/*****************************************************************************
* DecodeBlock:
*****************************************************************************/
static block_t *DecodeBlock( decoder_t *p_dec, block_t **pp_block )
{
decoder_sys_t *p_sys = p_dec->p_sys;
block_t *p_block;
if( !pp_block || !*pp_block ) return NULL;
p_block = *pp_block;
if( p_block->i_flags&(BLOCK_FLAG_DISCONTINUITY|BLOCK_FLAG_CORRUPTED) )
{
block_Release( p_block );
return NULL;
}
/* Remove ADTS header if we have decoder specific config */
if( p_dec->fmt_in.i_extra && p_block->i_buffer > 7 )
{
if( p_block->p_buffer[0] == 0xff &&
( p_block->p_buffer[1] & 0xf0 ) == 0xf0 ) /* syncword */
{ /* ADTS header present */
size_t i_header_size; /* 7 bytes (+ 2 bytes for crc) */
i_header_size = 7 + ( ( p_block->p_buffer[1] & 0x01 ) ? 0 : 2 );
/* FIXME: multiple blocks per frame */
if( p_block->i_buffer > i_header_size )
{
p_block->p_buffer += i_header_size;
p_block->i_buffer -= i_header_size;
}
}
}
/* Append the block to the temporary buffer */
if( p_sys->i_buffer_size < p_sys->i_buffer + p_block->i_buffer )
{
size_t i_buffer_size = p_sys->i_buffer + p_block->i_buffer;
uint8_t *p_buffer = realloc( p_sys->p_buffer, i_buffer_size );
if( p_buffer )
{
p_sys->i_buffer_size = i_buffer_size;
p_sys->p_buffer = p_buffer;
}
else
{
p_block->i_buffer = 0;
}
}
if( p_block->i_buffer > 0 )
{
memcpy( &p_sys->p_buffer[p_sys->i_buffer],
p_block->p_buffer, p_block->i_buffer );
p_sys->i_buffer += p_block->i_buffer;
p_block->i_buffer = 0;
}
if( p_dec->fmt_out.audio.i_rate == 0 && p_dec->fmt_in.i_extra > 0 )
{
/* We have a decoder config so init the handle */
unsigned long i_rate;
unsigned char i_channels;
if( faacDecInit2( p_sys->hfaad, p_dec->fmt_in.p_extra,
p_dec->fmt_in.i_extra,
&i_rate, &i_channels ) >= 0 )
{
p_dec->fmt_out.audio.i_rate = i_rate;
p_dec->fmt_out.audio.i_channels = i_channels;
p_dec->fmt_out.audio.i_physical_channels
= p_dec->fmt_out.audio.i_original_channels
= pi_channels_guessed[i_channels];
date_Init( &p_sys->date, i_rate, 1 );
}
}
if( p_dec->fmt_out.audio.i_rate == 0 && p_sys->i_buffer )
{
unsigned long i_rate;
unsigned char i_channels;
/* Init faad with the first frame */
if( faacDecInit( p_sys->hfaad,
p_sys->p_buffer, p_sys->i_buffer,
&i_rate, &i_channels ) < 0 )
{
block_Release( p_block );
return NULL;
}
p_dec->fmt_out.audio.i_rate = i_rate;
p_dec->fmt_out.audio.i_channels = i_channels;
p_dec->fmt_out.audio.i_physical_channels
= p_dec->fmt_out.audio.i_original_channels
= pi_channels_guessed[i_channels];
date_Init( &p_sys->date, i_rate, 1 );
}
if( p_block->i_pts > VLC_TS_INVALID && p_block->i_pts != date_Get( &p_sys->date ) )
{
date_Set( &p_sys->date, p_block->i_pts );
}
else if( !date_Get( &p_sys->date ) )
{
/* We've just started the stream, wait for the first PTS. */
block_Release( p_block );
p_sys->i_buffer = 0;
return NULL;
}
/* Decode all data */
if( p_sys->i_buffer )
{
void *samples;
faacDecFrameInfo frame;
block_t *p_out;
samples = faacDecDecode( p_sys->hfaad, &frame,
p_sys->p_buffer, p_sys->i_buffer );
if( frame.error > 0 )
{
msg_Warn( p_dec, "%s", faacDecGetErrorMessage( frame.error ) );
if( frame.error == 21 || frame.error == 12 )
{
/*
* Once an "Unexpected channel configuration change"
* or a "Invalid number of channels" error
* occurs, it will occurs afterwards, and we got no sound.
* Reinitialization of the decoder is required.
*/
unsigned long i_rate;
unsigned char i_channels;
faacDecHandle *hfaad;
faacDecConfiguration *cfg,*oldcfg;
oldcfg = faacDecGetCurrentConfiguration( p_sys->hfaad );
hfaad = faacDecOpen();
cfg = faacDecGetCurrentConfiguration( hfaad );
if( oldcfg->defSampleRate )
cfg->defSampleRate = oldcfg->defSampleRate;
cfg->defObjectType = oldcfg->defObjectType;
cfg->outputFormat = oldcfg->outputFormat;
faacDecSetConfiguration( hfaad, cfg );
if( faacDecInit( hfaad, p_sys->p_buffer, p_sys->i_buffer,
&i_rate,&i_channels ) < 0 )
{
/* reinitialization failed */
faacDecClose( hfaad );
faacDecSetConfiguration( p_sys->hfaad, oldcfg );
}
else
{
faacDecClose( p_sys->hfaad );
p_sys->hfaad = hfaad;
p_dec->fmt_out.audio.i_rate = i_rate;
p_dec->fmt_out.audio.i_channels = i_channels;
p_dec->fmt_out.audio.i_physical_channels
= p_dec->fmt_out.audio.i_original_channels
= pi_channels_guessed[i_channels];
date_Init( &p_sys->date, i_rate, 1 );
}
}
/* Flush the buffer */
p_sys->i_buffer = 0;
block_Release( p_block );
return NULL;
}
if( frame.channels <= 0 || frame.channels > 8 || frame.channels == 7 )
{
msg_Warn( p_dec, "invalid channels count: %i", frame.channels );
/* Flush the buffer */
p_sys->i_buffer -= frame.bytesconsumed;
if( p_sys->i_buffer > 0 )
{
memmove( p_sys->p_buffer,&p_sys->p_buffer[frame.bytesconsumed],
p_sys->i_buffer );
}
block_Release( p_block );
return NULL;
}
if( frame.samples <= 0 )
{
msg_Warn( p_dec, "decoded zero sample" );
/* Flush the buffer */
p_sys->i_buffer -= frame.bytesconsumed;
if( p_sys->i_buffer > 0 )
{
memmove( p_sys->p_buffer,&p_sys->p_buffer[frame.bytesconsumed],
p_sys->i_buffer );
}
block_Release( p_block );
return NULL;
}
/* We decoded a valid frame */
if( p_dec->fmt_out.audio.i_rate != frame.samplerate )
{
date_Init( &p_sys->date, frame.samplerate, 1 );
date_Set( &p_sys->date, p_block->i_pts );
}
p_block->i_pts = VLC_TS_INVALID; /* PTS is valid only once */
p_dec->fmt_out.audio.i_rate = frame.samplerate;
p_dec->fmt_out.audio.i_channels = frame.channels;
/* Adjust stream info when dealing with SBR/PS */
bool b_sbr = (frame.sbr == 1) || (frame.sbr == 2);
if( p_sys->b_sbr != b_sbr || p_sys->b_ps != frame.ps )
{
const char *psz_ext = (b_sbr && frame.ps) ? "SBR+PS" :
b_sbr ? "SBR" : "PS";
msg_Dbg( p_dec, "AAC %s (channels: %u, samplerate: %lu)",
psz_ext, frame.channels, frame.samplerate );
if( !p_dec->p_description )
p_dec->p_description = vlc_meta_New();
if( p_dec->p_description )
vlc_meta_AddExtra( p_dec->p_description, _("AAC extension"), psz_ext );
p_sys->b_sbr = b_sbr;
p_sys->b_ps = frame.ps;
}
/* Convert frame.channel_position to our own channel values */
p_dec->fmt_out.audio.i_physical_channels = 0;
const uint32_t nbChannels = frame.channels;
unsigned j;
for( unsigned i = 0; i < nbChannels; i++ )
{
/* Find the channel code */
for( j = 0; j < MAX_CHANNEL_POSITIONS; j++ )
{
if( frame.channel_position[i] == pi_channels_in[j] )
break;
}
if( j >= MAX_CHANNEL_POSITIONS )
{
msg_Warn( p_dec, "unknown channel ordering" );
/* Invent something */
j = i;
}
/* */
p_sys->pi_channel_positions[i] = pi_channels_out[j];
if( p_dec->fmt_out.audio.i_physical_channels & pi_channels_out[j] )
frame.channels--; /* We loose a duplicated channel */
else
p_dec->fmt_out.audio.i_physical_channels |= pi_channels_out[j];
}
if ( nbChannels != frame.channels )
{
p_dec->fmt_out.audio.i_physical_channels
= p_dec->fmt_out.audio.i_original_channels
= pi_channels_guessed[nbChannels];
}
else
{
p_dec->fmt_out.audio.i_original_channels =
p_dec->fmt_out.audio.i_physical_channels;
}
p_dec->fmt_out.audio.i_channels = nbChannels;
p_out = decoder_NewAudioBuffer( p_dec, frame.samples / nbChannels );
if( p_out == NULL )
{
p_sys->i_buffer = 0;
block_Release( p_block );
return NULL;
}
p_out->i_pts = date_Get( &p_sys->date );
p_out->i_length = date_Increment( &p_sys->date,
frame.samples / nbChannels )
- p_out->i_pts;
DoReordering( (uint32_t *)p_out->p_buffer, samples,
frame.samples / nbChannels, nbChannels,
p_sys->pi_channel_positions );
p_sys->i_buffer -= frame.bytesconsumed;
if( p_sys->i_buffer > 0 )
{
memmove( p_sys->p_buffer, &p_sys->p_buffer[frame.bytesconsumed],
p_sys->i_buffer );
}
return p_out;
}
block_Release( p_block );
return NULL;
}
static int init(sh_audio_t *sh)
{
unsigned long faac_samplerate;
unsigned char faac_channels;
int faac_init, pos = 0;
faac_hdec = faacDecOpen();
// If we don't get the ES descriptor, try manual config
if(!sh->codecdata_len && sh->wf) {
sh->codecdata_len = sh->wf->cbSize;
sh->codecdata = (char*)(sh->wf+1);
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: codecdata extracted from WAVEFORMATEX\n");
}
if(!sh->codecdata_len) {
#if 1
faacDecConfigurationPtr faac_conf;
/* Set the default object type and samplerate */
/* This is useful for RAW AAC files */
faac_conf = faacDecGetCurrentConfiguration(faac_hdec);
if(sh->samplerate)
faac_conf->defSampleRate = sh->samplerate;
/* XXX: FAAD support FLOAT output, how do we handle
* that (FAAD_FMT_FLOAT)? ::atmos
*/
if (audio_output_channels <= 2) faac_conf->downMatrix = 1;
switch(sh->samplesize){
case 1: // 8Bit
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: 8Bit samplesize not supported by FAAD, assuming 16Bit!\n");
default:
sh->samplesize=2;
case 2: // 16Bit
faac_conf->outputFormat = FAAD_FMT_16BIT;
break;
case 3: // 24Bit
faac_conf->outputFormat = FAAD_FMT_24BIT;
break;
case 4: // 32Bit
faac_conf->outputFormat = FAAD_FMT_32BIT;
break;
}
//faac_conf->defObjectType = LTP; // => MAIN, LC, SSR, LTP available.
faacDecSetConfiguration(faac_hdec, faac_conf);
#endif
sh->a_in_buffer_len = demux_read_data(sh->ds, sh->a_in_buffer, sh->a_in_buffer_size);
pos = aac_probe(sh->a_in_buffer, sh->a_in_buffer_len);
if(pos) {
sh->a_in_buffer_len -= pos;
memmove(sh->a_in_buffer, &(sh->a_in_buffer[pos]), sh->a_in_buffer_len);
sh->a_in_buffer_len +=
demux_read_data(sh->ds,&(sh->a_in_buffer[sh->a_in_buffer_len]),
sh->a_in_buffer_size - sh->a_in_buffer_len);
pos = 0;
}
/* init the codec */
faac_init = faacDecInit(faac_hdec, sh->a_in_buffer,
sh->a_in_buffer_len, &faac_samplerate, &faac_channels);
sh->a_in_buffer_len -= (faac_init > 0)?faac_init:0; // how many bytes init consumed
// XXX FIXME: shouldn't we memcpy() here in a_in_buffer ?? --A'rpi
} else { // We have ES DS in codecdata
faacDecConfigurationPtr faac_conf = faacDecGetCurrentConfiguration(faac_hdec);
if (audio_output_channels <= 2) {
faac_conf->downMatrix = 1;
faacDecSetConfiguration(faac_hdec, faac_conf);
}
/*int i;
for(i = 0; i < sh_audio->codecdata_len; i++)
printf("codecdata_dump %d: 0x%02X\n", i, sh_audio->codecdata[i]);*/
faac_init = faacDecInit2(faac_hdec, sh->codecdata,
sh->codecdata_len, &faac_samplerate, &faac_channels);
}
if(faac_init < 0) {
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to initialize the decoder!\n"); // XXX: deal with cleanup!
faacDecClose(faac_hdec);
// XXX: free a_in_buffer here or in uninit?
return 0;
} else {
mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Decoder init done (%dBytes)!\n", sh->a_in_buffer_len); // XXX: remove or move to debug!
mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Negotiated samplerate: %ldHz channels: %d\n", faac_samplerate, faac_channels);
sh->channels = faac_channels;
if (audio_output_channels <= 2) sh->channels = faac_channels > 1 ? 2 : 1;
sh->samplerate = faac_samplerate;
sh->samplesize=2;
//sh->o_bps = sh->samplesize*faac_channels*faac_samplerate;
if(!sh->i_bps) {
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: compressed input bitrate missing, assuming 128kbit/s!\n");
sh->i_bps = 128*1000/8; // XXX: HACK!!! ::atmos
} else
mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: got %dkbit/s bitrate from MP4 header!\n",sh->i_bps*8/1000);
}
return 1;
}
/*****************************************************************************
* OpenDecoder: probe the decoder and return score
*****************************************************************************/
static int Open( vlc_object_t *p_this )
{
decoder_t *p_dec = (decoder_t*)p_this;
decoder_sys_t *p_sys;
faacDecConfiguration *cfg;
if( p_dec->fmt_in.i_codec != VLC_CODEC_MP4A )
{
return VLC_EGENERIC;
}
/* Allocate the memory needed to store the decoder's structure */
if( ( p_dec->p_sys = p_sys = malloc( sizeof(*p_sys) ) ) == NULL )
return VLC_ENOMEM;
/* Open a faad context */
if( ( p_sys->hfaad = faacDecOpen() ) == NULL )
{
msg_Err( p_dec, "cannot initialize faad" );
free( p_sys );
return VLC_EGENERIC;
}
/* Misc init */
date_Set( &p_sys->date, 0 );
p_dec->fmt_out.i_cat = AUDIO_ES;
p_dec->fmt_out.i_codec = HAVE_FPU ? VLC_CODEC_FL32 : VLC_CODEC_S16N;
p_dec->pf_decode_audio = DecodeBlock;
p_dec->fmt_out.audio.i_physical_channels =
p_dec->fmt_out.audio.i_original_channels = 0;
if( p_dec->fmt_in.i_extra > 0 )
{
/* We have a decoder config so init the handle */
unsigned long i_rate;
unsigned char i_channels;
if( faacDecInit2( p_sys->hfaad, p_dec->fmt_in.p_extra,
p_dec->fmt_in.i_extra,
&i_rate, &i_channels ) < 0 )
{
msg_Err( p_dec, "Failed to initialize faad using extra data" );
faacDecClose( p_sys->hfaad );
free( p_sys );
return VLC_EGENERIC;
}
p_dec->fmt_out.audio.i_rate = i_rate;
p_dec->fmt_out.audio.i_channels = i_channels;
p_dec->fmt_out.audio.i_physical_channels
= p_dec->fmt_out.audio.i_original_channels
= pi_channels_guessed[i_channels];
date_Init( &p_sys->date, i_rate, 1 );
}
else
{
/* Will be initalised from first frame */
p_dec->fmt_out.audio.i_rate = 0;
p_dec->fmt_out.audio.i_channels = 0;
}
/* Set the faad config */
cfg = faacDecGetCurrentConfiguration( p_sys->hfaad );
cfg->outputFormat = HAVE_FPU ? FAAD_FMT_FLOAT : FAAD_FMT_16BIT;
faacDecSetConfiguration( p_sys->hfaad, cfg );
/* buffer */
p_sys->i_buffer = p_sys->i_buffer_size = 0;
p_sys->p_buffer = NULL;
/* Faad2 can't deal with truncated data (eg. from MPEG TS) */
p_dec->b_need_packetized = true;
p_sys->b_sbr = p_sys->b_ps = false;
return VLC_SUCCESS;
}
int decodeAACfile(char *sndfile, int def_srate, aac_dec_opt *opt)
{
int tagsize;
unsigned long samplerate;
unsigned char channels;
void *sample_buffer;
FILE *infile;
audio_file *aufile;
faacDecHandle hDecoder;
faacDecFrameInfo frameInfo;
faacDecConfigurationPtr config;
int first_time = 1;
/* declare variables for buffering */
DEC_BUFF_VARS
infile = fopen(opt->filename, "rb");
if (infile == NULL)
{
/* unable to open file */
error_handler("Error opening file: %s\n", opt->filename);
return 1;
}
INIT_BUFF(infile)
tagsize = id3v2_tag(buffer);
if (tagsize)
{
UPDATE_BUFF_SKIP(tagsize)
}
hDecoder = faacDecOpen();
/* Set the default object type and samplerate */
/* This is useful for RAW AAC files */
config = faacDecGetCurrentConfiguration(hDecoder);
if (def_srate)
config->defSampleRate = def_srate;
config->defObjectType = opt->object_type;
config->outputFormat = opt->output_format;
faacDecSetConfiguration(hDecoder, config);
if ((bytesconsumed = faacDecInit(hDecoder, buffer, bytes_in_buffer,
&samplerate, &channels)) < 0)
{
/* If some error initializing occured, skip the file */
error_handler("Error initializing decoder library.\n");
END_BUFF
faacDecClose(hDecoder);
fclose(infile);
return 1;
}
buffer_index += bytesconsumed;
do
{
/* update buffer */
UPDATE_BUFF_READ
sample_buffer = faacDecDecode(hDecoder, &frameInfo, buffer, bytes_in_buffer);
/* update buffer indices */
UPDATE_BUFF_IDX(frameInfo)
if (frameInfo.error > 0)
{
error_handler("Error: %s\n",
faacDecGetErrorMessage(frameInfo.error));
}
opt->progress_update((long)fileread, buffer_index);
/* open the sound file now that the number of channels are known */
if (first_time && !frameInfo.error)
{
if(opt->decode_mode == 0)
{
if (Set_WIN_Params (INVALID_FILEDESC, samplerate, SAMPLE_SIZE,
frameInfo.channels) < 0)
{
error_handler("\nCan't access %s\n", "WAVE OUT");
END_BUFF
faacDecClose(hDecoder);
fclose(infile);
return (0);
}
}
else
{
aufile = open_audio_file(sndfile, samplerate, frameInfo.channels,
opt->output_format, opt->file_type, aacChannelConfig2wavexChannelMask(&frameInfo));
if (aufile == NULL)
{
END_BUFF
faacDecClose(hDecoder);
fclose(infile);
return 0;
}
}
first_time = 0;
}
if ((frameInfo.error == 0) && (frameInfo.samples > 0))
{
if(opt->decode_mode == 0)
WIN_Play_Samples((short*)sample_buffer, frameInfo.channels*frameInfo.samples);
else
write_audio_file(aufile, sample_buffer, frameInfo.samples, 0);
}
if (buffer_index >= fileread)
sample_buffer = NULL; /* to make sure it stops now */
if(stop_decoding)
break;
} while (sample_buffer != NULL);
faacDecClose(hDecoder);
fclose(infile);
if(opt->decode_mode == 0)
WIN_Audio_close();
else
{
if (!first_time)
close_audio_file(aufile);
}
END_BUFF
return frameInfo.error;
}
static void *mp4Decode(void *args)
{
MP4FileHandle mp4file;
pthread_mutex_lock(&mutex);
seekPosition = -1;
bPlaying = TRUE;
if(!(mp4file = MP4Read(args, 0))){
mp4cfg.file_type = FILE_AAC;
MP4Close(mp4file);
}else{
mp4cfg.file_type = FILE_MP4;
}
if(mp4cfg.file_type == FILE_MP4){
// We are reading a MP4 file
gint mp4track;
if((mp4track = getAACTrack(mp4file)) < 0){
//TODO: check here for others Audio format.....
g_print("Unsupported Audio track type\n");
g_free(args);
MP4Close(mp4file);
bPlaying = FALSE;
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}else{
faacDecHandle decoder;
unsigned char *buffer = NULL;
guint bufferSize = 0;
gulong samplerate;
guchar channels;
guint avgBitrate;
MP4Duration duration;
gulong msDuration;
MP4SampleId numSamples;
MP4SampleId sampleID = 1;
decoder = faacDecOpen();
MP4GetTrackESConfiguration(mp4file, mp4track, &buffer, &bufferSize);
if(!buffer){
g_free(args);
faacDecClose(decoder);
MP4Close(mp4file);
bPlaying = FALSE;
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}
if(faacDecInit2(decoder, buffer, bufferSize, &samplerate, &channels)<0){
g_free(args);
faacDecClose(decoder);
MP4Close(mp4file);
bPlaying = FALSE;
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}
g_free(buffer);
if(channels == 0){
g_print("Number of Channels not supported\n");
g_free(args);
faacDecClose(decoder);
MP4Close(mp4file);
bPlaying = FALSE;
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}
duration = MP4GetTrackDuration(mp4file, mp4track);
msDuration = MP4ConvertFromTrackDuration(mp4file, mp4track, duration,
MP4_MSECS_TIME_SCALE);
numSamples = MP4GetTrackNumberOfSamples(mp4file, mp4track);
mp4_ip.output->open_audio(FMT_S16_NE, samplerate, channels);
mp4_ip.output->flush(0);
mp4_ip.set_info(args, msDuration, -1, samplerate/1000, channels);
g_print("MP4 - %d channels @ %d Hz\n", channels, samplerate);
while(bPlaying){
void* sampleBuffer;
faacDecFrameInfo frameInfo;
gint rc;
if(seekPosition!=-1){
duration = MP4ConvertToTrackDuration(mp4file,
mp4track,
seekPosition*1000,
MP4_MSECS_TIME_SCALE);
sampleID = MP4GetSampleIdFromTime(mp4file, mp4track, duration, 0);
mp4_ip.output->flush(seekPosition*1000);
seekPosition = -1;
}
buffer=NULL;
bufferSize=0;
if(sampleID > numSamples){
mp4_ip.output->close_audio();
g_free(args);
faacDecClose(decoder);
MP4Close(mp4file);
bPlaying = FALSE;
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}
rc = MP4ReadSample(mp4file, mp4track, sampleID++, &buffer, &bufferSize,
NULL, NULL, NULL, NULL);
//g_print("%d/%d\n", sampleID-1, numSamples);
if((rc==0) || (buffer== NULL)){
g_print("MP4: read error\n");
sampleBuffer = NULL;
sampleID=0;
mp4_ip.output->buffer_free();
mp4_ip.output->close_audio();
g_free(args);
faacDecClose(decoder);
MP4Close(mp4file);
bPlaying = FALSE;
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}else{
sampleBuffer = faacDecDecode(decoder, &frameInfo, buffer, bufferSize);
if(frameInfo.error > 0){
g_print("MP4: %s\n",
faacDecGetErrorMessage(frameInfo.error));
mp4_ip.output->close_audio();
g_free(args);
faacDecClose(decoder);
MP4Close(mp4file);
bPlaying = FALSE;
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}
if(buffer){
g_free(buffer); buffer=NULL; bufferSize=0;
}
while(bPlaying && mp4_ip.output->buffer_free()<frameInfo.samples<<1)
xmms_usleep(30000);
}
mp4_ip.add_vis_pcm(mp4_ip.output->written_time(),
FMT_S16_NE,
channels,
frameInfo.samples<<1,
sampleBuffer);
mp4_ip.output->write_audio(sampleBuffer, frameInfo.samples<<1);
}
while(bPlaying && mp4_ip.output->buffer_free()){
xmms_usleep(10000);
}
mp4_ip.output->close_audio();
g_free(args);
faacDecClose(decoder);
MP4Close(mp4file);
bPlaying = FALSE;
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}
} else{
// WE ARE READING AN AAC FILE
FILE *file = NULL;
faacDecHandle decoder = 0;
guchar *buffer = 0;
gulong bufferconsumed = 0;
gulong samplerate = 0;
guchar channels;
gulong buffervalid = 0;
TitleInput* input;
gchar *temp = g_strdup(args);
gchar *ext = strrchr(temp, '.');
gchar *xmmstitle = NULL;
faacDecConfigurationPtr config;
if((file = fopen(args, "rb")) == 0){
g_print("AAC: can't find file %s\n", args);
bPlaying = FALSE;
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}
if((decoder = faacDecOpen()) == NULL){
g_print("AAC: Open Decoder Error\n");
fclose(file);
bPlaying = FALSE;
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}
config = faacDecGetCurrentConfiguration(decoder);
config->useOldADTSFormat = 0;
faacDecSetConfiguration(decoder, config);
if((buffer = g_malloc(BUFFER_SIZE)) == NULL){
g_print("AAC: error g_malloc\n");
fclose(file);
bPlaying = FALSE;
faacDecClose(decoder);
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}
if((buffervalid = fread(buffer, 1, BUFFER_SIZE, file))==0){
g_print("AAC: Error reading file\n");
g_free(buffer);
fclose(file);
bPlaying = FALSE;
faacDecClose(decoder);
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}
XMMS_NEW_TITLEINPUT(input);
input->file_name = g_basename(temp);
input->file_ext = ext ? ext+1 : NULL;
input->file_path = temp;
if(!strncmp(buffer, "ID3", 3)){
gint size = 0;
fseek(file, 0, SEEK_SET);
size = (buffer[6]<<21) | (buffer[7]<<14) | (buffer[8]<<7) | buffer[9];
size+=10;
fread(buffer, 1, size, file);
buffervalid = fread(buffer, 1, BUFFER_SIZE, file);
}
xmmstitle = xmms_get_titlestring(xmms_get_gentitle_format(), input);
if(xmmstitle == NULL)
xmmstitle = g_strdup(input->file_name);
if(temp) g_free(temp);
if(input->performer) g_free(input->performer);
if(input->album_name) g_free(input->album_name);
if(input->track_name) g_free(input->track_name);
if(input->genre) g_free(input->genre);
g_free(input);
bufferconsumed = faacDecInit(decoder,
buffer,
buffervalid,
&samplerate,
&channels);
if(mp4_ip.output->open_audio(FMT_S16_NE,samplerate,channels) == FALSE){
g_print("AAC: Output Error\n");
g_free(buffer); buffer=0;
faacDecClose(decoder);
fclose(file);
mp4_ip.output->close_audio();
/*
if(positionTable){
g_free(positionTable); positionTable=0;
}
*/
g_free(xmmstitle);
bPlaying = FALSE;
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}
//if(bSeek){
//mp4_ip.set_info(xmmstitle, lenght*1000, -1, samplerate, channels);
//}else{
mp4_ip.set_info(xmmstitle, -1, -1, samplerate, channels);
//}
mp4_ip.output->flush(0);
while(bPlaying && buffervalid > 0){
faacDecFrameInfo finfo;
unsigned long samplesdecoded;
char* sample_buffer = NULL;
/*
if(bSeek && seekPosition!=-1){
fseek(file, positionTable[seekPosition], SEEK_SET);
bufferconsumed=0;
buffervalid = fread(buffer, 1, BUFFER_SIZE, file);
aac_ip.output->flush(seekPosition*1000);
seekPosition=-1;
}
*/
if(bufferconsumed > 0){
memmove(buffer, &buffer[bufferconsumed], buffervalid-bufferconsumed);
buffervalid -= bufferconsumed;
buffervalid += fread(&buffer[buffervalid], 1,
BUFFER_SIZE-buffervalid, file);
bufferconsumed = 0;
}
sample_buffer = faacDecDecode(decoder, &finfo, buffer, buffervalid);
if(finfo.error){
config = faacDecGetCurrentConfiguration(decoder);
if(config->useOldADTSFormat != 1){
faacDecClose(decoder);
decoder = faacDecOpen();
config = faacDecGetCurrentConfiguration(decoder);
config->useOldADTSFormat = 1;
faacDecSetConfiguration(decoder, config);
finfo.bytesconsumed=0;
finfo.samples = 0;
faacDecInit(decoder,
buffer,
buffervalid,
&samplerate,
&channels);
}else{
g_print("FAAD2 Warning %s\n", faacDecGetErrorMessage(finfo.error));
buffervalid = 0;
}
}
bufferconsumed += finfo.bytesconsumed;
samplesdecoded = finfo.samples;
if((samplesdecoded<=0) && !sample_buffer){
g_print("AAC: error sample decoding\n");
continue;
}
while(bPlaying && mp4_ip.output->buffer_free() < (samplesdecoded<<1)){
xmms_usleep(10000);
}
mp4_ip.add_vis_pcm(mp4_ip.output->written_time(),
FMT_S16_LE, channels,
samplesdecoded<<1, sample_buffer);
mp4_ip.output->write_audio(sample_buffer, samplesdecoded<<1);
}
while(bPlaying && mp4_ip.output->buffer_playing()){
xmms_usleep(10000);
}
mp4_ip.output->buffer_free();
mp4_ip.output->close_audio();
bPlaying = FALSE;
g_free(buffer);
faacDecClose(decoder);
g_free(xmmstitle);
fclose(file);
seekPosition = -1;
/*
if(positionTable){
g_free(positionTable); positionTable=0;
}
*/
bPlaying = FALSE;
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}
}
static gboolean
xmms_faad_init (xmms_xform_t *xform)
{
xmms_faad_data_t *data;
xmms_error_t error;
faacDecConfigurationPtr config;
gint bytes_read;
guint32 samplerate;
guchar channels;
g_return_val_if_fail (xform, FALSE);
data = g_new0 (xmms_faad_data_t, 1);
data->outbuf = g_string_new (NULL);
data->buffer_size = FAAD_BUFFER_SIZE;
xmms_xform_private_data_set (xform, data);
data->decoder = faacDecOpen ();
config = faacDecGetCurrentConfiguration (data->decoder);
config->defObjectType = LC;
config->defSampleRate = 44100;
config->outputFormat = FAAD_FMT_16BIT;
config->downMatrix = 0;
config->dontUpSampleImplicitSBR = 0;
faacDecSetConfiguration (data->decoder, config);
switch (config->outputFormat) {
case FAAD_FMT_16BIT:
data->sampleformat = XMMS_SAMPLE_FORMAT_S16;
break;
case FAAD_FMT_24BIT:
/* we don't have 24-bit format to use in xmms2 */
data->sampleformat = XMMS_SAMPLE_FORMAT_S32;
break;
case FAAD_FMT_32BIT:
data->sampleformat = XMMS_SAMPLE_FORMAT_S32;
break;
case FAAD_FMT_FLOAT:
data->sampleformat = XMMS_SAMPLE_FORMAT_FLOAT;
break;
case FAAD_FMT_DOUBLE:
data->sampleformat = XMMS_SAMPLE_FORMAT_DOUBLE;
break;
}
while (data->buffer_length < 8) {
xmms_error_reset (&error);
bytes_read = xmms_xform_read (xform,
(gchar *) data->buffer + data->buffer_length,
data->buffer_size - data->buffer_length,
&error);
data->buffer_length += bytes_read;
if (bytes_read < 0) {
xmms_log_error ("Error while trying to read data on init");
goto err;
} else if (bytes_read == 0) {
XMMS_DBG ("Not enough bytes to check the AAC header");
goto err;
}
}
/* which type of file are we dealing with? */
data->filetype = FAAD_TYPE_UNKNOWN;
if (xmms_xform_auxdata_has_val (xform, "decoder_config")) {
data->filetype = FAAD_TYPE_MP4;
} else if (!strncmp ((char *) data->buffer, "ADIF", 4)) {
data->filetype = FAAD_TYPE_ADIF;
} else {
int i;
/* ADTS mpeg file can be a stream and start in the middle of a
* frame so we need to have extra loop check here */
for (i=0; i<data->buffer_length-1; i++) {
if (data->buffer[i] == 0xff && (data->buffer[i+1]&0xf6) == 0xf0) {
data->filetype = FAAD_TYPE_ADTS;
g_memmove (data->buffer, data->buffer+i, data->buffer_length-i);
data->buffer_length -= i;
break;
}
}
}
if (data->filetype == FAAD_TYPE_ADTS || data->filetype == FAAD_TYPE_ADIF) {
bytes_read = faacDecInit (data->decoder, data->buffer,
data->buffer_length, &samplerate,
&channels);
} else if (data->filetype == FAAD_TYPE_MP4) {
const guchar *tmpbuf;
gssize tmpbuflen;
guchar *copy;
if (!xmms_xform_auxdata_get_bin (xform, "decoder_config", &tmpbuf,
&tmpbuflen)) {
XMMS_DBG ("AAC decoder config data found but it's wrong type! (something broken?)");
goto err;
}
copy = g_memdup (tmpbuf, tmpbuflen);
bytes_read = faacDecInit2 (data->decoder, copy, tmpbuflen,
&samplerate, &channels);
g_free (copy);
}
if (bytes_read < 0) {
XMMS_DBG ("Error initializing decoder library.");
goto err;
}
/* Get mediainfo and skip the possible header */
xmms_faad_get_mediainfo (xform);
g_memmove (data->buffer, data->buffer + bytes_read,
data->buffer_length - bytes_read);
data->buffer_length -= bytes_read;
data->samplerate = samplerate;
data->channels = channels;
/* Because for HE AAC files some versions of libfaad return the wrong
* samplerate in init, we have to do one read and let it decide the
* real parameters. After changing sample parameters and format is
* supported, this hack should be removed and handled in read instead. */
{
gchar tmpbuf[1024];
xmms_error_reset (&error);
bytes_read = xmms_faad_read (xform, tmpbuf, 1024, &error);
if (bytes_read <= 0) {
XMMS_DBG ("First read from faad decoder failed!");
return FALSE;
}
g_string_prepend_len (data->outbuf, tmpbuf, bytes_read);
}
xmms_xform_outdata_type_add (xform,
XMMS_STREAM_TYPE_MIMETYPE,
"audio/pcm",
XMMS_STREAM_TYPE_FMT_FORMAT,
data->sampleformat,
XMMS_STREAM_TYPE_FMT_CHANNELS,
data->channels,
XMMS_STREAM_TYPE_FMT_SAMPLERATE,
data->samplerate,
XMMS_STREAM_TYPE_END);
XMMS_DBG ("AAC decoder inited successfully!");
return TRUE;
err:
g_string_free (data->outbuf, TRUE);
g_free (data);
return FALSE;
}
int fdpl_decode_AAC(int def_srate, const char *file_path, size_t &c, size_t &sr, size_t &samples, std::vector<float> * ch1, std::vector<float> *ch2)
{
int tagsize;
uint32_t samplerate;
unsigned char channels;
void *sample_buffer;
FILE *infile;
faacDecHandle hDecoder;
faacDecFrameInfo frameInfo;
faacDecConfigurationPtr config;
/* declare variables for buffering */
DEC_BUFF_VARS
infile = fopen(file_path, "rb");
if (infile == NULL)
{
/* unable to open file */
std::cerr<<"Error opening file"<<std::endl;
return 1;
}
INIT_BUFF(infile)
tagsize = id3v2_tag(buffer);//avoids ID3 tag (useful for mp3 as well)
if (tagsize)
{
UPDATE_BUFF_SKIP(tagsize)
}
hDecoder = faacDecOpen();
/* Set the default object type and samplerate */
/* This is useful for RAW AAC files */
config = faacDecGetCurrentConfiguration(hDecoder);
if (def_srate)
config->defSampleRate = def_srate;
config->defObjectType = LC;//default
config->outputFormat = FAAD_FMT_FLOAT;
config -> downMatrix = 0; //set to 1
faacDecSetConfiguration(hDecoder, config);
if ((bytesconsumed = faacDecInit(hDecoder, buffer, bytes_in_buffer,
&samplerate, &channels)) < 0)
{
/* If some error initializing occured, skip the file */
std::cerr<<"Error initializing decoder library."<<std::endl;
END_BUFF
faacDecClose(hDecoder);
fclose(infile);
return 1;
}
buffer_index += bytesconsumed;
do
{
/* update buffer */
UPDATE_BUFF_READ
sample_buffer = faacDecDecode(hDecoder, &frameInfo, buffer, bytes_in_buffer);
/* update buffer indices */
UPDATE_BUFF_IDX(frameInfo)
if (frameInfo.error > 0)
{
std::cerr<<"Error: %s\n"<<faacDecGetErrorMessage(frameInfo.error)<<std::endl;
}
if ((frameInfo.error == 0) && (frameInfo.samples > 0) && (frameInfo.channels < 3))
{
size_t samples_channel = (size_t)frameInfo.samples/(size_t)frameInfo.channels;
samples += samples_channel;
float *pcm;
pcm = (float *)sample_buffer;
for(size_t i=0; i<samples_channel; i++){
//writing the samples of each channel in separate float vectors
for(size_t j=0; j<frameInfo.channels; j++){
if(j==0){
ch1 -> push_back(pcm[i*frameInfo.channels+j]);
}else if(j==1){
ch2 -> push_back(pcm[i*frameInfo.channels+j]);
}else{
std::cerr<<"Unexpected number of channels: "<<j<<std::endl;
return -1;
}
}
}
}
if (buffer_index >= fileread)
sample_buffer = NULL; /* to make sure it stops now */
} while (sample_buffer != NULL);
c = frameInfo.channels;
sr = frameInfo.samplerate;
faacDecClose(hDecoder);
fclose(infile);
END_BUFF
return frameInfo.error;
}
