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Commit 90a612c6 authored by Sam Hocevar's avatar Sam Hocevar
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* ./modules/audio_filter/converter/fixed32tofloat32.c: fixed conversion. 

  * ./modules/codec/mad/libmad.c: ported to the latest aout changes. Still
    doesn't work here, though.
parent 00724b73
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Showing with 217 additions and 496 deletions
modules/audio_filter/converter/fixed32tofloat32.c
View file @ 90a612c6
......@@ -2,7 +2,7 @@
* fixed32float32.c : converter from fixed32 to float32 bits integer
*****************************************************************************
* Copyright (C) 2002 VideoLAN
* $Id: fixed32tofloat32.c,v 1.6 2002/08/21 22:41:59 massiot Exp $
* $Id: fixed32tofloat32.c,v 1.7 2002/08/22 17:14:52 sam Exp $
*
* Authors: Jean-Paul Saman <jpsaman@wxs.nl>
*
......@@ -87,31 +87,7 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
for ( i = p_in_buf->i_nb_samples * p_filter->input.i_channels ; i-- ; )
{
/* convert vlc_fixed_t into s32 */
#if 0
s32 temp;
if ( *p_in >= 8 ) temp = 32767;
else if ( *p_in < -8 ) temp = -32768;
else temp = *p_in * (s32) 4096; // (32768/8);
#endif
/* convert s32 into float */
#if 0
if (temp >= 32768)
*p_out = (float) 1.0;
else if (temp <= -32768)
*p_out = (float) -1.0;
else *p_out = (float) (temp/32768.0);
#endif
/* combined conversion */
/* This has absolutely no chance of working. *p_in is s32, gcc
* doesn't know anything of vlc_fixed_t... --Meuuh */
if ( *p_in >= 8 ) *p_out = (float) 1.0;
else if ( *p_in < -8 ) *p_out = (float) -1.0;
else *p_out =(float) (*p_in/8.0);
p_in++; p_out++;
*p_out++ = (float)*p_in++ / -(float)FIXED32_MIN;
}
p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
......
modules/codec/mad/decoder.c
View file @ 90a612c6
......@@ -42,19 +42,19 @@
*****************************************************************************/
static int OpenDecoder ( vlc_object_t * );
static int RunDecoder ( decoder_fifo_t * );
static int InitThread ( mad_adec_thread_t * p_mad_adec );
static void EndThread ( mad_adec_thread_t * p_mad_adec );
static int InitThread ( mad_adec_thread_t * );
static void EndThread ( mad_adec_thread_t * );
/*****************************************************************************
* Module descriptor
*****************************************************************************/
#define DOWNSCALE_TEXT N_("Mad audio downscale routine (fast,mp321)")
#define DOWNSCALE_TEXT N_("mad audio downscale routine (fast,mpg321)")
#define DOWNSCALE_LONGTEXT N_( \
"Specify the mad audio downscale routine you want to use. By default " \
"the mad plugin will use the fastest routine.")
vlc_module_begin();
add_category_hint( N_("Miscellaneous"), NULL );
add_category_hint( N_("Libmad"), NULL );
add_string( "downscale", "fast", NULL, DOWNSCALE_TEXT, DOWNSCALE_LONGTEXT );
set_description( _("libmad MPEG 1/2/3 audio decoder") );
set_capability( "decoder", 100 );
......@@ -85,68 +85,71 @@ static int OpenDecoder( vlc_object_t *p_this )
*****************************************************************************/
static int RunDecoder( decoder_fifo_t *p_fifo )
{
mad_adec_thread_t * p_mad_adec;
mad_adec_thread_t * p_dec;
int i_ret;
/* Allocate the memory needed to store the thread's structure */
p_mad_adec = (mad_adec_thread_t *) malloc(sizeof(mad_adec_thread_t));
p_dec = (mad_adec_thread_t *) malloc(sizeof(mad_adec_thread_t));
if (p_mad_adec == NULL)
if (p_dec == NULL)
{
msg_Err( p_fifo, "out of memory" );
DecoderError( p_fifo );
return( -1 );
return VLC_ENOMEM;
}
/*
* Initialize the thread properties
*/
p_mad_adec->p_fifo = p_fifo;
if( InitThread( p_mad_adec ) )
p_dec->p_fifo = p_fifo;
if( InitThread( p_dec ) )
{
msg_Err( p_fifo, "could not initialize thread" );
DecoderError( p_fifo );
free( p_mad_adec );
return( -1 );
free( p_dec );
return VLC_ETHREAD;
}
/* mad decoder thread's main loop */
while ((!p_mad_adec->p_fifo->b_die) && (!p_mad_adec->p_fifo->b_error))
while( !p_dec->p_fifo->b_die && !p_dec->p_fifo->b_error )
{
msg_Dbg( p_mad_adec->p_fifo, "starting libmad decoder" );
if (mad_decoder_run(p_mad_adec->libmad_decoder, MAD_DECODER_MODE_SYNC)==-1)
msg_Dbg( p_dec->p_fifo, "starting libmad decoder" );
i_ret = mad_decoder_run( &p_dec->libmad_decoder,
MAD_DECODER_MODE_SYNC );
if( i_ret == -1 )
{
msg_Err( p_mad_adec->p_fifo, "libmad decoder returned abnormally" );
DecoderError( p_mad_adec->p_fifo );
EndThread(p_mad_adec);
return( -1 );
msg_Err( p_dec->p_fifo, "libmad decoder returned abnormally" );
DecoderError( p_dec->p_fifo );
EndThread(p_dec);
return VLC_EGENERIC;
}
}
/* If b_error is set, the mad decoder thread enters the error loop */
if (p_mad_adec->p_fifo->b_error)
if (p_dec->p_fifo->b_error)
{
DecoderError( p_mad_adec->p_fifo );
DecoderError( p_dec->p_fifo );
}
/* End of the mad decoder thread */
EndThread (p_mad_adec);
EndThread (p_dec);
return( 0 );
return VLC_SUCCESS;
}
/*****************************************************************************
* InitThread: initialize data before entering main loop
*****************************************************************************/
static int InitThread( mad_adec_thread_t * p_mad_adec )
static int InitThread( mad_adec_thread_t * p_dec )
{
decoder_fifo_t * p_fifo = p_mad_adec->p_fifo;
decoder_fifo_t * p_fifo = p_dec->p_fifo;
char *psz_downscale = NULL;
/* Initialize the thread properties */
p_mad_adec->p_aout = NULL;
p_mad_adec->p_aout_input = NULL;
p_mad_adec->output_format.i_format = AOUT_FMT_FIXED32;
p_mad_adec->output_format.i_channels = 2; /* FIXME ! */
p_dec->p_aout = NULL;
p_dec->p_aout_input = NULL;
p_dec->output_format.i_format = AOUT_FMT_FIXED32;
p_dec->output_format.i_channels = 2; /* FIXME ! */
/*
* Properties of audio for libmad
......@@ -157,45 +160,43 @@ static int InitThread( mad_adec_thread_t * p_mad_adec )
if ( strncmp(psz_downscale,"fast",4)==0 )
{
p_mad_adec->audio_scaling = FAST_SCALING;
p_dec->audio_scaling = FAST_SCALING;
msg_Dbg( p_fifo, "downscale fast selected" );
}
else if ( strncmp(psz_downscale,"mpg321",7)==0 )
{
p_mad_adec->audio_scaling = MPG321_SCALING;
p_dec->audio_scaling = MPG321_SCALING;
msg_Dbg( p_fifo, "downscale mpg321 selected" );
}
else
{
p_mad_adec->audio_scaling = FAST_SCALING;
p_dec->audio_scaling = FAST_SCALING;
msg_Dbg( p_fifo, "downscale default fast selected" );
}
if (psz_downscale) free(psz_downscale);
/* Initialize the libmad decoder structures */
p_mad_adec->libmad_decoder = (struct mad_decoder*) malloc(sizeof(struct mad_decoder));
if (p_mad_adec->libmad_decoder == NULL)
{
msg_Err( p_mad_adec->p_fifo, "out of memory" );
return -1;
}
p_mad_adec->i_current_pts = p_mad_adec->i_next_pts = 0;
p_dec->i_current_pts = p_dec->i_next_pts = 0;
mad_decoder_init( p_mad_adec->libmad_decoder,
p_mad_adec, /* vlc's thread structure and p_fifo playbuffer */
mad_decoder_init( &p_dec->libmad_decoder,
p_dec, /* vlc's thread structure and p_fifo playbuffer */
libmad_input, /* input_func */
0, /* header_func */
0, /* filter */
libmad_output3, /* output_func */
0, /* error */
0); /* message */
NULL, /* header_func */
NULL, /* filter */
libmad_output, /* output_func */
NULL, /* error */
NULL ); /* message */
mad_decoder_options(p_mad_adec->libmad_decoder, MAD_OPTION_IGNORECRC);
mad_decoder_options( &p_dec->libmad_decoder, MAD_OPTION_IGNORECRC );
/*
* Initialize the input properties
*/
/* Init the Bitstream */
InitBitstream( &p_dec->bit_stream, p_dec->p_fifo, NULL, NULL );
/* Get the first data packet. */
vlc_mutex_lock( &p_fifo->data_lock );
while ( p_fifo->p_first == NULL )
......@@ -203,32 +204,30 @@ static int InitThread( mad_adec_thread_t * p_mad_adec )
if ( p_fifo->b_die )
{
vlc_mutex_unlock( &p_fifo->data_lock );
return( -1 );
return VLC_EGENERIC;
}
vlc_cond_wait( &p_fifo->data_wait, &p_fifo->data_lock );
}
vlc_mutex_unlock( &p_fifo->data_lock );
p_mad_adec->p_data = p_fifo->p_first->p_first;
p_dec->p_data = p_fifo->p_first->p_first;
return( 0 );
return VLC_SUCCESS;
}
/*****************************************************************************
* EndThread : libmad decoder thread destruction
*****************************************************************************/
static void EndThread (mad_adec_thread_t * p_mad_adec)
static void EndThread (mad_adec_thread_t * p_dec)
{
/* If the audio output fifo was created, we destroy it */
if (p_mad_adec->p_aout_input != NULL)
if (p_dec->p_aout_input != NULL)
{
aout_InputDelete( p_mad_adec->p_aout, p_mad_adec->p_aout_input );
aout_InputDelete( p_dec->p_aout, p_dec->p_aout_input );
}
/* mad_decoder_finish releases the memory allocated inside the struct */
mad_decoder_finish( p_mad_adec->libmad_decoder );
mad_decoder_finish( &p_dec->libmad_decoder );
/* Unlock the modules, p_mad_adec->p_fifo is released by the decoder subsystem */
free( p_mad_adec->libmad_decoder );
free( p_mad_adec );
free( p_dec );
}
modules/codec/mad/decoder.h
View file @ 90a612c6
......@@ -29,7 +29,7 @@ typedef struct mad_adec_thread_s
/*
* Decoder properties
*/
struct mad_decoder *libmad_decoder;
struct mad_decoder libmad_decoder;
mad_timer_t libmad_timer;
byte_t buffer[MAD_BUFFER_MDLEN];
......@@ -38,6 +38,9 @@ typedef struct mad_adec_thread_s
*/
vlc_thread_t thread_id; /* id for thread functions */
/* The bit stream structure handles the PES stream at the bit level */
bit_stream_t bit_stream;
/*
* Input properties
*/
......@@ -47,19 +50,14 @@ typedef struct mad_adec_thread_s
/* Store i_pts for syncing audio frames */
mtime_t i_current_pts, i_next_pts;
/*
* Output properties
*/
//old way aout_fifo_t * p_aout_fifo; /* stores the decompressed audio frames */
/*
* Output properties
*/
aout_instance_t * p_aout; /* opaque */
aout_input_t * p_aout_input; /* opaque */
audio_sample_format_t output_format;
audio_date_t end_date;
mtime_t last_date;
enum mad_scaling audio_scaling;
} mad_adec_thread_t;
......
modules/codec/mad/libmad.c
View file @ 90a612c6
......@@ -41,452 +41,198 @@ static void PrintFrameInfo(struct mad_header *Header);
* libmad_input: this function is called by libmad when the input buffer needs
* to be filled.
*****************************************************************************/
enum mad_flow libmad_input(void *data, struct mad_stream *p_libmad_stream)
enum mad_flow libmad_input( void *p_data, struct mad_stream *p_stream )
{
mad_adec_thread_t *p_mad_adec = (mad_adec_thread_t *) data;
size_t ReadSize, Remaining;
unsigned char *ReadStart;
mad_adec_thread_t * p_dec = (mad_adec_thread_t *) p_data;
size_t i_wanted, i_left;
if ( p_mad_adec->p_fifo->b_die == 1 ) {
msg_Dbg( p_mad_adec->p_fifo, "libmad_input stopping libmad decoder" );
if ( p_dec->p_fifo->b_die )
{
msg_Dbg( p_dec->p_fifo, "stopping libmad decoder" );
return MAD_FLOW_STOP;
}
if ( p_mad_adec->p_fifo->b_error == 1 ) {
msg_Warn( p_mad_adec->p_fifo, "libmad_input ignoring current audio frame" );
if ( p_dec->p_fifo->b_error )
{
msg_Warn( p_dec->p_fifo, "ignoring current audio frame" );
return MAD_FLOW_IGNORE;
}
/* libmad_stream_buffer does not consume the total buffer, it consumes only data
* for one frame only. So all data left in the buffer should be put back in front.
*/
if ((p_libmad_stream->buffer==NULL) || (p_libmad_stream->error==MAD_ERROR_BUFLEN))
/* libmad_stream_buffer does not consume the total buffer, it consumes
* only data for one frame only. So all data left in the buffer should
* be put back in front. */
if ( !p_stream->buffer || p_stream->error == MAD_ERROR_BUFLEN )
{
/* libmad does not consume all the buffer it's given. Some
* datas, part of a truncated frame, is left unused at the
* end of the buffer. Those datas must be put back at the
* beginning of the buffer and taken in account for
* refilling the buffer. This means that the input buffer
* must be large enough to hold a complete frame at the
* highest observable bit-rate (currently 448 kb/s). XXX=XXX
* Is 2016 bytes the size of the largest frame?
* (448000*(1152/32000))/8
*/
if(p_libmad_stream->next_frame!=NULL)
/* libmad does not consume all the buffer it's given. Some data,
* part of a truncated frame, is left unused at the end of the
* buffer. Those datas must be put back at the beginning of the
* buffer and taken in account for refilling the buffer. This
* means that the input buffer must be large enough to hold a
* complete frame at the highest observable bit-rate (currently
* 448 kb/s). XXX=XXX Is 2016 bytes the size of the largest frame?
* (448000*(1152/32000))/8 */
if( p_stream->next_frame )
{
Remaining=p_libmad_stream->bufend-p_libmad_stream->next_frame;
if( p_mad_adec->buffer != p_libmad_stream->next_frame )
i_left = p_stream->bufend - p_stream->next_frame;
if( p_dec->buffer != p_stream->next_frame )
{
memcpy( p_mad_adec->buffer,
p_libmad_stream->next_frame, Remaining );
memcpy( p_dec->buffer, p_stream->next_frame, i_left );
}
ReadStart=p_mad_adec->buffer+Remaining;
ReadSize=(MAD_BUFFER_MDLEN)-Remaining;
i_wanted = MAD_BUFFER_MDLEN - i_left;
/* Store time stamp of next frame */
p_mad_adec->i_current_pts = p_mad_adec->i_next_pts;
p_mad_adec->i_next_pts = p_mad_adec->p_fifo->p_first->i_pts;
/* Store timestamp for next frame */
p_dec->i_next_pts = p_dec->p_fifo->p_first->i_pts;
}
else
{
ReadSize=(MAD_BUFFER_MDLEN);
ReadStart=p_mad_adec->buffer;
Remaining=0;
p_mad_adec->i_next_pts = 0;
p_mad_adec->i_current_pts = p_mad_adec->p_fifo->p_first->i_pts;
i_wanted = MAD_BUFFER_MDLEN;
i_left = 0;
/* Store timestamp for this frame */
p_dec->i_current_pts = p_dec->p_fifo->p_first->i_pts;
}
/* Fill-in the buffer. If an error occurs print a message
* and leave the decoding loop. If the end of stream is
* reached we also leave the loop but the return status is
* left untouched.
*/
if( ReadSize > p_mad_adec->p_data->p_payload_end
- p_mad_adec->p_data->p_payload_start )
/* Fill-in the buffer. If an error occurs print a message and leave
* the decoding loop. If the end of stream is reached we also leave
* the loop but the return status is left untouched. */
if( i_wanted > p_dec->p_data->p_payload_end
- p_dec->p_data->p_payload_start )
{
ReadSize = p_mad_adec->p_data->p_payload_end
- p_mad_adec->p_data->p_payload_start;
memcpy( ReadStart, p_mad_adec->p_data->p_payload_start, ReadSize );
NextDataPacket( p_mad_adec->p_fifo, &p_mad_adec->p_data );
i_wanted = p_dec->p_data->p_payload_end
- p_dec->p_data->p_payload_start;
memcpy( p_dec->buffer + i_left,
p_dec->p_data->p_payload_start, i_wanted );
NextDataPacket( p_dec->p_fifo, &p_dec->p_data );
}
else
{
memcpy( ReadStart, p_mad_adec->p_data->p_payload_start, ReadSize );
p_mad_adec->p_data->p_payload_start += ReadSize;
memcpy( p_dec->buffer + i_left,
p_dec->p_data->p_payload_start, i_wanted );
p_dec->p_data->p_payload_start += i_wanted;
}
if ( p_mad_adec->p_fifo->b_die == 1 )
if ( p_dec->p_fifo->b_die )
{
msg_Dbg( p_mad_adec->p_fifo, "libmad_input stopping libmad decoder" );
msg_Dbg( p_dec->p_fifo, "stopping libmad decoder" );
return MAD_FLOW_STOP;
}
if ( p_mad_adec->p_fifo->b_error == 1 )
if ( p_dec->p_fifo->b_error )
{
msg_Warn( p_mad_adec->p_fifo, "libmad_input ignoring current audio frame" );
msg_Warn( p_dec->p_fifo, "ignoring current audio frame" );
return MAD_FLOW_IGNORE;
}
/* Pipe the new buffer content to libmad's stream decoder facility.
* Libmad never copies the buffer, but just references it. So keep it in
* mad_adec_thread_t structure.
*/
mad_stream_buffer(p_libmad_stream,(unsigned char*) &p_mad_adec->buffer,
Remaining + ReadSize);
p_libmad_stream->error=0;
* Libmad never copies the buffer, but just references it. So keep
* it in mad_adec_thread_t structure. */
mad_stream_buffer( p_stream, (unsigned char*) &p_dec->buffer,
i_left + i_wanted );
p_stream->error = 0;
}
return MAD_FLOW_CONTINUE;
}
/*****************************************************************************
* libmad_header: this function is called just after the header of a frame is
* decoded
*****************************************************************************/
/*
*enum mad_flow libmad_header(void *data, struct mad_header const *p_libmad_header)
*{
* mad_adec_thread_t *p_mad_adec = (mad_adec_thread_t *) data;
*
* msg_Err( p_mad_adec->p_fifo, "libmad_header samplerate %d", p_libmad_header->samplerate);
*
* PrintFrameInfo(p_limad_mad_header)
* return MAD_FLOW_CONTINUE;
*}
*/
/*****************************************************************************
* lib_mad_filter: this function is called to filter data of a frame
* libmad_output: this function is called just after the frame is decoded
*****************************************************************************/
/* enum mad_flow libmad_filter(void *data, struct mad_stream const *p_libmad_stream, struct mad_frame *p_libmad_frame)
* {
* return MAD_FLOW_CONTINUE;
* }
*/
///*****************************************************************************
// * support routines borrowed from mpg321 (file: mad.c), which is distributed
// * under GPL license
// *
// * mpg321 was written by Joe Drew <drew@debian.org>, and based upon 'plaympeg'
// * from the smpeg sources, which was written by various people from Loki Software
// * (http://www.lokigames.com).
// *
// * It also incorporates some source from mad, written by Robert Leslie
// *****************************************************************************/
//
///* The following two routines and data structure are from the ever-brilliant
// Rob Leslie.
//*/
//
//struct audio_dither {
// mad_fixed_t error[3];
// mad_fixed_t random;
//};
//
///*
//* NAME: prng()
//* DESCRIPTION: 32-bit pseudo-random number generator
//*/
//static inline unsigned long prng(unsigned long state)
//{
// return (state * 0x0019660dL + 0x3c6ef35fL) & 0xffffffffL;
//}
//
///*
//* NAME: mpg321_s24_to_s16_pcm()
//* DESCRIPTION: generic linear sample quantize and dither routine
//*/
//static inline signed int mpg321_s24_to_s16_pcm(unsigned int bits, mad_fixed_t sample,
// struct audio_dither *dither)
//{
// unsigned int scalebits;
// mad_fixed_t output, mask, random;
//
// enum {
// MIN = -MAD_F_ONE,
// MAX = MAD_F_ONE - 1
// };
//
// /* noise shape */
// sample += dither->error[0] - dither->error[1] + dither->error[2];
//
// dither->error[2] = dither->error[1];
// dither->error[1] = dither->error[0] / 2;
//
// /* bias */
// output = sample + (1L << (MAD_F_FRACBITS + 1 - bits - 1));
//
// scalebits = MAD_F_FRACBITS + 1 - bits;
// mask = (1L << scalebits) - 1;
//
// /* dither */
// random = prng(dither->random);
// output += (random & mask) - (dither->random & mask);
//
// dither->random = random;
//
// /* clip */
// if (output > MAX) {
// output = MAX;
//
// if (sample > MAX)
// sample = MAX;
// }
// else if (output < MIN) {
// output = MIN;
//
// if (sample < MIN)
// sample = MIN;
// }
//
// /* quantize */
// output &= ~mask;
//
// /* error feedback */
// dither->error[0] = sample - output;
//
// /* scale */
// return output >> scalebits;
//}
//
///*****************************************************************************
// * s24_to_s16_pcm: Scale a 24 bit pcm sample to a 16 bit pcm sample.
// *****************************************************************************/
//static inline mad_fixed_t s24_to_s16_pcm(mad_fixed_t sample)
//{
// /* round */
// sample += (1L << (MAD_F_FRACBITS - 16));
//
// /* clip */
// if (sample >= MAD_F_ONE)
// sample = MAD_F_ONE - 1;
// else if (sample < -MAD_F_ONE)
// sample = -MAD_F_ONE;
//
// /* quantize */
// return sample >> (MAD_F_FRACBITS + 1 - 16);
//}
//
/*****************************************************************************
* libmad_ouput: this function is called just after the frame is decoded
*****************************************************************************/
//enum mad_flow libmad_output(void *data, struct mad_header const *p_libmad_header, struct mad_pcm *p_libmad_pcm)
//{
// mad_adec_thread_t *p_mad_adec= (mad_adec_thread_t *) data;
// byte_t *buffer=NULL;
// mad_fixed_t const *left_ch = p_libmad_pcm->samples[0], *right_ch = p_libmad_pcm->samples[1];
// register int nsamples = p_libmad_pcm->length;
// mad_fixed_t sample;
// static struct audio_dither dither;
//
// /* Creating the audio output fifo.
// * Assume the samplerate and nr of channels from the first decoded frame is right for the entire audio track.
// */
// if (p_mad_adec->p_aout_fifo==NULL)
// {
// p_mad_adec->p_aout_fifo = aout_CreateFifo(
// p_mad_adec->p_fifo,
// AOUT_FIFO_PCM, /* fifo type */
// 2, /*p_libmad_pcm->channels,*/ /* nr. of channels */
// p_libmad_pcm->samplerate, /* frame rate in Hz ?*/
// p_libmad_pcm->length*2, /* length of output buffer *2 channels*/
// NULL ); /* buffer */
//
// if ( p_mad_adec->p_aout_fifo == NULL )
// {
// return MAD_FLOW_BREAK;
// }
//
// msg_Dbg( p_mad_adec->p_fifo, "aout fifo created");
// }
//
// if (p_mad_adec->p_aout_fifo->i_rate != p_libmad_pcm->samplerate)
// {
// msg_Warn( p_mad_adec->p_fifo, "samplerate is changing from [%d] Hz "
// "to [%d] Hz, sample size [%d], error_code [%0x]",
// p_mad_adec->p_aout_fifo->i_rate, p_libmad_pcm->samplerate,
// p_libmad_pcm->length,
// p_mad_adec->libmad_decoder->sync->stream.error );
// p_mad_adec->p_aout_fifo->i_rate = p_libmad_pcm->samplerate;
// }
//
// if( p_mad_adec->i_current_pts )
// {
// p_mad_adec->p_aout_fifo->date[p_mad_adec->p_aout_fifo->i_end_frame]
// = p_mad_adec->i_current_pts;
// }
// else
// {
// p_mad_adec->p_aout_fifo->date[p_mad_adec->p_aout_fifo->i_end_frame]
// = LAST_MDATE;
// }
//// mad_timer_add(&p_mad_adec->libmad_timer,p_libmad_header->duration);
//
// buffer = ((byte_t *)p_mad_adec->p_aout_fifo->buffer) + (p_mad_adec->p_aout_fifo->i_end_frame * (p_libmad_pcm->length*4));
//
// while (nsamples--)
// {
// switch (p_mad_adec->audio_scaling)
// {
// case MPG321_SCALING:
// sample = mpg321_s24_to_s16_pcm(16, *left_ch++, &dither);
// break;
// case FAST_SCALING: /* intended fall through */
// default:
// sample = s24_to_s16_pcm(*left_ch++);
// break;
// }
//
// /* left audio channel */
//#ifndef WORDS_BIGENDIAN
// *buffer++ = (byte_t) (sample >> 0);
// *buffer++ = (byte_t) (sample >> 8);
//#else
// *buffer++ = (byte_t) (sample >> 8);
// *buffer++ = (byte_t) (sample >> 0);
//#endif
// if (p_libmad_pcm->channels == 2)
// {
// /* right audio channel */
// switch (p_mad_adec->audio_scaling)
// {
// case MPG321_SCALING:
// sample = mpg321_s24_to_s16_pcm(16, *right_ch++, &dither);
// break;
// case FAST_SCALING: /* intended fall through */
// default:
// sample = s24_to_s16_pcm(*right_ch++);
// break;
// }
// }
// /* else reuse left_ch */
//#ifndef WORDS_BIGENDIAN
// *buffer++ = (byte_t) (sample >> 0);
// *buffer++ = (byte_t) (sample >> 8);
//#else
// *buffer++ = (byte_t) (sample >> 8);
// *buffer++ = (byte_t) (sample >> 0);
//#endif
// }
//
// /* DEBUG */
// /*
// if (p_libmad_pcm->channels == 1) {
// msg_Dbg( p_mad_adec->p_fifo, "libmad_output channels [%d]", p_libmad_pcm->channels);
// }
// */
//
// vlc_mutex_lock (&p_mad_adec->p_aout_fifo->data_lock);
// p_mad_adec->p_aout_fifo->i_end_frame = (p_mad_adec->p_aout_fifo->i_end_frame + 1) & AOUT_FIFO_SIZE;
// vlc_cond_signal (&p_mad_adec->p_aout_fifo->data_wait);
// vlc_mutex_unlock (&p_mad_adec->p_aout_fifo->data_lock);
//
// return MAD_FLOW_CONTINUE;
//}
/*****************************************************************************
* libmad_ouput3: this function is called just after the frame is decoded
*****************************************************************************/
enum mad_flow libmad_output3(void *data, struct mad_header const *p_libmad_header, struct mad_pcm *p_libmad_pcm)
enum mad_flow libmad_output( void *p_data, struct mad_header const *p_header,
struct mad_pcm *p_pcm )
{
mad_adec_thread_t *p_mad_adec= (mad_adec_thread_t *) data;
mad_adec_thread_t * p_dec = (mad_adec_thread_t *) p_data;
aout_buffer_t * p_buffer;
mad_fixed_t const *left_ch = p_libmad_pcm->samples[0], *right_ch = p_libmad_pcm->samples[1];
register int nsamples = p_libmad_pcm->length;
mad_fixed_t sample;
/* Creating the audio output fifo.
* Assume the samplerate and nr of channels from the first decoded frame
* is right for the entire audio track.
*/
if( (p_mad_adec->p_aout_input != NULL) &&
(p_mad_adec->output_format.i_rate != p_libmad_pcm->samplerate) )
mad_fixed_t const * p_left = p_pcm->samples[0];
mad_fixed_t const * p_right = p_pcm->samples[1];
register int i_samples = p_pcm->length;
mad_fixed_t * p_samples;
/* Creating the audio output fifo. Assume the samplerate and nr of channels
* from the first decoded frame is right for the entire audio track. */
if( (p_dec->p_aout_input != NULL) &&
(p_dec->output_format.i_rate != p_pcm->samplerate) )
{
/* Parameters changed - this should not happen. */
aout_InputDelete( p_mad_adec->p_aout, p_mad_adec->p_aout_input );
p_mad_adec->p_aout_input = NULL;
aout_InputDelete( p_dec->p_aout, p_dec->p_aout_input );
p_dec->p_aout_input = NULL;
}
/* Creating the audio input if not created yet. */
if( p_mad_adec->p_aout_input == NULL )
if( p_dec->p_aout_input == NULL )
{
p_mad_adec->output_format.i_rate = p_libmad_pcm->samplerate;
/* p_mad_adec->output_format.i_channels = p_libmad_pcm->channels; */
p_mad_adec->p_aout_input = aout_InputNew( p_mad_adec->p_fifo,
&p_mad_adec->p_aout,
&p_mad_adec->output_format );
if ( p_mad_adec->p_aout_input == NULL )
p_dec->output_format.i_rate = p_pcm->samplerate;
/* p_dec->output_format.i_channels = p_pcm->channels; */
aout_DateInit( &p_dec->end_date, p_pcm->samplerate );
p_dec->p_aout_input = aout_InputNew( p_dec->p_fifo,
&p_dec->p_aout,
&p_dec->output_format );
if ( p_dec->p_aout_input == NULL )
{
p_mad_adec->p_fifo->b_error = 1;
p_dec->p_fifo->b_error = VLC_TRUE;
return MAD_FLOW_BREAK;
}
msg_Dbg( p_mad_adec->p_fifo, "aout3 input created");
}
if (p_mad_adec->output_format.i_rate != p_libmad_pcm->samplerate)
if( p_dec->i_current_pts )
{
msg_Warn( p_mad_adec->p_fifo, "samplerate is changing from [%d] Hz "
"to [%d] Hz, sample size [%d], error_code [%0x]",
p_mad_adec->output_format.i_rate, p_libmad_pcm->samplerate,
p_libmad_pcm->length,
p_mad_adec->libmad_decoder->sync->stream.error );
p_mad_adec->output_format.i_rate = p_libmad_pcm->samplerate;
}
/* Set the Presentation Time Stamp */
p_buffer = aout_BufferNew( p_mad_adec->p_aout,
p_mad_adec->p_aout_input,
(p_libmad_pcm->length*2) );
if( p_dec->i_current_pts != aout_DateGet( &p_dec->end_date ) )
{
aout_DateSet( &p_dec->end_date, p_dec->i_current_pts );
}
if ( p_buffer == NULL )
p_dec->i_current_pts = 0;
}
else if( p_dec->i_next_pts )
{
msg_Dbg( p_mad_adec->p_fifo, "allocating new buffer failed");
return MAD_FLOW_BREAK;
/* No PTS this time, but it'll be for next frame */
p_dec->i_current_pts = p_dec->i_next_pts;
p_dec->i_next_pts = 0;
}
/* Add accurate PTS to buffer. */
if ( p_mad_adec->i_current_pts )
if( !aout_DateGet( &p_dec->end_date ) )
{
p_buffer->start_date = p_mad_adec->i_current_pts;
/* No date available yet, wait for the first PTS. */
return MAD_FLOW_CONTINUE;
}
else
p_buffer = aout_BufferNew( p_dec->p_aout, p_dec->p_aout_input, i_samples );
if ( p_buffer == NULL )
{
p_buffer->start_date = LAST_MDATE;
msg_Err( p_dec->p_fifo, "allocating new buffer failed" );
return MAD_FLOW_BREAK;
}
p_mad_adec->last_date += (mtime_t)(p_libmad_pcm->length*2)
/ p_mad_adec->output_format.i_rate;
p_buffer->end_date = p_mad_adec->last_date;
p_buffer->start_date = aout_DateGet( &p_dec->end_date );
p_buffer->end_date = aout_DateIncrement( &p_dec->end_date, i_samples );
/* Interleave and keep buffers in mad_fixed_t format */
while (nsamples--)
p_samples = (mad_fixed_t *)p_buffer->p_buffer;
switch( p_pcm->channels )
{
/* left audio channel */
sample = *left_ch++;
#ifndef WORDS_BIGENDIAN
*(p_buffer->p_buffer)++ = (byte_t) (sample);
// *(p_buffer->p_buffer)++ = (byte_t) (sample >> 0);
// *(p_buffer->p_buffer)++ = (byte_t) (sample >> 8);
#else
// *(p_buffer->p_buffer)++ = (byte_t) (sample >> 8);
// *(p_buffer->p_buffer)++ = (byte_t) (sample >> 0);
#endif
/* right audio channel */
if (p_libmad_pcm->channels == 2)
case 2:
while( i_samples-- )
{
sample = *right_ch++;
} /* else reuse left audio channel */
#ifndef WORDS_BIGENDIAN
*(p_buffer->p_buffer)++ = (byte_t) (sample);
// *(p_buffer->p_buffer)++ = (byte_t) (sample >> 0);
// *(p_buffer->p_buffer)++ = (byte_t) (sample >> 8);
#else
// *(p_buffer->p_buffer)++ = (byte_t) (sample >> 8);
// *(p_buffer->p_buffer)++ = (byte_t) (sample >> 0);
#endif
*p_samples++ = *p_left++;
*p_samples++ = *p_right++;
}
break;
case 1:
while( i_samples-- )
{
*p_samples++ = *p_left;
*p_samples++ = *p_left++;
}
break;
default:
msg_Err( p_dec->p_fifo, "cannot interleave %i channels",
p_pcm->channels );
}
aout_BufferPlay( p_mad_adec->p_aout, p_mad_adec->p_aout_input, p_buffer );
aout_BufferPlay( p_dec->p_aout, p_dec->p_aout_input, p_buffer );
return MAD_FLOW_CONTINUE;
}
......@@ -494,95 +240,97 @@ enum mad_flow libmad_output3(void *data, struct mad_header const *p_libmad_heade
/*****************************************************************************
* libmad_error: this function is called when an error occurs during decoding
*****************************************************************************/
enum mad_flow libmad_error(void *data, struct mad_stream *p_libmad_stream, struct mad_frame *p_libmad_frame)
enum mad_flow libmad_error( void *data, struct mad_stream *p_libmad_stream,
struct mad_frame *p_libmad_frame )
{
mad_adec_thread_t *p_dec = (mad_adec_thread_t *) data;
enum mad_flow result = MAD_FLOW_CONTINUE;
switch (p_libmad_stream->error)
{
case MAD_ERROR_BUFLEN: /* input buffer too small (or EOF) */
//X msg_Err("libmad error: input buffer too small (or EOF)");
msg_Err( p_dec->p_fifo, "input buffer too small (or EOF)" );
result = MAD_FLOW_CONTINUE;
break;
case MAD_ERROR_BUFPTR: /* invalid (null) buffer pointer */
//X msg_Err("libmad error: invalid (null) buffer pointer");
msg_Err( p_dec->p_fifo, "invalid (null) buffer pointer" );
result = MAD_FLOW_STOP;
break;
case MAD_ERROR_NOMEM: /* not enough memory */
//X msg_Err("libmad error: invalid (null) buffer pointer");
msg_Err( p_dec->p_fifo, "invalid (null) buffer pointer" );
result = MAD_FLOW_STOP;
break;
case MAD_ERROR_LOSTSYNC: /* lost synchronization */
//X msg_Err("libmad error: lost synchronization");
msg_Err( p_dec->p_fifo, "lost synchronization" );
mad_stream_sync(p_libmad_stream);
result = MAD_FLOW_CONTINUE;
break;
case MAD_ERROR_BADLAYER: /* reserved header layer value */
//X msg_Err("libmad error: reserved header layer value");
msg_Err( p_dec->p_fifo, "reserved header layer value" );
result = MAD_FLOW_CONTINUE;
break;
case MAD_ERROR_BADBITRATE: /* forbidden bitrate value */
//X msg_Err("libmad error: forbidden bitrate value");
msg_Err( p_dec->p_fifo, "forbidden bitrate value" );
result = MAD_FLOW_CONTINUE;
break;
case MAD_ERROR_BADSAMPLERATE: /* reserved sample frequency value */
//X msg_Err("libmad error: reserved sample frequency value");
msg_Err( p_dec->p_fifo, "reserved sample frequency value" );
result = MAD_FLOW_CONTINUE;
break;
case MAD_ERROR_BADEMPHASIS: /* reserved emphasis value */
//X msg_Err("libmad error: reserverd emphasis value");
msg_Err( p_dec->p_fifo, "reserverd emphasis value" );
result = MAD_FLOW_CONTINUE;
break;
case MAD_ERROR_BADCRC: /* CRC check failed */
//X msg_Err("libmad error: CRC check failed");
msg_Err( p_dec->p_fifo, "CRC check failed" );
result = MAD_FLOW_CONTINUE;
break;
case MAD_ERROR_BADBITALLOC: /* forbidden bit allocation value */
//X msg_Err("libmad error: forbidden bit allocation value");
msg_Err( p_dec->p_fifo, "forbidden bit allocation value" );
result = MAD_FLOW_IGNORE;
break;
case MAD_ERROR_BADSCALEFACTOR:/* bad scalefactor index */
//X msg_Err("libmad error: bad scalefactor index");
msg_Err( p_dec->p_fifo, "bad scalefactor index" );
result = MAD_FLOW_CONTINUE;
break;
case MAD_ERROR_BADFRAMELEN: /* bad frame length */
//X msg_Err("libmad error: bad frame length");
msg_Err( p_dec->p_fifo, "bad frame length" );
result = MAD_FLOW_CONTINUE;
break;
case MAD_ERROR_BADBIGVALUES: /* bad big_values count */
//X msg_Err("libmad error: bad big values count");
msg_Err( p_dec->p_fifo, "bad big values count" );
result = MAD_FLOW_IGNORE;
break;
case MAD_ERROR_BADBLOCKTYPE: /* reserved block_type */
//X msg_Err("libmad error: reserverd block_type");
msg_Err( p_dec->p_fifo, "reserverd block_type" );
result = MAD_FLOW_IGNORE;
break;
case MAD_ERROR_BADSCFSI: /* bad scalefactor selection info */
//X msg_Err("libmad error: bad scalefactor selection info");
msg_Err( p_dec->p_fifo, "bad scalefactor selection info" );
result = MAD_FLOW_CONTINUE;
break;
case MAD_ERROR_BADDATAPTR: /* bad main_data_begin pointer */
//X msg_Err("libmad error: bad main_data_begin pointer");
msg_Err( p_dec->p_fifo, "bad main_data_begin pointer" );
result = MAD_FLOW_STOP;
break;
case MAD_ERROR_BADPART3LEN: /* bad audio data length */
//X msg_Err("libmad error: bad audio data length");
msg_Err( p_dec->p_fifo, "bad audio data length" );
result = MAD_FLOW_IGNORE;
break;
case MAD_ERROR_BADHUFFTABLE: /* bad Huffman table select */
//X msg_Err("libmad error: bad Huffman table select");
msg_Err( p_dec->p_fifo, "bad Huffman table select" );
result = MAD_FLOW_IGNORE;
break;
case MAD_ERROR_BADHUFFDATA: /* Huffman data overrun */
//X msg_Err("libmad error: Huffman data overrun");
msg_Err( p_dec->p_fifo, "Huffman data overrun" );
result = MAD_FLOW_IGNORE;
break;
case MAD_ERROR_BADSTEREO: /* incompatible block_type for JS */
//X msg_Err("libmad error: incompatible block_type for JS");
msg_Err( p_dec->p_fifo, "incompatible block_type for JS" );
result = MAD_FLOW_IGNORE;
break;
default:
//X msg_Err("libmad error: unknown error occured stopping decoder");
msg_Err( p_dec->p_fifo, "unknown error occured stopping decoder" );
result = MAD_FLOW_STOP;
break;
}
......@@ -611,7 +359,7 @@ static void PrintFrameInfo(struct mad_header *Header)
*Mode,
*Emphasis;
/* Convert the layer number to it's printed representation. */
/* Convert the layer number to its printed representation. */
switch(Header->layer)
{
case MAD_LAYER_I:
......@@ -628,7 +376,7 @@ static void PrintFrameInfo(struct mad_header *Header)
break;
}
/* Convert the audio mode to it's printed representation. */
/* Convert the audio mode to its printed representation. */
switch(Header->mode)
{
case MAD_MODE_SINGLE_CHANNEL:
......@@ -648,7 +396,7 @@ static void PrintFrameInfo(struct mad_header *Header)
break;
}
/* Convert the emphasis to it's printed representation. */
/* Convert the emphasis to its printed representation. */
switch(Header->emphasis)
{
case MAD_EMPHASIS_NONE:
......
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