mpeg3_ac3_t* mpeg3_new_ac3()
{
mpeg3_ac3_t *result = calloc(1, sizeof(mpeg3_ac3_t));
result->stream = mpeg3bits_new_stream(0, 0);
#ifdef A52_INIT_NEEDS_ARG
result->state = a52_init(0);
#else
result->state = a52_init();
#endif
result->output = a52_samples(result->state);
return result;
}
static GF_Err AC3_AttachStream(GF_BaseDecoder *ifcg, GF_ESD *esd)
{
A52CTX();
if (ctx->ES_ID && ctx->ES_ID!=esd->ESID) return GF_NOT_SUPPORTED;
GF_LOG(GF_LOG_DEBUG, GF_LOG_CODEC, ("[A52] Attaching stream %d\n", esd->ESID));
if (ctx->codec) a52_free(ctx->codec);
ctx->codec = a52_init(MM_ACCEL_DJBFFT);
if (!ctx->codec) {
GF_LOG(GF_LOG_ERROR, GF_LOG_CODEC, ("[A52] Error initializing decoder\n"));
return GF_IO_ERR;
}
ctx->samples = a52_samples(ctx->codec);
if (!ctx->samples) {
a52_free(ctx->codec);
GF_LOG(GF_LOG_ERROR, GF_LOG_CODEC, ("[A52] Error initializing decoder\n"));
return GF_IO_ERR;
}
ctx->num_channels = 0;
ctx->sample_rate = 0;
ctx->out_size = 0;
ctx->num_samples = 1536;
ctx->ES_ID = esd->ESID;
return GF_OK;
}
ADM_AudiocodecAC3::ADM_AudiocodecAC3( uint32_t fourcc, WAVHeader *info,uint32_t extraLength,uint8_t *extraData)
: ADM_Audiocodec(fourcc)
{
int flags=0;
ADM_assert(fourcc==WAV_AC3);
ac3_handle=NULL;
ac3_sample=NULL;
#ifdef ADM_CPU_X86
#define CHK(x,y) if(CpuCaps::has##x()) flags|=MM_ACCEL_X86_##y;
CHK(MMX,MMX);
CHK(3DNOW,3DNOW);
CHK(MMXEXT,MMXEXT);
#endif
ac3_handle=(void *)a52_init(flags);
if(!ac3_handle)
{
printf("Cannot init a52\n");
ADM_assert(0);
}
ac3_sample=(sample_t *)a52_samples(AC3_HANDLE);
if(!ac3_sample)
{
printf("Cannot init a52 sample\n");
ADM_assert(0);
}
_downmix=0;
_wavHeader = info;
ADM_assert(_wavHeader);
}
static int liba52_init(adec_feeder_t * feeder)
{
uint32_t a52_accel = 0;
int flags = 0;
c_feeder = feeder;
if (feeder == NULL)
return -1;
memset(&a52_buffer, 0, sizeof(a52_buffer));
a52_buffer.a_in_buffer = frame_buffer;
a52_state = a52_init(a52_accel);
if (a52_state == NULL) {
printf("A52 init failed !\n");
return -1;
}
if (a52_fillbuff(&a52_buffer) < 0) {
printf("A52 sync failed !\n");
return -1;
}
a52_buffer.channels = audio_output_channels;
switch (a52_buffer.channels) {
case 1:
a52_flags = A52_MONO;
break;
/*case 2: a52_flags=A52_STEREO; break; */
case 2:
a52_flags = A52_DOLBY;
break;
/*case 3: a52_flags=A52_3F; break; */
case 3:
a52_flags = A52_2F1R;
break;
case 4:
a52_flags = A52_2F2R;
break; /* 2+2 */
case 5:
a52_flags = A52_3F2R;
break;
case 6:
a52_flags = A52_3F2R | A52_LFE;
break; /* 5.1 */
}
flags |= a52_flags;
if (a52_resample_init(a52_accel, flags, a52_buffer.channels) < 0) {
printf("a52_resample_init failed!\n");
a52_free(a52_state);
return -1;
}
feeder->channel_num = audio_output_channels;
feeder->sample_rate = a52_buffer.samplerate;
return 0;
}
/* this is called for each file to process */
enum codec_status codec_run(void)
{
size_t n;
unsigned char *filebuf;
int sample_loc;
intptr_t param;
if (codec_init())
return CODEC_ERROR;
ci->configure(DSP_SET_FREQUENCY, ci->id3->frequency);
codec_set_replaygain(ci->id3);
/* Intialise the A52 decoder and check for success */
state = a52_init(0);
samplesdone = 0;
/* The main decoding loop */
if (ci->id3->offset) {
if (ci->seek_buffer(ci->id3->offset)) {
samplesdone = (ci->id3->offset / ci->id3->bytesperframe) *
A52_SAMPLESPERFRAME;
ci->set_elapsed(samplesdone/(ci->id3->frequency / 1000));
}
}
else {
ci->seek_buffer(ci->id3->first_frame_offset);
ci->set_elapsed(0);
}
while (1) {
enum codec_command_action action = ci->get_command(¶m);
if (action == CODEC_ACTION_HALT)
break;
if (action == CODEC_ACTION_SEEK_TIME) {
sample_loc = param/1000 * ci->id3->frequency;
if (ci->seek_buffer((sample_loc/A52_SAMPLESPERFRAME)*ci->id3->bytesperframe)) {
samplesdone = sample_loc;
ci->set_elapsed(samplesdone/(ci->id3->frequency/1000));
}
ci->seek_complete();
}
filebuf = ci->request_buffer(&n, BUFFER_SIZE);
if (n == 0) /* End of Stream */
break;
a52_decode_data(filebuf, filebuf + n);
ci->advance_buffer(n);
}
return CODEC_OK;
}
static GstStateChangeReturn
gst_a52dec_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstA52Dec *a52dec = GST_A52DEC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:{
GstA52DecClass *klass;
klass = GST_A52DEC_CLASS (G_OBJECT_GET_CLASS (a52dec));
a52dec->state = a52_init (klass->a52_cpuflags);
break;
}
case GST_STATE_CHANGE_READY_TO_PAUSED:
a52dec->samples = a52_samples (a52dec->state);
a52dec->bit_rate = -1;
a52dec->sample_rate = -1;
a52dec->stream_channels = A52_CHANNEL;
a52dec->using_channels = A52_CHANNEL;
a52dec->level = 1;
a52dec->bias = 0;
a52dec->time = 0;
a52dec->sent_segment = FALSE;
a52dec->flag_update = TRUE;
gst_segment_init (&a52dec->segment, GST_FORMAT_UNDEFINED);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
a52dec->samples = NULL;
if (a52dec->cache) {
gst_buffer_unref (a52dec->cache);
a52dec->cache = NULL;
}
clear_queued (a52dec);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
a52_free (a52dec->state);
a52dec->state = NULL;
break;
default:
break;
}
return ret;
}
static gboolean
gst_a52dec_start (GstAudioDecoder * dec)
{
GstA52Dec *a52dec = GST_A52DEC (dec);
GstA52DecClass *klass;
GST_DEBUG_OBJECT (dec, "start");
klass = GST_A52DEC_CLASS (G_OBJECT_GET_CLASS (a52dec));
#if defined(A52_ACCEL_DETECT)
a52dec->state = a52_init ();
/* This line is just to avoid being accused of not using klass */
a52_accel (klass->a52_cpuflags & A52_ACCEL_DETECT);
#else
a52dec->state = a52_init (klass->a52_cpuflags);
#endif
if (!a52dec->state) {
GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), LIBRARY, INIT, (NULL),
("failed to initialize a52 state"));
return FALSE;
}
a52dec->samples = a52_samples (a52dec->state);
a52dec->bit_rate = -1;
a52dec->sample_rate = -1;
a52dec->stream_channels = A52_CHANNEL;
a52dec->using_channels = A52_CHANNEL;
a52dec->level = 1;
a52dec->bias = 0;
a52dec->flag_update = TRUE;
/* call upon legacy upstream byte support (e.g. seeking) */
gst_audio_decoder_set_estimate_rate (dec, TRUE);
return TRUE;
}
bool TaudioCodecLiba52::init(const CMediaType &mt)
{
dll=new Tdll(dllname,config);
dll->loadFunction(a52_init,"a52_init");
dll->loadFunction(a52_samples,"a52_samples");
dll->loadFunction(a52_syncinfo,"a52_syncinfo");
dll->loadFunction(a52_dynrng,"a52_dynrng");
dll->loadFunction(a52_dynrngsetlevel,"a52_dynrngsetlevel");
dll->loadFunction(a52_frame,"a52_frame");
dll->loadFunction(a52_block,"a52_block");
dll->loadFunction(a52_free,"a52_free");
if (dll->ok) {
state=a52_init(Tconfig::cpu_flags);
fmt.sf=TsampleFormat::SF_FLOAT32;
inited=true;
return true;
} else {
return false;
}
}
/*****************************************************************************
* Open:
*****************************************************************************/
static int Open( vlc_object_t *p_this, filter_sys_t *p_sys,
audio_format_t input, audio_format_t output )
{
p_sys->b_dynrng = var_InheritBool( p_this, "a52-dynrng" );
p_sys->b_dontwarn = 0;
/* No upmixing: it's not necessary and some other filters may want to do
* it themselves. */
if ( aout_FormatNbChannels( &output ) > aout_FormatNbChannels( &input ) )
{
if ( ! var_InheritBool( p_this, "a52-upmix" ) )
{
return VLC_EGENERIC;
}
}
/* We'll do our own downmixing, thanks. */
p_sys->i_nb_channels = aout_FormatNbChannels( &output );
switch ( (output.i_physical_channels & AOUT_CHAN_PHYSMASK)
& ~AOUT_CHAN_LFE )
{
case AOUT_CHAN_CENTER:
if ( (output.i_original_channels & AOUT_CHAN_CENTER)
|| (output.i_original_channels
& (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT)) )
{
p_sys->i_flags = A52_MONO;
}
else if ( output.i_original_channels & AOUT_CHAN_LEFT )
{
p_sys->i_flags = A52_CHANNEL1;
}
else
{
p_sys->i_flags = A52_CHANNEL2;
}
break;
case AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT:
if ( output.i_original_channels & AOUT_CHAN_DOLBYSTEREO )
{
p_sys->i_flags = A52_DOLBY;
}
else if ( input.i_original_channels == AOUT_CHAN_CENTER )
{
p_sys->i_flags = A52_MONO;
}
else if ( input.i_original_channels & AOUT_CHAN_DUALMONO )
{
p_sys->i_flags = A52_CHANNEL;
}
else if ( !(output.i_original_channels & AOUT_CHAN_RIGHT) )
{
p_sys->i_flags = A52_CHANNEL1;
}
else if ( !(output.i_original_channels & AOUT_CHAN_LEFT) )
{
p_sys->i_flags = A52_CHANNEL2;
}
else
{
p_sys->i_flags = A52_STEREO;
}
break;
case AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_CENTER:
p_sys->i_flags = A52_3F;
break;
case AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_REARCENTER:
p_sys->i_flags = A52_2F1R;
break;
case AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_CENTER
| AOUT_CHAN_REARCENTER:
p_sys->i_flags = A52_3F1R;
break;
case AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT
| AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT:
p_sys->i_flags = A52_2F2R;
break;
case AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_CENTER
| AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT:
p_sys->i_flags = A52_3F2R;
break;
default:
msg_Warn( p_this, "unknown sample format!" );
free( p_sys );
return VLC_EGENERIC;
}
if ( output.i_physical_channels & AOUT_CHAN_LFE )
{
p_sys->i_flags |= A52_LFE;
}
p_sys->i_flags |= A52_ADJUST_LEVEL;
/* Initialize liba52 */
p_sys->p_liba52 = a52_init( 0 );
if( p_sys->p_liba52 == NULL )
{
msg_Err( p_this, "unable to initialize liba52" );
free( p_sys );
return VLC_EGENERIC;
}
aout_CheckChannelReorder( pi_channels_in, NULL,
output.i_physical_channels & AOUT_CHAN_PHYSMASK,
p_sys->i_nb_channels,
p_sys->pi_chan_table );
return VLC_SUCCESS;
}
static int init(sh_audio_t *sh_audio)
{
uint32_t a52_accel=0;
sample_t level=a52_level, bias=384;
int flags=0;
/* Dolby AC3 audio:*/
#ifdef MM_ACCEL_X86_SSE
if(gCpuCaps.hasSSE) a52_accel|=MM_ACCEL_X86_SSE;
#endif
if(gCpuCaps.hasMMX) a52_accel|=MM_ACCEL_X86_MMX;
if(gCpuCaps.hasMMX2) a52_accel|=MM_ACCEL_X86_MMXEXT;
if(gCpuCaps.has3DNow) a52_accel|=MM_ACCEL_X86_3DNOW;
#ifdef MM_ACCEL_X86_3DNOWEXT
if(gCpuCaps.has3DNowExt) a52_accel|=MM_ACCEL_X86_3DNOWEXT;
#endif
#ifdef MM_ACCEL_PPC_ALTIVEC
if(gCpuCaps.hasAltiVec) a52_accel|=MM_ACCEL_PPC_ALTIVEC;
#endif
a52_state=a52_init (a52_accel);
if (a52_state == NULL) {
mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 init failed\n");
return 0;
}
sh_audio->sample_format = AF_FORMAT_FLOAT_NE;
if(a52_fillbuff(sh_audio)<0){
mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 sync failed\n");
return 0;
}
/* Init a52 dynrng */
if (drc_level < 0.001) {
/* level == 0 --> no compression, init library without callback */
a52_drc_action = DRC_NO_COMPRESSION;
} else if (drc_level > 0.999) {
/* level == 1 --> full compression, do nothing at all (library default = full compression) */
a52_drc_action = DRC_NO_ACTION;
} else {
a52_drc_action = DRC_CALLBACK;
}
/* Library init for dynrng has to be done for each frame, see decode_audio() */
/* 'a52 cannot upmix' hotfix:*/
a52_printinfo(sh_audio);
sh_audio->channels=audio_output_channels;
while(sh_audio->channels>0){
switch(sh_audio->channels){
case 1: a52_flags=A52_MONO; break;
/* case 2: a52_flags=A52_STEREO; break;*/
case 2: a52_flags=A52_DOLBY; break;
/* case 3: a52_flags=A52_3F; break;*/
case 3: a52_flags=A52_2F1R; break;
case 4: a52_flags=A52_2F2R; break; /* 2+2*/
case 5: a52_flags=A52_3F2R; break;
case 6: a52_flags=A52_3F2R|A52_LFE; break; /* 5.1*/
}
/* test:*/
flags=a52_flags|A52_ADJUST_LEVEL;
mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags before a52_frame: 0x%X\n",flags);
if (a52_frame (a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){
mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: error decoding frame -> nosound\n");
return 0;
}
mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags after a52_frame: 0x%X\n",flags);
/* frame decoded, let's init resampler:*/
channel_map = 0;
if (sh_audio->sample_format == AF_FORMAT_FLOAT_NE) {
if (!(flags & A52_LFE)) {
switch ((flags<<3) | sh_audio->channels) {
case (A52_MONO << 3) | 1: channel_map = 0x1; break;
case (A52_CHANNEL << 3) | 2:
case (A52_STEREO << 3) | 2:
case (A52_DOLBY << 3) | 2: channel_map = 0x21; break;
case (A52_2F1R << 3) | 3: channel_map = 0x321; break;
case (A52_2F2R << 3) | 4: channel_map = 0x4321; break;
case (A52_3F << 3) | 5: channel_map = 0x2ff31; break;
case (A52_3F2R << 3) | 5: channel_map = 0x25431; break;
}
} else if (sh_audio->channels == 6) {
switch (flags & ~A52_LFE) {
case A52_MONO : channel_map = 0x12ffff; break;
case A52_CHANNEL:
case A52_STEREO :
case A52_DOLBY : channel_map = 0x1fff32; break;
case A52_3F : channel_map = 0x13ff42; break;
case A52_2F1R : channel_map = 0x1f4432; break;
case A52_2F2R : channel_map = 0x1f5432; break;
case A52_3F2R : channel_map = 0x136542; break;
}
}
if (channel_map) {
a52_resample = a52_resample_float;
break;
}
} else
break;
}
if(sh_audio->channels<=0){
mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: no resampler. try different channel setup!\n");
return 0;
}
return 1;
}
/* this is the codec entry point */
enum codec_status codec_main(void)
{
size_t n;
unsigned char *filebuf;
int sample_loc;
int retval;
/* Generic codec initialisation */
ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
ci->configure(DSP_SET_SAMPLE_DEPTH, 28);
next_track:
retval = CODEC_OK;
if (codec_init()) {
retval = CODEC_ERROR;
goto exit;
}
if (codec_wait_taginfo() != 0)
goto request_next_track;
ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
codec_set_replaygain(ci->id3);
/* Intialise the A52 decoder and check for success */
state = a52_init(0);
/* The main decoding loop */
if (ci->id3->offset) {
if (ci->seek_buffer(ci->id3->offset)) {
samplesdone = (ci->id3->offset / ci->id3->bytesperframe) *
A52_SAMPLESPERFRAME;
ci->set_elapsed(samplesdone/(ci->id3->frequency / 1000));
}
}
else {
samplesdone = 0;
}
while (1) {
if (ci->stop_codec || ci->new_track)
break;
if (ci->seek_time) {
sample_loc = (ci->seek_time - 1)/1000 * ci->id3->frequency;
if (ci->seek_buffer((sample_loc/A52_SAMPLESPERFRAME)*ci->id3->bytesperframe)) {
samplesdone = sample_loc;
ci->set_elapsed(samplesdone/(ci->id3->frequency/1000));
}
ci->seek_complete();
}
filebuf = ci->request_buffer(&n, BUFFER_SIZE);
if (n == 0) /* End of Stream */
break;
a52_decode_data(filebuf, filebuf + n);
ci->advance_buffer(n);
}
request_next_track:
if (ci->request_next_track())
goto next_track;
exit:
a52_free(state);
return retval;
}
/* this is called for each file to process */
enum codec_status codec_run(void)
{
size_t n;
uint8_t *filebuf;
int consumed, packet_offset;
int playback_on = -1;
size_t resume_offset;
intptr_t param;
enum codec_command_action action = CODEC_ACTION_NULL;
if (codec_init()) {
return CODEC_ERROR;
}
resume_offset = ci->id3->offset;
ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
codec_set_replaygain(ci->id3);
ci->seek_buffer(ci->id3->first_frame_offset);
/* Intializations */
state = a52_init(0);
ci->memset(&rmctx,0,sizeof(RMContext));
ci->memset(&pkt,0,sizeof(RMPacket));
init_rm(&rmctx);
samplesdone = 0;
/* check for a mid-track resume and force a seek time accordingly */
if(resume_offset > rmctx.data_offset + DATA_HEADER_SIZE) {
resume_offset -= rmctx.data_offset + DATA_HEADER_SIZE;
/* put number of subpackets to skip in resume_offset */
resume_offset /= (rmctx.block_align + PACKET_HEADER_SIZE);
param = (int)resume_offset * ((rmctx.block_align * 8 * 1000)/rmctx.bit_rate);
action = CODEC_ACTION_SEEK_TIME;
}
else {
/* Seek to the first packet */
ci->set_elapsed(0);
ci->advance_buffer(rmctx.data_offset + DATA_HEADER_SIZE );
}
/* The main decoding loop */
while((unsigned)rmctx.audio_pkt_cnt < rmctx.nb_packets) {
if (action == CODEC_ACTION_NULL)
action = ci->get_command(¶m);
if (action == CODEC_ACTION_HALT)
break;
if (action == CODEC_ACTION_SEEK_TIME) {
packet_offset = param / ((rmctx.block_align*8*1000)/rmctx.bit_rate);
ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE +
packet_offset*(rmctx.block_align + PACKET_HEADER_SIZE));
rmctx.audio_pkt_cnt = packet_offset;
samplesdone = (rmctx.sample_rate/1000 * param);
ci->set_elapsed(samplesdone/(frequency/1000));
ci->seek_complete();
}
action = CODEC_ACTION_NULL;
filebuf = ci->request_buffer(&n, rmctx.block_align + PACKET_HEADER_SIZE);
consumed = rm_get_packet(&filebuf, &rmctx, &pkt);
if(consumed < 0 && playback_on != 0) {
if(playback_on == -1) {
/* Error only if packet-parsing failed and playback hadn't started */
DEBUGF("rm_get_packet failed\n");
return CODEC_ERROR;
}
else {
break;
}
}
playback_on = 1;
a52_decode_data(filebuf, filebuf + rmctx.block_align);
ci->advance_buffer(pkt.length);
}
return CODEC_OK;
}