// to set/get/query special features/parameters
static int control(int cmd,void *arg){
switch(cmd){
case AOCONTROL_SET_DEVICE:
dsp=(char*)arg;
return CONTROL_OK;
case AOCONTROL_GET_DEVICE:
*(char**)arg=dsp;
return CONTROL_OK;
#ifdef SNDCTL_DSP_GETFMTS
case AOCONTROL_QUERY_FORMAT:
{
int format;
if (!ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &format))
if ((unsigned int)format & (unsigned long)arg)
return CONTROL_TRUE;
return CONTROL_FALSE;
}
#endif
case AOCONTROL_GET_VOLUME:
case AOCONTROL_SET_VOLUME:
{
ao_control_vol_t *vol = (ao_control_vol_t *)arg;
int fd, v, devs;
if(AF_FORMAT_IS_AC3(ao_data.format))
return CONTROL_TRUE;
if ((fd = open(oss_mixer_device, O_RDONLY)) > 0)
{
ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs);
if (devs & (1 << oss_mixer_channel))
{
if (cmd == AOCONTROL_GET_VOLUME)
{
ioctl(fd, MIXER_READ(oss_mixer_channel), &v);
vol->right = (v & 0xFF00) >> 8;
vol->left = v & 0x00FF;
}
else
{
v = ((int)vol->right << 8) | (int)vol->left;
ioctl(fd, MIXER_WRITE(oss_mixer_channel), &v);
}
}
else
{
int af_fmt2bits(int format)
{
if (AF_FORMAT_IS_AC3(format)) return 16;
return (format & AF_FORMAT_BITS_MASK)+8;
// return (((format & AF_FORMAT_BITS_MASK)>>3)+1) * 8;
#if 0
switch(format & AF_FORMAT_BITS_MASK)
{
case AF_FORMAT_8BIT:
return 8;
case AF_FORMAT_16BIT:
return 16;
case AF_FORMAT_24BIT:
return 24;
case AF_FORMAT_32BIT:
return 32;
case AF_FORMAT_48BIT:
return 48;
}
#endif
return -1;
}
// open & setup audio device
// return: 1=success 0=fail
static int init(int rate,int channels,int format,int flags)
{
WAVEFORMATEXTENSIBLE wformat;
MMRESULT result;
unsigned char* buffer;
int i;
if (AF_FORMAT_IS_AC3(format))
format = AF_FORMAT_AC3_NE;
switch(format){
case AF_FORMAT_AC3_NE:
case AF_FORMAT_S24_LE:
case AF_FORMAT_S16_LE:
case AF_FORMAT_U8:
break;
default:
mp_msg(MSGT_AO, MSGL_V,"ao_win32: format %s not supported defaulting to Signed 16-bit Little-Endian\n",af_fmt2str_short(format));
format=AF_FORMAT_S16_LE;
}
//fill global ao_data
ao_data.channels=channels;
ao_data.samplerate=rate;
ao_data.format=format;
ao_data.bps=channels*rate;
ao_data.bps*=af_fmt2bits(format)/8;
if(ao_data.buffersize==-1)
{
ao_data.buffersize=af_fmt2bits(format)/8;
ao_data.buffersize*= channels;
ao_data.buffersize*= SAMPLESIZE;
}
ao_data.outburst = ao_data.buffersize;
mp_msg(MSGT_AO, MSGL_V,"ao_win32: Samplerate:%iHz Channels:%i Format:%s\n",rate, channels, af_fmt2str_short(format));
mp_msg(MSGT_AO, MSGL_V,"ao_win32: Buffersize:%d\n",ao_data.buffersize);
//fill waveformatex
ZeroMemory( &wformat, sizeof(WAVEFORMATEXTENSIBLE));
wformat.Format.cbSize = (channels>2)?sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX):0;
wformat.Format.nChannels = channels;
wformat.Format.nSamplesPerSec = rate;
wformat.Format.wBitsPerSample = af_fmt2bits(format);
if(AF_FORMAT_IS_AC3(format))
{
wformat.Format.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF;
wformat.Format.nBlockAlign = 4;
}
else
{
wformat.Format.wFormatTag = (channels>2)?WAVE_FORMAT_EXTENSIBLE:WAVE_FORMAT_PCM;
wformat.Format.nBlockAlign = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3);
}
if(channels>2)
{
wformat.dwChannelMask = channel_mask[channels-3];
wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
wformat.Samples.wValidBitsPerSample=af_fmt2bits(format);
}
wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign;
//open sound device
//WAVE_MAPPER always points to the default wave device on the system
result = waveOutOpen(&hWaveOut,WAVE_MAPPER,(WAVEFORMATEX*)&wformat,(DWORD_PTR)waveOutProc,0,CALLBACK_FUNCTION);
if(result == WAVERR_BADFORMAT)
{
mp_msg(MSGT_AO, MSGL_ERR,"ao_win32: format not supported switching to default\n");
ao_data.channels = wformat.Format.nChannels = 2;
ao_data.samplerate = wformat.Format.nSamplesPerSec = 44100;
ao_data.format = AF_FORMAT_S16_LE;
ao_data.bps=ao_data.channels * ao_data.samplerate*2;
wformat.Format.wBitsPerSample=16;
wformat.Format.wFormatTag=WAVE_FORMAT_PCM;
wformat.Format.nBlockAlign = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3);
wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign;
ao_data.buffersize=(wformat.Format.wBitsPerSample>>3)*wformat.Format.nChannels*SAMPLESIZE;
result = waveOutOpen(&hWaveOut,WAVE_MAPPER,(WAVEFORMATEX*)&wformat,(DWORD_PTR)waveOutProc,0,CALLBACK_FUNCTION);
}
/**
\brief setup sound device
\param rate samplerate
\param channels number of channels
\param format format
\param flags unused
\return 0=success -1=fail
*/
static int init(struct ao *ao)
{
struct priv *p = ao->priv;
int res;
if (!InitDirectSound(ao))
return -1;
ao->no_persistent_volume = true;
p->audio_volume = 100;
// ok, now create the buffers
WAVEFORMATEXTENSIBLE wformat;
DSBUFFERDESC dsbpridesc;
DSBUFFERDESC dsbdesc;
int format = af_fmt_from_planar(ao->format);
int rate = ao->samplerate;
if (AF_FORMAT_IS_AC3(format))
format = AF_FORMAT_AC3;
else {
struct mp_chmap_sel sel = {0};
mp_chmap_sel_add_waveext(&sel);
if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
return -1;
}
switch (format) {
case AF_FORMAT_AC3:
case AF_FORMAT_S24_LE:
case AF_FORMAT_S16_LE:
case AF_FORMAT_U8:
break;
default:
MP_VERBOSE(ao, "format %s not supported defaulting to Signed 16-bit Little-Endian\n",
af_fmt_to_str(format));
format = AF_FORMAT_S16_LE;
}
//set our audio parameters
ao->samplerate = rate;
ao->format = format;
ao->bps = ao->channels.num * rate * (af_fmt2bits(format) >> 3);
int buffersize = ao->bps; // space for 1 sec
MP_VERBOSE(ao, "Samplerate:%iHz Channels:%i Format:%s\n", rate,
ao->channels.num, af_fmt_to_str(format));
MP_VERBOSE(ao, "Buffersize:%d bytes (%d msec)\n",
buffersize, buffersize / ao->bps * 1000);
//fill waveformatex
ZeroMemory(&wformat, sizeof(WAVEFORMATEXTENSIBLE));
wformat.Format.cbSize = (ao->channels.num > 2)
? sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX) : 0;
wformat.Format.nChannels = ao->channels.num;
wformat.Format.nSamplesPerSec = rate;
if (AF_FORMAT_IS_AC3(format)) {
wformat.Format.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF;
wformat.Format.wBitsPerSample = 16;
wformat.Format.nBlockAlign = 4;
} else {
wformat.Format.wFormatTag = (ao->channels.num > 2)
? WAVE_FORMAT_EXTENSIBLE : WAVE_FORMAT_PCM;
wformat.Format.wBitsPerSample = af_fmt2bits(format);
wformat.Format.nBlockAlign = wformat.Format.nChannels *
(wformat.Format.wBitsPerSample >> 3);
}
// fill in primary sound buffer descriptor
memset(&dsbpridesc, 0, sizeof(DSBUFFERDESC));
dsbpridesc.dwSize = sizeof(DSBUFFERDESC);
dsbpridesc.dwFlags = DSBCAPS_PRIMARYBUFFER;
dsbpridesc.dwBufferBytes = 0;
dsbpridesc.lpwfxFormat = NULL;
// fill in the secondary sound buffer (=stream buffer) descriptor
memset(&dsbdesc, 0, sizeof(DSBUFFERDESC));
dsbdesc.dwSize = sizeof(DSBUFFERDESC);
dsbdesc.dwFlags = DSBCAPS_GETCURRENTPOSITION2 /** Better position accuracy */
| DSBCAPS_GLOBALFOCUS /** Allows background playing */
| DSBCAPS_CTRLVOLUME; /** volume control enabled */
if (ao->channels.num > 2) {
wformat.dwChannelMask = mp_chmap_to_waveext(&ao->channels);
wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
wformat.Samples.wValidBitsPerSample = wformat.Format.wBitsPerSample;
// Needed for 5.1 on emu101k - shit soundblaster
dsbdesc.dwFlags |= DSBCAPS_LOCHARDWARE;
}
wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec *
wformat.Format.nBlockAlign;
dsbdesc.dwBufferBytes = buffersize;
dsbdesc.lpwfxFormat = (WAVEFORMATEX *)&wformat;
p->buffer_size = dsbdesc.dwBufferBytes;
p->write_offset = 0;
p->min_free_space = wformat.Format.nBlockAlign;
p->outburst = wformat.Format.nBlockAlign * 512;
// create primary buffer and set its format
res = IDirectSound_CreateSoundBuffer(p->hds, &dsbpridesc, &p->hdspribuf, NULL);
if (res != DS_OK) {
UninitDirectSound(ao);
MP_ERR(ao, "cannot create primary buffer (%s)\n", dserr2str(res));
return -1;
}
res = IDirectSoundBuffer_SetFormat(p->hdspribuf, (WAVEFORMATEX *)&wformat);
if (res != DS_OK) {
MP_WARN(ao, "cannot set primary buffer format (%s), using "
"standard setting (bad quality)", dserr2str(res));
}
MP_VERBOSE(ao, "primary buffer created\n");
// now create the stream buffer
res = IDirectSound_CreateSoundBuffer(p->hds, &dsbdesc, &p->hdsbuf, NULL);
if (res != DS_OK) {
if (dsbdesc.dwFlags & DSBCAPS_LOCHARDWARE) {
// Try without DSBCAPS_LOCHARDWARE
dsbdesc.dwFlags &= ~DSBCAPS_LOCHARDWARE;
res = IDirectSound_CreateSoundBuffer(p->hds, &dsbdesc, &p->hdsbuf, NULL);
}
if (res != DS_OK) {
UninitDirectSound(ao);
MP_ERR(ao, "cannot create secondary (stream)buffer (%s)\n",
dserr2str(res));
return -1;
}
}
MP_VERBOSE(ao, "secondary (stream)buffer created\n");
return 0;
}
// Initialization and runtime control
static int control(struct af_instance_s* af, int cmd, void* arg)
{
switch(cmd){
case AF_CONTROL_REINIT:{
char buf1[256];
char buf2[256];
af_data_t *data = arg;
// Make sure this filter isn't redundant
if(af->data->format == data->format &&
af->data->bps == data->bps)
return AF_DETACH;
// Allow trivial AC3-endianness conversion
if (!AF_FORMAT_IS_AC3(af->data->format) || !AF_FORMAT_IS_AC3(data->format))
// 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;
mp_msg(MSGT_AFILTER, MSGL_V, "[format] Changing sample format from %s to %s\n",
af_fmt2str(data->format,buf1,256),
af_fmt2str(af->data->format,buf2,256));
af->data->rate = data->rate;
af->data->nch = data->nch;
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",
af_fmt2str(data->format,buf1,256),
af_fmt2str(af->data->format,buf2,256));
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",
af_fmt2str(data->format,buf1,256),
af_fmt2str(af->data->format,buf2,256));
af->play = play_s16_float;
}
return AF_OK;
}
case AF_CONTROL_COMMAND_LINE:{
int format = af_str2fmt_short(bstr0(arg));
if (format == -1) {
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;
af->data->format = *(int*)arg;
af->data->bps = af_fmt2bits(af->data->format)/8;
return AF_OK;
}
}
return AF_UNKNOWN;
}
// Initialization and runtime control
static int control(struct af_instance_s *af, int cmd, void *arg)
{
af_ac3enc_t *s = (af_ac3enc_t *)af->setup;
af_data_t *data = (af_data_t *)arg;
int i, bit_rate, test_output_res;
static const int default_bit_rate[AC3_MAX_CHANNELS+1] = \
{0, 96000, 192000, 256000, 384000, 448000, 448000};
switch (cmd){
case AF_CONTROL_REINIT:
if (AF_FORMAT_IS_AC3(data->format) || data->nch < s->min_channel_num)
return AF_DETACH;
s->pending_len = 0;
s->expect_len = AC3_FRAME_SIZE * data->nch * data->bps;
if (s->add_iec61937_header)
af->mul = (double)AC3_FRAME_SIZE * 2 * 2 / s->expect_len;
else
af->mul = (double)AC3_MAX_CODED_FRAME_SIZE / s->expect_len;
mp_msg(MSGT_AFILTER, MSGL_DBG2, "af_lavcac3enc reinit: %d, %d, %f, %d.\n",
data->nch, data->rate, af->mul, s->expect_len);
af->data->format = AF_FORMAT_S16_NE;
if (data->rate == 48000 || data->rate == 44100 || data->rate == 32000)
af->data->rate = data->rate;
else
af->data->rate = 48000;
if (data->nch > AC3_MAX_CHANNELS)
af->data->nch = AC3_MAX_CHANNELS;
else
af->data->nch = data->nch;
af->data->bps = 2;
test_output_res = af_test_output(af, data);
bit_rate = s->bit_rate ? s->bit_rate : default_bit_rate[af->data->nch];
if (s->lavc_actx->channels != af->data->nch ||
s->lavc_actx->sample_rate != af->data->rate ||
s->lavc_actx->bit_rate != bit_rate) {
if (s->lavc_actx->codec)
avcodec_close(s->lavc_actx);
// Put sample parameters
s->lavc_actx->channels = af->data->nch;
s->lavc_actx->sample_rate = af->data->rate;
s->lavc_actx->bit_rate = bit_rate;
if(avcodec_open(s->lavc_actx, s->lavc_acodec) < 0) {
mp_msg(MSGT_AFILTER, MSGL_ERR, MSGTR_CouldntOpenCodec, "ac3", bit_rate);
return AF_ERROR;
}
}
af->data->format = AF_FORMAT_AC3_BE;
af->data->nch = 2;
return test_output_res;
case AF_CONTROL_COMMAND_LINE:
mp_msg(MSGT_AFILTER, MSGL_DBG2, "af_lavcac3enc cmdline: %s.\n", (char*)arg);
s->bit_rate = 0;
s->min_channel_num = 0;
s->add_iec61937_header = 0;
sscanf((char*)arg,"%d:%d:%d", &s->add_iec61937_header, &s->bit_rate,
&s->min_channel_num);
if (s->bit_rate < 1000)
s->bit_rate *= 1000;
if (s->bit_rate) {
for (i = 0; i < 19; ++i)
if (ff_ac3_bitrate_tab[i] * 1000 == s->bit_rate)
break;
if (i >= 19) {
mp_msg(MSGT_AFILTER, MSGL_WARN, "af_lavcac3enc unable set unsupported "
"bitrate %d, use default bitrate (check manpage to see "
"supported bitrates).\n", s->bit_rate);
s->bit_rate = 0;
}
}
if (s->min_channel_num == 0)
s->min_channel_num = 5;
mp_msg(MSGT_AFILTER, MSGL_V, "af_lavcac3enc config spdif:%d, bitrate:%d, "
"minchnum:%d.\n", s->add_iec61937_header, s->bit_rate,
s->min_channel_num);
return AF_OK;
}
return AF_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;
}
/* 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_AC3(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;
}
static int init(int rate,int channels,int format,int flags)
{
AudioStreamBasicDescription inDesc;
ComponentDescription desc;
Component comp;
AURenderCallbackStruct renderCallback;
OSStatus err;
UInt32 size, maxFrames, b_alive;
char *psz_name;
AudioDeviceID devid_def = 0;
int device_id, display_help = 0;
const opt_t subopts[] = {
{"device_id", OPT_ARG_INT, &device_id, NULL},
{"help", OPT_ARG_BOOL, &display_help, NULL},
{NULL}
};
// set defaults
device_id = 0;
if (subopt_parse(ao_subdevice, subopts) != 0 || display_help) {
print_help();
if (!display_help)
return 0;
}
ao_msg(MSGT_AO,MSGL_V, "init([%dHz][%dch][%s][%d])\n", rate, channels, af_fmt2str_short(format), flags);
ao = calloc(1, sizeof(ao_coreaudio_t));
ao->i_selected_dev = 0;
ao->b_supports_digital = 0;
ao->b_digital = 0;
ao->b_muted = 0;
ao->b_stream_format_changed = 0;
ao->i_hog_pid = -1;
ao->i_stream_id = 0;
ao->i_stream_index = -1;
ao->b_revert = 0;
ao->b_changed_mixing = 0;
if (device_id == 0) {
/* Find the ID of the default Device. */
err = GetAudioProperty(kAudioObjectSystemObject,
kAudioHardwarePropertyDefaultOutputDevice,
sizeof(UInt32), &devid_def);
if (err != noErr)
{
ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device: [%4.4s]\n", (char *)&err);
goto err_out;
}
} else {
devid_def = device_id;
}
/* Retrieve the name of the device. */
err = GetAudioPropertyString(devid_def,
kAudioObjectPropertyName,
&psz_name);
if (err != noErr)
{
ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device name: [%4.4s]\n", (char *)&err);
goto err_out;
}
ao_msg(MSGT_AO,MSGL_V, "got audio output device ID: %"PRIu32" Name: %s\n", devid_def, psz_name );
/* Probe whether device support S/PDIF stream output if input is AC3 stream. */
if (AF_FORMAT_IS_AC3(format)) {
if (AudioDeviceSupportsDigital(devid_def))
{
ao->b_supports_digital = 1;
}
ao_msg(MSGT_AO, MSGL_V,
"probe default audio output device about support for digital s/pdif output: %d\n",
ao->b_supports_digital );
}
free(psz_name);
// Save selected device id
ao->i_selected_dev = devid_def;
// Build Description for the input format
inDesc.mSampleRate=rate;
inDesc.mFormatID=ao->b_supports_digital ? kAudioFormat60958AC3 : kAudioFormatLinearPCM;
inDesc.mChannelsPerFrame=channels;
inDesc.mBitsPerChannel=af_fmt2bits(format);
if((format&AF_FORMAT_POINT_MASK)==AF_FORMAT_F) {
// float
inDesc.mFormatFlags = kAudioFormatFlagIsFloat|kAudioFormatFlagIsPacked;
}
else if((format&AF_FORMAT_SIGN_MASK)==AF_FORMAT_SI) {
// signed int
inDesc.mFormatFlags = kAudioFormatFlagIsSignedInteger|kAudioFormatFlagIsPacked;
}
else {
// unsigned int
inDesc.mFormatFlags = kAudioFormatFlagIsPacked;
}
if ((format & AF_FORMAT_END_MASK) == AF_FORMAT_BE)
inDesc.mFormatFlags |= kAudioFormatFlagIsBigEndian;
inDesc.mFramesPerPacket = 1;
ao->packetSize = inDesc.mBytesPerPacket = inDesc.mBytesPerFrame = inDesc.mFramesPerPacket*channels*(inDesc.mBitsPerChannel/8);
print_format(MSGL_V, "source:",&inDesc);
if (ao->b_supports_digital)
{
b_alive = 1;
err = GetAudioProperty(ao->i_selected_dev,
kAudioDevicePropertyDeviceIsAlive,
sizeof(UInt32), &b_alive);
if (err != noErr)
ao_msg(MSGT_AO, MSGL_WARN, "could not check whether device is alive: [%4.4s]\n", (char *)&err);
if (!b_alive)
ao_msg(MSGT_AO, MSGL_WARN, "device is not alive\n" );
/* S/PDIF output need device in HogMode. */
err = GetAudioProperty(ao->i_selected_dev,
kAudioDevicePropertyHogMode,
sizeof(pid_t), &ao->i_hog_pid);
if (err != noErr)
{
/* This is not a fatal error. Some drivers simply don't support this property. */
ao_msg(MSGT_AO, MSGL_WARN, "could not check whether device is hogged: [%4.4s]\n",
(char *)&err);
ao->i_hog_pid = -1;
}
if (ao->i_hog_pid != -1 && ao->i_hog_pid != getpid())
{
ao_msg(MSGT_AO, MSGL_WARN, "Selected audio device is exclusively in use by another program.\n" );
goto err_out;
}
ao->stream_format = inDesc;
return OpenSPDIF();
}
/* original analog output code */
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = (device_id == 0) ? kAudioUnitSubType_DefaultOutput : kAudioUnitSubType_HALOutput;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
comp = FindNextComponent(NULL, &desc); //Finds an component that meets the desc spec's
if (comp == NULL) {
ao_msg(MSGT_AO, MSGL_WARN, "Unable to find Output Unit component\n");
goto err_out;
}
err = OpenAComponent(comp, &(ao->theOutputUnit)); //gains access to the services provided by the component
if (err) {
ao_msg(MSGT_AO, MSGL_WARN, "Unable to open Output Unit component: [%4.4s]\n", (char *)&err);
goto err_out;
}
// Initialize AudioUnit
err = AudioUnitInitialize(ao->theOutputUnit);
if (err) {
ao_msg(MSGT_AO, MSGL_WARN, "Unable to initialize Output Unit component: [%4.4s]\n", (char *)&err);
goto err_out1;
}
size = sizeof(AudioStreamBasicDescription);
err = AudioUnitSetProperty(ao->theOutputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &inDesc, size);
if (err) {
ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the input format: [%4.4s]\n", (char *)&err);
goto err_out2;
}
size = sizeof(UInt32);
err = AudioUnitGetProperty(ao->theOutputUnit, kAudioDevicePropertyBufferSize, kAudioUnitScope_Input, 0, &maxFrames, &size);
if (err)
{
ao_msg(MSGT_AO,MSGL_WARN, "AudioUnitGetProperty returned [%4.4s] when getting kAudioDevicePropertyBufferSize\n", (char *)&err);
goto err_out2;
}
//Set the Current Device to the Default Output Unit.
err = AudioUnitSetProperty(ao->theOutputUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &ao->i_selected_dev, sizeof(ao->i_selected_dev));
ao->chunk_size = maxFrames;//*inDesc.mBytesPerFrame;
ao_data.samplerate = inDesc.mSampleRate;
ao_data.channels = inDesc.mChannelsPerFrame;
ao_data.bps = ao_data.samplerate * inDesc.mBytesPerFrame;
ao_data.outburst = ao->chunk_size;
ao_data.buffersize = ao_data.bps;
ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size;
ao->buffer_len = ao->num_chunks * ao->chunk_size;
ao->buffer = av_fifo_alloc(ao->buffer_len);
ao_msg(MSGT_AO,MSGL_V, "using %5d chunks of %d bytes (buffer len %d bytes)\n", (int)ao->num_chunks, (int)ao->chunk_size, (int)ao->buffer_len);
renderCallback.inputProc = theRenderProc;
renderCallback.inputProcRefCon = 0;
err = AudioUnitSetProperty(ao->theOutputUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &renderCallback, sizeof(AURenderCallbackStruct));
if (err) {
ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the render callback: [%4.4s]\n", (char *)&err);
goto err_out2;
}
reset();
return CONTROL_OK;
err_out2:
AudioUnitUninitialize(ao->theOutputUnit);
err_out1:
CloseComponent(ao->theOutputUnit);
err_out:
av_fifo_free(ao->buffer);
free(ao);
ao = NULL;
return CONTROL_FALSE;
}
// Initialization and runtime control
static int control(struct af_instance *af, int cmd, void *arg)
{
af_ac3enc_t *s = af->setup;
static const int default_bit_rate[AC3_MAX_CHANNELS+1] = \
{0, 96000, 192000, 256000, 384000, 448000, 448000};
switch (cmd){
case AF_CONTROL_REINIT: {
struct mp_audio *in = arg;
struct mp_audio orig_in = *in;
if (AF_FORMAT_IS_AC3(in->format) || in->nch < s->cfg_min_channel_num)
return AF_DETACH;
mp_audio_set_format(in, s->in_sampleformat);
if (in->rate != 48000 && in->rate != 44100 && in->rate != 32000)
in->rate = 48000;
af->data->rate = in->rate;
mp_chmap_reorder_to_lavc(&in->channels);
if (in->nch > AC3_MAX_CHANNELS)
mp_audio_set_num_channels(in, AC3_MAX_CHANNELS);
mp_audio_set_format(af->data, AF_FORMAT_AC3_BE);
mp_audio_set_num_channels(af->data, 2);
if (!mp_audio_config_equals(in, &orig_in))
return AF_FALSE;
s->in_samples = AC3_FRAME_SIZE;
if (s->cfg_add_iec61937_header) {
s->out_samples = AC3_FRAME_SIZE;
} else {
s->out_samples = AC3_MAX_CODED_FRAME_SIZE / af->data->sstride;
}
af->mul = s->out_samples / (double)s->in_samples;
mp_audio_buffer_reinit(s->pending, in);
mp_msg(MSGT_AFILTER, MSGL_DBG2, "af_lavcac3enc reinit: %d, %d, %f, %d.\n",
in->nch, in->rate, af->mul, s->in_samples);
int bit_rate = s->bit_rate ? s->bit_rate : default_bit_rate[in->nch];
if (s->lavc_actx->channels != in->nch ||
s->lavc_actx->sample_rate != in->rate ||
s->lavc_actx->bit_rate != bit_rate)
{
avcodec_close(s->lavc_actx);
// Put sample parameters
s->lavc_actx->channels = in->nch;
s->lavc_actx->channel_layout = mp_chmap_to_lavc(&in->channels);
s->lavc_actx->sample_rate = in->rate;
s->lavc_actx->bit_rate = bit_rate;
if (avcodec_open2(s->lavc_actx, s->lavc_acodec, NULL) < 0) {
mp_tmsg(MSGT_AFILTER, MSGL_ERR, "Couldn't open codec %s, br=%d.\n", "ac3", bit_rate);
return AF_ERROR;
}
}
if (s->lavc_actx->frame_size != AC3_FRAME_SIZE) {
mp_msg(MSGT_AFILTER, MSGL_ERR, "lavcac3enc: unexpected ac3 "
"encoder frame size %d\n", s->lavc_actx->frame_size);
return AF_ERROR;
}
return AF_OK;
}
}
return AF_UNKNOWN;
}
