From f4d3c8500bd118237f9b37800f02a19f419deb5c Mon Sep 17 00:00:00 2001 From: =?utf8?q?R=C3=A9mi=20Denis-Courmont?= Date: Sun, 27 Sep 2009 20:20:10 +0300 Subject: [PATCH] band-limited resampler: switch to audio filter2 --- modules/audio_filter/resampler/bandlimited.c | 213 ++++--------------- 1 file changed, 37 insertions(+), 176 deletions(-) diff --git a/modules/audio_filter/resampler/bandlimited.c b/modules/audio_filter/resampler/bandlimited.c index 3e1b7915b3..6511898f8e 100644 --- a/modules/audio_filter/resampler/bandlimited.c +++ b/modules/audio_filter/resampler/bandlimited.c @@ -48,10 +48,6 @@ /***************************************************************************** * Local prototypes *****************************************************************************/ -static int Create ( vlc_object_t * ); -static void Close ( vlc_object_t * ); -static void DoWork ( aout_instance_t *, aout_filter_t *, aout_buffer_t *, - aout_buffer_t * ); /* audio filter2 */ static int OpenFilter ( vlc_object_t * ); @@ -81,10 +77,9 @@ struct filter_sys_t int i_old_wing; unsigned int i_remainder; /* remainder of previous sample */ + bool b_first; date_t end_date; - - bool b_filter2; }; /***************************************************************************** @@ -94,112 +89,29 @@ vlc_module_begin () set_category( CAT_AUDIO ) set_subcategory( SUBCAT_AUDIO_MISC ) set_description( N_("Audio filter for band-limited interpolation resampling") ) - set_capability( "audio filter", 20 ) - set_callbacks( Create, Close ) - - add_submodule () - set_description( N_("Audio filter for band-limited interpolation resampling") ) set_capability( "audio filter2", 20 ) set_callbacks( OpenFilter, CloseFilter ) vlc_module_end () /***************************************************************************** - * Create: allocate linear resampler + * Resample: convert a buffer *****************************************************************************/ -static int Create( vlc_object_t *p_this ) +static block_t *Resample( filter_t * p_filter, block_t * p_in_buf ) { - aout_filter_t * p_filter = (aout_filter_t *)p_this; - struct filter_sys_t * p_sys; - double d_factor; - int i_filter_wing; - - if ( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate - || p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format - || p_filter->fmt_in.audio.i_physical_channels - != p_filter->fmt_out.audio.i_physical_channels - || p_filter->fmt_in.audio.i_original_channels - != p_filter->fmt_out.audio.i_original_channels - || p_filter->fmt_in.audio.i_format != VLC_CODEC_FL32 ) - { - return VLC_EGENERIC; - } - -#if !defined( __APPLE__ ) - if( !config_GetInt( p_this, "hq-resampling" ) ) + if( !p_in_buf || !p_in_buf->i_nb_samples ) { - return VLC_EGENERIC; - } -#endif - - /* Allocate the memory needed to store the module's structure */ - p_sys = malloc( sizeof(filter_sys_t) ); - if( p_sys == NULL ) - return VLC_ENOMEM; - p_filter->p_sys = (struct aout_filter_sys_t *)p_sys; - - /* Calculate worst case for the length of the filter wing */ - d_factor = (double)p_filter->fmt_out.audio.i_rate - / p_filter->fmt_in.audio.i_rate / AOUT_MAX_INPUT_RATE; - i_filter_wing = ((SMALL_FILTER_NMULT + 1)/2.0) - * __MAX(1.0, 1.0/d_factor) + 10; - p_sys->i_buf_size = aout_FormatNbChannels( &p_filter->fmt_in.audio ) * - sizeof(int32_t) * 2 * i_filter_wing; - - /* Allocate enough memory to buffer previous samples */ - p_sys->p_buf = malloc( p_sys->i_buf_size ); - if( p_sys->p_buf == NULL ) - { - free( p_sys ); - return VLC_ENOMEM; + if( p_in_buf ) + block_Release( p_in_buf ); + return NULL; } - p_sys->i_old_wing = 0; - p_sys->b_filter2 = false; /* It seams to be a good valuefor this module */ - p_filter->pf_do_work = DoWork; - - /* We don't want a new buffer to be created because we're not sure we'll - * actually need to resample anything. */ - p_filter->b_in_place = true; - - return VLC_SUCCESS; -} - -/***************************************************************************** - * Close: free our resources - *****************************************************************************/ -static void Close( vlc_object_t * p_this ) -{ - aout_filter_t * p_filter = (aout_filter_t *)p_this; - filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys; - free( p_sys->p_buf ); - free( p_sys ); -} - -/***************************************************************************** - * DoWork: convert a buffer - *****************************************************************************/ -static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter, - aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf ) -{ - filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys; - float *p_out = (float *)p_out_buf->p_buffer; - + filter_sys_t *p_sys = p_filter->p_sys; + unsigned int i_out_rate = p_filter->fmt_out.audio.i_rate; int i_nb_channels = aout_FormatNbChannels( &p_filter->fmt_in.audio ); - int i_in_nb = p_in_buf->i_nb_samples; - int i_in, i_out = 0; - unsigned int i_out_rate; - double d_factor, d_scale_factor, d_old_scale_factor; - int i_filter_wing; - - if( p_sys->b_filter2 ) - i_out_rate = p_filter->fmt_out.audio.i_rate; - else - i_out_rate = p_aout->mixer_format.i_rate; /* Check if we really need to run the resampler */ if( i_out_rate == p_filter->fmt_in.audio.i_rate ) { -#if 0 /* FIXME: needs audio filter2 to use block_Realloc */ if( /*p_filter->b_continuity && /--* What difference does it make ? :) */ p_sys->i_old_wing ) { @@ -208,30 +120,35 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter, p_sys->i_old_wing * p_filter->fmt_in.audio.i_bytes_per_frame, p_in_buf->i_buffer ); if( !p_in_buf ) - abort(); + return NULL; memcpy( p_in_buf->p_buffer, p_sys->p_buf + i_nb_channels * p_sys->i_old_wing, p_sys->i_old_wing * p_filter->fmt_in.audio.i_bytes_per_frame ); - p_out_buf->i_nb_samples = p_in_buf->i_nb_samples + - p_sys->i_old_wing; + p_in_buf->i_nb_samples += p_sys->i_old_wing; - p_out_buf->i_pts = date_Get( &p_sys->end_date ); - p_out_buf->i_length = + p_in_buf->i_pts = date_Get( &p_sys->end_date ); + p_in_buf->i_length = date_Increment( &p_sys->end_date, - p_out_buf->i_nb_samples ) - p_out_buf->i_pts; - - p_out_buf->i_buffer = p_out_buf->i_nb_samples * - p_filter->fmt_in.audio.i_bytes_per_frame; + p_in_buf->i_nb_samples ) - p_in_buf->i_pts; } -#endif - p_out_buf->i_flags |= BLOCK_FLAG_DISCONTINUITY; + p_in_buf->i_flags |= BLOCK_FLAG_DISCONTINUITY; p_sys->i_old_wing = 0; - return; + return p_in_buf; } - if( p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY ) + unsigned i_bytes_per_frame = p_filter->fmt_out.audio.i_channels * + p_filter->fmt_out.audio.i_bitspersample / 8; + size_t i_out_size = i_bytes_per_frame * ( 1 + ( p_in_buf->i_nb_samples * + p_filter->fmt_out.audio.i_rate / p_filter->fmt_in.audio.i_rate) ) + + p_filter->p_sys->i_buf_size; + block_t *p_out_buf = filter_NewAudioBuffer( p_filter, i_out_size ); + if( !p_out_buf ) + return NULL; + float *p_out = (float *)p_out_buf->p_buffer; + + if( (p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY) || p_sys->b_first ) { /* Continuity in sound samples has been broken, we'd better reset * everything. */ @@ -241,8 +158,14 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter, date_Set( &p_sys->end_date, p_in_buf->i_pts ); p_sys->d_old_factor = 1; p_sys->i_old_wing = 0; + p_sys->b_first = false; } + int i_in_nb = p_in_buf->i_nb_samples; + int i_in, i_out = 0; + double d_factor, d_scale_factor, d_old_scale_factor; + int i_filter_wing; + #if 0 msg_Err( p_filter, "old rate: %i, old factor: %f, old wing: %i, i_in: %i", p_sys->i_old_rate, p_sys->d_old_factor, @@ -262,9 +185,11 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter, p_sys->i_old_wing * 2 * p_filter->fmt_in.audio.i_bytes_per_frame ); } + /* XXX: why i_nb_channels instead of i_bytes_per_frame??? */ vlc_memcpy( p_in + p_sys->i_old_wing * 2 * i_nb_channels, p_in_buf->p_buffer, p_in_buf->i_nb_samples * p_filter->fmt_in.audio.i_bytes_per_frame ); + block_Release( p_in_buf ); /* Make sure the output buffer is reset */ memset( p_out, 0, p_out_buf->i_buffer ); @@ -457,7 +382,7 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter, p_out_buf->i_buffer = p_out_buf->i_nb_samples * i_nb_channels * sizeof(int32_t); - + return p_out_buf; } /***************************************************************************** @@ -505,7 +430,7 @@ static int OpenFilter( vlc_object_t *p_this ) } p_sys->i_old_wing = 0; - p_sys->b_filter2 = true; + p_sys->b_first = true; p_filter->pf_audio_filter = Resample; msg_Dbg( p_this, "%4.4s/%iKHz/%i->%4.4s/%iKHz/%i", @@ -532,70 +457,6 @@ static void CloseFilter( vlc_object_t *p_this ) free( p_filter->p_sys ); } -/***************************************************************************** - * Resample - *****************************************************************************/ -static block_t *Resample( filter_t *p_filter, block_t *p_block ) -{ - aout_filter_t aout_filter; - aout_buffer_t in_buf, out_buf; - block_t *p_out; - int i_out_size; - int i_bytes_per_frame; - - if( !p_block || !p_block->i_nb_samples ) - { - if( p_block ) - block_Release( p_block ); - return NULL; - } - - i_bytes_per_frame = p_filter->fmt_out.audio.i_channels * - p_filter->fmt_out.audio.i_bitspersample / 8; - - i_out_size = i_bytes_per_frame * ( 1 + ( p_block->i_nb_samples * - p_filter->fmt_out.audio.i_rate / - p_filter->fmt_in.audio.i_rate) ) + - p_filter->p_sys->i_buf_size; - - p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size ); - if( !p_out ) - { - msg_Warn( p_filter, "can't get output buffer" ); - block_Release( p_block ); - return NULL; - } - - p_out->i_nb_samples = i_out_size / i_bytes_per_frame; - p_out->i_dts = p_block->i_dts; - p_out->i_pts = p_block->i_pts; - p_out->i_length = p_block->i_length; - - aout_filter.p_sys = (struct aout_filter_sys_t *)p_filter->p_sys; - aout_filter.fmt_in.audio = p_filter->fmt_in.audio; - aout_filter.fmt_in.audio.i_bytes_per_frame = p_filter->fmt_in.audio.i_channels * - p_filter->fmt_in.audio.i_bitspersample / 8; - aout_filter.fmt_out.audio = p_filter->fmt_out.audio; - aout_filter.fmt_out.audio.i_bytes_per_frame = p_filter->fmt_out.audio.i_channels * - p_filter->fmt_out.audio.i_bitspersample / 8; - - in_buf.p_buffer = p_block->p_buffer; - in_buf.i_buffer = p_block->i_buffer; - in_buf.i_nb_samples = p_block->i_nb_samples; - out_buf.p_buffer = p_out->p_buffer; - out_buf.i_buffer = p_out->i_buffer; - out_buf.i_nb_samples = p_out->i_nb_samples; - - DoWork( (aout_instance_t *)p_filter, &aout_filter, &in_buf, &out_buf ); - - block_Release( p_block ); - - p_out->i_buffer = out_buf.i_buffer; - p_out->i_nb_samples = out_buf.i_nb_samples; - - return p_out; -} - void FilterFloatUP( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in, float *p_out, uint32_t ui_remainder, uint32_t ui_output_rate, int16_t Inc, int i_nb_channels ) -- 2.39.2