Commit cece0cc7 authored by Rémi Denis-Courmont's avatar Rémi Denis-Courmont

Remove filter_t.pf_audio_buffer_new

parent 6cdef672
...@@ -89,10 +89,8 @@ struct filter_t ...@@ -89,10 +89,8 @@ struct filter_t
struct struct
{ {
block_t * (*pf_filter) ( filter_t *, block_t * ); block_t * (*pf_filter) ( filter_t *, block_t * );
block_t * (*pf_buffer_new) ( filter_t *, int );
} audio; } audio;
#define pf_audio_filter u.audio.pf_filter #define pf_audio_filter u.audio.pf_filter
#define pf_audio_buffer_new u.audio.pf_buffer_new
struct struct
{ {
...@@ -211,23 +209,7 @@ static inline void filter_DeleteSubpicture( filter_t *p_filter, subpicture_t *p_ ...@@ -211,23 +209,7 @@ static inline void filter_DeleteSubpicture( filter_t *p_filter, subpicture_t *p_
p_filter->pf_sub_buffer_del( p_filter, p_subpicture ); p_filter->pf_sub_buffer_del( p_filter, p_subpicture );
} }
/** #define filter_NewAudioBuffer block_New
* This function will return a new audio buffer usable by p_filter as an
* output buffer. You have to release it using block_Release or by returning
* it to the caller as a pf_audio_filter return value.
* Provided for convenience.
*
* \param p_filter filter_t object
* \param i_size size of audio buffer requested
* \return block to be used as audio output buffer
*/
static inline block_t *filter_NewAudioBuffer( filter_t *p_filter, int i_size )
{
block_t *p_block = p_filter->pf_audio_buffer_new( p_filter, i_size );
if( !p_block )
msg_Warn( p_filter, "can't get output block" );
return p_block;
}
/** /**
* This function gives all input attachments at once. * This function gives all input attachments at once.
......
...@@ -466,7 +466,7 @@ static block_t *Convert( filter_t *p_filter, block_t *p_block ) ...@@ -466,7 +466,7 @@ static block_t *Convert( filter_t *p_filter, block_t *p_block )
i_out_size = p_block->i_nb_samples * p_filter->p_sys->i_bitspersample/8 * i_out_size = p_block->i_nb_samples * p_filter->p_sys->i_bitspersample/8 *
aout_FormatNbChannels( &(p_filter->fmt_out.audio) ); aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size ); p_out = block_Alloc( i_out_size );
if( !p_out ) if( !p_out )
{ {
msg_Warn( p_filter, "can't get output buffer" ); msg_Warn( p_filter, "can't get output buffer" );
......
...@@ -369,7 +369,7 @@ static block_t *Convert( filter_t *p_filter, block_t *p_block ) ...@@ -369,7 +369,7 @@ static block_t *Convert( filter_t *p_filter, block_t *p_block )
p_filter->fmt_out.audio.i_bitspersample * p_filter->fmt_out.audio.i_bitspersample *
p_filter->fmt_out.audio.i_channels / 8; p_filter->fmt_out.audio.i_channels / 8;
block_t *p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size ); block_t *p_out = block_Alloc( i_out_size );
if( !p_out ) if( !p_out )
{ {
msg_Warn( p_filter, "can't get output buffer" ); msg_Warn( p_filter, "can't get output buffer" );
......
...@@ -236,7 +236,7 @@ static block_t *Convert( filter_t *p_filter, block_t *p_block ) ...@@ -236,7 +236,7 @@ static block_t *Convert( filter_t *p_filter, block_t *p_block )
p_filter->fmt_out.audio.i_bitspersample * p_filter->fmt_out.audio.i_bitspersample *
p_filter->fmt_out.audio.i_channels / 8; p_filter->fmt_out.audio.i_channels / 8;
block_t *p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size ); block_t *p_out = block_Alloc( i_out_size );
if( !p_out ) if( !p_out )
{ {
msg_Warn( p_filter, "can't get output buffer" ); msg_Warn( p_filter, "can't get output buffer" );
......
...@@ -62,12 +62,6 @@ static inline void audio_timer_close( encoder_t * p_encoder ) ...@@ -62,12 +62,6 @@ static inline void audio_timer_close( encoder_t * p_encoder )
stats_TimerClean( p_encoder, STATS_TIMER_AUDIO_FRAME_ENCODING ); stats_TimerClean( p_encoder, STATS_TIMER_AUDIO_FRAME_ENCODING );
} }
static block_t *transcode_audio_alloc( filter_t *p_filter, int size )
{
VLC_UNUSED( p_filter );
return block_Alloc( size );
}
static aout_buffer_t *audio_new_buffer( decoder_t *p_dec, int i_samples ) static aout_buffer_t *audio_new_buffer( decoder_t *p_dec, int i_samples )
{ {
block_t *p_block; block_t *p_block;
...@@ -97,8 +91,8 @@ static aout_buffer_t *audio_new_buffer( decoder_t *p_dec, int i_samples ) ...@@ -97,8 +91,8 @@ static aout_buffer_t *audio_new_buffer( decoder_t *p_dec, int i_samples )
static int transcode_audio_filter_allocation_init( filter_t *p_filter, static int transcode_audio_filter_allocation_init( filter_t *p_filter,
void *data ) void *data )
{ {
VLC_UNUSED(p_filter);
VLC_UNUSED(data); VLC_UNUSED(data);
p_filter->pf_audio_buffer_new = transcode_audio_alloc;
return VLC_SUCCESS; return VLC_SUCCESS;
} }
......
...@@ -46,8 +46,6 @@ struct filter_owner_sys_t ...@@ -46,8 +46,6 @@ struct filter_owner_sys_t
aout_input_t *p_input; aout_input_t *p_input;
}; };
block_t *aout_FilterBufferNew( filter_t *, int );
/** an input stream for the audio output */ /** an input stream for the audio output */
struct aout_input_t struct aout_input_t
{ {
......
...@@ -40,12 +40,6 @@ ...@@ -40,12 +40,6 @@
#include "aout_internal.h" #include "aout_internal.h"
#include <libvlc.h> #include <libvlc.h>
block_t *aout_FilterBufferNew( filter_t *p_filter, int size )
{
(void) p_filter;
return block_Alloc( size );
}
/***************************************************************************** /*****************************************************************************
* FindFilter: find an audio filter for a specific transformation * FindFilter: find an audio filter for a specific transformation
*****************************************************************************/ *****************************************************************************/
...@@ -66,7 +60,6 @@ static filter_t * FindFilter( vlc_object_t *obj, ...@@ -66,7 +60,6 @@ static filter_t * FindFilter( vlc_object_t *obj,
memcpy( &p_filter->fmt_out.audio, p_output_format, memcpy( &p_filter->fmt_out.audio, p_output_format,
sizeof(audio_sample_format_t) ); sizeof(audio_sample_format_t) );
p_filter->fmt_out.i_codec = p_output_format->i_format; p_filter->fmt_out.i_codec = p_output_format->i_format;
p_filter->pf_audio_buffer_new = aout_FilterBufferNew;
p_filter->p_owner = NULL; p_filter->p_owner = NULL;
p_filter->p_module = module_need( p_filter, "audio filter", NULL, false ); p_filter->p_module = module_need( p_filter, "audio filter", NULL, false );
......
...@@ -159,7 +159,6 @@ aout_input_t *aout_InputNew (audio_output_t * p_aout, ...@@ -159,7 +159,6 @@ aout_input_t *aout_InputNew (audio_output_t * p_aout,
memcpy( &p_filter->fmt_out.audio, &chain_output_format, memcpy( &p_filter->fmt_out.audio, &chain_output_format,
sizeof(audio_sample_format_t) ); sizeof(audio_sample_format_t) );
p_filter->fmt_out.i_codec = chain_output_format.i_format; p_filter->fmt_out.i_codec = chain_output_format.i_format;
p_filter->pf_audio_buffer_new = aout_FilterBufferNew;
/* try to find the requested filter */ /* try to find the requested filter */
if( i_visual == 2 ) /* this can only be a visualization module */ if( i_visual == 2 ) /* this can only be a visualization module */
......
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