Skip to content

V
vlc-1.1
Commit a4fcb5e2 authored by Jean-Paul Saman's avatar Jean-Paul Saman
Browse files
Options

Added mad plugin downscale config option for selecting downscaling routine.... 

Added mad plugin downscale config option for selecting downscaling routine. Choices are: fast, mpg321. Default the fast version is used. The downscale option mpg321 uses an algorithm that adapts to dither errors. This routine is borrowed from MPG321 code.
Mono audio streams get now duplicated on the right channel instead of silencing the right channel (untested).
parent f2c5927d
Hide whitespace changes
Inline Side-by-side
Showing with 138 additions and 39 deletions
plugins/mad/mad_adec.c
View file @ a4fcb5e2
......@@ -60,7 +60,14 @@ void _M( adec_getfunctions )( function_list_t * p_function_list )
/*****************************************************************************
* Build configuration tree.
*****************************************************************************/
#define DOWNSCALE_TEXT N_("Mad audio downscale routine (fast,mp321)")
#define DOWNSCALE_LONGTEXT N_( \
"Specify the mad audio downscale routine you want to use.\nBy default mad plugins will " \
"use the fastest routine.")
MODULE_CONFIG_START
ADD_CATEGORY_HINT( N_("Miscellaneous"), NULL )
ADD_STRING ( "downscale", "fast", NULL, DOWNSCALE_TEXT, DOWNSCALE_LONGTEXT )
MODULE_CONFIG_STOP
MODULE_INIT_START
......@@ -156,11 +163,33 @@ static int decoder_Run ( decoder_config_t * p_config )
static int InitThread( mad_adec_thread_t * p_mad_adec )
{
decoder_fifo_t * p_fifo = p_mad_adec->p_fifo;
char *psz_downscale = NULL;
/*
* Properties of audio for libmad
*/
/* Look what scaling method was requested by the user */
psz_downscale = config_GetPszVariable( "downscale" );
if ( strncmp(psz_downscale,"fast",4)==0 )
{
p_mad_adec->audio_scaling = FAST_SCALING;
intf_ErrMsg( "mad_adec debug: downscale fast selected" );
}
else if ( strncmp(psz_downscale,"mpg321",7)==0 )
{
p_mad_adec->audio_scaling = MPG321_SCALING;
intf_ErrMsg( "mad_adec debug: downscale mpg321 selected" );
}
else
{
p_mad_adec->audio_scaling = FAST_SCALING;
intf_ErrMsg( "mad_adec debug: 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)
......@@ -172,7 +201,7 @@ static int InitThread( mad_adec_thread_t * p_mad_adec )
p_mad_adec->i_current_pts = p_mad_adec->i_next_pts = 0;
mad_decoder_init( p_mad_adec->libmad_decoder,
p_mad_adec, /* vlc's thread structure and p_fifo playbuffer */
p_mad_adec, /* vlc's thread structure and p_fifo playbuffer */
libmad_input, /* input_func */
0, /* header_func */
0, /* filter */
......
plugins/mad/mad_adec.h
View file @ a4fcb5e2
......@@ -22,6 +22,8 @@
* mad_adec_thread_t : mad decoder thread descriptor
*****************************************************************************/
enum mad_scaling { FAST_SCALING, MPG321_SCALING };
typedef struct mad_adec_thread_s
{
/*
......@@ -50,6 +52,7 @@ typedef struct mad_adec_thread_s
* Output properties
*/
aout_fifo_t * p_aout_fifo; /* stores the decompressed audio frames */
enum mad_scaling audio_scaling;
} mad_adec_thread_t;
......
plugins/mad/mad_libmad.c
View file @ a4fcb5e2
......@@ -38,6 +38,8 @@
#include "mad_adec.h"
#include "mad_libmad.h"
static void PrintFrameInfo(struct mad_header *Header);
/*****************************************************************************
* libmad_input: this function is called by libmad when the input buffer needs
* to be filled.
......@@ -153,9 +155,7 @@ enum mad_flow libmad_input(void *data, struct mad_stream *p_libmad_stream)
*
* intf_ErrMsg( "mad_adec: libmad_header samplerate %d", p_libmad_header->samplerate);
*
* p_mad_adec->p_aout_fifo->i_rate = p_libmad_header->samplerate;
* // mad_timer_add(&p_mad_adec->libmad_timer,p_libmad_header->duration);
*
* PrintFrameInfo(p_limad_mad_header)
* return MAD_FLOW_CONTINUE;
*}
*/
......@@ -169,8 +169,6 @@ enum mad_flow libmad_input(void *data, struct mad_stream *p_libmad_stream)
* }
*/
//#define MPG321_ROUTINES 1
#ifdef MPG321_ROUTINES
/*****************************************************************************
* support routines borrowed from mpg321 (file: mad.c), which is distributed
* under GPL license
......@@ -201,10 +199,10 @@ static __inline__ unsigned long prng(unsigned long state)
}
/*
* NAME: audio_linear_dither()
* NAME: mpg321_s24_to_s16_pcm()
* DESCRIPTION: generic linear sample quantize and dither routine
*/
static __inline__ signed int audio_linear_dither(unsigned int bits, mad_fixed_t sample,
static __inline__ signed int mpg321_s24_to_s16_pcm(unsigned int bits, mad_fixed_t sample,
struct audio_dither *dither)
{
unsigned int scalebits;
......@@ -256,7 +254,6 @@ static __inline__ signed int audio_linear_dither(unsigned int bits, mad_fixed_t
/* scale */
return output >> scalebits;
}
#else
/*****************************************************************************
* s24_to_s16_pcm: Scale a 24 bit pcm sample to a 16 bit pcm sample.
......@@ -275,7 +272,6 @@ static __inline__ mad_fixed_t s24_to_s16_pcm(mad_fixed_t sample)
/* quantize */
return sample >> (MAD_F_FRACBITS + 1 - 16);
}
#endif
/*****************************************************************************
* libmad_ouput: this function is called just after the frame is decoded
......@@ -284,13 +280,10 @@ enum mad_flow libmad_output(void *data, struct mad_header const *p_libmad_header
{
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;
#ifdef MPG321_ROUTINES
static struct audio_dither dither;
#endif
/* Creating the audio output fifo.
* Assume the samplerate and nr of channels from the first decoded frame is right for the entire audio track.
......@@ -336,12 +329,18 @@ enum mad_flow libmad_output(void *data, struct mad_header const *p_libmad_header
while (nsamples--)
{
#ifdef MPG321_ROUTINES
sample = audio_linear_dither(16, *left_ch++, &dither);
#else
sample = s24_to_s16_pcm(*left_ch++);
#endif
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);
......@@ -352,29 +351,27 @@ enum mad_flow libmad_output(void *data, struct mad_header const *p_libmad_header
if (p_libmad_pcm->channels == 2)
{
/* right audio channel */
#ifdef MPG321_ROUTINES
sample = audio_linear_dither(16, *right_ch++, &dither);
#else
sample = s24_to_s16_pcm(*right_ch++);
#endif
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);
*buffer++ = (byte_t) (sample >> 0);
*buffer++ = (byte_t) (sample >> 8);
#else
*buffer++ = (byte_t) (sample >> 8);
*buffer++ = (byte_t) (sample >> 0);
*buffer++ = (byte_t) (sample >> 8);
*buffer++ = (byte_t) (sample >> 0);
#endif
}
else {
/* Somethimes a single channel frame is found, while the rest of the movie are
* stereo channel frames. How to deal with this ??
* One solution is to silence the second channel.
*/
*buffer++ = (byte_t) (0);
*buffer++ = (byte_t) (0);
}
}
/* DEBUG */
if (p_libmad_pcm->channels == 1) {
intf_ErrMsg( "mad debug: libmad_output channels [%d]", p_libmad_pcm->channels);
......@@ -498,3 +495,73 @@ enum mad_flow libmad_error(void *data, struct mad_stream *p_libmad_stream, struc
*/
/****************************************************************************
* Print human readable informations about an audio MPEG frame. *
****************************************************************************/
static void PrintFrameInfo(struct mad_header *Header)
{
const char *Layer,
*Mode,
*Emphasis;
/* Convert the layer number to it's printed representation. */
switch(Header->layer)
{
case MAD_LAYER_I:
Layer="I";
break;
case MAD_LAYER_II:
Layer="II";
break;
case MAD_LAYER_III:
Layer="III";
break;
default:
Layer="(unexpected layer value)";
break;
}
/* Convert the audio mode to it's printed representation. */
switch(Header->mode)
{
case MAD_MODE_SINGLE_CHANNEL:
Mode="single channel";
break;
case MAD_MODE_DUAL_CHANNEL:
Mode="dual channel";
break;
case MAD_MODE_JOINT_STEREO:
Mode="joint (MS/intensity) stereo";
break;
case MAD_MODE_STEREO:
Mode="normal LR stereo";
break;
default:
Mode="(unexpected mode value)";
break;
}
/* Convert the emphasis to it's printed representation. */
switch(Header->emphasis)
{
case MAD_EMPHASIS_NONE:
Emphasis="no";
break;
case MAD_EMPHASIS_50_15_US:
Emphasis="50/15 us";
break;
case MAD_EMPHASIS_CCITT_J_17:
Emphasis="CCITT J.17";
break;
default:
Emphasis="(unexpected emphasis value)";
break;
}
intf_ErrMsg("statistics: %lu kb/s audio mpeg layer %s stream %s crc, "
"%s with %s emphasis at %d Hz sample rate\n",
Header->bitrate,Layer,
Header->flags&MAD_FLAG_PROTECTION?"with":"without",
Mode,Emphasis,Header->samplerate);
}
Markdown is supported
0% or
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment