]> git.sesse.net Git - vlc/blobdiff - modules/codec/vorbis.c
vout: remove unimplemented GET_OPENGL controls
[vlc] / modules / codec / vorbis.c
index 0e493771973def1b1546261413d0aac35166f044..077a1c3becf6877ba00da410f44446a469a8aaf5 100644 (file)
@@ -1,7 +1,7 @@
 /*****************************************************************************
  * vorbis.c: vorbis decoder/encoder/packetizer module using of libvorbis.
  *****************************************************************************
- * Copyright (C) 2001-2012 the VideoLAN team
+ * Copyright (C) 2001-2012 VLC authors and VideoLAN
  * Copyright (C) 2007 Société des arts technologiques
  * Copyright (C) 2007 Savoir-faire Linux
  *
@@ -9,19 +9,19 @@
  *
  * Authors: 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.
  *****************************************************************************/
 
 /*****************************************************************************
@@ -34,6 +34,7 @@
 #include <vlc_common.h>
 #include <vlc_plugin.h>
 #include <vlc_codec.h>
+#include <vlc_charset.h>
 #include <vlc_aout.h>
 #include <vlc_input.h>
 #include <vlc_sout.h>
@@ -88,7 +89,7 @@ struct decoder_sys_t
     /*
     ** Channel reordering
     */
-    int pi_chan_table[AOUT_CHAN_MAX];
+    uint8_t pi_chan_table[AOUT_CHAN_MAX];
 };
 
 static const int pi_channels_maps[9] =
@@ -156,7 +157,7 @@ static block_t *SendPacket( decoder_t *, ogg_packet *, block_t * );
 
 static void ParseVorbisComments( decoder_t * );
 
-static void ConfigureChannelOrder(int *, int, uint32_t, bool );
+static void ConfigureChannelOrder(uint8_t *, int, uint32_t, bool );
 
 #ifdef HAVE_VORBIS_ENCODER
 static int OpenEncoder   ( vlc_object_t * );
@@ -253,7 +254,7 @@ static int OpenDecoder( vlc_object_t *p_this )
     /* Set output properties */
     p_dec->fmt_out.i_cat = AUDIO_ES;
 #ifdef MODULE_NAME_IS_tremor
-    p_dec->fmt_out.i_codec = VLC_CODEC_FI32;
+    p_dec->fmt_out.i_codec = VLC_CODEC_S32N;
 #else
     p_dec->fmt_out.i_codec = VLC_CODEC_FL32;
 #endif
@@ -317,7 +318,8 @@ static block_t *DecodeBlock( decoder_t *p_dec, block_t **pp_block )
     {
         if( ProcessHeaders( p_dec ) )
         {
-            block_Release( *pp_block );
+            if( *pp_block )
+                block_Release( *pp_block );
             return NULL;
         }
         p_sys->b_has_headers = true;
@@ -335,13 +337,13 @@ static int ProcessHeaders( decoder_t *p_dec )
     ogg_packet oggpacket;
 
     unsigned pi_size[XIPH_MAX_HEADER_COUNT];
-    void     *pp_data[XIPH_MAX_HEADER_COUNT];
+    void *pp_data[XIPH_MAX_HEADER_COUNT];
     unsigned i_count;
     if( xiph_SplitHeaders( pi_size, pp_data, &i_count,
                            p_dec->fmt_in.i_extra, p_dec->fmt_in.p_extra) )
         return VLC_EGENERIC;
     if( i_count < 3 )
-        goto error;
+        return VLC_EGENERIC;
 
     oggpacket.granulepos = -1;
     oggpacket.e_o_s = 0;
@@ -354,29 +356,30 @@ static int ProcessHeaders( decoder_t *p_dec )
     if( vorbis_synthesis_headerin( &p_sys->vi, &p_sys->vc, &oggpacket ) < 0 )
     {
         msg_Err( p_dec, "this bitstream does not contain Vorbis audio data");
-        goto error;
+        return VLC_EGENERIC;
     }
 
     /* Setup the format */
     p_dec->fmt_out.audio.i_rate     = p_sys->vi.rate;
     p_dec->fmt_out.audio.i_channels = p_sys->vi.channels;
 
-    if( p_dec->fmt_out.audio.i_channels > 9 )
+    if( p_dec->fmt_out.audio.i_channels >= ARRAY_SIZE(pi_channels_maps) )
     {
-        msg_Err( p_dec, "invalid number of channels (not between 1 and 9): %i",
+        msg_Err( p_dec, "invalid number of channels (1-%zu): %i",
+                 ARRAY_SIZE(pi_channels_maps),
                  p_dec->fmt_out.audio.i_channels );
-        goto error;
+        return VLC_EGENERIC;
     }
 
     p_dec->fmt_out.audio.i_physical_channels =
         p_dec->fmt_out.audio.i_original_channels =
             pi_channels_maps[p_sys->vi.channels];
-    p_dec->fmt_out.i_bitrate = p_sys->vi.bitrate_nominal;
+    p_dec->fmt_out.i_bitrate = __MAX( 0, (int32_t) p_sys->vi.bitrate_nominal );
 
     date_Init( &p_sys->end_date, p_sys->vi.rate, 1 );
 
-    msg_Dbg( p_dec, "channels:%d samplerate:%ld bitrate:%ld",
-             p_sys->vi.channels, p_sys->vi.rate, p_sys->vi.bitrate_nominal );
+    msg_Dbg( p_dec, "channels:%d samplerate:%ld bitrate:%ud",
+             p_sys->vi.channels, p_sys->vi.rate, p_dec->fmt_out.i_bitrate );
 
     /* The next packet in order is the comments header */
     oggpacket.b_o_s  = 0;
@@ -385,7 +388,7 @@ static int ProcessHeaders( decoder_t *p_dec )
     if( vorbis_synthesis_headerin( &p_sys->vi, &p_sys->vc, &oggpacket ) < 0 )
     {
         msg_Err( p_dec, "2nd Vorbis header is corrupted" );
-        goto error;
+        return VLC_EGENERIC;
     }
     ParseVorbisComments( p_dec );
 
@@ -419,14 +422,7 @@ static int ProcessHeaders( decoder_t *p_dec )
     ConfigureChannelOrder(p_sys->pi_chan_table, p_sys->vi.channels,
             p_dec->fmt_out.audio.i_physical_channels, true);
 
-    for( unsigned i = 0; i < i_count; i++ )
-        free( pp_data[i] );
     return VLC_SUCCESS;
-
-error:
-    for( unsigned i = 0; i < i_count; i++ )
-        free( pp_data[i] );
-    return VLC_EGENERIC;
 }
 
 /*****************************************************************************
@@ -471,15 +467,20 @@ static void *ProcessPacket( decoder_t *p_dec, ogg_packet *p_oggpacket,
  * Interleave: helper function to interleave channels
  *****************************************************************************/
 static void Interleave( INTERLEAVE_TYPE *p_out, const INTERLEAVE_TYPE **pp_in,
-                        int i_nb_channels, int i_samples, int *pi_chan_table)
+                        int i_nb_channels, int i_samples, uint8_t *pi_chan_table)
 {
     for( int j = 0; j < i_samples; j++ )
         for( int i = 0; i < i_nb_channels; i++ )
-            p_out[j * i_nb_channels + pi_chan_table[i]] = pp_in[i][j]
+        {
 #ifdef MODULE_NAME_IS_tremor
-                * (FIXED32_ONE >> 24)
+            union { int32_t i; uint32_t u;} spl;
+
+            spl.u = ((uint32_t)pp_in[i][j]) << 8;
+            p_out[j * i_nb_channels + pi_chan_table[i]] = spl.i;
+#else
+            p_out[j * i_nb_channels + pi_chan_table[i]] = pp_in[i][j];
 #endif
-            ;
+        }
 }
 
 /*****************************************************************************
@@ -513,7 +514,8 @@ static block_t *DecodePacket( decoder_t *p_dec, ogg_packet *p_oggpacket )
 
         /* Interleave the samples */
         Interleave( (INTERLEAVE_TYPE*)p_aout_buffer->p_buffer,
-                    (const INTERLEAVE_TYPE**)pp_pcm, p_sys->vi.channels, i_samples, p_sys->pi_chan_table);
+                    (const INTERLEAVE_TYPE**)pp_pcm, p_sys->vi.channels, i_samples,
+                    p_sys->pi_chan_table);
 
         /* Tell libvorbis how many samples we actually consumed */
         vorbis_synthesis_read( &p_sys->vd, i_samples );
@@ -567,22 +569,19 @@ static void ParseVorbisComments( decoder_t *p_dec )
             break;
         psz_name = psz_comment;
         psz_value = strchr( psz_comment, '=' );
-        if( psz_value )
+        /* Don't add empty values */
+        if( psz_value && psz_value[1] != '\0')
         {
             *psz_value = '\0';
             psz_value++;
 
-            /* Don't add empty values */
-            if( *psz_value == '\0' )
-                break;
-
             if( !strcasecmp( psz_name, "REPLAYGAIN_TRACK_GAIN" ) ||
                 !strcasecmp( psz_name, "RG_RADIO" ) )
             {
                 audio_replay_gain_t *r = &p_dec->fmt_out.audio_replay_gain;
 
                 r->pb_gain[AUDIO_REPLAY_GAIN_TRACK] = true;
-                r->pf_gain[AUDIO_REPLAY_GAIN_TRACK] = atof( psz_value );
+                r->pf_gain[AUDIO_REPLAY_GAIN_TRACK] = us_atof( psz_value );
             }
             else if( !strcasecmp( psz_name, "REPLAYGAIN_TRACK_PEAK" ) ||
                      !strcasecmp( psz_name, "RG_PEAK" ) )
@@ -590,7 +589,7 @@ static void ParseVorbisComments( decoder_t *p_dec )
                 audio_replay_gain_t *r = &p_dec->fmt_out.audio_replay_gain;
 
                 r->pb_peak[AUDIO_REPLAY_GAIN_TRACK] = true;
-                r->pf_peak[AUDIO_REPLAY_GAIN_TRACK] = atof( psz_value );
+                r->pf_peak[AUDIO_REPLAY_GAIN_TRACK] = us_atof( psz_value );
             }
             else if( !strcasecmp( psz_name, "REPLAYGAIN_ALBUM_GAIN" ) ||
                      !strcasecmp( psz_name, "RG_AUDIOPHILE" ) )
@@ -598,14 +597,14 @@ static void ParseVorbisComments( decoder_t *p_dec )
                 audio_replay_gain_t *r = &p_dec->fmt_out.audio_replay_gain;
 
                 r->pb_gain[AUDIO_REPLAY_GAIN_ALBUM] = true;
-                r->pf_gain[AUDIO_REPLAY_GAIN_ALBUM] = atof( psz_value );
+                r->pf_gain[AUDIO_REPLAY_GAIN_ALBUM] = us_atof( psz_value );
             }
             else if( !strcasecmp( psz_name, "REPLAYGAIN_ALBUM_PEAK" ) )
             {
                 audio_replay_gain_t *r = &p_dec->fmt_out.audio_replay_gain;
 
                 r->pb_peak[AUDIO_REPLAY_GAIN_ALBUM] = true;
-                r->pf_peak[AUDIO_REPLAY_GAIN_ALBUM] = atof( psz_value );
+                r->pf_peak[AUDIO_REPLAY_GAIN_ALBUM] = us_atof( psz_value );
             }
             else if( !strcasecmp( psz_name, "METADATA_BLOCK_PICTURE" ) )
             { /* Do nothing, for now */ }
@@ -626,7 +625,8 @@ static void ParseVorbisComments( decoder_t *p_dec )
 /*****************************************************************************
  *
  *****************************************************************************/
-static void ConfigureChannelOrder(int *pi_chan_table, int i_channels, uint32_t i_channel_mask, bool b_decode)
+static void ConfigureChannelOrder(uint8_t *pi_chan_table, int i_channels,
+                                  uint32_t i_channel_mask, bool b_decode)
 {
     const uint32_t *pi_channels_in;
     switch( i_channels )
@@ -660,14 +660,10 @@ static void ConfigureChannelOrder(int *pi_chan_table, int i_channels, uint32_t i
 
     if( b_decode )
         aout_CheckChannelReorder( pi_channels_in, NULL,
-                                  i_channel_mask,
-                                  i_channels,
-                                  pi_chan_table );
+                                  i_channel_mask, pi_chan_table );
     else
         aout_CheckChannelReorder( NULL, pi_channels_in,
-                                  i_channel_mask,
-                                  i_channels,
-                                  pi_chan_table );
+                                  i_channel_mask, pi_chan_table );
 }
 
 /*****************************************************************************
@@ -714,7 +710,7 @@ struct encoder_sys_t
     /*
     ** Channel reordering
     */
-    int pi_chan_table[AOUT_CHAN_MAX];
+    uint8_t pi_chan_table[AOUT_CHAN_MAX];
 
 };
 
@@ -848,6 +844,9 @@ static block_t *Encode( encoder_t *p_enc, block_t *p_aout_buf )
     block_t *p_block, *p_chain = NULL;
     float **buffer;
 
+    /* FIXME: flush buffers in here */
+    if( unlikely( !p_aout_buf ) ) return NULL;
+
     mtime_t i_pts = p_aout_buf->i_pts -
                 (mtime_t)1000000 * (mtime_t)p_sys->i_samples_delay /
                 (mtime_t)p_enc->fmt_in.audio.i_rate;
@@ -878,7 +877,7 @@ static block_t *Encode( encoder_t *p_enc, block_t *p_aout_buf )
         while( vorbis_bitrate_flushpacket( &p_sys->vd, &oggpacket ) )
         {
             int i_block_size;
-            p_block = block_New( p_enc, oggpacket.bytes );
+            p_block = block_Alloc( oggpacket.bytes );
             memcpy( p_block->p_buffer, oggpacket.packet, oggpacket.bytes );
 
             i_block_size = vorbis_packet_blocksize( &p_sys->vi, &oggpacket );