]> git.sesse.net Git - vlc/blobdiff - modules/codec/avcodec/audio.c
Simplify FLAC extradata (streaminfo) parsing
[vlc] / modules / codec / avcodec / audio.c
index bb680e384200f8b86d0d2c9a58d1bfda4619c2ee..b2f70ffe9341aea04db3a84a6563d867562de25a 100644 (file)
@@ -1,25 +1,25 @@
 /*****************************************************************************
- * audio.c: audio decoder using ffmpeg library
+ * audio.c: audio decoder using libavcodec library
  *****************************************************************************
- * Copyright (C) 1999-2003 the VideoLAN team
+ * Copyright (C) 1999-2003 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.
  *****************************************************************************/
 
 /*****************************************************************************
 # include "config.h"
 #endif
 
+#include <assert.h>
+
 #include <vlc_common.h>
 #include <vlc_aout.h>
 #include <vlc_codec.h>
 #include <vlc_avcodec.h>
 
-/* ffmpeg header */
-#ifdef HAVE_LIBAVCODEC_AVCODEC_H
-#   include <libavcodec/avcodec.h>
-#elif defined(HAVE_FFMPEG_AVCODEC_H)
-#   include <ffmpeg/avcodec.h>
-#else
-#   include <avcodec.h>
-#endif
+#include <libavcodec/avcodec.h>
+#include <libavutil/mem.h>
+
+#include <libavutil/audioconvert.h>
 
 #include "avcodec.h"
 
  *****************************************************************************/
 struct decoder_sys_t
 {
-    FFMPEG_COMMON_MEMBERS
-
-    /* Temporary buffer for libavcodec */
-    int     i_output_max;
-    uint8_t *p_output;
+    AVCODEC_COMMON_MEMBERS
 
     /*
      * Output properties
      */
     audio_sample_format_t aout_format;
-    audio_date_t          end_date;
-
-    /*
-     *
-     */
-    uint8_t *p_samples;
-    int     i_samples;
+    date_t                end_date;
 
     /* */
     int     i_reject_count;
@@ -78,54 +66,21 @@ struct decoder_sys_t
     int64_t i_previous_layout;
 };
 
+#define BLOCK_FLAG_PRIVATE_REALLOCATED (1 << BLOCK_FLAG_PRIVATE_SHIFT)
+
 static void SetupOutputFormat( decoder_t *p_dec, bool b_trust );
+static int GetAudioBuf( struct AVCodecContext *, AVFrame * );
 
-/*****************************************************************************
- * InitAudioDec: initialize audio decoder
- *****************************************************************************
- * The ffmpeg codec will be opened, some memory allocated.
- *****************************************************************************/
-int InitAudioDec( decoder_t *p_dec, AVCodecContext *p_context,
-                      AVCodec *p_codec, int i_codec_id, const char *psz_namecodec )
+static void InitDecoderConfig( decoder_t *p_dec, AVCodecContext *p_context )
 {
-    decoder_sys_t *p_sys;
-
-    /* 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;
-    }
-
-    p_sys->p_context = p_context;
-    p_sys->p_codec = p_codec;
-    p_sys->i_codec_id = i_codec_id;
-    p_sys->psz_namecodec = psz_namecodec;
-    p_sys->b_delayed_open = false;
-
-    /* ***** Fill p_context with init values ***** */
-    p_sys->p_context->sample_rate = p_dec->fmt_in.audio.i_rate;
-    p_sys->p_context->channels = p_dec->fmt_in.audio.i_channels;
-
-    p_sys->p_context->block_align = p_dec->fmt_in.audio.i_blockalign;
-    p_sys->p_context->bit_rate = p_dec->fmt_in.i_bitrate;
-#if LIBAVCODEC_VERSION_INT < ((52<<16)+(0<<8)+0)
-    p_sys->p_context->bits_per_sample = p_dec->fmt_in.audio.i_bitspersample;
-#else
-    p_sys->p_context->bits_per_coded_sample = p_dec->fmt_in.audio.i_bitspersample;
-#endif
-
     if( p_dec->fmt_in.i_extra > 0 )
     {
         const uint8_t * const p_src = p_dec->fmt_in.p_extra;
-        int i_offset;
-        int i_size;
 
-        if( p_dec->fmt_in.i_codec == VLC_CODEC_FLAC )
-        {
-            i_offset = 8;
-            i_size = p_dec->fmt_in.i_extra - 8;
-        }
-        else if( p_dec->fmt_in.i_codec == VLC_CODEC_ALAC )
+        int i_offset = 0;
+        int i_size = p_dec->fmt_in.i_extra;
+
+        if( p_dec->fmt_in.i_codec == VLC_CODEC_ALAC )
         {
             static const uint8_t p_pattern[] = { 0, 0, 0, 36, 'a', 'l', 'a', 'c' };
             /* Find alac atom XXX it is a bit ugly */
@@ -138,21 +93,16 @@ int InitAudioDec( decoder_t *p_dec, AVCodecContext *p_context,
             if( i_size < 36 )
                 i_size = 0;
         }
-        else
-        {
-            i_offset = 0;
-            i_size = p_dec->fmt_in.i_extra;
-        }
 
         if( i_size > 0 )
         {
-            p_sys->p_context->extradata =
+            p_context->extradata =
                 malloc( i_size + FF_INPUT_BUFFER_PADDING_SIZE );
-            if( p_sys->p_context->extradata )
+            if( p_context->extradata )
             {
-                uint8_t *p_dst = p_sys->p_context->extradata;
+                uint8_t *p_dst = p_context->extradata;
 
-                p_sys->p_context->extradata_size = i_size;
+                p_context->extradata_size = i_size;
 
                 memcpy( &p_dst[0],            &p_src[i_offset], i_size );
                 memset( &p_dst[i_size], 0, FF_INPUT_BUFFER_PADDING_SIZE );
@@ -161,16 +111,42 @@ int InitAudioDec( decoder_t *p_dec, AVCodecContext *p_context,
     }
     else
     {
-        p_sys->p_context->extradata_size = 0;
-        p_sys->p_context->extradata = NULL;
+        p_context->extradata_size = 0;
+        p_context->extradata = NULL;
+    }
+}
+
+/*****************************************************************************
+ * InitAudioDec: initialize audio decoder
+ *****************************************************************************
+ * The avcodec codec will be opened, some memory allocated.
+ *****************************************************************************/
+int InitAudioDec( decoder_t *p_dec, AVCodecContext *p_context,
+                      AVCodec *p_codec, int i_codec_id, const char *psz_namecodec )
+{
+    decoder_sys_t *p_sys;
+
+    /* 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;
     }
 
+    p_codec->type = AVMEDIA_TYPE_AUDIO;
+    p_context->codec_type = AVMEDIA_TYPE_AUDIO;
+    p_context->codec_id = i_codec_id;
+    p_context->get_buffer = GetAudioBuf;
+    p_sys->p_context = p_context;
+    p_sys->p_codec = p_codec;
+    p_sys->i_codec_id = i_codec_id;
+    p_sys->psz_namecodec = psz_namecodec;
+    p_sys->b_delayed_open = true;
+
+    // Initialize decoder extradata
+    InitDecoderConfig( p_dec, p_context);
+
     /* ***** Open the codec ***** */
-    int ret;
-    vlc_avcodec_lock();
-    ret = avcodec_open( p_sys->p_context, p_sys->p_codec );
-    vlc_avcodec_unlock();
-    if( ret < 0 )
+    if( ffmpeg_OpenCodec( p_dec ) < 0 )
     {
         msg_Err( p_dec, "cannot open codec (%s)", p_sys->psz_namecodec );
         free( p_sys->p_context->extradata );
@@ -178,340 +154,306 @@ int InitAudioDec( decoder_t *p_dec, AVCodecContext *p_context,
         return VLC_EGENERIC;
     }
 
-    msg_Dbg( p_dec, "ffmpeg codec (%s) started", p_sys->psz_namecodec );
-
-    switch( i_codec_id )
-    {
-#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT( 51, 16, 0 )
-    case CODEC_ID_WAVPACK:
-        p_sys->i_output_max = 8 * sizeof(int32_t) * 131072;
-        break;
-#endif
-    case CODEC_ID_FLAC:
-        p_sys->i_output_max = 8 * sizeof(int32_t) * 65535;
-        break;
-    default:
-        p_sys->i_output_max = 0;
-        break;
-    }
-    if( p_sys->i_output_max < AVCODEC_MAX_AUDIO_FRAME_SIZE )
-        p_sys->i_output_max = AVCODEC_MAX_AUDIO_FRAME_SIZE;
-    msg_Dbg( p_dec, "Using %d bytes output buffer", p_sys->i_output_max );
-    p_sys->p_output = malloc( p_sys->i_output_max );
-
-    p_sys->p_samples = NULL;
-    p_sys->i_samples = 0;
     p_sys->i_reject_count = 0;
     p_sys->b_extract = false;
     p_sys->i_previous_channels = 0;
     p_sys->i_previous_layout = 0;
 
-    aout_DateSet( &p_sys->end_date, 0 );
-    if( p_dec->fmt_in.audio.i_rate )
-        aout_DateInit( &p_sys->end_date, p_dec->fmt_in.audio.i_rate );
-
     /* */
     p_dec->fmt_out.i_cat = AUDIO_ES;
-    /* Try to set as much informations as possible but do not trust it */
+    /* Try to set as much information as possible but do not trust it */
     SetupOutputFormat( p_dec, false );
 
+    date_Set( &p_sys->end_date, 0 );
+    if( p_dec->fmt_out.audio.i_rate )
+        date_Init( &p_sys->end_date, p_dec->fmt_out.audio.i_rate, 1 );
+    else if( p_dec->fmt_in.audio.i_rate )
+        date_Init( &p_sys->end_date, p_dec->fmt_in.audio.i_rate, 1 );
+
     return VLC_SUCCESS;
 }
 
-/*****************************************************************************
- * SplitBuffer: Needed because aout really doesn't like big audio chunk and
- * wma produces easily > 30000 samples...
- *****************************************************************************/
-static aout_buffer_t *SplitBuffer( decoder_t *p_dec )
+/**
+ * Allocates decoded audio buffer for libavcodec to use.
+ */
+static int GetAudioBuf( AVCodecContext *ctx, AVFrame *buf )
 {
-    decoder_sys_t *p_sys = p_dec->p_sys;
-    int i_samples = __MIN( p_sys->i_samples, 4096 );
-    aout_buffer_t *p_buffer;
-
-    if( i_samples == 0 ) return NULL;
-
-    if( ( p_buffer = decoder_NewAudioBuffer( p_dec, i_samples ) ) == NULL )
-        return NULL;
-
-    p_buffer->start_date = aout_DateGet( &p_sys->end_date );
-    p_buffer->end_date = aout_DateIncrement( &p_sys->end_date, i_samples );
-
-    if( p_sys->b_extract )
-        aout_ChannelExtract( p_buffer->p_buffer, p_dec->fmt_out.audio.i_channels,
-                             p_sys->p_samples, p_sys->p_context->channels, i_samples,
-                             p_sys->pi_extraction, p_dec->fmt_out.audio.i_bitspersample );
+    block_t *block;
+    bool planar = av_sample_fmt_is_planar( ctx->sample_fmt );
+    unsigned channels = planar ? 1 : ctx->channels;
+    unsigned planes = planar ? ctx->channels : 1;
+
+    int bytes = av_samples_get_buffer_size( &buf->linesize[0], channels,
+                                            buf->nb_samples, ctx->sample_fmt,
+                                            16 );
+    assert( bytes >= 0 );
+    block = block_Alloc( bytes * planes );
+    if( unlikely(block == NULL) )
+        return AVERROR(ENOMEM);
+
+    block->i_nb_samples = buf->nb_samples;
+    buf->opaque = block;
+
+    if( planes > AV_NUM_DATA_POINTERS )
+    {
+        uint8_t **ext = malloc( sizeof( *ext ) * planes );
+        if( unlikely(ext == NULL) )
+        {
+            block_Release( block );
+            return AVERROR(ENOMEM);
+        }
+        buf->extended_data = ext;
+    }
     else
-        memcpy( p_buffer->p_buffer, p_sys->p_samples, p_buffer->i_nb_bytes );
+        buf->extended_data = buf->data;
 
-    p_sys->p_samples += p_buffer->i_nb_bytes;
-    p_sys->i_samples -= i_samples;
+    uint8_t *buffer = block->p_buffer;
+    for( unsigned i = 0; i < planes; i++ )
+    {
+        buf->linesize[i] = buf->linesize[0];
+        buf->extended_data[i] = buffer;
+        buffer += bytes;
+    }
 
-    return p_buffer;
+    return 0;
 }
 
 /*****************************************************************************
  * DecodeAudio: Called to decode one frame
  *****************************************************************************/
-aout_buffer_t * DecodeAudio ( decoder_t *p_dec, block_t **pp_block )
+block_t * DecodeAudio ( decoder_t *p_dec, block_t **pp_block )
 {
     decoder_sys_t *p_sys = p_dec->p_sys;
-    int i_used, i_output;
-    aout_buffer_t *p_buffer;
-    block_t *p_block;
+    AVCodecContext *ctx = p_sys->p_context;
+
+    if( !pp_block || !*pp_block )
+        return NULL;
+
+    block_t *p_block = *pp_block;
 
-    if( !pp_block || !*pp_block ) return NULL;
+    if( !ctx->extradata_size && p_dec->fmt_in.i_extra && p_sys->b_delayed_open)
+    {
+        InitDecoderConfig( p_dec, ctx );
+        if( ffmpeg_OpenCodec( p_dec ) )
+            msg_Err( p_dec, "Cannot open decoder %s", p_sys->psz_namecodec );
+    }
 
-    p_block = *pp_block;
+    if( p_sys->b_delayed_open )
+        goto end;
 
     if( p_block->i_flags & (BLOCK_FLAG_DISCONTINUITY|BLOCK_FLAG_CORRUPTED) )
     {
-        block_Release( p_block );
-        avcodec_flush_buffers( p_sys->p_context );
-        p_sys->i_samples = 0;
-        aout_DateSet( &p_sys->end_date, 0 );
+        avcodec_flush_buffers( ctx );
+        date_Set( &p_sys->end_date, 0 );
 
-        if( p_sys->i_codec_id == CODEC_ID_MP2 || p_sys->i_codec_id == CODEC_ID_MP3 )
+        if( p_sys->i_codec_id == AV_CODEC_ID_MP2 || p_sys->i_codec_id == AV_CODEC_ID_MP3 )
             p_sys->i_reject_count = 3;
-        return NULL;
-    }
 
-    if( p_sys->i_samples > 0 )
-    {
-        /* More data */
-        p_buffer = SplitBuffer( p_dec );
-        if( !p_buffer ) block_Release( p_block );
-        return p_buffer;
+        goto end;
     }
 
-    if( !aout_DateGet( &p_sys->end_date ) && !p_block->i_pts )
-    {
-        /* We've just started the stream, wait for the first PTS. */
-        block_Release( p_block );
-        return NULL;
-    }
+    /* We've just started the stream, wait for the first PTS. */
+    if( !date_Get( &p_sys->end_date ) && p_block->i_pts <= VLC_TS_INVALID )
+        goto end;
 
     if( p_block->i_buffer <= 0 )
-    {
-        block_Release( p_block );
-        return NULL;
-    }
+        goto end;
 
-    i_output = __MAX( p_block->i_buffer, p_sys->i_output_max );
-    if( i_output > p_sys->i_output_max )
+    if( (p_block->i_flags & BLOCK_FLAG_PRIVATE_REALLOCATED) == 0 )
     {
-        /* Grow output buffer if necessary (eg. for PCM data) */
-        p_sys->p_output = realloc( p_sys->p_output, i_output );
+        p_block = block_Realloc( p_block, 0, p_block->i_buffer + FF_INPUT_BUFFER_PADDING_SIZE );
+        if( !p_block )
+            return NULL;
+        p_block->i_buffer -= FF_INPUT_BUFFER_PADDING_SIZE;
+        memset( &p_block->p_buffer[p_block->i_buffer], 0, FF_INPUT_BUFFER_PADDING_SIZE );
+
+        p_block->i_flags |= BLOCK_FLAG_PRIVATE_REALLOCATED;
     }
 
-    *pp_block = p_block = block_Realloc( p_block, 0, p_block->i_buffer + FF_INPUT_BUFFER_PADDING_SIZE );
-    if( !p_block )
-        return NULL;
-    p_block->i_buffer -= FF_INPUT_BUFFER_PADDING_SIZE;
-    memset( &p_block->p_buffer[p_block->i_buffer], 0, FF_INPUT_BUFFER_PADDING_SIZE );
-
-#if LIBAVCODEC_VERSION_INT >= ((52<<16)+(0<<8)+0)
-    i_used = avcodec_decode_audio2( p_sys->p_context,
-                                   (int16_t*)p_sys->p_output, &i_output,
-                                   p_block->p_buffer, p_block->i_buffer );
-#else
-    i_used = avcodec_decode_audio( p_sys->p_context,
-                                   (int16_t*)p_sys->p_output, &i_output,
-                                   p_block->p_buffer, p_block->i_buffer );
-#endif
+    AVFrame frame;
+    memset( &frame, 0, sizeof( frame ) );
 
-    if( i_used < 0 || i_output < 0 )
+    for( int got_frame = 0; !got_frame; )
     {
-        if( i_used < 0 )
+        if( p_block->i_buffer == 0 )
+            goto end;
+
+        AVPacket pkt;
+        av_init_packet( &pkt );
+        pkt.data = p_block->p_buffer;
+        pkt.size = p_block->i_buffer;
+
+        int used = avcodec_decode_audio4( ctx, &frame, &got_frame, &pkt );
+        if( used < 0 )
+        {
             msg_Warn( p_dec, "cannot decode one frame (%zu bytes)",
                       p_block->i_buffer );
+            goto end;
+        }
 
-        block_Release( p_block );
-        return NULL;
-    }
-    else if( (size_t)i_used > p_block->i_buffer )
-    {
-        i_used = p_block->i_buffer;
+        assert( p_block->i_buffer >= (unsigned)used );
+        p_block->p_buffer += used;
+        p_block->i_buffer -= used;
     }
 
-    p_block->i_buffer -= i_used;
-    p_block->p_buffer += i_used;
-
-    if( p_sys->p_context->channels <= 0 || p_sys->p_context->channels > 8 ||
-        p_sys->p_context->sample_rate <= 0 )
+    if( ctx->channels <= 0 || ctx->channels > 8 || ctx->sample_rate <= 0 )
     {
         msg_Warn( p_dec, "invalid audio properties channels count %d, sample rate %d",
-                  p_sys->p_context->channels, p_sys->p_context->sample_rate );
-        block_Release( p_block );
-        return NULL;
+                  ctx->channels, ctx->sample_rate );
+        goto end;
     }
 
-    if( p_dec->fmt_out.audio.i_rate != (unsigned int)p_sys->p_context->sample_rate )
+    if( p_dec->fmt_out.audio.i_rate != (unsigned int)ctx->sample_rate )
+        date_Init( &p_sys->end_date, ctx->sample_rate, 1 );
+
+    if( p_block->i_pts > VLC_TS_INVALID &&
+        p_block->i_pts > date_Get( &p_sys->end_date ) )
     {
-        aout_DateInit( &p_sys->end_date, p_sys->p_context->sample_rate );
-        aout_DateSet( &p_sys->end_date, p_block->i_pts );
+        date_Set( &p_sys->end_date, p_block->i_pts );
     }
 
-    /* **** Set audio output parameters **** */
-    SetupOutputFormat( p_dec, true );
-
-    if( p_block->i_pts != 0 &&
-        p_block->i_pts != aout_DateGet( &p_sys->end_date ) )
-    {
-        aout_DateSet( &p_sys->end_date, p_block->i_pts );
+    if( p_block->i_buffer == 0 )
+    {   /* Done with this buffer */
+        block_Release( p_block );
+        *pp_block = NULL;
     }
-    p_block->i_pts = 0;
 
-    /* **** Now we can output these samples **** */
-    p_sys->i_samples = i_output / (p_dec->fmt_out.audio.i_bitspersample / 8) / p_sys->p_context->channels;
-    p_sys->p_samples = p_sys->p_output;
+    /* NOTE WELL: Beyond this point, p_block now refers to the DECODED block */
+    p_block = frame.opaque;
+    SetupOutputFormat( p_dec, true );
 
     /* Silent unwanted samples */
     if( p_sys->i_reject_count > 0 )
     {
-        memset( p_sys->p_output, 0, i_output );
+        memset( p_block->p_buffer, 0, p_block->i_buffer );
         p_sys->i_reject_count--;
     }
 
-    p_buffer = SplitBuffer( p_dec );
-    if( !p_buffer ) block_Release( p_block );
-    return p_buffer;
-}
+    block_t *p_buffer = decoder_NewAudioBuffer( p_dec, p_block->i_nb_samples );
+    if (!p_buffer)
+        return NULL;
+    assert( p_block->i_nb_samples >= (unsigned)frame.nb_samples );
+    assert( p_block->i_nb_samples == p_buffer->i_nb_samples );
+    p_block->i_buffer = p_buffer->i_buffer; /* drop buffer padding */
 
-/*****************************************************************************
- * EndAudioDec: audio decoder destruction
- *****************************************************************************/
-void EndAudioDec( decoder_t *p_dec )
-{
-    decoder_sys_t *p_sys = p_dec->p_sys;
+    /* Interleave audio if required */
+    if( av_sample_fmt_is_planar( ctx->sample_fmt ) )
+    {
+        aout_Interleave( p_buffer->p_buffer, p_block->p_buffer,
+                         p_block->i_nb_samples, ctx->channels,
+                         p_dec->fmt_out.audio.i_format );
+        if( ctx->channels > AV_NUM_DATA_POINTERS )
+            free( frame.extended_data );
+        block_Release( p_block );
+        p_block = p_buffer;
+    }
+    else /* FIXME: improve decoder_NewAudioBuffer(), avoid useless buffer... */
+        block_Release( p_buffer );
+
+    if (p_sys->b_extract)
+    {   /* TODO: do not drop channels... at least not here */
+        p_buffer = block_Alloc( p_dec->fmt_out.audio.i_bytes_per_frame
+                                * frame.nb_samples );
+        if( unlikely(p_buffer == NULL) )
+        {
+            block_Release( p_block );
+            return NULL;
+        }
+        aout_ChannelExtract( p_buffer->p_buffer,
+                             p_dec->fmt_out.audio.i_channels,
+                             p_block->p_buffer, ctx->channels,
+                             frame.nb_samples, p_sys->pi_extraction,
+                             p_dec->fmt_out.audio.i_bitspersample );
+        block_Release( p_block );
+        p_block = p_buffer;
+    }
 
-    free( p_sys->p_output );
+    p_block->i_nb_samples = frame.nb_samples;
+    p_block->i_buffer = frame.nb_samples
+                        * p_dec->fmt_out.audio.i_bytes_per_frame;
+    p_block->i_pts = date_Get( &p_sys->end_date );
+    p_block->i_length = date_Increment( &p_sys->end_date, frame.nb_samples )
+                        - p_block->i_pts;
+    return p_block;
+
+end:
+    block_Release(p_block);
+    *pp_block = NULL;
+    return NULL;
 }
 
 /*****************************************************************************
  *
  *****************************************************************************/
-#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT( 52, 2, 0 )
-#   define LIBAVCODEC_AUDIO_LAYOUT
-#else
-#   warning "Audio channel layout is unsupported by your avcodec version."
-#endif
 
-#if defined(LIBAVCODEC_AUDIO_LAYOUT)
-static const uint64_t pi_channels_map[][2] =
+vlc_fourcc_t GetVlcAudioFormat( int fmt )
 {
-    { CH_FRONT_LEFT,        AOUT_CHAN_LEFT },
-    { CH_FRONT_RIGHT,       AOUT_CHAN_RIGHT },
-    { CH_FRONT_CENTER,      AOUT_CHAN_CENTER },
-    { CH_LOW_FREQUENCY,     AOUT_CHAN_LFE },
-    { CH_BACK_LEFT,         AOUT_CHAN_REARLEFT },
-    { CH_BACK_RIGHT,        AOUT_CHAN_REARRIGHT },
-    { CH_FRONT_LEFT_OF_CENTER, 0 },
-    { CH_FRONT_RIGHT_OF_CENTER, 0 },
-    { CH_BACK_CENTER,       AOUT_CHAN_REARCENTER },
-    { CH_SIDE_LEFT,         AOUT_CHAN_MIDDLELEFT },
-    { CH_SIDE_RIGHT,        AOUT_CHAN_MIDDLERIGHT },
-    { CH_TOP_CENTER,        0 },
-    { CH_TOP_FRONT_LEFT,    0 },
-    { CH_TOP_FRONT_CENTER,  0 },
-    { CH_TOP_FRONT_RIGHT,   0 },
-    { CH_TOP_BACK_LEFT,     0 },
-    { CH_TOP_BACK_CENTER,   0 },
-    { CH_TOP_BACK_RIGHT,    0 },
-    { CH_STEREO_LEFT,       0 },
-    { CH_STEREO_RIGHT,      0 },
-};
-#else
+    static const vlc_fourcc_t fcc[] = {
+        [AV_SAMPLE_FMT_U8]    = VLC_CODEC_U8,
+        [AV_SAMPLE_FMT_S16]   = VLC_CODEC_S16N,
+        [AV_SAMPLE_FMT_S32]   = VLC_CODEC_S32N,
+        [AV_SAMPLE_FMT_FLT]   = VLC_CODEC_FL32,
+        [AV_SAMPLE_FMT_DBL]   = VLC_CODEC_FL64,
+        [AV_SAMPLE_FMT_U8P]   = VLC_CODEC_U8,
+        [AV_SAMPLE_FMT_S16P]  = VLC_CODEC_S16N,
+        [AV_SAMPLE_FMT_S32P]  = VLC_CODEC_S32N,
+        [AV_SAMPLE_FMT_FLTP]  = VLC_CODEC_FL32,
+        [AV_SAMPLE_FMT_DBLP]  = VLC_CODEC_FL64,
+    };
+    if( (sizeof(fcc) / sizeof(fcc[0])) > (unsigned)fmt )
+        return fcc[fmt];
+    return VLC_CODEC_S16N;
+}
+
 static const uint64_t pi_channels_map[][2] =
 {
-    { 0, AOUT_CHAN_LEFT },
-    { 0, AOUT_CHAN_RIGHT },
-    { 0, AOUT_CHAN_CENTER },
-    { 0, AOUT_CHAN_LFE },
-    { 0, AOUT_CHAN_REARLEFT },
-    { 0, AOUT_CHAN_REARRIGHT },
-    { 0, 0 },
-    { 0, 0 },
-    { 0, AOUT_CHAN_REARCENTER },
-    { 0, AOUT_CHAN_MIDDLELEFT },
-    { 0, AOUT_CHAN_MIDDLERIGHT },
-    { 0, 0 },
-    { 0, 0 },
-    { 0, 0 },
-    { 0, 0 },
-    { 0, 0 },
-    { 0, 0 },
-    { 0, 0 },
-    { 0, 0 },
-    { 0, 0 },
+    { AV_CH_FRONT_LEFT,        AOUT_CHAN_LEFT },
+    { AV_CH_FRONT_RIGHT,       AOUT_CHAN_RIGHT },
+    { AV_CH_FRONT_CENTER,      AOUT_CHAN_CENTER },
+    { AV_CH_LOW_FREQUENCY,     AOUT_CHAN_LFE },
+    { AV_CH_BACK_LEFT,         AOUT_CHAN_REARLEFT },
+    { AV_CH_BACK_RIGHT,        AOUT_CHAN_REARRIGHT },
+    { AV_CH_FRONT_LEFT_OF_CENTER, 0 },
+    { AV_CH_FRONT_RIGHT_OF_CENTER, 0 },
+    { AV_CH_BACK_CENTER,       AOUT_CHAN_REARCENTER },
+    { AV_CH_SIDE_LEFT,         AOUT_CHAN_MIDDLELEFT },
+    { AV_CH_SIDE_RIGHT,        AOUT_CHAN_MIDDLERIGHT },
+    { AV_CH_TOP_CENTER,        0 },
+    { AV_CH_TOP_FRONT_LEFT,    0 },
+    { AV_CH_TOP_FRONT_CENTER,  0 },
+    { AV_CH_TOP_FRONT_RIGHT,   0 },
+    { AV_CH_TOP_BACK_LEFT,     0 },
+    { AV_CH_TOP_BACK_CENTER,   0 },
+    { AV_CH_TOP_BACK_RIGHT,    0 },
+    { AV_CH_STEREO_LEFT,       0 },
+    { AV_CH_STEREO_RIGHT,      0 },
 };
-#endif
 
 static void SetupOutputFormat( decoder_t *p_dec, bool b_trust )
 {
     decoder_sys_t *p_sys = p_dec->p_sys;
 
-#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT( 51, 65, 0 )
-    switch( p_sys->p_context->sample_fmt )
-    {
-    case SAMPLE_FMT_U8:
-        p_dec->fmt_out.i_codec = VLC_CODEC_U8;
-        p_dec->fmt_out.audio.i_bitspersample = 8;
-        break;
-    case SAMPLE_FMT_S32:
-        p_dec->fmt_out.i_codec = AOUT_FMT_S32_NE;
-        p_dec->fmt_out.audio.i_bitspersample = 32;
-        break;
-    case SAMPLE_FMT_FLT:
-        p_dec->fmt_out.i_codec = VLC_CODEC_FL32;
-        p_dec->fmt_out.audio.i_bitspersample = 32;
-        break;
-    case SAMPLE_FMT_DBL:
-        p_dec->fmt_out.i_codec = VLC_CODEC_FL64;
-        p_dec->fmt_out.audio.i_bitspersample = 64;
-        break;
-
-    case SAMPLE_FMT_S16:
-    default:
-        p_dec->fmt_out.i_codec = AOUT_FMT_S16_NE;
-        p_dec->fmt_out.audio.i_bitspersample = 16;
-        break;
-    }
-#else
-    p_dec->fmt_out.i_codec = AOUT_FMT_S16_NE;
-    p_dec->fmt_out.audio.i_bitspersample = 16;
-#endif
-    p_dec->fmt_out.audio.i_rate     = p_sys->p_context->sample_rate;
-    p_dec->fmt_out.audio.i_channels = p_sys->p_context->channels;
+    p_dec->fmt_out.i_codec = GetVlcAudioFormat( p_sys->p_context->sample_fmt );
+    p_dec->fmt_out.audio.i_format = p_dec->fmt_out.i_codec;
+    p_dec->fmt_out.audio.i_rate = p_sys->p_context->sample_rate;
 
     /* */
-#if defined(LIBAVCODEC_AUDIO_LAYOUT)
     if( p_sys->i_previous_channels == p_sys->p_context->channels &&
         p_sys->i_previous_layout == p_sys->p_context->channel_layout )
         return;
-#else
-    if( p_sys->i_previous_channels == p_sys->p_context->channels )
-        return;
-#endif
     if( b_trust )
     {
         p_sys->i_previous_channels = p_sys->p_context->channels;
-#if defined(LIBAVCODEC_AUDIO_LAYOUT)
         p_sys->i_previous_layout = p_sys->p_context->channel_layout;
-#endif
     }
 
     /* Specified order
      * FIXME should we use fmt_in.audio.i_physical_channels or not ?
      */
-#if defined(LIBAVCODEC_AUDIO_LAYOUT)
     const unsigned i_order_max = 8 * sizeof(p_sys->p_context->channel_layout);
-#else
-    const unsigned i_order_max = 64;
-#endif
     uint32_t pi_order_src[i_order_max];
     int i_channels_src = 0;
 
-#if defined(LIBAVCODEC_AUDIO_LAYOUT)
     if( p_sys->p_context->channel_layout )
     {
         for( unsigned i = 0; i < sizeof(pi_channels_map)/sizeof(*pi_channels_map); i++ )
@@ -521,7 +463,6 @@ static void SetupOutputFormat( decoder_t *p_dec, bool b_trust )
         }
     }
     else
-#endif
     {
         /* Create default order  */
         if( b_trust )
@@ -545,6 +486,6 @@ static void SetupOutputFormat( decoder_t *p_dec, bool b_trust )
 
     p_dec->fmt_out.audio.i_physical_channels =
     p_dec->fmt_out.audio.i_original_channels = i_layout_dst;
-    p_dec->fmt_out.audio.i_channels = i_channels_dst;
+    aout_FormatPrepare( &p_dec->fmt_out.audio );
 }