static int setup_format(sh_audio_t *sh_audio, const AVCodecContext *lavc_context)
{
int broken_srate = 0;
int samplerate = lavc_context->sample_rate;
int sample_format = samplefmt2affmt(lavc_context->sample_fmt);
if (!sample_format)
sample_format = sh_audio->sample_format;
if(sh_audio->wf){
// If the decoder uses the wrong number of channels all is lost anyway.
// sh_audio->channels=sh_audio->wf->nChannels;
if (lavc_context->codec_id == CODEC_ID_AAC &&
samplerate == 2*sh_audio->wf->nSamplesPerSec) {
broken_srate = 1;
} else if (sh_audio->wf->nSamplesPerSec)
samplerate=sh_audio->wf->nSamplesPerSec;
}
if (lavc_context->channels != sh_audio->channels ||
samplerate != sh_audio->samplerate ||
sample_format != sh_audio->sample_format) {
sh_audio->channels=lavc_context->channels;
sh_audio->samplerate=samplerate;
sh_audio->sample_format = sample_format;
sh_audio->samplesize=af_fmt2bits(sh_audio->sample_format)/ 8;
if (broken_srate)
mp_msg(MSGT_DECAUDIO, MSGL_WARN,
"Ignoring broken container sample rate for AAC with SBR\n");
return 1;
}
return 0;
}
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);
}
static demuxer_t* demux_rawaudio_open(demuxer_t* demuxer) {
sh_audio_t* sh_audio;
WAVEFORMATEX* w;
if ((format & AF_FORMAT_SPECIAL_MASK) != 0)
return NULL;
sh_audio = new_sh_audio(demuxer,0);
sh_audio->gsh->codec = "mp-pcm";
sh_audio->format = format;
sh_audio->wf = w = malloc(sizeof(*w));
w->wFormatTag = 0;
w->nChannels = sh_audio->channels = channels;
w->nSamplesPerSec = sh_audio->samplerate = samplerate;
int samplesize = (af_fmt2bits(format) + 7) / 8;
w->nAvgBytesPerSec = samplerate * samplesize * channels;
w->nBlockAlign = channels * samplesize;
w->wBitsPerSample = 8 * samplesize;
w->cbSize = 0;
demuxer->movi_start = demuxer->stream->start_pos;
demuxer->movi_end = demuxer->stream->end_pos;
demuxer->audio->id = 0;
demuxer->audio->sh = sh_audio;
sh_audio->ds = demuxer->audio;
sh_audio->needs_parsing = 1;
return demuxer;
}
int af_fmt_seconds_to_bytes(int format, float seconds, int channels, int samplerate)
{
assert(!af_fmt_is_planar(format));
int bps = (af_fmt2bits(format) / 8);
int framelen = channels * bps;
int bytes = seconds * bps * samplerate;
if (bytes % framelen)
bytes += framelen - (bytes % framelen);
return bytes;
}
static int init_vqf_audio_codec(sh_audio_t *sh_audio){
WAVEFORMATEX *in_fmt=sh_audio->wf;
vqf_priv_t*priv=sh_audio->context;
int ver;
mp_msg(MSGT_DECAUDIO, MSGL_INFO, "======= Win32 (TWinVQ) AUDIO Codec init =======\n");
sh_audio->channels=in_fmt->nChannels;
sh_audio->samplerate=in_fmt->nSamplesPerSec;
sh_audio->sample_format=AF_FORMAT_S16_NE;
// sh_audio->sample_format=AF_FORMAT_FLOAT_NE;
sh_audio->samplesize=af_fmt2bits(sh_audio->sample_format)/8;
priv->o_wf.nChannels=in_fmt->nChannels;
priv->o_wf.nSamplesPerSec=in_fmt->nSamplesPerSec;
priv->o_wf.nBlockAlign=sh_audio->samplesize*in_fmt->nChannels;
priv->o_wf.nAvgBytesPerSec=in_fmt->nBlockAlign*in_fmt->nChannels;
priv->o_wf.wFormatTag=0x01;
priv->o_wf.wBitsPerSample=in_fmt->wBitsPerSample;
priv->o_wf.cbSize=0;
if( mp_msg_test(MSGT_DECAUDIO,MSGL_V) )
{
mp_msg(MSGT_DECAUDIO, MSGL_V, "Input format:\n");
print_wave_header(in_fmt, MSGL_V);
mp_msg(MSGT_DECAUDIO, MSGL_V, "Output fmt:\n");
print_wave_header(&priv->o_wf, MSGL_V);
}
memcpy(&priv->hi,&in_fmt[1],sizeof(headerInfo));
if((ver=TvqInitialize(&priv->hi,&priv->index,0))){
const char *tvqe[]={
"No errors",
"General error",
"Wrong version",
"Channel setting error",
"Wrong coding mode",
"Inner parameter setting error",
"Wrong number of VQ pre-selection candidates, used only in encoder" };
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "Tvq initialization error: %s\n",ver>=0&&ver<7?tvqe[ver]:"Unknown");
return 0;
}
ver=TvqCheckVersion(priv->hi.ID);
if(ver==TVQ_UNKNOWN_VERSION){
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "Tvq unknown version of stream\n" );
return 0;
}
TvqGetConfInfo(&priv->cf);
TvqGetVectorInfo(priv->bits_0,priv->bits_1);
priv->framesize=TvqGetFrameSize();
sh_audio->audio_in_minsize=priv->framesize*in_fmt->nChannels;
sh_audio->a_in_buffer_size=4*sh_audio->audio_in_minsize;
sh_audio->a_in_buffer=av_malloc(sh_audio->a_in_buffer_size);
sh_audio->a_in_buffer_len=0;
return 1;
}
/* Convert format to str input str is a buffer for the
converted string, size is the size of the buffer */
char* af_fmt2str(int format, char* str, int size) {
int i=0;
if (size < 1)
return NULL;
size--; // reserve one for terminating 0
// Endianness
if(AF_FORMAT_LE == (format & AF_FORMAT_END_MASK))
i+=snprintf(str,size-i,"little-endian ");
else
i+=snprintf(str,size-i,"big-endian ");
if(format & AF_FORMAT_SPECIAL_MASK) {
switch(format & AF_FORMAT_SPECIAL_MASK) {
case(AF_FORMAT_MU_LAW):
i+=snprintf(&str[i],size-i,"mu-law ");
break;
case(AF_FORMAT_A_LAW):
i+=snprintf(&str[i],size-i,"A-law ");
break;
case(AF_FORMAT_MPEG2):
i+=snprintf(&str[i],size-i,"MPEG-2 ");
break;
case(AF_FORMAT_AC3):
i+=snprintf(&str[i],size-i,"AC3 ");
break;
case(AF_FORMAT_IMA_ADPCM):
i+=snprintf(&str[i],size-i,"IMA-ADPCM ");
break;
default:
i+=snprintf(&str[i],size-i,"unknown format ");
}
} else {
// Bits
i+=snprintf(&str[i],size-i,"%d-bit ", af_fmt2bits(format));
// Point
if(AF_FORMAT_F == (format & AF_FORMAT_POINT_MASK)) {
i+=snprintf(&str[i],size-i,"float ");
} else {
// Sign
if(AF_FORMAT_US == (format & AF_FORMAT_SIGN_MASK))
i+=snprintf(&str[i],size-i,"unsigned ");
else
i+=snprintf(&str[i],size-i,"signed ");
i+=snprintf(&str[i],size-i,"int ");
}
}
// remove trailing space
if (i > 0 && str[i - 1] == ' ')
i--;
str[i] = 0; // make sure it is 0 terminated.
return str;
}
static int af_open(af_instance_t* af){
af_ac3enc_t *s = calloc(1,sizeof(af_ac3enc_t));
af->control=control;
af->uninit=uninit;
af->play=play;
af->mul=1;
af->data=calloc(1,sizeof(af_data_t));
af->setup=s;
s->lavc_acodec = avcodec_find_encoder_by_name("ac3");
if (!s->lavc_acodec) {
mp_tmsg(MSGT_AFILTER, MSGL_ERR, "Audio LAVC, couldn't find encoder for codec %s.\n", "ac3");
return AF_ERROR;
}
s->lavc_actx = avcodec_alloc_context3(s->lavc_acodec);
if (!s->lavc_actx) {
mp_tmsg(MSGT_AFILTER, MSGL_ERR, "Audio LAVC, couldn't allocate context!\n");
return AF_ERROR;
}
const enum AVSampleFormat *fmts = s->lavc_acodec->sample_fmts;
for (int i = 0; ; i++) {
if (fmts[i] == AV_SAMPLE_FMT_NONE) {
mp_msg(MSGT_AFILTER, MSGL_ERR, "Audio LAVC, encoder doesn't "
"support expected sample formats!\n");
return AF_ERROR;
} else if (fmts[i] == AV_SAMPLE_FMT_S16) {
s->in_sampleformat = AF_FORMAT_S16_NE;
s->lavc_actx->sample_fmt = fmts[i];
break;
} else if (fmts[i] == AV_SAMPLE_FMT_FLT) {
s->in_sampleformat = AF_FORMAT_FLOAT_NE;
s->lavc_actx->sample_fmt = fmts[i];
break;
}
}
char buf[100];
mp_msg(MSGT_AFILTER, MSGL_V, "[af_lavcac3enc]: in sample format: %s\n",
af_fmt2str(s->in_sampleformat, buf, 100));
s->pending_data_size = AF_NCH * AC3_FRAME_SIZE *
af_fmt2bits(s->in_sampleformat) / 8;
s->pending_data = malloc(s->pending_data_size);
return AF_OK;
}
/* Prefer playing audio with the samplerate given in container data
* if available, but take number the number of channels and sample format
* from the codec, since if the codec isn't using the correct values for
* those everything breaks anyway.
*/
static int setup_format(sh_audio_t *sh_audio,
const AVCodecContext *lavc_context)
{
int sample_format = sh_audio->sample_format;
switch (lavc_context->sample_fmt) {
case AV_SAMPLE_FMT_U8: sample_format = AF_FORMAT_U8; break;
case AV_SAMPLE_FMT_S16: sample_format = AF_FORMAT_S16_NE; break;
case AV_SAMPLE_FMT_S32: sample_format = AF_FORMAT_S32_NE; break;
case AV_SAMPLE_FMT_FLT: sample_format = AF_FORMAT_FLOAT_NE; break;
default:
mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Unsupported sample format\n");
sample_format = AF_FORMAT_UNKNOWN;
}
bool broken_srate = false;
int samplerate = lavc_context->sample_rate;
int container_samplerate = sh_audio->container_out_samplerate;
if (!container_samplerate && sh_audio->wf)
container_samplerate = sh_audio->wf->nSamplesPerSec;
if (lavc_context->codec_id == CODEC_ID_AAC
&& samplerate == 2 * container_samplerate)
broken_srate = true;
else if (container_samplerate)
samplerate = container_samplerate;
if (lavc_context->channels != sh_audio->channels ||
samplerate != sh_audio->samplerate ||
sample_format != sh_audio->sample_format) {
sh_audio->channels = lavc_context->channels;
sh_audio->samplerate = samplerate;
sh_audio->sample_format = sample_format;
sh_audio->samplesize = af_fmt2bits(sh_audio->sample_format) / 8;
if (broken_srate)
mp_msg(MSGT_DECAUDIO, MSGL_WARN,
"Ignoring broken container sample rate for AAC with SBR\n");
return 1;
}
return 0;
}
static int write_to_ao(struct MPContext *mpctx, void *data, int len, int flags,
double pts)
{
if (mpctx->paused)
return 0;
struct ao *ao = mpctx->ao;
double bps = ao->bps / mpctx->opts->playback_speed;
int unitsize = ao->channels.num * af_fmt2bits(ao->format) / 8;
ao->pts = pts;
int played = ao_play(mpctx->ao, data, len, flags);
assert(played <= len);
assert(played % unitsize == 0);
if (played > 0) {
mpctx->shown_aframes += played / unitsize;
mpctx->delay += played / bps;
// Keep correct pts for remaining data - could be used to flush
// remaining buffer when closing ao.
ao->pts += played / bps;
return played;
}
return 0;
}
// Return pts value corresponding to the end point of audio written to the
// ao so far.
double written_audio_pts(struct MPContext *mpctx)
{
sh_audio_t *sh_audio = mpctx->sh_audio;
if (!sh_audio)
return MP_NOPTS_VALUE;
double bps = sh_audio->channels.num * sh_audio->samplerate *
(af_fmt2bits(sh_audio->sample_format) / 8);
// first calculate the end pts of audio that has been output by decoder
double a_pts = sh_audio->pts;
if (a_pts == MP_NOPTS_VALUE)
return MP_NOPTS_VALUE;
// sh_audio->pts is the timestamp of the latest input packet with
// known pts that the decoder has decoded. sh_audio->pts_bytes is
// the amount of bytes the decoder has written after that timestamp.
a_pts += sh_audio->pts_bytes / bps;
// Now a_pts hopefully holds the pts for end of audio from decoder.
// Subtract data in buffers between decoder and audio out.
// Decoded but not filtered
a_pts -= sh_audio->a_buffer_len / bps;
// Data buffered in audio filters, measured in bytes of "missing" output
double buffered_output = af_calc_delay(sh_audio->afilter);
// Data that was ready for ao but was buffered because ao didn't fully
// accept everything to internal buffers yet
buffered_output += mpctx->ao->buffer.len;
// Filters divide audio length by playback_speed, so multiply by it
// to get the length in original units without speedup or slowdown
a_pts -= buffered_output * mpctx->opts->playback_speed / mpctx->ao->bps;
return a_pts + mpctx->video_offset;
}
/* libmpg123 has a new format ready; query and store, return return value
of mpg123_getformat() */
static int set_format(struct dec_audio *da)
{
struct ad_mpg123_context *con = da->priv;
int ret;
long rate;
int channels;
int encoding;
ret = mpg123_getformat(con->handle, &rate, &channels, &encoding);
if (ret == MPG123_OK) {
mp_audio_set_num_channels(&da->decoded, channels);
da->decoded.rate = rate;
int af = mpg123_format_to_af(encoding);
if (!af) {
/* This means we got a funny custom build of libmpg123 that only supports an unknown format. */
MP_ERR(da, "Bad encoding from mpg123: %i.\n", encoding);
return MPG123_ERR;
}
mp_audio_set_format(&da->decoded, af);
con->sample_size = channels * (af_fmt2bits(af) / 8);
con->new_format = 0;
}
return ret;
}
// open & setup audio device
static int init(struct ao *ao)
{
struct priv *ac = talloc_zero(ao, struct priv);
AVCodec *codec;
ao->priv = ac;
if (!encode_lavc_available(ao->encode_lavc_ctx)) {
MP_ERR(ao, "the option --o (output file) must be specified\n");
return -1;
}
pthread_mutex_lock(&ao->encode_lavc_ctx->lock);
ac->stream = encode_lavc_alloc_stream(ao->encode_lavc_ctx,
AVMEDIA_TYPE_AUDIO);
if (!ac->stream) {
MP_ERR(ao, "could not get a new audio stream\n");
goto fail;
}
codec = encode_lavc_get_codec(ao->encode_lavc_ctx, ac->stream);
// ac->stream->time_base.num = 1;
// ac->stream->time_base.den = ao->samplerate;
// doing this breaks mpeg2ts in ffmpeg
// which doesn't properly force the time base to be 90000
// furthermore, ffmpeg.c doesn't do this either and works
ac->stream->codec->time_base.num = 1;
ac->stream->codec->time_base.den = ao->samplerate;
ac->stream->codec->sample_rate = ao->samplerate;
struct mp_chmap_sel sel = {0};
mp_chmap_sel_add_any(&sel);
if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
goto fail;
mp_chmap_reorder_to_lavc(&ao->channels);
ac->stream->codec->channels = ao->channels.num;
ac->stream->codec->channel_layout = mp_chmap_to_lavc(&ao->channels);
ac->stream->codec->sample_fmt = AV_SAMPLE_FMT_NONE;
select_format(ao, codec);
ac->sample_size = af_fmt2bits(ao->format) / 8;
ac->stream->codec->sample_fmt = af_to_avformat(ao->format);
ac->stream->codec->bits_per_raw_sample = ac->sample_size * 8;
if (encode_lavc_open_codec(ao->encode_lavc_ctx, ac->stream) < 0)
goto fail;
ac->pcmhack = 0;
if (ac->stream->codec->frame_size <= 1)
ac->pcmhack = av_get_bits_per_sample(ac->stream->codec->codec_id) / 8;
if (ac->pcmhack) {
ac->aframesize = 16384; // "enough"
ac->buffer_size =
ac->aframesize * ac->pcmhack * ao->channels.num * 2 + 200;
} else {
ac->aframesize = ac->stream->codec->frame_size;
ac->buffer_size =
ac->aframesize * ac->sample_size * ao->channels.num * 2 + 200;
}
if (ac->buffer_size < FF_MIN_BUFFER_SIZE)
ac->buffer_size = FF_MIN_BUFFER_SIZE;
ac->buffer = talloc_size(ac, ac->buffer_size);
// enough frames for at least 0.25 seconds
ac->framecount = ceil(ao->samplerate * 0.25 / ac->aframesize);
// but at least one!
ac->framecount = FFMAX(ac->framecount, 1);
ac->savepts = MP_NOPTS_VALUE;
ac->lastpts = MP_NOPTS_VALUE;
ao->untimed = true;
pthread_mutex_unlock(&ao->encode_lavc_ctx->lock);
return 0;
fail:
pthread_mutex_unlock(&ao->encode_lavc_ctx->lock);
return -1;
}
/*
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
static int init(sh_audio_t *sh_audio)
{
int tries = 0;
int x;
AVCodecContext *lavc_context;
AVCodec *lavc_codec;
mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n");
if(!avcodec_initialized){
avcodec_init();
avcodec_register_all();
avcodec_initialized=1;
}
lavc_codec = (AVCodec *)avcodec_find_decoder_by_name(sh_audio->codec->dll);
if(!lavc_codec){
mp_tmsg(MSGT_DECAUDIO,MSGL_ERR,"Cannot find codec '%s' in libavcodec...\n",sh_audio->codec->dll);
return 0;
}
lavc_context = avcodec_alloc_context();
sh_audio->context=lavc_context;
lavc_context->sample_rate = sh_audio->samplerate;
lavc_context->bit_rate = sh_audio->i_bps * 8;
if(sh_audio->wf){
lavc_context->channels = sh_audio->wf->nChannels;
lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec;
lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8;
lavc_context->block_align = sh_audio->wf->nBlockAlign;
lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample;
}
lavc_context->request_channels = audio_output_channels;
lavc_context->codec_tag = sh_audio->format; //FOURCC
lavc_context->codec_type = CODEC_TYPE_AUDIO;
lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi
/* alloc extra data */
if (sh_audio->wf && sh_audio->wf->cbSize > 0) {
lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE);
lavc_context->extradata_size = sh_audio->wf->cbSize;
memcpy(lavc_context->extradata, (char *)sh_audio->wf + sizeof(WAVEFORMATEX),
lavc_context->extradata_size);
}
// for QDM2
if (sh_audio->codecdata_len && sh_audio->codecdata && !lavc_context->extradata)
{
lavc_context->extradata = av_malloc(sh_audio->codecdata_len);
lavc_context->extradata_size = sh_audio->codecdata_len;
memcpy(lavc_context->extradata, (char *)sh_audio->codecdata,
lavc_context->extradata_size);
}
/* open it */
if (avcodec_open(lavc_context, lavc_codec) < 0) {
mp_tmsg(MSGT_DECAUDIO,MSGL_ERR, "Could not open codec.\n");
return 0;
}
mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec \"%s\" init OK!\n", lavc_codec->name);
// printf("\nFOURCC: 0x%X\n",sh_audio->format);
if(sh_audio->format==0x3343414D){
// MACE 3:1
sh_audio->ds->ss_div = 2*3; // 1 samples/packet
sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
} else
if(sh_audio->format==0x3643414D){
// MACE 6:1
sh_audio->ds->ss_div = 2*6; // 1 samples/packet
sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
}
// Decode at least 1 byte: (to get header filled)
do {
x=decode_audio(sh_audio,sh_audio->a_buffer,1,sh_audio->a_buffer_size);
} while (x <= 0 && tries++ < 5);
if(x>0) sh_audio->a_buffer_len=x;
sh_audio->channels=lavc_context->channels;
sh_audio->samplerate=lavc_context->sample_rate;
sh_audio->i_bps=lavc_context->bit_rate/8;
switch (lavc_context->sample_fmt) {
case SAMPLE_FMT_U8: sh_audio->sample_format = AF_FORMAT_U8; break;
case SAMPLE_FMT_S16: sh_audio->sample_format = AF_FORMAT_S16_NE; break;
case SAMPLE_FMT_S32: sh_audio->sample_format = AF_FORMAT_S32_NE; break;
case SAMPLE_FMT_FLT: sh_audio->sample_format = AF_FORMAT_FLOAT_NE; break;
default:
mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Unsupported sample format\n");
return 0;
}
/* If the audio is AAC the container level data may be unreliable
* because of SBR handling problems (possibly half real sample rate at
* container level). Default AAC decoding with ad_faad has used codec-level
* values for a long time without generating complaints so it should be OK.
*/
if (sh_audio->wf && lavc_context->codec_id != CODEC_ID_AAC) {
// If the decoder uses the wrong number of channels all is lost anyway.
// sh_audio->channels=sh_audio->wf->nChannels;
if (sh_audio->wf->nSamplesPerSec)
sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
if (sh_audio->wf->nAvgBytesPerSec)
sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
}
sh_audio->samplesize=af_fmt2bits(sh_audio->sample_format)/ 8;
return 1;
}
void mp_audio_set_format(struct mp_audio *mpa, int format)
{
mpa->format = format;
mpa->bps = af_fmt2bits(format) / 8;
}
int format_2_bps(int format) {
return af_fmt2bits(format)/8;
}
static int init(struct ao *ao)
{
struct priv *priv = ao->priv;
if (!check_pa_ret(Pa_Initialize()))
return -1;
pthread_mutex_init(&priv->ring_mutex, NULL);
int pa_device = Pa_GetDefaultOutputDevice();
if (priv->cfg_device && priv->cfg_device[0])
pa_device = find_device(priv->cfg_device);
if (pa_device == paNoDevice)
goto error_exit;
// The actual channel order probably depends on the platform.
struct mp_chmap_sel sel = {0};
mp_chmap_sel_add_waveext_def(&sel);
if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
goto error_exit;
PaStreamParameters sp = {
.device = pa_device,
.channelCount = ao->channels.num,
.suggestedLatency
= Pa_GetDeviceInfo(pa_device)->defaultHighOutputLatency,
};
ao->format = af_fmt_from_planar(ao->format);
const struct format_map *fmt = format_maps;
while (fmt->pa_format) {
if (fmt->mp_format == ao->format) {
PaStreamParameters test = sp;
test.sampleFormat = fmt->pa_format;
if (Pa_IsFormatSupported(NULL, &test, ao->samplerate) == paNoError)
break;
}
fmt++;
}
if (!fmt->pa_format) {
MP_VERBOSE(ao, "Unsupported format, using default.\n");
fmt = format_maps;
}
ao->format = fmt->mp_format;
sp.sampleFormat = fmt->pa_format;
priv->framelen = ao->channels.num * (af_fmt2bits(ao->format) / 8);
ao->bps = ao->samplerate * priv->framelen;
if (!check_pa_ret(Pa_IsFormatSupported(NULL, &sp, ao->samplerate)))
goto error_exit;
if (!check_pa_ret(Pa_OpenStream(&priv->stream, NULL, &sp, ao->samplerate,
paFramesPerBufferUnspecified, paNoFlag,
stream_callback, ao)))
goto error_exit;
priv->ring = mp_ring_new(priv, seconds_to_bytes(ao, 0.5));
return 0;
error_exit:
uninit(ao, true);
return -1;
}
static int play(struct ao *ao, void **data, int samples, int flags)
{
struct priv *priv = ao->priv;
pthread_mutex_lock(&priv->ring_mutex);
int write_len = mp_ring_write(priv->ring, data[0], samples * ao->sstride);
if (flags & AOPLAY_FINAL_CHUNK)
priv->play_remaining = true;
pthread_mutex_unlock(&priv->ring_mutex);
if (Pa_IsStreamStopped(priv->stream) == 1)
check_pa_ret(Pa_StartStream(priv->stream));
return write_len / ao->sstride;
}
static int get_space(struct ao *ao)
{
struct priv *priv = ao->priv;
pthread_mutex_lock(&priv->ring_mutex);
int free = mp_ring_available(priv->ring);
pthread_mutex_unlock(&priv->ring_mutex);
return free / ao->sstride;
}
static float get_delay(struct ao *ao)
{
struct priv *priv = ao->priv;
double stream_time = Pa_GetStreamTime(priv->stream);
pthread_mutex_lock(&priv->ring_mutex);
float frame_time = priv->play_time ? priv->play_time - stream_time : 0;
float buffer_latency = (mp_ring_buffered(priv->ring) + priv->play_silence)
/ (float)ao->bps;
pthread_mutex_unlock(&priv->ring_mutex);
return buffer_latency + frame_time;
}
static void reset(struct ao *ao)
{
struct priv *priv = ao->priv;
if (Pa_IsStreamStopped(priv->stream) != 1)
check_pa_ret(Pa_AbortStream(priv->stream));
pthread_mutex_lock(&priv->ring_mutex);
mp_ring_reset(priv->ring);
priv->play_remaining = false;
priv->play_time = 0;
priv->play_silence = 0;
pthread_mutex_unlock(&priv->ring_mutex);
}
static void pause(struct ao *ao)
{
struct priv *priv = ao->priv;
check_pa_ret(Pa_AbortStream(priv->stream));
double stream_time = Pa_GetStreamTime(priv->stream);
pthread_mutex_lock(&priv->ring_mutex);
// When playback resumes, replace the lost audio (due to dropping the
// portaudio/driver/hardware internal buffers) with silence.
float frame_time = priv->play_time ? priv->play_time - stream_time : 0;
priv->play_silence += seconds_to_bytes(ao, FFMAX(frame_time, 0));
priv->play_time = 0;
pthread_mutex_unlock(&priv->ring_mutex);
}
static void resume(struct ao *ao)
{
struct priv *priv = ao->priv;
check_pa_ret(Pa_StartStream(priv->stream));
}
#define OPT_BASE_STRUCT struct priv
const struct ao_driver audio_out_portaudio = {
.description = "PortAudio",
.name = "portaudio",
.init = init,
.uninit = uninit,
.reset = reset,
.get_space = get_space,
.play = play,
.get_delay = get_delay,
.pause = pause,
.resume = resume,
.priv_size = sizeof(struct priv),
.options = (const struct m_option[]) {
OPT_STRING_VALIDATE("device", cfg_device, 0, validate_device_opt),
{0}
},
};
static int demux_open_tv(demuxer_t *demuxer, enum demux_check check)
{
tvi_handle_t *tvh;
sh_video_t *sh_video;
sh_audio_t *sh_audio = NULL;
const tvi_functions_t *funcs;
if (check > DEMUX_CHECK_REQUEST || demuxer->stream->type != STREAMTYPE_TV)
return -1;
demuxer->priv=NULL;
if(!(tvh=tv_begin(demuxer->stream->priv, demuxer->log))) return -1;
if (!tvh->functions->init(tvh->priv)) return -1;
tvh->demuxer = demuxer;
if (!open_tv(tvh)){
tv_uninit(tvh);
return -1;
}
funcs = tvh->functions;
demuxer->priv=tvh;
struct sh_stream *sh_v = new_sh_stream(demuxer, STREAM_VIDEO);
sh_video = sh_v->video;
/* get IMAGE FORMAT */
int fourcc;
funcs->control(tvh->priv, TVI_CONTROL_VID_GET_FORMAT, &fourcc);
if (fourcc == MP_FOURCC_MJPEG) {
sh_v->codec = "mjpeg";
} else {
sh_v->codec = "rawvideo";
sh_v->format = fourcc;
}
/* set FPS and FRAMETIME */
if(!sh_video->fps)
{
float tmp;
if (funcs->control(tvh->priv, TVI_CONTROL_VID_GET_FPS, &tmp) != TVI_CONTROL_TRUE)
sh_video->fps = 25.0f; /* on PAL */
else sh_video->fps = tmp;
}
if (tvh->tv_param->fps != -1.0f)
sh_video->fps = tvh->tv_param->fps;
/* If playback only mode, go to immediate mode, fail silently */
if(tvh->tv_param->immediate == 1)
{
funcs->control(tvh->priv, TVI_CONTROL_IMMEDIATE, 0);
tvh->tv_param->noaudio = 1;
}
/* set width */
funcs->control(tvh->priv, TVI_CONTROL_VID_GET_WIDTH, &sh_video->disp_w);
/* set height */
funcs->control(tvh->priv, TVI_CONTROL_VID_GET_HEIGHT, &sh_video->disp_h);
demuxer->seekable = 0;
/* here comes audio init */
if (tvh->tv_param->noaudio == 0 && funcs->control(tvh->priv, TVI_CONTROL_IS_AUDIO, 0) == TVI_CONTROL_TRUE)
{
int audio_format;
/* yeah, audio is present */
funcs->control(tvh->priv, TVI_CONTROL_AUD_SET_SAMPLERATE,
&tvh->tv_param->audiorate);
if (funcs->control(tvh->priv, TVI_CONTROL_AUD_GET_FORMAT, &audio_format) != TVI_CONTROL_TRUE)
goto no_audio;
switch(audio_format)
{
case AF_FORMAT_U8:
case AF_FORMAT_S8:
case AF_FORMAT_U16_LE:
case AF_FORMAT_U16_BE:
case AF_FORMAT_S16_LE:
case AF_FORMAT_S16_BE:
case AF_FORMAT_S32_LE:
case AF_FORMAT_S32_BE:
break;
case AF_FORMAT_MPEG2:
default:
MP_ERR(tvh, "Audio type '%s' unsupported!\n",
af_fmt_to_str(audio_format));
goto no_audio;
}
struct sh_stream *sh_a = new_sh_stream(demuxer, STREAM_AUDIO);
sh_audio = sh_a->audio;
funcs->control(tvh->priv, TVI_CONTROL_AUD_GET_SAMPLERATE,
&sh_audio->samplerate);
int nchannels = sh_audio->channels.num;
funcs->control(tvh->priv, TVI_CONTROL_AUD_GET_CHANNELS,
&nchannels);
mp_chmap_from_channels(&sh_audio->channels, nchannels);
sh_a->codec = "mp-pcm";
sh_a->format = audio_format;
int samplesize = af_fmt2bits(audio_format) / 8;
sh_audio->i_bps =
sh_audio->samplerate * samplesize * sh_audio->channels.num;
// emulate WF for win32 codecs:
sh_audio->wf = talloc_zero(sh_audio, MP_WAVEFORMATEX);
sh_audio->wf->wFormatTag = sh_a->format;
sh_audio->wf->nChannels = sh_audio->channels.num;
sh_audio->wf->wBitsPerSample = samplesize * 8;
sh_audio->wf->nSamplesPerSec = sh_audio->samplerate;
sh_audio->wf->nBlockAlign = samplesize * sh_audio->channels.num;
sh_audio->wf->nAvgBytesPerSec = sh_audio->i_bps;
MP_VERBOSE(tvh, " TV audio: %d channels, %d bits, %d Hz\n",
sh_audio->wf->nChannels, sh_audio->wf->wBitsPerSample,
sh_audio->wf->nSamplesPerSec);
}
no_audio:
if(!(funcs->start(tvh->priv))){
// start failed :(
tv_uninit(tvh);
return -1;
}
/* set color eq */
tv_set_color_options(tvh, TV_COLOR_BRIGHTNESS, tvh->tv_param->brightness);
tv_set_color_options(tvh, TV_COLOR_HUE, tvh->tv_param->hue);
tv_set_color_options(tvh, TV_COLOR_SATURATION, tvh->tv_param->saturation);
tv_set_color_options(tvh, TV_COLOR_CONTRAST, tvh->tv_param->contrast);
if(tvh->tv_param->gain!=-1)
if(funcs->control(tvh->priv,TVI_CONTROL_VID_SET_GAIN,&tvh->tv_param->gain)!=TVI_CONTROL_TRUE)
MP_WARN(tvh, "Unable to set gain control!\n");
return 0;
}
int fill_audio_out_buffers(struct MPContext *mpctx, double endpts)
{
struct MPOpts *opts = mpctx->opts;
struct ao *ao = mpctx->ao;
int playsize;
int playflags = 0;
bool audio_eof = false;
bool signal_eof = false;
bool partial_fill = false;
sh_audio_t * const sh_audio = mpctx->sh_audio;
bool modifiable_audio_format = !(ao->format & AF_FORMAT_SPECIAL_MASK);
int unitsize = ao->channels.num * af_fmt2bits(ao->format) / 8;
if (mpctx->paused)
playsize = 1; // just initialize things (audio pts at least)
else
playsize = ao_get_space(ao);
// Coming here with hrseek_active still set means audio-only
if (!mpctx->sh_video || !mpctx->sync_audio_to_video)
mpctx->syncing_audio = false;
if (!opts->initial_audio_sync || !modifiable_audio_format) {
mpctx->syncing_audio = false;
mpctx->hrseek_active = false;
}
int res;
if (mpctx->syncing_audio || mpctx->hrseek_active)
res = audio_start_sync(mpctx, playsize);
else
res = decode_audio(sh_audio, &ao->buffer, playsize);
if (res < 0) { // EOF, error or format change
if (res == -2) {
/* The format change isn't handled too gracefully. A more precise
* implementation would require draining buffered old-format audio
* while displaying video, then doing the output format switch.
*/
if (!mpctx->opts->gapless_audio)
uninit_player(mpctx, INITIALIZED_AO);
reinit_audio_chain(mpctx);
return -1;
} else if (res == ASYNC_PLAY_DONE)
return 0;
else if (demux_stream_eof(mpctx->sh_audio->gsh))
audio_eof = true;
}
if (endpts != MP_NOPTS_VALUE && modifiable_audio_format) {
double bytes = (endpts - written_audio_pts(mpctx) + mpctx->audio_delay)
* ao->bps / opts->playback_speed;
if (playsize > bytes) {
playsize = MPMAX(bytes, 0);
audio_eof = true;
partial_fill = true;
}
}
assert(ao->buffer.len % unitsize == 0);
if (playsize > ao->buffer.len) {
partial_fill = true;
playsize = ao->buffer.len;
}
playsize -= playsize % unitsize;
if (!playsize)
return partial_fill && audio_eof ? -2 : -partial_fill;
if (audio_eof && partial_fill) {
if (opts->gapless_audio) {
// With gapless audio, delay this to ao_uninit. There must be only
// 1 final chunk, and that is handled when calling ao_uninit().
signal_eof = true;
} else {
playflags |= AOPLAY_FINAL_CHUNK;
}
}
assert(ao->buffer_playable_size <= ao->buffer.len);
int played = write_to_ao(mpctx, ao->buffer.start, playsize, playflags,
written_audio_pts(mpctx));
ao->buffer_playable_size = playsize - played;
if (played > 0) {
ao->buffer.len -= played;
memmove(ao->buffer.start, ao->buffer.start + played, ao->buffer.len);
} else if (!mpctx->paused && audio_eof && ao_get_delay(ao) < .04) {
// Sanity check to avoid hanging in case current ao doesn't output
// partial chunks and doesn't check for AOPLAY_FINAL_CHUNK
signal_eof = true;
}
return signal_eof ? -2 : -partial_fill;
}
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;
}
static int audio_start_sync(struct MPContext *mpctx, int playsize)
{
struct ao *ao = mpctx->ao;
struct MPOpts *opts = mpctx->opts;
sh_audio_t * const sh_audio = mpctx->sh_audio;
int res;
// Timing info may not be set without
res = decode_audio(sh_audio, &ao->buffer, 1);
if (res < 0)
return res;
int bytes;
bool did_retry = false;
double written_pts;
double bps = ao->bps / opts->playback_speed;
bool hrseek = mpctx->hrseek_active; // audio only hrseek
mpctx->hrseek_active = false;
while (1) {
written_pts = written_audio_pts(mpctx);
double ptsdiff;
if (hrseek)
ptsdiff = written_pts - mpctx->hrseek_pts;
else
ptsdiff = written_pts - mpctx->sh_video->pts - mpctx->delay
- mpctx->audio_delay;
bytes = ptsdiff * bps;
bytes -= bytes % (ao->channels.num * af_fmt2bits(ao->format) / 8);
// ogg demuxers give packets without timing
if (written_pts <= 1 && sh_audio->pts == MP_NOPTS_VALUE) {
if (!did_retry) {
// Try to read more data to see packets that have pts
res = decode_audio(sh_audio, &ao->buffer, ao->bps);
if (res < 0)
return res;
did_retry = true;
continue;
}
bytes = 0;
}
if (fabs(ptsdiff) > 300 || isnan(ptsdiff)) // pts reset or just broken?
bytes = 0;
if (bytes > 0)
break;
mpctx->syncing_audio = false;
int a = MPMIN(-bytes, MPMAX(playsize, 20000));
res = decode_audio(sh_audio, &ao->buffer, a);
bytes += ao->buffer.len;
if (bytes >= 0) {
memmove(ao->buffer.start,
ao->buffer.start + ao->buffer.len - bytes, bytes);
ao->buffer.len = bytes;
if (res < 0)
return res;
return decode_audio(sh_audio, &ao->buffer, playsize);
}
ao->buffer.len = 0;
if (res < 0)
return res;
}
if (hrseek)
// Don't add silence in audio-only case even if position is too late
return 0;
int fillbyte = 0;
if ((ao->format & AF_FORMAT_SIGN_MASK) == AF_FORMAT_US)
fillbyte = 0x80;
if (bytes >= playsize) {
/* This case could fall back to the one below with
* bytes = playsize, but then silence would keep accumulating
* in a_out_buffer if the AO accepts less data than it asks for
* in playsize. */
char *p = malloc(playsize);
memset(p, fillbyte, playsize);
write_to_ao(mpctx, p, playsize, 0, written_pts - bytes / bps);
free(p);
return ASYNC_PLAY_DONE;
}
mpctx->syncing_audio = false;
decode_audio_prepend_bytes(&ao->buffer, bytes, fillbyte);
return decode_audio(sh_audio, &ao->buffer, playsize);
}
// Filter data through filter
static af_data_t* play(struct af_instance_s* af, af_data_t* data)
{
af_ac3enc_t *s = af->setup;
af_data_t *c = data; // Current working data
af_data_t *l;
int len, left, outsize = 0, destsize;
char *buf, *src, *dest;
int max_output_len;
int frame_num = (data->len + s->pending_len) / s->expect_len;
int samplesize = af_fmt2bits(s->in_sampleformat) / 8;
if (s->add_iec61937_header)
max_output_len = AC3_FRAME_SIZE * 2 * 2 * frame_num;
else
max_output_len = AC3_MAX_CODED_FRAME_SIZE * frame_num;
if (af->data->len < max_output_len) {
mp_msg(MSGT_AFILTER, MSGL_V, "[libaf] Reallocating memory in module %s, "
"old len = %i, new len = %i\n", af->info->name, af->data->len,
max_output_len);
free(af->data->audio);
af->data->audio = malloc(max_output_len);
if (!af->data->audio) {
mp_msg(MSGT_AFILTER, MSGL_FATAL, "[libaf] Could not allocate memory \n");
return NULL;
}
af->data->len = max_output_len;
}
l = af->data; // Local data
buf = (char *)l->audio;
src = (char *)c->audio;
left = c->len;
while (left > 0) {
if (left + s->pending_len < s->expect_len) {
memcpy(s->pending_data + s->pending_len, src, left);
src += left;
s->pending_len += left;
left = 0;
break;
}
dest = s->add_iec61937_header ? buf + 8 : buf;
destsize = (char *)l->audio + l->len - buf;
if (s->pending_len) {
int needs = s->expect_len - s->pending_len;
if (needs > 0) {
memcpy(s->pending_data + s->pending_len, src, needs);
src += needs;
left -= needs;
}
if (c->nch >= 5)
reorder_channel_nch(s->pending_data,
AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
AF_CHANNEL_LAYOUT_LAVC_DEFAULT,
c->nch,
s->expect_len / samplesize, samplesize);
len = avcodec_encode_audio(s->lavc_actx, dest, destsize,
(void *)s->pending_data);
s->pending_len = 0;
}
else {
if (c->nch >= 5)
reorder_channel_nch(src,
AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
AF_CHANNEL_LAYOUT_LAVC_DEFAULT,
c->nch,
s->expect_len / samplesize, samplesize);
len = avcodec_encode_audio(s->lavc_actx,dest,destsize,(void *)src);
src += s->expect_len;
left -= s->expect_len;
}
mp_msg(MSGT_AFILTER, MSGL_DBG2, "avcodec_encode_audio got %d, pending %d.\n",
len, s->pending_len);
if (s->add_iec61937_header) {
int bsmod = dest[5] & 0x7;
AV_WB16(buf, 0xF872); // iec 61937 syncword 1
AV_WB16(buf + 2, 0x4E1F); // iec 61937 syncword 2
buf[4] = bsmod; // bsmod
buf[5] = 0x01; // data-type ac3
AV_WB16(buf + 6, len << 3); // number of bits in payload
memset(buf + 8 + len, 0, AC3_FRAME_SIZE * 2 * 2 - 8 - len);
len = AC3_FRAME_SIZE * 2 * 2;
}
outsize += len;
buf += len;
}
c->audio = l->audio;
c->nch = 2;
c->bps = 2;
c->len = outsize;
mp_msg(MSGT_AFILTER, MSGL_DBG2, "play return size %d, pending %d\n",
outsize, s->pending_len);
return c;
}
// 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;
af->data->format = s->in_sampleformat;
af->data->bps = af_fmt2bits(s->in_sampleformat) / 8;
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;
test_output_res = af_test_output(af, data);
s->pending_len = 0;
s->expect_len = AC3_FRAME_SIZE * data->nch * af->data->bps;
assert(s->expect_len <= s->pending_data_size);
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);
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) {
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_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;
}
af->data->format = AF_FORMAT_AC3_BE;
af->data->bps = 2;
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 (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;
}
/**
\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;
}
/* Prefer playing audio with the samplerate given in container data
* if available, but take number the number of channels and sample format
* from the codec, since if the codec isn't using the correct values for
* those everything breaks anyway.
*/
static int setup_format(sh_audio_t *sh_audio)
{
struct priv *priv = sh_audio->context;
AVCodecContext *codec = priv->avctx;
int sample_format = sample_fmt_lavc2native(codec->sample_fmt);
if (sample_format == AF_FORMAT_UNKNOWN) {
#ifndef CONFIG_LIBAVRESAMPLE
if (av_sample_fmt_is_planar(codec->sample_fmt))
mp_msg(MSGT_DECAUDIO, MSGL_ERR,
"The player has been compiled without libavresample "
"support,\nwhich is needed with this libavcodec decoder "
"version.\nCompile with libavresample enabled to make "
"audio decoding work!\n");
else
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "Unsupported sample format\n");
goto error;
#else
if (priv->avr && (priv->resample_fmt != codec->sample_fmt ||
priv->resample_channels != codec->channels))
avresample_free(&priv->avr);
if (!priv->avr) {
int ret;
uint8_t error[128];
enum AVSampleFormat out_fmt =
av_get_packed_sample_fmt(codec->sample_fmt);
uint64_t ch_layout = codec->channel_layout;
mp_msg(MSGT_DECAUDIO, MSGL_V,
"(Re)initializing libavresample format conversion...\n");
if (!ch_layout)
ch_layout = av_get_default_channel_layout(codec->channels);
/* if lavc format is planar, try just getting packed equivalent */
sample_format = sample_fmt_lavc2native(out_fmt);
if (sample_format == AF_FORMAT_UNKNOWN) {
/* fallback to s16 */
out_fmt = AV_SAMPLE_FMT_S16;
sample_format = AF_FORMAT_S16_NE;
}
priv->avr = avresample_alloc_context();
if (!priv->avr) {
mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Out of memory.\n");
abort();
}
av_opt_set_int(priv->avr, "in_channel_layout", ch_layout, 0);
av_opt_set_int(priv->avr, "out_channel_layout", ch_layout, 0);
av_opt_set_int(priv->avr, "in_sample_rate", codec->sample_rate, 0);
av_opt_set_int(priv->avr, "out_sample_rate", codec->sample_rate, 0);
av_opt_set_int(priv->avr, "in_sample_fmt", codec->sample_fmt, 0);
av_opt_set_int(priv->avr, "out_sample_fmt", out_fmt, 0);
if ((ret = avresample_open(priv->avr)) < 0) {
av_strerror(ret, error, sizeof(error));
mp_msg(MSGT_DECAUDIO, MSGL_ERR,
"Error opening libavresample: %s.\n", error);
goto error;
}
priv->resample_fmt = codec->sample_fmt;
priv->resample_channels = codec->channels;
priv->out_fmt = out_fmt;
priv->unitsize = av_get_bytes_per_sample(out_fmt) *
codec->channels;
} else
sample_format = sh_audio->sample_format;
} else if (priv->avr) {
avresample_free(&priv->avr);
#endif
}
bool broken_srate = false;
int samplerate = codec->sample_rate;
int container_samplerate = sh_audio->container_out_samplerate;
if (!container_samplerate && sh_audio->wf)
container_samplerate = sh_audio->wf->nSamplesPerSec;
if (codec->codec_id == AV_CODEC_ID_AAC
&& samplerate == 2 * container_samplerate)
broken_srate = true;
else if (container_samplerate)
samplerate = container_samplerate;
if (codec->channels != sh_audio->channels ||
samplerate != sh_audio->samplerate ||
sample_format != sh_audio->sample_format) {
sh_audio->channels = codec->channels;
sh_audio->samplerate = samplerate;
sh_audio->sample_format = sample_format;
sh_audio->samplesize = af_fmt2bits(sh_audio->sample_format) / 8;
if (broken_srate)
mp_msg(MSGT_DECAUDIO, MSGL_WARN,
"Ignoring broken container sample rate for AAC with SBR\n");
return 1;
}
return 0;
error:
#ifdef CONFIG_LIBAVRESAMPLE
avresample_free(&priv->avr);
#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);
}
void af_fix_parameters(af_data_t *data) {
data->bps = af_fmt2bits(data->format)/8;
}
