]> git.sesse.net Git - vlc/blobdiff - modules/codec/avcodec/audio.c
AVCodec : Set audio related parameters in ffmpeg_OpenCodec and use it for audio decod...
[vlc] / modules / codec / avcodec / audio.c
index 92c62080259b61ac3fd24af05cb5bc8f59f11787..483c3067c84515774f7747c3b45705de54cfad2c 100644 (file)
@@ -32,6 +32,7 @@
 #include <vlc_common.h>
 #include <vlc_aout.h>
 #include <vlc_codec.h>
+#include <vlc_avcodec.h>
 
 /* ffmpeg header */
 #ifdef HAVE_LIBAVCODEC_AVCODEC_H
 
 #include "avcodec.h"
 
-static const unsigned int pi_channels_maps[7] =
-{
-    0,
-    AOUT_CHAN_CENTER,   AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT,
-    AOUT_CHAN_CENTER | AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT,
-    AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_REARLEFT
-     | AOUT_CHAN_REARRIGHT,
-    AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_CENTER
-     | AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT,
-    AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_CENTER
-     | AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT | AOUT_CHAN_LFE
-};
-
 /*****************************************************************************
  * decoder_sys_t : decoder descriptor
  *****************************************************************************/
@@ -65,13 +53,14 @@ struct decoder_sys_t
     FFMPEG_COMMON_MEMBERS
 
     /* Temporary buffer for libavcodec */
+    int     i_output_max;
     uint8_t *p_output;
 
     /*
      * Output properties
      */
     audio_sample_format_t aout_format;
-    audio_date_t          end_date;
+    date_t                end_date;
 
     /*
      *
@@ -81,65 +70,32 @@ struct decoder_sys_t
 
     /* */
     int     i_reject_count;
-};
-
-/*****************************************************************************
- * 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 )
-{
-    decoder_sys_t *p_sys;
 
-    /* Allocate the memory needed to store the decoder's structure */
-    if( ( p_dec->p_sys = p_sys =
-          (decoder_sys_t *)malloc(sizeof(decoder_sys_t)) ) == 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;
-
-    /* ***** 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;
-    if( !p_dec->fmt_in.audio.i_physical_channels )
-    {
-        msg_Warn( p_dec, "Physical channel configuration not set : guessing" );
-        p_dec->fmt_in.audio.i_original_channels =
-            p_dec->fmt_in.audio.i_physical_channels =
-                pi_channels_maps[p_sys->p_context->channels];
-    }
+    /* */
+    bool    b_extract;
+    int     pi_extraction[AOUT_CHAN_MAX];
+    int     i_previous_channels;
+    int64_t i_previous_layout;
+};
 
-    p_dec->fmt_out.audio.i_physical_channels =
-        p_dec->fmt_out.audio.i_original_channels =
-        p_dec->fmt_in.audio.i_physical_channels;
+#define BLOCK_FLAG_PRIVATE_REALLOCATED (1 << BLOCK_FLAG_PRIVATE_SHIFT)
 
-    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
+static void SetupOutputFormat( decoder_t *p_dec, bool b_trust );
 
+static void InitDecoderConfig( decoder_t *p_dec, AVCodecContext *p_context )
+{
     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_FOURCC( 'f', 'l', 'a', 'c' ) )
+        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_FOURCC( 'a', 'l', 'a', 'c' ) )
+        else 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 */
@@ -160,13 +116,13 @@ int InitAudioDec( decoder_t *p_dec, AVCodecContext *p_context,
 
         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 );
@@ -175,38 +131,90 @@ 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;
     }
+}
 
-    /* ***** Open the codec ***** */
-    vlc_mutex_lock( &avcodec_lock );
+/*****************************************************************************
+ * 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 )
+{
+    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 = CODEC_TYPE_AUDIO;
+    p_context->codec_type = CODEC_TYPE_AUDIO;
+    p_context->codec_id = i_codec_id;
+    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;
+
+    // Initialize decoder extradata
+    InitDecoderConfig( p_dec, p_context);
 
-    if (avcodec_open( p_sys->p_context, p_sys->p_codec ) < 0)
+    /* ***** Open the codec ***** */
+    if( ffmpeg_OpenCodec( p_dec ) < 0 )
     {
-        vlc_mutex_unlock( &avcodec_lock );
         msg_Err( p_dec, "cannot open codec (%s)", p_sys->psz_namecodec );
         free( p_sys->p_context->extradata );
         free( p_sys );
         return VLC_EGENERIC;
     }
-    vlc_mutex_unlock( &avcodec_lock );
 
-    msg_Dbg( p_dec, "ffmpeg codec (%s) started", p_sys->psz_namecodec );
+    switch( i_codec_id )
+    {
+    case CODEC_ID_WAVPACK:
+        p_sys->i_output_max = 8 * sizeof(int32_t) * 131072;
+        break;
+    case CODEC_ID_TTA:
+        p_sys->i_output_max = p_sys->p_context->channels * sizeof(int32_t) * p_sys->p_context->sample_rate * 2;
+        break;
+    case CODEC_ID_FLAC:
+        p_sys->i_output_max = 8 * sizeof(int32_t) * 65535;
+        break;
+#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT( 52, 35, 0 )
+    case CODEC_ID_WMAPRO:
+        p_sys->i_output_max = 8 * sizeof(float) * 6144; /* (1 << 12) * 3/2 */
+        break;
+#endif
+    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 = av_malloc( p_sys->i_output_max );
 
-    p_sys->p_output = malloc( AVCODEC_MAX_AUDIO_FRAME_SIZE );
     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 );
-
-    /* Set output properties */
+    /* */
     p_dec->fmt_out.i_cat = AUDIO_ES;
-    p_dec->fmt_out.i_codec = AOUT_FMT_S16_NE;
-    p_dec->fmt_out.audio.i_bitspersample = 16;
+    /* 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;
 }
@@ -223,18 +231,21 @@ static aout_buffer_t *SplitBuffer( decoder_t *p_dec )
 
     if( i_samples == 0 ) return NULL;
 
-    if( ( p_buffer = p_dec->pf_aout_buffer_new( p_dec, i_samples ) ) == NULL )
-    {
-        msg_Err( p_dec, "cannot get aout buffer" );
+    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 );
+    p_buffer->i_pts = date_Get( &p_sys->end_date );
+    p_buffer->i_length = date_Increment( &p_sys->end_date, i_samples )
+                         - p_buffer->i_pts;
 
-    memcpy( p_buffer->p_buffer, p_sys->p_samples, p_buffer->i_nb_bytes );
+    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 );
+    else
+        memcpy( p_buffer->p_buffer, p_sys->p_samples, p_buffer->i_buffer );
 
-    p_sys->p_samples += p_buffer->i_nb_bytes;
+    p_sys->p_samples += i_samples * p_sys->p_context->channels * ( p_dec->fmt_out.audio.i_bitspersample / 8 );
     p_sys->i_samples -= i_samples;
 
     return p_buffer;
@@ -258,6 +269,8 @@ aout_buffer_t * DecodeAudio ( decoder_t *p_dec, block_t **pp_block )
     {
         block_Release( p_block );
         avcodec_flush_buffers( p_sys->p_context );
+        p_sys->i_samples = 0;
+        date_Set( &p_sys->end_date, 0 );
 
         if( p_sys->i_codec_id == CODEC_ID_MP2 || p_sys->i_codec_id == CODEC_ID_MP3 )
             p_sys->i_reject_count = 3;
@@ -272,7 +285,7 @@ aout_buffer_t * DecodeAudio ( decoder_t *p_dec, block_t **pp_block )
         return p_buffer;
     }
 
-    if( !aout_DateGet( &p_sys->end_date ) && !p_block->i_pts )
+    if( !date_Get( &p_sys->end_date ) && !p_block->i_pts )
     {
         /* We've just started the stream, wait for the first PTS. */
         block_Release( p_block );
@@ -284,47 +297,51 @@ aout_buffer_t * DecodeAudio ( decoder_t *p_dec, block_t **pp_block )
         block_Release( p_block );
         return NULL;
     }
-    if( p_block->i_buffer > AVCODEC_MAX_AUDIO_FRAME_SIZE )
-    {
-        /* Grow output buffer if necessary (eg. for PCM data) */
-        p_sys->p_output = realloc(p_sys->p_output, p_block->i_buffer);
-    }
 
-    *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_output = __MAX( AVCODEC_MAX_AUDIO_FRAME_SIZE, p_block->i_buffer );
-    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
-
-    if( i_used < 0 || i_output < 0 )
+    if( (p_block->i_flags & BLOCK_FLAG_PRIVATE_REALLOCATED) == 0 )
     {
-        if( i_used < 0 )
-            msg_Warn( p_dec, "cannot decode one frame (%zu bytes)",
-                      p_block->i_buffer );
+        *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 );
 
-        block_Release( p_block );
-        return NULL;
+        p_block->i_flags |= BLOCK_FLAG_PRIVATE_REALLOCATED;
     }
-    else if( (size_t)i_used > p_block->i_buffer )
+
+    do
     {
-        i_used = p_block->i_buffer;
-    }
+        i_output = __MAX( p_block->i_buffer, p_sys->i_output_max );
+        if( i_output > p_sys->i_output_max )
+        {
+            /* Grow output buffer if necessary (eg. for PCM data) */
+            p_sys->p_output = av_realloc( p_sys->p_output, i_output );
+        }
 
-    p_block->i_buffer -= i_used;
-    p_block->p_buffer += i_used;
+        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 );
 
-    if( p_sys->p_context->channels <= 0 || p_sys->p_context->channels > 6 ||
+        if( i_used < 0 || i_output < 0 )
+        {
+            if( i_used < 0 )
+                msg_Warn( p_dec, "cannot decode one frame (%zu bytes)",
+                          p_block->i_buffer );
+
+            block_Release( p_block );
+            return NULL;
+        }
+        else if( (size_t)i_used > p_block->i_buffer )
+        {
+            i_used = p_block->i_buffer;
+        }
+
+        p_block->i_buffer -= i_used;
+        p_block->p_buffer += i_used;
+
+    } while( p_block->i_buffer > 0 && i_output <= 0 );
+
+    if( p_sys->p_context->channels <= 0 || p_sys->p_context->channels > 8 ||
         p_sys->p_context->sample_rate <= 0 )
     {
         msg_Warn( p_dec, "invalid audio properties channels count %d, sample rate %d",
@@ -335,26 +352,22 @@ aout_buffer_t * DecodeAudio ( decoder_t *p_dec, block_t **pp_block )
 
     if( p_dec->fmt_out.audio.i_rate != (unsigned int)p_sys->p_context->sample_rate )
     {
-        aout_DateInit( &p_sys->end_date, p_sys->p_context->sample_rate );
-        aout_DateSet( &p_sys->end_date, p_block->i_pts );
+        date_Init( &p_sys->end_date, p_sys->p_context->sample_rate, 1 );
+        date_Set( &p_sys->end_date, p_block->i_pts );
     }
 
     /* **** Set audio output parameters **** */
-    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.audio.i_original_channels =
-        p_dec->fmt_out.audio.i_physical_channels =
-            pi_channels_maps[p_sys->p_context->channels];
+    SetupOutputFormat( p_dec, true );
 
     if( p_block->i_pts != 0 &&
-        p_block->i_pts != aout_DateGet( &p_sys->end_date ) )
+        p_block->i_pts != date_Get( &p_sys->end_date ) )
     {
-        aout_DateSet( &p_sys->end_date, p_block->i_pts );
+        date_Set( &p_sys->end_date, p_block->i_pts );
     }
     p_block->i_pts = 0;
 
     /* **** Now we can output these samples **** */
-    p_sys->i_samples = i_output / sizeof(int16_t) / p_sys->p_context->channels;
+    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;
 
     /* Silent unwanted samples */
@@ -376,5 +389,124 @@ void EndAudioDec( decoder_t *p_dec )
 {
     decoder_sys_t *p_sys = p_dec->p_sys;
 
-    free( p_sys->p_output );
+    av_free( p_sys->p_output );
+}
+
+/*****************************************************************************
+ *
+ *****************************************************************************/
+
+void GetVlcAudioFormat( vlc_fourcc_t *pi_codec, unsigned *pi_bits, int i_sample_fmt )
+{
+    switch( i_sample_fmt )
+    {
+    case SAMPLE_FMT_U8:
+        *pi_codec = VLC_CODEC_U8;
+        *pi_bits = 8;
+        break;
+    case SAMPLE_FMT_S32:
+        *pi_codec = VLC_CODEC_S32N;
+        *pi_bits = 32;
+        break;
+    case SAMPLE_FMT_FLT:
+        *pi_codec = VLC_CODEC_FL32;
+        *pi_bits = 32;
+        break;
+    case SAMPLE_FMT_DBL:
+        *pi_codec = VLC_CODEC_FL64;
+        *pi_bits = 64;
+        break;
+
+    case SAMPLE_FMT_S16:
+    default:
+        *pi_codec = VLC_CODEC_S16N;
+        *pi_bits = 16;
+        break;
+    }
+}
+
+static const uint64_t pi_channels_map[][2] =
+{
+    { 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 },
+};
+
+static void SetupOutputFormat( decoder_t *p_dec, bool b_trust )
+{
+    decoder_sys_t *p_sys = p_dec->p_sys;
+
+    GetVlcAudioFormat( &p_dec->fmt_out.i_codec,
+                       &p_dec->fmt_out.audio.i_bitspersample,
+                       p_sys->p_context->sample_fmt );
+    p_dec->fmt_out.audio.i_rate = p_sys->p_context->sample_rate;
+
+    /* */
+    if( p_sys->i_previous_channels == p_sys->p_context->channels &&
+        p_sys->i_previous_layout == p_sys->p_context->channel_layout )
+        return;
+    if( b_trust )
+    {
+        p_sys->i_previous_channels = p_sys->p_context->channels;
+        p_sys->i_previous_layout = p_sys->p_context->channel_layout;
+    }
+
+    /* Specified order
+     * FIXME should we use fmt_in.audio.i_physical_channels or not ?
+     */
+    const unsigned i_order_max = 8 * sizeof(p_sys->p_context->channel_layout);
+    uint32_t pi_order_src[i_order_max];
+    int i_channels_src = 0;
+
+    if( p_sys->p_context->channel_layout )
+    {
+        for( unsigned i = 0; i < sizeof(pi_channels_map)/sizeof(*pi_channels_map); i++ )
+        {
+            if( p_sys->p_context->channel_layout & pi_channels_map[i][0] )
+                pi_order_src[i_channels_src++] = pi_channels_map[i][1];
+        }
+    }
+    else
+    {
+        /* Create default order  */
+        if( b_trust )
+            msg_Warn( p_dec, "Physical channel configuration not set : guessing" );
+        for( unsigned int i = 0; i < __MIN( i_order_max, (unsigned)p_sys->p_context->channels ); i++ )
+        {
+            if( i < sizeof(pi_channels_map)/sizeof(*pi_channels_map) )
+                pi_order_src[i_channels_src++] = pi_channels_map[i][1];
+        }
+    }
+    if( i_channels_src != p_sys->p_context->channels && b_trust )
+        msg_Err( p_dec, "Channel layout not understood" );
+
+    uint32_t i_layout_dst;
+    int      i_channels_dst;
+    p_sys->b_extract = aout_CheckChannelExtraction( p_sys->pi_extraction,
+                                                    &i_layout_dst, &i_channels_dst,
+                                                    NULL, pi_order_src, i_channels_src );
+    if( i_channels_dst != i_channels_src && b_trust )
+        msg_Warn( p_dec, "%d channels are dropped", i_channels_src - i_channels_dst );
+
+    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;
 }
+