OK;
}
} //end switch
return CONTROL_UNKNOWN;
}
/*
open & setup audio device
return: 1=success 0=fail
*/
static int init(int rate_hz, int channels, int format, int flags)
{
unsigned int alsa_buffer_time = 500000; /* 0.5 s */
unsigned int alsa_fragcount = 16;
int err;
int block;
strarg_t device;
snd_pcm_uframes_t chunk_size;
snd_pcm_uframes_t bufsize;
snd_pcm_uframes_t boundary;
const opt_t subopts[] = {
{"block", OPT_ARG_BOOL, &block, NULL},
{"device", OPT_ARG_STR, &device, str_maxlen},
{NULL}
};
char alsa_device[ALSA_DEVICE_SIZE + 1];
// make sure alsa_device is null-terminated even when using strncpy etc.
memset(alsa_device, 0, ALSA_DEVICE_SIZE + 1);
mp_msg(MSGT_AO,MSGL_V,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz,
channels, format);
alsa_handler = NULL;
#if SND_LIB_VERSION >= 0x010005
mp_msg(MSGT_AO,MSGL_V,"alsa-init: using ALSA %s\n", snd_asoundlib_version());
#else
mp_msg(MSGT_AO,MSGL_V,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR);
#endif
snd_lib_error_set_handler(alsa_error_handler);
ao_data.samplerate = rate_hz;
ao_data.format = format;
ao_data.channels = channels;
switch (format)
{
case AF_FORMAT_S8:
alsa_format = SND_PCM_FORMAT_S8;
break;
case AF_FORMAT_U8:
alsa_format = SND_PCM_FORMAT_U8;
break;
case AF_FORMAT_U16_LE:
alsa_format = SND_PCM_FORMAT_U16_LE;
break;
case AF_FORMAT_U16_BE:
alsa_format = SND_PCM_FORMAT_U16_BE;
break;
case AF_FORMAT_AC3_LE:
case AF_FORMAT_S16_LE:
case AF_FORMAT_IEC61937_LE:
alsa_format = SND_PCM_FORMAT_S16_LE;
break;
case AF_FORMAT_AC3_BE:
case AF_FORMAT_S16_BE:
case AF_FORMAT_IEC61937_BE:
alsa_format = SND_PCM_FORMAT_S16_BE;
break;
case AF_FORMAT_U32_LE:
alsa_format = SND_PCM_FORMAT_U32_LE;
break;
case AF_FORMAT_U32_BE:
alsa_format = SND_PCM_FORMAT_U32_BE;
break;
case AF_FORMAT_S32_LE:
alsa_format = SND_PCM_FORMAT_S32_LE;
break;
case AF_FORMAT_S32_BE:
alsa_format = SND_PCM_FORMAT_S32_BE;
break;
case AF_FORMAT_U24_LE:
alsa_format = SND_PCM_FORMAT_U24_3LE;
break;
case AF_FORMAT_U24_BE:
alsa_format = SND_PCM_FORMAT_U24_3BE;
break;
case AF_FORMAT_S24_LE:
alsa_format = SND_PCM_FORMAT_S24_3LE;
break;
case AF_FORMAT_S24_BE:
alsa_format = SND_PCM_FORMAT_S24_3BE;
break;
case AF_FORMAT_FLOAT_LE:
alsa_format = SND_PCM_FORMAT_FLOAT_LE;
break;
case AF_FORMAT_FLOAT_BE:
alsa_format = SND_PCM_FORMAT_FLOAT_BE;
break;
case AF_FORMAT_MU_LAW:
alsa_format = SND_PCM_FORMAT_MU_LAW;
break;
case AF_FORMAT_A_LAW:
alsa_format = SND_PCM_FORMAT_A_LAW;
break;
default:
alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1
break;
}
//subdevice parsing
// set defaults
block = 1;
/* switch for spdif
* sets opening sequence for SPDIF
* sets also the playback and other switches 'on the fly'
* while opening the abstract alias for the spdif subdevice
* 'iec958'
*/
if (AF_FORMAT_IS_IEC61937(format)) {
device.str = "iec958";
mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3/iec61937/iec958, %i channels\n", channels);
}
else
/* in any case for multichannel playback we should select
* appropriate device
*/
switch (channels) {
case 1:
case 2:
device.str = "default";
mp_msg(MSGT_AO,MSGL_V,"alsa-init: setup for 1/2 channel(s)\n");
break;
case 4:
if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
// hack - use the converter plugin
device.str = "plug:surround40";
else
device.str = "surround40";
mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround40\n");
break;
case 6:
if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
device.str = "plug:surround51";
else
device.str = "surround51";
mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround51\n");
break;
case 8:
if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
device.str = "plug:surround71";
else
device.str = "surround71";
mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround71\n");
break;
default:
device.str = "default";
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ChannelsNotSupported,channels);
}
device.len = strlen(device.str);
if (subopt_parse(ao_subdevice, subopts) != 0) {
print_help();
return 0;
}
parse_device(alsa_device, device.str, device.len);
mp_msg(MSGT_AO,MSGL_V,"alsa-init: using device %s\n", alsa_device);
if (!alsa_handler) {
int open_mode = block ? 0 : SND_PCM_NONBLOCK;
int isac3 = AF_FORMAT_IS_IEC61937(format);
//modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
mp_msg(MSGT_AO,MSGL_V,"alsa-init: opening device in %sblocking mode\n", block ? "" : "non-");
if ((err = try_open_device(alsa_device, open_mode, isac3)) < 0)
{
if (err != -EBUSY && !block) {
mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_OpenInNonblockModeFailed);
if ((err = try_open_device(alsa_device, 0, isac3)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err));
return 0;
}
} else {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err));
return 0;
}
}
if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSetBlockMode, snd_strerror(err));
} else {
mp_msg(MSGT_AO,MSGL_V,"alsa-init: device reopened in blocking mode\n");
}
snd_pcm_hw_params_alloca(&alsa_hwparams);
snd_pcm_sw_params_alloca(&alsa_swparams);
// setting hw-parameters
if ((err = snd_pcm_hw_params_any(alsa_handler, alsa_hwparams)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetInitialParameters,
snd_strerror(err));
return 0;
}
err = snd_pcm_hw_params_set_access(alsa_handler, alsa_hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetAccessType,
snd_strerror(err));
return 0;
}
/* workaround for nonsupported formats
sets default format to S16_LE if the given formats aren't supported */
if ((err = snd_pcm_hw_params_test_format(alsa_handler, alsa_hwparams,
alsa_format)) < 0)
{
mp_msg(MSGT_AO,MSGL_INFO,
MSGTR_AO_ALSA_FormatNotSupportedByHardware, af_fmt2str_short(format));
alsa_format = SND_PCM_FORMAT_S16_LE;
if (AF_FORMAT_IS_AC3(ao_data.format))
ao_data.format = AF_FORMAT_AC3_LE;
else if (AF_FORMAT_IS_IEC61937(ao_data.format))
ao_data.format = AF_FORMAT_IEC61937_LE;
else
ao_data.format = AF_FORMAT_S16_LE;
}
if ((err = snd_pcm_hw_params_set_format(alsa_handler, alsa_hwparams,
alsa_format)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetFormat,
snd_strerror(err));
return 0;
}
if ((err = snd_pcm_hw_params_set_channels_near(alsa_handler, alsa_hwparams,
&ao_data.channels)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetChannels,
snd_strerror(err));
return 0;
}
/* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11)
prefer our own resampler, since that allows users to choose the resampler,
even per file if desired */
#if SND_LIB_VERSION >= 0x010009
if ((err = snd_pcm_hw_params_set_rate_resample(alsa_handler, alsa_hwparams,
0)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToDisableResampling,
snd_strerror(err));
return 0;
}
#endif
if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams,
&ao_data.samplerate, NULL)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetSamplerate2,
snd_strerror(err));
return 0;
}
bytes_per_sample = af_fmt2bits(ao_data.format) / 8;
bytes_per_sample *= ao_data.channels;
ao_data.bps = ao_data.samplerate * bytes_per_sample;
if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams,
&alsa_buffer_time, NULL)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetBufferTimeNear,
snd_strerror(err));
return 0;
}
if ((err = snd_pcm_hw_params_set_periods_near(alsa_handler, alsa_hwparams,
&alsa_fragcount, NULL)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetPeriods,
snd_strerror(err));
return 0;
}
/* finally install hardware parameters */
if ((err = snd_pcm_hw_params(alsa_handler, alsa_hwparams)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetHwParameters,
snd_strerror(err));
return 0;
}
// end setting hw-params
// gets buffersize for control
if ((err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetBufferSize, snd_strerror(err));
return 0;
}
else {
ao_data.buffersize = bufsize * bytes_per_sample;
mp_msg(MSGT_AO,MSGL_V,"alsa-init: got buffersize=%i\n", ao_data.buffersize);
}
if ((err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetPeriodSize, snd_strerror(err));
return 0;
} else {
mp_msg(MSGT_AO,MSGL_V,"alsa-init: got period size %li\n", chunk_size);
}
ao_data.outburst = chunk_size * bytes_per_sample;
/* setting software parameters */
if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetSwParameters,
snd_strerror(err));
return 0;
}
#if SND_LIB_VERSION >= 0x000901
if ((err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetBoundary,
snd_strerror(err));
return 0;
}
#else
boundary = 0x7fffffff;
#endif
/* start playing when one period has been written */
if ((err = snd_pcm_sw_params_set_start_threshold(alsa_handler, alsa_swparams, chunk_size)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetStartThreshold,
snd_strerror(err));
return 0;
}
/* disable underrun reporting */
if ((err = snd_pcm_sw_params_set_stop_threshold(alsa_handler, alsa_swparams, boundary)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetStopThreshold,
snd_strerror(err));
return 0;
}
#if SND_LIB_VERSION >= 0x000901
/* play silence when there is an underrun */
if ((err = snd_pcm_sw_params_set_silence_size(alsa_handler, alsa_swparams, boundary)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetSilenceSize,
snd_strerror(err));
return 0;
}
#endif
if ((err = snd_pcm_sw_params(alsa_handler, alsa_swparams)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetSwParameters,
snd_strerror(err));
return 0;
}
/* end setting sw-params */
mp_msg(MSGT_AO,MSGL_V,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
ao_data.samplerate, ao_data.channels, (int)bytes_per_sample, ao_data.buffersize,
snd_pcm_format_description(alsa_format));
} // end switch alsa_handler (spdif)
alsa_can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);
return 1;
} // end init
// Initialization and runtime control
static int control(struct af_instance* af, int cmd, void* arg)
{
switch(cmd){
case AF_CONTROL_REINIT:{
char buf1[256];
char buf2[256];
struct mp_audio *data = arg;
int supported_ac3 = 0;
// Make sure this filter isn't redundant
if(af->data->format == data->format)
return AF_DETACH;
// A bit complex because we can convert AC3
// to generic iec61937 but not the other way
// round.
if (AF_FORMAT_IS_AC3(af->data->format))
supported_ac3 = AF_FORMAT_IS_AC3(data->format);
else if (AF_FORMAT_IS_IEC61937(af->data->format))
supported_ac3 = AF_FORMAT_IS_IEC61937(data->format);
// Allow trivial AC3-endianness conversion
if (!supported_ac3)
// Check for errors in configuration
if((AF_OK != check_bps(data->bps)) ||
(AF_OK != check_format(data->format)) ||
(AF_OK != check_bps(af->data->bps)) ||
(AF_OK != check_format(af->data->format)))
return AF_ERROR;
af_fmt2str(data->format,buf1,256);
af_fmt2str(af->data->format,buf2,256);
mp_msg(MSGT_AFILTER, MSGL_V, "[format] Changing sample format from %s to %s\n",
buf1, buf2);
af->data->rate = data->rate;
mp_audio_set_channels(af->data, &data->channels);
af->mul = (double)af->data->bps / data->bps;
af->play = play; // set default
// look whether only endianness differences are there
if ((af->data->format & ~AF_FORMAT_END_MASK) ==
(data->format & ~AF_FORMAT_END_MASK))
{
mp_msg(MSGT_AFILTER, MSGL_V, "[format] Accelerated endianness conversion only\n");
af->play = play_swapendian;
}
if ((data->format == AF_FORMAT_FLOAT_NE) &&
(af->data->format == AF_FORMAT_S16_NE))
{
mp_msg(MSGT_AFILTER, MSGL_V, "[format] Accelerated %s to %s conversion\n",
buf1, buf2);
af->play = play_float_s16;
}
if ((data->format == AF_FORMAT_S16_NE) &&
(af->data->format == AF_FORMAT_FLOAT_NE))
{
mp_msg(MSGT_AFILTER, MSGL_V, "[format] Accelerated %s to %s conversion\n",
buf1, buf2);
af->play = play_s16_float;
}
return AF_OK;
}
case AF_CONTROL_COMMAND_LINE:{
int format = af_str2fmt_short(bstr0(arg));
if (!format) {
mp_msg(MSGT_AFILTER, MSGL_ERR, "[format] %s is not a valid format\n", (char *)arg);
return AF_ERROR;
}
if(AF_OK != af->control(af, AF_CONTROL_FORMAT_FMT | AF_CONTROL_SET,&format))
return AF_ERROR;
return AF_OK;
}
case AF_CONTROL_FORMAT_FMT | AF_CONTROL_SET:{
// Check for errors in configuration
if(!AF_FORMAT_IS_AC3(*(int*)arg) && AF_OK != check_format(*(int*)arg))
return AF_ERROR;
mp_audio_set_format(af->data, *(int*)arg);
return AF_OK;
}
}
return AF_UNKNOWN;
}
