/*****************************************************************************
- * decoder.c: AAC decoder using libfaad2
+ * faad.c: AAC decoder using libfaad2
*****************************************************************************
- * Copyright (C) 2001, 2003 the VideoLAN team
+ * Copyright (C) 2001, 2003 VLC authors and VideoLAN
* $Id$
*
* Authors: Laurent Aimar <fenrir@via.ecp.fr>
* Gildas Bazin <gbazin@videolan.org>
*
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation; either version 2.1 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Lesser General Public License for more details.
*
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with this program; if not, write to the Free Software Foundation,
+ * Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
*****************************************************************************/
+/*****************************************************************************
+ * NOTA BENE: this module requires the linking against a library which is
+ * known to require licensing under the GNU General Public License version 2
+ * (or later). Therefore, the result of compiling this module will normally
+ * be subject to the terms of that later license.
+ *****************************************************************************/
+
+
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <vlc_common.h>
#include <vlc_plugin.h>
#include <vlc_input.h>
-#include <vlc_aout.h>
#include <vlc_codec.h>
#include <vlc_cpu.h>
/****************************************************************************
* Local prototypes
****************************************************************************/
-static aout_buffer_t *DecodeBlock( decoder_t *, block_t ** );
+static block_t *DecodeBlock( decoder_t *, block_t ** );
static void DoReordering( uint32_t *, uint32_t *, int, int, uint32_t * );
#define MAX_CHANNEL_POSITIONS 9
date_Set( &p_sys->date, 0 );
p_dec->fmt_out.i_cat = AUDIO_ES;
- if (HAVE_FPU)
- p_dec->fmt_out.i_codec = VLC_CODEC_FL32;
- else
- p_dec->fmt_out.i_codec = VLC_CODEC_S16N;
+ 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 =
/* Set the faad config */
cfg = faacDecGetCurrentConfiguration( p_sys->hfaad );
- if (HAVE_FPU)
- cfg->outputFormat = FAAD_FMT_FLOAT;
- else
- cfg->outputFormat = FAAD_FMT_16BIT;
+ cfg->outputFormat = HAVE_FPU ? FAAD_FMT_FLOAT : FAAD_FMT_16BIT;
faacDecSetConfiguration( p_sys->hfaad, cfg );
/* buffer */
/*****************************************************************************
* DecodeBlock:
*****************************************************************************/
-static aout_buffer_t *DecodeBlock( decoder_t *p_dec, block_t **pp_block )
+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( p_block->i_buffer > 0 )
{
- vlc_memcpy( &p_sys->p_buffer[p_sys->i_buffer],
+ 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;
{
void *samples;
faacDecFrameInfo frame;
- aout_buffer_t *p_out;
- int i, j;
+ block_t *p_out;
samples = faacDecDecode( p_sys->hfaad, &frame,
p_sys->p_buffer, p_sys->i_buffer );
{
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 );
p_dec->fmt_out.audio.i_rate = frame.samplerate;
p_dec->fmt_out.audio.i_channels = frame.channels;
- p_dec->fmt_out.audio.i_physical_channels
- = p_dec->fmt_out.audio.i_original_channels
- = pi_channels_guessed[frame.channels];
/* Adjust stream info when dealing with SBR/PS */
bool b_sbr = (frame.sbr == 1) || (frame.sbr == 2);
/* Convert frame.channel_position to our own channel values */
p_dec->fmt_out.audio.i_physical_channels = 0;
- for( i = 0; i < frame.channels; i++ )
+ 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++ )
else
p_dec->fmt_out.audio.i_physical_channels |= pi_channels_out[j];
}
- p_dec->fmt_out.audio.i_original_channels =
- p_dec->fmt_out.audio.i_physical_channels;
-
- p_out = decoder_NewAudioBuffer(p_dec, frame.samples/frame.channels);
+ 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;
p_out->i_pts = date_Get( &p_sys->date );
p_out->i_length = date_Increment( &p_sys->date,
- frame.samples / frame.channels )
+ frame.samples / nbChannels )
- p_out->i_pts;
DoReordering( (uint32_t *)p_out->p_buffer, samples,
- frame.samples / frame.channels, frame.channels,
+ frame.samples / nbChannels, nbChannels,
p_sys->pi_channel_positions );
p_sys->i_buffer -= frame.bytesconsumed;
static void DoReordering( uint32_t *p_out, uint32_t *p_in, int i_samples,
int i_nb_channels, uint32_t *pi_chan_positions )
{
- int pi_chan_table[MAX_CHANNEL_POSITIONS];
+ int pi_chan_table[MAX_CHANNEL_POSITIONS] = {0};
int i, j, k;
/* Find the channels mapping */
projects / vlc / blobdiff