X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=modules%2Faudio_filter%2Fchannel_mixer%2Fheadphone.c;h=de6858698635f3cee4f8d214123aa29aee17cd31;hb=2aa61dc55a77f535c3e70104b6ec2167766f8499;hp=c659cdf11d6e96a54aea567799775b62992f74f5;hpb=fd7f8f854a13f7b65cb2b8622755aa47ee932919;p=vlc diff --git a/modules/audio_filter/channel_mixer/headphone.c b/modules/audio_filter/channel_mixer/headphone.c index c659cdf11d..de68586986 100644 --- a/modules/audio_filter/channel_mixer/headphone.c +++ b/modules/audio_filter/channel_mixer/headphone.c @@ -7,19 +7,19 @@ * * Authors: Boris Dorès * - * 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 - * (at your option) any later version. + * 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 (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. + * 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. * - * 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 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. *****************************************************************************/ /***************************************************************************** @@ -41,13 +41,6 @@ /***************************************************************************** * Local prototypes *****************************************************************************/ -static int Create ( vlc_object_t * ); -static void Destroy ( 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 * ); static void CloseFilter( vlc_object_t * ); static block_t *Convert( filter_t *, block_t * ); @@ -87,19 +80,12 @@ vlc_module_begin () add_integer( "headphone-dim", 10, NULL, HEADPHONE_DIM_TEXT, HEADPHONE_DIM_LONGTEXT, false ) - add_bool( "headphone-compensate", 0, NULL, HEADPHONE_COMPENSATE_TEXT, + add_bool( "headphone-compensate", false, NULL, HEADPHONE_COMPENSATE_TEXT, HEADPHONE_COMPENSATE_LONGTEXT, true ) - add_bool( "headphone-dolby", 0, NULL, HEADPHONE_DOLBY_TEXT, + add_bool( "headphone-dolby", false, NULL, HEADPHONE_DOLBY_TEXT, HEADPHONE_DOLBY_LONGTEXT, true ) set_capability( "audio filter", 0 ) - set_callbacks( Create, Destroy ) - add_shortcut( "headphone" ) - - /* Audio filter 2 */ - add_submodule () - set_description( N_("Headphone virtual spatialization effect") ) - set_capability( "audio filter2", 0 ) set_callbacks( OpenFilter, CloseFilter ) add_shortcut( "headphone" ) vlc_module_end () @@ -116,14 +102,6 @@ struct atomic_operation_t double d_amplitude_factor; }; -struct aout_filter_sys_t -{ - size_t i_overflow_buffer_size;/* in bytes */ - uint8_t * p_overflow_buffer; - unsigned int i_nb_atomic_operations; - struct atomic_operation_t * p_atomic_operations; -}; - struct filter_sys_t { size_t i_overflow_buffer_size;/* in bytes */ @@ -150,7 +128,7 @@ struct filter_sys_t * * x-axis * */ -static void ComputeChannelOperations( struct aout_filter_sys_t * p_data +static void ComputeChannelOperations( struct filter_sys_t * p_data , unsigned int i_rate, unsigned int i_next_atomic_operation , int i_source_channel_offset, double d_x, double d_z , double d_compensation_length, double d_channel_amplitude_factor ) @@ -207,7 +185,7 @@ static void ComputeChannelOperations( struct aout_filter_sys_t * p_data } } -static int Init( vlc_object_t *p_this, struct aout_filter_sys_t * p_data +static int Init( vlc_object_t *p_this, struct filter_sys_t * p_data , unsigned int i_nb_channels, uint32_t i_physical_channels , unsigned int i_rate ) { @@ -348,108 +326,15 @@ static int Init( vlc_object_t *p_this, struct aout_filter_sys_t * p_data return 0; } -/***************************************************************************** - * Create: allocate headphone downmixer - *****************************************************************************/ -static int Create( vlc_object_t *p_this ) -{ - aout_filter_t * p_filter = (aout_filter_t *)p_this; - bool b_fit = true; - - /* Activate this filter only with stereo devices */ - if( p_filter->output.i_physical_channels - != (AOUT_CHAN_LEFT|AOUT_CHAN_RIGHT) ) - { - msg_Dbg( p_filter, "filter discarded (incompatible format)" ); - return VLC_EGENERIC; - } - - /* Request a specific format if not already compatible */ - if( p_filter->input.i_original_channels - != p_filter->output.i_original_channels ) - { - b_fit = false; - p_filter->input.i_original_channels = - p_filter->output.i_original_channels; - } - if( p_filter->input.i_format != VLC_CODEC_FL32 - || p_filter->output.i_format != VLC_CODEC_FL32 ) - { - b_fit = false; - p_filter->input.i_format = VLC_CODEC_FL32; - p_filter->output.i_format = VLC_CODEC_FL32; - } - if( p_filter->input.i_rate != p_filter->output.i_rate ) - { - b_fit = false; - p_filter->input.i_rate = p_filter->output.i_rate; - } - if( p_filter->input.i_physical_channels == (AOUT_CHAN_LEFT|AOUT_CHAN_RIGHT) - && ( p_filter->input.i_original_channels & AOUT_CHAN_DOLBYSTEREO ) - && ! config_GetInt ( p_filter , "headphone-dolby" ) ) - { - b_fit = false; - p_filter->input.i_physical_channels = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | - AOUT_CHAN_CENTER | - AOUT_CHAN_REARLEFT | - AOUT_CHAN_REARRIGHT; - } - - if( !b_fit ) - { - msg_Dbg( p_filter, "requesting specific format" ); - return VLC_EGENERIC; - } - - /* Allocate the memory needed to store the module's structure */ - p_filter->p_sys = malloc( sizeof(struct aout_filter_sys_t) ); - if( p_filter->p_sys == NULL ) - return VLC_ENOMEM; - p_filter->p_sys->i_overflow_buffer_size = 0; - p_filter->p_sys->p_overflow_buffer = NULL; - p_filter->p_sys->i_nb_atomic_operations = 0; - p_filter->p_sys->p_atomic_operations = NULL; - - if( Init( VLC_OBJECT(p_filter), p_filter->p_sys - , aout_FormatNbChannels ( &p_filter->input ) - , p_filter->input.i_physical_channels - , p_filter->input.i_rate ) < 0 ) - { - free( p_filter->p_sys ); - return VLC_EGENERIC; - } - - p_filter->pf_do_work = DoWork; - p_filter->b_in_place = 0; - - return VLC_SUCCESS; -} - -/***************************************************************************** - * Destroy: deallocate resources associated with headphone downmixer - *****************************************************************************/ -static void Destroy( vlc_object_t *p_this ) -{ - aout_filter_t * p_filter = (aout_filter_t *)p_this; - - if( p_filter->p_sys != NULL ) - { - free( p_filter->p_sys->p_overflow_buffer ); - free( p_filter->p_sys->p_atomic_operations ); - free( p_filter->p_sys ); - p_filter->p_sys = NULL; - } -} - /***************************************************************************** * DoWork: convert a buffer *****************************************************************************/ -static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter, +static void DoWork( filter_t * p_filter, aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf ) { - VLC_UNUSED(p_aout); - int i_input_nb = aout_FormatNbChannels( &p_filter->input ); - int i_output_nb = aout_FormatNbChannels( &p_filter->output ); + filter_sys_t *p_sys = p_filter->p_sys; + int i_input_nb = aout_FormatNbChannels( &p_filter->fmt_in.audio ); + int i_output_nb = aout_FormatNbChannels( &p_filter->fmt_out.audio ); float * p_in = (float*) p_in_buf->p_buffer; uint8_t * p_out; @@ -468,89 +353,82 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter, /* out buffer characterisitcs */ p_out_buf->i_nb_samples = p_in_buf->i_nb_samples; - p_out_buf->i_nb_bytes = p_in_buf->i_nb_bytes * i_output_nb / i_input_nb; + p_out_buf->i_buffer = p_in_buf->i_buffer * i_output_nb / i_input_nb; p_out = p_out_buf->p_buffer; - i_out_size = p_out_buf->i_nb_bytes; + i_out_size = p_out_buf->i_buffer; - if( p_filter->p_sys != NULL ) - { - /* Slide the overflow buffer */ - p_overflow = p_filter->p_sys->p_overflow_buffer; - i_overflow_size = p_filter->p_sys->i_overflow_buffer_size; + /* Slide the overflow buffer */ + p_overflow = p_sys->p_overflow_buffer; + i_overflow_size = p_sys->i_overflow_buffer_size; + + memset( p_out, 0, i_out_size ); + if ( i_out_size > i_overflow_size ) + memcpy( p_out, p_overflow, i_overflow_size ); + else + memcpy( p_out, p_overflow, i_out_size ); - memset( p_out, 0, i_out_size ); - if ( i_out_size > i_overflow_size ) - memcpy( p_out, p_overflow, i_overflow_size ); + p_slide = p_sys->p_overflow_buffer; + while( p_slide < p_overflow + i_overflow_size ) + { + if( p_slide + i_out_size < p_overflow + i_overflow_size ) + { + memset( p_slide, 0, i_out_size ); + if( p_slide + 2 * i_out_size < p_overflow + i_overflow_size ) + memcpy( p_slide, p_slide + i_out_size, i_out_size ); + else + memcpy( p_slide, p_slide + i_out_size, + p_overflow + i_overflow_size - ( p_slide + i_out_size ) ); + } else - memcpy( p_out, p_overflow, i_out_size ); + { + memset( p_slide, 0, p_overflow + i_overflow_size - p_slide ); + } + p_slide += i_out_size; + } + + /* apply the atomic operations */ + for( i = 0; i < p_sys->i_nb_atomic_operations; i++ ) + { + /* shorter variable names */ + i_source_channel_offset + = p_sys->p_atomic_operations[i].i_source_channel_offset; + i_dest_channel_offset + = p_sys->p_atomic_operations[i].i_dest_channel_offset; + i_delay = p_sys->p_atomic_operations[i].i_delay; + d_amplitude_factor + = p_sys->p_atomic_operations[i].d_amplitude_factor; - p_slide = p_filter->p_sys->p_overflow_buffer; - while( p_slide < p_overflow + i_overflow_size ) + if( p_out_buf->i_nb_samples > i_delay ) { - if( p_slide + i_out_size < p_overflow + i_overflow_size ) + /* current buffer coefficients */ + for( j = 0; j < p_out_buf->i_nb_samples - i_delay; j++ ) { - memset( p_slide, 0, i_out_size ); - if( p_slide + 2 * i_out_size < p_overflow + i_overflow_size ) - memcpy( p_slide, p_slide + i_out_size, i_out_size ); - else - memcpy( p_slide, p_slide + i_out_size, - p_overflow + i_overflow_size - ( p_slide + i_out_size ) ); + ((float*)p_out)[ (i_delay+j)*i_output_nb + i_dest_channel_offset ] + += p_in[ j * i_input_nb + i_source_channel_offset ] + * d_amplitude_factor; } - else + + /* overflow buffer coefficients */ + for( j = 0; j < i_delay; j++ ) { - memset( p_slide, 0, p_overflow + i_overflow_size - p_slide ); + ((float*)p_overflow)[ j*i_output_nb + i_dest_channel_offset ] + += p_in[ (p_out_buf->i_nb_samples - i_delay + j) + * i_input_nb + i_source_channel_offset ] + * d_amplitude_factor; } - p_slide += i_out_size; } - - /* apply the atomic operations */ - for( i = 0; i < p_filter->p_sys->i_nb_atomic_operations; i++ ) + else { - /* shorter variable names */ - i_source_channel_offset - = p_filter->p_sys->p_atomic_operations[i].i_source_channel_offset; - i_dest_channel_offset - = p_filter->p_sys->p_atomic_operations[i].i_dest_channel_offset; - i_delay = p_filter->p_sys->p_atomic_operations[i].i_delay; - d_amplitude_factor - = p_filter->p_sys->p_atomic_operations[i].d_amplitude_factor; - - if( p_out_buf->i_nb_samples > i_delay ) + /* overflow buffer coefficients only */ + for( j = 0; j < p_out_buf->i_nb_samples; j++ ) { - /* current buffer coefficients */ - for( j = 0; j < p_out_buf->i_nb_samples - i_delay; j++ ) - { - ((float*)p_out)[ (i_delay+j)*i_output_nb + i_dest_channel_offset ] - += p_in[ j * i_input_nb + i_source_channel_offset ] - * d_amplitude_factor; - } - - /* overflow buffer coefficients */ - for( j = 0; j < i_delay; j++ ) - { - ((float*)p_overflow)[ j*i_output_nb + i_dest_channel_offset ] - += p_in[ (p_out_buf->i_nb_samples - i_delay + j) - * i_input_nb + i_source_channel_offset ] - * d_amplitude_factor; - } - } - else - { - /* overflow buffer coefficients only */ - for( j = 0; j < p_out_buf->i_nb_samples; j++ ) - { - ((float*)p_overflow)[ (i_delay - p_out_buf->i_nb_samples + j) - * i_output_nb + i_dest_channel_offset ] - += p_in[ j * i_input_nb + i_source_channel_offset ] - * d_amplitude_factor; - } + ((float*)p_overflow)[ (i_delay - p_out_buf->i_nb_samples + j) + * i_output_nb + i_dest_channel_offset ] + += p_in[ j * i_input_nb + i_source_channel_offset ] + * d_amplitude_factor; } } } - else - { - memset( p_out, 0, i_out_size ); - } } /* @@ -562,6 +440,7 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter, static int OpenFilter( vlc_object_t *p_this ) { filter_t *p_filter = (filter_t *)p_this; + filter_sys_t *p_sys; bool b_fit = true; /* Activate this filter only with stereo devices */ @@ -609,20 +488,20 @@ static int OpenFilter( vlc_object_t *p_this ) } /* Allocate the memory needed to store the module's structure */ - p_filter->p_sys = malloc( sizeof(struct filter_sys_t) ); - if( p_filter->p_sys == NULL ) + p_sys = p_filter->p_sys = malloc( sizeof(struct filter_sys_t) ); + if( p_sys == NULL ) return VLC_ENOMEM; - p_filter->p_sys->i_overflow_buffer_size = 0; - p_filter->p_sys->p_overflow_buffer = NULL; - p_filter->p_sys->i_nb_atomic_operations = 0; - p_filter->p_sys->p_atomic_operations = NULL; + p_sys->i_overflow_buffer_size = 0; + p_sys->p_overflow_buffer = NULL; + p_sys->i_nb_atomic_operations = 0; + p_sys->p_atomic_operations = NULL; - if( Init( VLC_OBJECT(p_filter), (struct aout_filter_sys_t *)p_filter->p_sys + if( Init( VLC_OBJECT(p_filter), p_sys , aout_FormatNbChannels ( &(p_filter->fmt_in.audio) ) , p_filter->fmt_in.audio.i_physical_channels , p_filter->fmt_in.audio.i_rate ) < 0 ) { - free( p_filter->p_sys ); + free( p_sys ); return VLC_EGENERIC; } @@ -639,34 +518,25 @@ static void CloseFilter( vlc_object_t *p_this ) { filter_t *p_filter = (filter_t *)p_this; - if( p_filter->p_sys != NULL ) - { - free( p_filter->p_sys->p_overflow_buffer ); - free( p_filter->p_sys->p_atomic_operations ); - free( p_filter->p_sys ); - p_filter->p_sys = NULL; - } + free( p_filter->p_sys->p_overflow_buffer ); + free( p_filter->p_sys->p_atomic_operations ); + free( p_filter->p_sys ); } static block_t *Convert( 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; - - if( !p_block || !p_block->i_samples ) + if( !p_block || !p_block->i_nb_samples ) { if( p_block ) block_Release( p_block ); return NULL; } - i_out_size = p_block->i_samples * + size_t i_out_size = p_block->i_nb_samples * p_filter->fmt_out.audio.i_bitspersample/8 * aout_FormatNbChannels( &(p_filter->fmt_out.audio) ); - p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size ); + block_t *p_out = filter_NewAudioBuffer( p_filter, i_out_size ); if( !p_out ) { msg_Warn( p_filter, "can't get output buffer" ); @@ -674,29 +544,12 @@ static block_t *Convert( filter_t *p_filter, block_t *p_block ) return NULL; } - p_out->i_samples = p_block->i_samples; + p_out->i_nb_samples = p_block->i_nb_samples; 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_format = p_filter->fmt_in.i_codec; - aout_filter.output = p_filter->fmt_out.audio; - aout_filter.output.i_format = p_filter->fmt_out.i_codec; - aout_filter.b_in_place = 0; - - in_buf.p_buffer = p_block->p_buffer; - in_buf.i_nb_bytes = 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 = 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 ); - - p_out->i_buffer = out_buf.i_nb_bytes; - p_out->i_samples = out_buf.i_nb_samples; + DoWork( p_filter, p_block, p_out ); block_Release( p_block ); return p_out;