X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=modules%2Faudio_filter%2Fresampler%2Fbandlimited.c;h=0fba703f447eaeed348dae654a1472c6550b575d;hb=d8e4bc56fd3fe95a18abccddc7f3943b00575c5e;hp=485f320150f86368c0ec538b9c6d4f7a03587ed5;hpb=14f37b2101842fa6e427f962f689db74eff6faba;p=vlc diff --git a/modules/audio_filter/resampler/bandlimited.c b/modules/audio_filter/resampler/bandlimited.c index 485f320150..0fba703f44 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 * ); @@ -77,16 +73,13 @@ struct filter_sys_t int32_t *p_buf; /* this filter introduces a delay */ int i_buf_size; - int i_old_rate; double d_old_factor; int i_old_wing; unsigned int i_remainder; /* remainder of previous sample */ - - audio_date_t end_date; - bool b_first; - bool b_filter2; + + date_t end_date; }; /***************************************************************************** @@ -96,155 +89,82 @@ 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->input.i_rate == p_filter->output.i_rate - || p_filter->input.i_format != p_filter->output.i_format - || p_filter->input.i_physical_channels - != p_filter->output.i_physical_channels - || p_filter->input.i_original_channels - != p_filter->output.i_original_channels - || p_filter->input.i_format != VLC_CODEC_FL32 ) + if( !p_in_buf || !p_in_buf->i_nb_samples ) { - return VLC_EGENERIC; - } - -#if !defined( __APPLE__ ) - if( !config_GetInt( p_this, "hq-resampling" ) ) - { - 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->output.i_rate - / p_filter->input.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->input ) * - 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; - - int i_nb_channels = aout_FormatNbChannels( &p_filter->input ); - 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->output.i_rate; - else - i_out_rate = p_aout->mixer.mixer.i_rate; + 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 ); /* Check if we really need to run the resampler */ - if( i_out_rate == p_filter->input.i_rate ) + if( i_out_rate == p_filter->fmt_in.audio.i_rate ) { - if( /*p_filter->b_continuity && /--* What difference does it make ? :) */ - p_sys->i_old_wing && - p_in_buf->i_size >= - p_in_buf->i_nb_bytes + p_sys->i_old_wing * - p_filter->input.i_bytes_per_frame ) + if( !(p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY) && + p_sys->i_old_wing ) { /* output the whole thing with the samples from last time */ - memmove( ((float *)(p_in_buf->p_buffer)) + - i_nb_channels * p_sys->i_old_wing, - p_in_buf->p_buffer, p_in_buf->i_nb_bytes ); + p_in_buf = block_Realloc( p_in_buf, + p_sys->i_old_wing * p_filter->fmt_in.audio.i_bytes_per_frame, + p_in_buf->i_buffer ); + if( !p_in_buf ) + 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->input.i_bytes_per_frame ); - - p_out_buf->i_nb_samples = p_in_buf->i_nb_samples + - p_sys->i_old_wing; + p_filter->fmt_in.audio.i_bytes_per_frame ); - p_out_buf->start_date = aout_DateGet( &p_sys->end_date ); - p_out_buf->end_date = - aout_DateIncrement( &p_sys->end_date, - p_out_buf->i_nb_samples ); + p_in_buf->i_nb_samples += p_sys->i_old_wing; - p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples * - p_filter->input.i_bytes_per_frame; + p_in_buf->i_pts = date_Get( &p_sys->end_date ); + p_in_buf->i_length = + date_Increment( &p_sys->end_date, + p_in_buf->i_nb_samples ) - p_in_buf->i_pts; } - p_filter->b_continuity = false; p_sys->i_old_wing = 0; - return; + return p_in_buf; } - if( !p_filter->b_continuity ) + 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. */ - p_filter->b_continuity = true; + p_out_buf->i_flags |= BLOCK_FLAG_DISCONTINUITY; p_sys->i_remainder = 0; - aout_DateInit( &p_sys->end_date, i_out_rate ); - aout_DateSet( &p_sys->end_date, p_in_buf->start_date ); - p_sys->i_old_rate = p_filter->input.i_rate; + date_Init( &p_sys->end_date, i_out_rate, 1 ); + 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, @@ -254,7 +174,7 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter, /* Prepare the source buffer */ i_in_nb += (p_sys->i_old_wing * 2); - float p_in_orig[i_in_nb * p_filter->input.i_bytes_per_frame / 4], + float p_in_orig[i_in_nb * p_filter->fmt_in.audio.i_bytes_per_frame / 4], *p_in = p_in_orig; /* Copy all our samples in p_in */ @@ -262,17 +182,19 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter, { vlc_memcpy( p_in, p_sys->p_buf, p_sys->i_old_wing * 2 * - p_filter->input.i_bytes_per_frame ); + 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->input.i_bytes_per_frame ); + 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_size ); + memset( p_out, 0, p_out_buf->i_buffer ); /* Calculate the new length of the filter wing */ - d_factor = (double)i_out_rate / p_filter->input.i_rate; + d_factor = (double)i_out_rate / p_filter->fmt_in.audio.i_rate; i_filter_wing = ((SMALL_FILTER_NMULT+1)/2.0) * __MAX(1.0,1.0/d_factor) + 1; /* Account for increased filter gain when using factors less than 1 */ @@ -290,14 +212,14 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter, { /* Just copy the samples */ memcpy( p_out, p_in, - p_filter->input.i_bytes_per_frame ); + p_filter->fmt_in.audio.i_bytes_per_frame ); p_in += i_nb_channels; p_out += i_nb_channels; i_out++; continue; } - while( p_sys->i_remainder < p_filter->output.i_rate ) + while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate ) { if( p_sys->d_old_factor >= 1 ) @@ -308,14 +230,14 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter, FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD, SMALL_FILTER_NWING, p_in, p_out, p_sys->i_remainder, - p_filter->output.i_rate, + p_filter->fmt_out.audio.i_rate, -1, i_nb_channels ); /* Perform right-wing inner product */ FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD, SMALL_FILTER_NWING, p_in + i_nb_channels, p_out, - p_filter->output.i_rate - + p_filter->fmt_out.audio.i_rate - p_sys->i_remainder, - p_filter->output.i_rate, + p_filter->fmt_out.audio.i_rate, 1, i_nb_channels ); #if 0 @@ -327,12 +249,12 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter, #endif /* Sanity check */ - if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame + if( p_out_buf->i_buffer/p_filter->fmt_in.audio.i_bytes_per_frame <= (unsigned int)i_out+1 ) { p_out += i_nb_channels; i_out++; - p_sys->i_remainder += p_filter->input.i_rate; + p_sys->i_remainder += p_filter->fmt_in.audio.i_rate; break; } } @@ -342,37 +264,36 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter, FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD, SMALL_FILTER_NWING, p_in, p_out, p_sys->i_remainder, - p_filter->output.i_rate, p_filter->input.i_rate, + p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate, -1, i_nb_channels ); /* Perform right-wing inner product */ FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD, SMALL_FILTER_NWING, p_in + i_nb_channels, p_out, - p_filter->output.i_rate - + p_filter->fmt_out.audio.i_rate - p_sys->i_remainder, - p_filter->output.i_rate, p_filter->input.i_rate, + p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate, 1, i_nb_channels ); } p_out += i_nb_channels; i_out++; - p_sys->i_remainder += p_filter->input.i_rate; + p_sys->i_remainder += p_filter->fmt_in.audio.i_rate; } p_in += i_nb_channels; - p_sys->i_remainder -= p_filter->output.i_rate; + p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate; } /* Apply the new rate for the rest of the samples */ if( i_in < i_in_nb - i_filter_wing ) { - p_sys->i_old_rate = p_filter->input.i_rate; p_sys->d_old_factor = d_factor; p_sys->i_old_wing = i_filter_wing; } for( ; i_in < i_in_nb - i_filter_wing; i_in++ ) { - while( p_sys->i_remainder < p_filter->output.i_rate ) + while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate ) { if( d_factor >= 1 ) @@ -383,15 +304,15 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter, FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD, SMALL_FILTER_NWING, p_in, p_out, p_sys->i_remainder, - p_filter->output.i_rate, + p_filter->fmt_out.audio.i_rate, -1, i_nb_channels ); /* Perform right-wing inner product */ FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD, SMALL_FILTER_NWING, p_in + i_nb_channels, p_out, - p_filter->output.i_rate - + p_filter->fmt_out.audio.i_rate - p_sys->i_remainder, - p_filter->output.i_rate, + p_filter->fmt_out.audio.i_rate, 1, i_nb_channels ); #if 0 @@ -402,12 +323,12 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter, } #endif /* Sanity check */ - if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame + if( p_out_buf->i_buffer/p_filter->fmt_in.audio.i_bytes_per_frame <= (unsigned int)i_out+1 ) { p_out += i_nb_channels; i_out++; - p_sys->i_remainder += p_filter->input.i_rate; + p_sys->i_remainder += p_filter->fmt_in.audio.i_rate; break; } } @@ -417,25 +338,25 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter, FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD, SMALL_FILTER_NWING, p_in, p_out, p_sys->i_remainder, - p_filter->output.i_rate, p_filter->input.i_rate, + p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate, -1, i_nb_channels ); /* Perform right-wing inner product */ FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD, SMALL_FILTER_NWING, p_in + i_nb_channels, p_out, - p_filter->output.i_rate - + p_filter->fmt_out.audio.i_rate - p_sys->i_remainder, - p_filter->output.i_rate, p_filter->input.i_rate, + p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate, 1, i_nb_channels ); } p_out += i_nb_channels; i_out++; - p_sys->i_remainder += p_filter->input.i_rate; + p_sys->i_remainder += p_filter->fmt_in.audio.i_rate; } p_in += i_nb_channels; - p_sys->i_remainder -= p_filter->output.i_rate; + p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate; } /* Buffer i_filter_wing * 2 samples for next time */ @@ -444,23 +365,23 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter, memcpy( p_sys->p_buf, p_in_orig + (i_in_nb - 2 * p_sys->i_old_wing) * i_nb_channels, (2 * p_sys->i_old_wing) * - p_filter->input.i_bytes_per_frame ); + p_filter->fmt_in.audio.i_bytes_per_frame ); } #if 0 - msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_size, - i_out * p_filter->input.i_bytes_per_frame ); + msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_buffer, + i_out * p_filter->fmt_in.audio.i_bytes_per_frame ); #endif /* Finalize aout buffer */ p_out_buf->i_nb_samples = i_out; - p_out_buf->start_date = aout_DateGet( &p_sys->end_date ); - p_out_buf->end_date = aout_DateIncrement( &p_sys->end_date, - p_out_buf->i_nb_samples ); + p_out_buf->i_pts = date_Get( &p_sys->end_date ); + p_out_buf->i_length = date_Increment( &p_sys->end_date, + p_out_buf->i_nb_samples ) - p_out_buf->i_pts; - p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples * + p_out_buf->i_buffer = p_out_buf->i_nb_samples * i_nb_channels * sizeof(int32_t); - + return p_out_buf; } /***************************************************************************** @@ -496,20 +417,19 @@ static int OpenFilter( vlc_object_t *p_this ) d_factor = (double)i_out_rate / p_filter->fmt_in.audio.i_rate; i_filter_wing = ((SMALL_FILTER_NMULT + 1)/2.0) * __MAX(1.0, 1.0/d_factor) + 10; - p_filter->p_sys->i_buf_size = p_filter->fmt_in.audio.i_channels * + p_sys->i_buf_size = p_filter->fmt_in.audio.i_channels * sizeof(int32_t) * 2 * i_filter_wing; /* Allocate enough memory to buffer previous samples */ - p_filter->p_sys->p_buf = malloc( p_filter->p_sys->i_buf_size ); - if( p_filter->p_sys->p_buf == NULL ) + p_sys->p_buf = malloc( p_sys->i_buf_size ); + if( p_sys->p_buf == NULL ) { free( p_sys ); return VLC_ENOMEM; } - p_filter->p_sys->i_old_wing = 0; + p_sys->i_old_wing = 0; p_sys->b_first = true; - p_sys->b_filter2 = true; p_filter->pf_audio_filter = Resample; msg_Dbg( p_this, "%4.4s/%iKHz/%i->%4.4s/%iKHz/%i", @@ -536,72 +456,6 @@ 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_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_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_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.input = p_filter->fmt_in.audio; - aout_filter.input.i_bytes_per_frame = p_filter->fmt_in.audio.i_channels * - p_filter->fmt_in.audio.i_bitspersample / 8; - aout_filter.output = p_filter->fmt_out.audio; - aout_filter.output.i_bytes_per_frame = p_filter->fmt_out.audio.i_channels * - p_filter->fmt_out.audio.i_bitspersample / 8; - aout_filter.b_continuity = !p_filter->p_sys->b_first; - p_filter->p_sys->b_first = false; - - in_buf.p_buffer = p_block->p_buffer; - in_buf.i_nb_bytes = in_buf.i_size = p_block->i_buffer; - in_buf.i_nb_samples = p_block->i_samples; - out_buf.p_buffer = p_out->p_buffer; - out_buf.i_nb_bytes = out_buf.i_size = p_out->i_buffer; - out_buf.i_nb_samples = p_out->i_samples; - - DoWork( (aout_instance_t *)p_filter, &aout_filter, &in_buf, &out_buf ); - - block_Release( p_block ); - - p_out->i_buffer = out_buf.i_nb_bytes; - p_out->i_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 )