X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=src%2Faudio_output%2Finput.c;h=b394eab2d9e85d3a2e7883415da90cc4693e9e36;hb=6b15eba41eb55231031a8325248065a11793f7ef;hp=d9cb774f4468d13ac6287b5c41e2ef69c6d6eac5;hpb=d1296b32da7293d0f780cdb9b4782c84e39fb1da;p=vlc diff --git a/src/audio_output/input.c b/src/audio_output/input.c index d9cb774f44..b394eab2d9 100644 --- a/src/audio_output/input.c +++ b/src/audio_output/input.c @@ -1,8 +1,8 @@ /***************************************************************************** * input.c : internal management of input streams for the audio output ***************************************************************************** - * Copyright (C) 2002 VideoLAN - * $Id: input.c,v 1.21 2002/11/13 20:51:04 sam Exp $ + * Copyright (C) 2002-2007 the VideoLAN team + * $Id$ * * Authors: Christophe Massiot * @@ -10,7 +10,7 @@ * 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 @@ -18,112 +18,459 @@ * * 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., 59 Temple Place - Suite 330, Boston, MA 02111, USA. + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA. *****************************************************************************/ /***************************************************************************** * Preamble *****************************************************************************/ -#include /* calloc(), malloc(), free() */ + +#ifdef HAVE_CONFIG_H +# include "config.h" +#endif + +#include + +#include #include +#include +#include -#include +#include +#include /* for vout_Request */ #ifdef HAVE_ALLOCA_H # include #endif +#include +#include -#include "audio_output.h" #include "aout_internal.h" +#define AOUT_ASSERT_MIXER_LOCKED vlc_assert_locked( &p_aout->mixer_lock ) +#define AOUT_ASSERT_INPUT_LOCKED vlc_assert_locked( &p_input->lock ) + +static void inputFailure( aout_instance_t *, aout_input_t *, const char * ); +static void inputDrop( aout_input_t *, aout_buffer_t * ); +static void inputResamplingStop( aout_input_t *p_input ); + +static int VisualizationCallback( vlc_object_t *, char const *, + vlc_value_t, vlc_value_t, void * ); +static int EqualizerCallback( vlc_object_t *, char const *, + vlc_value_t, vlc_value_t, void * ); +static int ReplayGainCallback( vlc_object_t *, char const *, + vlc_value_t, vlc_value_t, void * ); +static void ReplayGainSelect( aout_instance_t *, aout_input_t * ); + +static vout_thread_t *RequestVout( void *, + vout_thread_t *, video_format_t *, bool ); +static vout_thread_t *RequestVoutFromFilter( void *, + vout_thread_t *, video_format_t *, bool ); + /***************************************************************************** * aout_InputNew : allocate a new input and rework the filter pipeline *****************************************************************************/ -int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input ) +int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const aout_request_vout_t *p_request_vout ) { - audio_sample_format_t intermediate_format; + audio_sample_format_t chain_input_format; + audio_sample_format_t chain_output_format; + vlc_value_t val, text; + char *psz_filters, *psz_visual, *psz_scaletempo; + int i_visual; aout_FormatPrint( p_aout, "input", &p_input->input ); + p_input->i_nb_resamplers = p_input->i_nb_filters = 0; + /* Prepare FIFO. */ aout_FifoInit( p_aout, &p_input->fifo, p_aout->mixer.mixer.i_rate ); p_input->p_first_byte_to_mix = NULL; - /* Create filters. */ - memcpy( &intermediate_format, &p_aout->mixer.mixer, + /* */ + if( p_request_vout ) + { + p_input->request_vout = *p_request_vout; + } + else + { + p_input->request_vout.pf_request_vout = RequestVout; + p_input->request_vout.p_private = p_aout; + } + + /* Prepare format structure */ + memcpy( &chain_input_format, &p_input->input, + sizeof(audio_sample_format_t) ); + memcpy( &chain_output_format, &p_aout->mixer.mixer, sizeof(audio_sample_format_t) ); - intermediate_format.i_rate = p_input->input.i_rate; - if ( aout_FiltersCreatePipeline( p_aout, p_input->pp_filters, - &p_input->i_nb_filters, &p_input->input, - &intermediate_format ) < 0 ) + chain_output_format.i_rate = p_input->input.i_rate; + aout_FormatPrepare( &chain_output_format ); + + /* Now add user filters */ + if( var_Type( p_aout, "visual" ) == 0 ) + { + var_Create( p_aout, "visual", VLC_VAR_STRING | VLC_VAR_HASCHOICE ); + text.psz_string = _("Visualizations"); + var_Change( p_aout, "visual", VLC_VAR_SETTEXT, &text, NULL ); + val.psz_string = (char*)""; text.psz_string = _("Disable"); + var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text ); + val.psz_string = (char*)"spectrometer"; text.psz_string = _("Spectrometer"); + var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text ); + val.psz_string = (char*)"scope"; text.psz_string = _("Scope"); + var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text ); + val.psz_string = (char*)"spectrum"; text.psz_string = _("Spectrum"); + var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text ); + val.psz_string = (char*)"vuMeter"; text.psz_string = _("Vu meter"); + var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text ); + + /* Look for goom plugin */ + if( module_exists( "goom" ) ) + { + val.psz_string = (char*)"goom"; text.psz_string = (char*)"Goom"; + var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text ); + } + + /* Look for galaktos plugin */ + if( module_exists( "galaktos" ) ) + { + val.psz_string = (char*)"galaktos"; text.psz_string = (char*)"GaLaktos"; + var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text ); + } + + /* Look for libprojectM plugin */ + if( module_exists( "projectm" ) ) + { + val.psz_string = (char*)"projectm"; text.psz_string = (char*)"projectM"; + var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text ); + } + + if( var_Get( p_aout, "effect-list", &val ) == VLC_SUCCESS ) + { + var_SetString( p_aout, "visual", val.psz_string ); + free( val.psz_string ); + } + var_AddCallback( p_aout, "visual", VisualizationCallback, NULL ); + } + + if( var_Type( p_aout, "equalizer" ) == 0 ) + { + module_config_t *p_config; + int i; + + p_config = config_FindConfig( VLC_OBJECT(p_aout), "equalizer-preset" ); + if( p_config && p_config->i_list ) + { + var_Create( p_aout, "equalizer", + VLC_VAR_STRING | VLC_VAR_HASCHOICE ); + text.psz_string = _("Equalizer"); + var_Change( p_aout, "equalizer", VLC_VAR_SETTEXT, &text, NULL ); + + val.psz_string = (char*)""; text.psz_string = _("Disable"); + var_Change( p_aout, "equalizer", VLC_VAR_ADDCHOICE, &val, &text ); + + for( i = 0; i < p_config->i_list; i++ ) + { + val.psz_string = (char *)p_config->ppsz_list[i]; + text.psz_string = (char *)p_config->ppsz_list_text[i]; + var_Change( p_aout, "equalizer", VLC_VAR_ADDCHOICE, + &val, &text ); + } + + var_AddCallback( p_aout, "equalizer", EqualizerCallback, NULL ); + } + } + + if( var_Type( p_aout, "audio-filter" ) == 0 ) + { + var_Create( p_aout, "audio-filter", + VLC_VAR_STRING | VLC_VAR_DOINHERIT ); + text.psz_string = _("Audio filters"); + var_Change( p_aout, "audio-filter", VLC_VAR_SETTEXT, &text, NULL ); + } + if( var_Type( p_aout, "audio-visual" ) == 0 ) + { + var_Create( p_aout, "audio-visual", + VLC_VAR_STRING | VLC_VAR_DOINHERIT ); + text.psz_string = _("Audio visualizations"); + var_Change( p_aout, "audio-visual", VLC_VAR_SETTEXT, &text, NULL ); + } + + if( var_Type( p_aout, "audio-replay-gain-mode" ) == 0 ) + { + module_config_t *p_config; + int i; + + p_config = config_FindConfig( VLC_OBJECT(p_aout), "audio-replay-gain-mode" ); + if( p_config && p_config->i_list ) + { + var_Create( p_aout, "audio-replay-gain-mode", + VLC_VAR_STRING | VLC_VAR_DOINHERIT ); + + text.psz_string = _("Replay gain"); + var_Change( p_aout, "audio-replay-gain-mode", VLC_VAR_SETTEXT, &text, NULL ); + + for( i = 0; i < p_config->i_list; i++ ) + { + val.psz_string = (char *)p_config->ppsz_list[i]; + text.psz_string = (char *)p_config->ppsz_list_text[i]; + var_Change( p_aout, "audio-replay-gain-mode", VLC_VAR_ADDCHOICE, + &val, &text ); + } + + var_AddCallback( p_aout, "audio-replay-gain-mode", ReplayGainCallback, NULL ); + } + } + if( var_Type( p_aout, "audio-replay-gain-preamp" ) == 0 ) + { + var_Create( p_aout, "audio-replay-gain-preamp", + VLC_VAR_FLOAT | VLC_VAR_DOINHERIT ); + } + if( var_Type( p_aout, "audio-replay-gain-default" ) == 0 ) + { + var_Create( p_aout, "audio-replay-gain-default", + VLC_VAR_FLOAT | VLC_VAR_DOINHERIT ); + } + if( var_Type( p_aout, "audio-replay-gain-peak-protection" ) == 0 ) + { + var_Create( p_aout, "audio-replay-gain-peak-protection", + VLC_VAR_BOOL | VLC_VAR_DOINHERIT ); + } + if( var_Type( p_aout, "audio-time-stretch" ) == 0 ) + { + var_Create( p_aout, "audio-time-stretch", + VLC_VAR_BOOL | VLC_VAR_DOINHERIT ); + } + + psz_filters = var_GetString( p_aout, "audio-filter" ); + psz_visual = var_GetString( p_aout, "audio-visual"); + psz_scaletempo = var_GetBool( p_aout, "audio-time-stretch" ) ? strdup( "scaletempo" ) : NULL; + + p_input->b_recycle_vout = psz_visual && *psz_visual; + + /* parse user filter lists */ + char *const ppsz_array[] = { psz_scaletempo, psz_filters, psz_visual }; + p_input->p_playback_rate_filter = NULL; + + for( i_visual = 0; i_visual < 3 && !AOUT_FMT_NON_LINEAR(&chain_output_format); i_visual++ ) { - msg_Err( p_aout, "couldn't set an input pipeline" ); + char *psz_next = NULL; + char *psz_parser = ppsz_array[i_visual]; - aout_FifoDestroy( p_aout, &p_input->fifo ); - p_input->b_error = 1; + if( psz_parser == NULL || !*psz_parser ) + continue; - return -1; + while( psz_parser && *psz_parser ) + { + aout_filter_t * p_filter = NULL; + + if( p_input->i_nb_filters >= AOUT_MAX_FILTERS ) + { + msg_Dbg( p_aout, "max filters reached (%d)", AOUT_MAX_FILTERS ); + break; + } + + while( *psz_parser == ' ' && *psz_parser == ':' ) + { + psz_parser++; + } + if( ( psz_next = strchr( psz_parser , ':' ) ) ) + { + *psz_next++ = '\0'; + } + if( *psz_parser =='\0' ) + { + break; + } + + /* Create a VLC object */ + static const char typename[] = "audio filter"; + p_filter = vlc_custom_create( p_aout, sizeof(*p_filter), + VLC_OBJECT_GENERIC, typename ); + if( p_filter == NULL ) + { + msg_Err( p_aout, "cannot add user filter %s (skipped)", + psz_parser ); + psz_parser = psz_next; + continue; + } + + vlc_object_attach( p_filter , p_aout ); + + p_filter->request_vout.pf_request_vout = RequestVoutFromFilter; + p_filter->request_vout.p_private = p_input; + + p_filter->p_owner = malloc( sizeof(*p_filter->p_owner) ); + p_filter->p_owner->p_aout = p_aout; + p_filter->p_owner->p_input = p_input; + + /* request format */ + memcpy( &p_filter->input, &chain_output_format, + sizeof(audio_sample_format_t) ); + memcpy( &p_filter->output, &chain_output_format, + sizeof(audio_sample_format_t) ); + + + /* try to find the requested filter */ + if( i_visual == 2 ) /* this can only be a visualization module */ + { + + p_filter->p_module = module_need( p_filter, "visualization", + psz_parser, true ); + } + else /* this can be a audio filter module as well as a visualization module */ + { + p_filter->p_module = module_need( p_filter, "audio filter", + psz_parser, true ); + + if ( p_filter->p_module == NULL ) + { + /* if the filter requested a special format, retry */ + if ( !( AOUT_FMTS_IDENTICAL( &p_filter->input, + &chain_input_format ) + && AOUT_FMTS_IDENTICAL( &p_filter->output, + &chain_output_format ) ) ) + { + aout_FormatPrepare( &p_filter->input ); + aout_FormatPrepare( &p_filter->output ); + p_filter->p_module = module_need( p_filter, + "audio filter", + psz_parser, true ); + } + /* try visual filters */ + else + { + memcpy( &p_filter->input, &chain_output_format, + sizeof(audio_sample_format_t) ); + memcpy( &p_filter->output, &chain_output_format, + sizeof(audio_sample_format_t) ); + p_filter->p_module = module_need( p_filter, + "visualization", + psz_parser, true ); + } + } + } + + /* failure */ + if ( p_filter->p_module == NULL ) + { + msg_Err( p_aout, "cannot add user filter %s (skipped)", + psz_parser ); + + free( p_filter->p_owner ); + vlc_object_detach( p_filter ); + vlc_object_release( p_filter ); + + psz_parser = psz_next; + continue; + } + + /* complete the filter chain if necessary */ + if ( !AOUT_FMTS_IDENTICAL( &chain_input_format, &p_filter->input ) ) + { + if ( aout_FiltersCreatePipeline( p_aout, p_input->pp_filters, + &p_input->i_nb_filters, + &chain_input_format, + &p_filter->input ) < 0 ) + { + msg_Err( p_aout, "cannot add user filter %s (skipped)", + psz_parser ); + + module_unneed( p_filter, p_filter->p_module ); + free( p_filter->p_owner ); + vlc_object_detach( p_filter ); + vlc_object_release( p_filter ); + + psz_parser = psz_next; + continue; + } + } + + /* success */ + p_filter->b_continuity = false; + p_input->pp_filters[p_input->i_nb_filters++] = p_filter; + memcpy( &chain_input_format, &p_filter->output, + sizeof( audio_sample_format_t ) ); + + if( i_visual == 0 ) /* scaletempo */ + p_input->p_playback_rate_filter = p_filter; + + /* next filter if any */ + psz_parser = psz_next; + } + } + free( psz_visual ); + free( psz_filters ); + free( psz_scaletempo ); + + /* complete the filter chain if necessary */ + if ( !AOUT_FMTS_IDENTICAL( &chain_input_format, &chain_output_format ) ) + { + if ( aout_FiltersCreatePipeline( p_aout, p_input->pp_filters, + &p_input->i_nb_filters, + &chain_input_format, + &chain_output_format ) < 0 ) + { + inputFailure( p_aout, p_input, "couldn't set an input pipeline" ); + return -1; + } } /* Prepare hints for the buffer allocator. */ p_input->input_alloc.i_alloc_type = AOUT_ALLOC_HEAP; p_input->input_alloc.i_bytes_per_sec = -1; - if ( AOUT_FMT_NON_LINEAR( &p_aout->mixer.mixer ) ) - { - p_input->i_nb_resamplers = 0; - } - else + /* Create resamplers. */ + if ( !AOUT_FMT_NON_LINEAR( &p_aout->mixer.mixer ) ) { - /* Create resamplers. */ - intermediate_format.i_rate = (p_input->input.i_rate + chain_output_format.i_rate = (__MAX(p_input->input.i_rate, + p_aout->mixer.mixer.i_rate) * (100 + AOUT_MAX_RESAMPLING)) / 100; - if ( intermediate_format.i_rate == p_aout->mixer.mixer.i_rate ) + if ( chain_output_format.i_rate == p_aout->mixer.mixer.i_rate ) { /* Just in case... */ - intermediate_format.i_rate++; + chain_output_format.i_rate++; } if ( aout_FiltersCreatePipeline( p_aout, p_input->pp_resamplers, &p_input->i_nb_resamplers, - &intermediate_format, + &chain_output_format, &p_aout->mixer.mixer ) < 0 ) { - msg_Err( p_aout, "couldn't set a resampler pipeline" ); - - aout_FiltersDestroyPipeline( p_aout, p_input->pp_filters, - p_input->i_nb_filters ); - aout_FifoDestroy( p_aout, &p_input->fifo ); - p_input->b_error = 1; - + inputFailure( p_aout, p_input, "couldn't set a resampler pipeline"); return -1; } aout_FiltersHintBuffers( p_aout, p_input->pp_resamplers, p_input->i_nb_resamplers, &p_input->input_alloc ); + p_input->input_alloc.i_alloc_type = AOUT_ALLOC_HEAP; /* Setup the initial rate of the resampler */ p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate; - p_input->i_resampling_type = AOUT_RESAMPLING_NONE; + } + p_input->i_resampling_type = AOUT_RESAMPLING_NONE; + if( ! p_input->p_playback_rate_filter && p_input->i_nb_resamplers > 0 ) + { + p_input->p_playback_rate_filter = p_input->pp_resamplers[0]; } - p_input->input_alloc.i_alloc_type = AOUT_ALLOC_HEAP; - p_input->input_alloc.i_bytes_per_sec = -1; aout_FiltersHintBuffers( p_aout, p_input->pp_filters, p_input->i_nb_filters, &p_input->input_alloc ); + p_input->input_alloc.i_alloc_type = AOUT_ALLOC_HEAP; /* i_bytes_per_sec is still == -1 if no filters */ p_input->input_alloc.i_bytes_per_sec = __MAX( p_input->input_alloc.i_bytes_per_sec, - p_input->input.i_bytes_per_frame + (int)(p_input->input.i_bytes_per_frame * p_input->input.i_rate - / p_input->input.i_frame_length ); - /* Allocate in the heap, it is more convenient for the decoder. */ - p_input->input_alloc.i_alloc_type = AOUT_ALLOC_HEAP; + / p_input->input.i_frame_length) ); - p_input->b_error = 0; + ReplayGainSelect( p_aout, p_input ); + + /* Success */ + p_input->b_error = false; + p_input->b_restart = false; + p_input->i_last_input_rate = INPUT_RATE_DEFAULT; return 0; } @@ -135,12 +482,24 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input ) *****************************************************************************/ int aout_InputDelete( aout_instance_t * p_aout, aout_input_t * p_input ) { - if ( p_input->b_error ) return 0; + AOUT_ASSERT_MIXER_LOCKED; + if ( p_input->b_error ) + return 0; + + /* XXX We need to update b_recycle_vout before calling aout_FiltersDestroyPipeline. + * FIXME They can be a race condition if audio-visual is updated between + * aout_InputDelete and aout_InputNew. + */ + char *psz_visual = var_GetString( p_aout, "audio-visual"); + p_input->b_recycle_vout = psz_visual && *psz_visual; + free( psz_visual ); aout_FiltersDestroyPipeline( p_aout, p_input->pp_filters, p_input->i_nb_filters ); + p_input->i_nb_filters = 0; aout_FiltersDestroyPipeline( p_aout, p_input->pp_resamplers, p_input->i_nb_resamplers ); + p_input->i_nb_resamplers = 0; aout_FifoDestroy( p_aout, &p_input->fifo ); return 0; @@ -151,33 +510,104 @@ int aout_InputDelete( aout_instance_t * p_aout, aout_input_t * p_input ) ***************************************************************************** * This function must be entered with the input lock. *****************************************************************************/ +/* XXX Do not activate it !! */ +//#define AOUT_PROCESS_BEFORE_CHEKS int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, - aout_buffer_t * p_buffer ) + aout_buffer_t * p_buffer, int i_input_rate ) { mtime_t start_date; + AOUT_ASSERT_INPUT_LOCKED; + + if( p_input->b_restart ) + { + aout_fifo_t fifo; + uint8_t *p_first_byte_to_mix; + bool b_paused; + mtime_t i_pause_date; + + aout_lock_mixer( p_aout ); + aout_lock_input_fifos( p_aout ); + + /* A little trick to avoid loosing our input fifo and properties */ + + p_first_byte_to_mix = p_input->p_first_byte_to_mix; + fifo = p_input->fifo; + b_paused = p_input->b_paused; + i_pause_date = p_input->i_pause_date; + + aout_FifoInit( p_aout, &p_input->fifo, p_aout->mixer.mixer.i_rate ); + + aout_InputDelete( p_aout, p_input ); + + aout_InputNew( p_aout, p_input, &p_input->request_vout ); + p_input->p_first_byte_to_mix = p_first_byte_to_mix; + p_input->fifo = fifo; + p_input->b_paused = b_paused; + p_input->i_pause_date = i_pause_date; + + aout_unlock_input_fifos( p_aout ); + aout_unlock_mixer( p_aout ); + } + + if( i_input_rate != INPUT_RATE_DEFAULT && p_input->p_playback_rate_filter == NULL ) + { + inputDrop( p_input, p_buffer ); + return 0; + } + +#ifdef AOUT_PROCESS_BEFORE_CHEKS + /* Run pre-filters. */ + aout_FiltersPlay( p_aout, p_input->pp_filters, p_input->i_nb_filters, + &p_buffer ); + + /* Actually run the resampler now. */ + if ( p_input->i_nb_resamplers > 0 ) + { + const mtime_t i_date = p_buffer->start_date; + aout_FiltersPlay( p_aout, p_input->pp_resamplers, + p_input->i_nb_resamplers, + &p_buffer ); + } + + if( p_buffer->i_nb_samples <= 0 ) + { + aout_BufferFree( p_buffer ); + return 0; + } +#endif + + /* Handle input rate change, but keep drift correction */ + if( i_input_rate != p_input->i_last_input_rate ) + { + unsigned int * const pi_rate = &p_input->p_playback_rate_filter->input.i_rate; +#define F(r,ir) ( INPUT_RATE_DEFAULT * (r) / (ir) ) + const int i_delta = *pi_rate - F(p_input->input.i_rate,p_input->i_last_input_rate); + *pi_rate = F(p_input->input.i_rate + i_delta, i_input_rate); +#undef F + p_input->i_last_input_rate = i_input_rate; + } /* We don't care if someone changes the start date behind our back after * this. We'll deal with that when pushing the buffer, and compensate * with the next incoming buffer. */ - vlc_mutex_lock( &p_aout->input_fifos_lock ); + aout_lock_input_fifos( p_aout ); start_date = aout_FifoNextStart( p_aout, &p_input->fifo ); - vlc_mutex_unlock( &p_aout->input_fifos_lock ); + aout_unlock_input_fifos( p_aout ); if ( start_date != 0 && start_date < mdate() ) { /* The decoder is _very_ late. This can only happen if the user * pauses the stream (or if the decoder is buggy, which cannot * happen :). */ - msg_Warn( p_aout, "computed PTS is out of range ("I64Fd"), " + msg_Warn( p_aout, "computed PTS is out of range (%"PRId64"), " "clearing out", mdate() - start_date ); - vlc_mutex_lock( &p_aout->input_fifos_lock ); + aout_lock_input_fifos( p_aout ); aout_FifoSet( p_aout, &p_input->fifo, 0 ); - vlc_mutex_unlock( &p_aout->input_fifos_lock ); + p_input->p_first_byte_to_mix = NULL; + aout_unlock_input_fifos( p_aout ); if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE ) msg_Warn( p_aout, "timing screwed, stopping resampling" ); - p_input->i_resampling_type = AOUT_RESAMPLING_NONE; - p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate; - p_input->pp_resamplers[0]->b_reinit = VLC_TRUE; + inputResamplingStop( p_input ); start_date = 0; } @@ -185,24 +615,54 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, { /* The decoder gives us f*cked up PTS. It's its business, but we * can't present it anyway, so drop the buffer. */ - msg_Warn( p_aout, "PTS is out of range ("I64Fd"), dropping buffer", + msg_Warn( p_aout, "PTS is out of range (%"PRId64"), dropping buffer", mdate() - p_buffer->start_date ); - aout_BufferFree( p_buffer ); + inputDrop( p_input, p_buffer ); + inputResamplingStop( p_input ); + return 0; + } + + /* If the audio drift is too big then it's not worth trying to resample + * the audio. */ + mtime_t i_pts_tolerance = 3 * AOUT_PTS_TOLERANCE * i_input_rate / INPUT_RATE_DEFAULT; + if ( start_date != 0 && + ( start_date < p_buffer->start_date - i_pts_tolerance ) ) + { + msg_Warn( p_aout, "audio drift is too big (%"PRId64"), clearing out", + start_date - p_buffer->start_date ); + aout_lock_input_fifos( p_aout ); + aout_FifoSet( p_aout, &p_input->fifo, 0 ); + p_input->p_first_byte_to_mix = NULL; + aout_unlock_input_fifos( p_aout ); + if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE ) + msg_Warn( p_aout, "timing screwed, stopping resampling" ); + inputResamplingStop( p_input ); + start_date = 0; + } + else if ( start_date != 0 && + ( start_date > p_buffer->start_date + i_pts_tolerance) ) + { + msg_Warn( p_aout, "audio drift is too big (%"PRId64"), dropping buffer", + start_date - p_buffer->start_date ); + inputDrop( p_input, p_buffer ); return 0; } if ( start_date == 0 ) start_date = p_buffer->start_date; +#ifndef AOUT_PROCESS_BEFORE_CHEKS /* Run pre-filters. */ aout_FiltersPlay( p_aout, p_input->pp_filters, p_input->i_nb_filters, &p_buffer ); +#endif /* Run the resampler if needed. * We first need to calculate the output rate of this resampler. */ if ( ( p_input->i_resampling_type == AOUT_RESAMPLING_NONE ) && ( start_date < p_buffer->start_date - AOUT_PTS_TOLERANCE - || start_date > p_buffer->start_date + AOUT_PTS_TOLERANCE ) ) + || start_date > p_buffer->start_date + AOUT_PTS_TOLERANCE ) && + p_input->i_nb_resamplers > 0 ) { /* Can happen in several circumstances : * 1. A problem at the input (clock drift) @@ -221,7 +681,7 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, else p_input->i_resampling_type = AOUT_RESAMPLING_UP; - msg_Warn( p_aout, "buffer is "I64Fd" %s, triggering %ssampling", + msg_Warn( p_aout, "buffer is %"PRId64" %s, triggering %ssampling", drift > 0 ? drift : -drift, drift > 0 ? "in advance" : "late", drift > 0 ? "down" : "up"); @@ -235,21 +695,26 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, if( p_input->i_resampling_type == AOUT_RESAMPLING_UP ) { - p_input->pp_resamplers[0]->input.i_rate += 10; /* Hz */ + p_input->pp_resamplers[0]->input.i_rate += 2; /* Hz */ } else { - p_input->pp_resamplers[0]->input.i_rate -= 10; /* Hz */ + p_input->pp_resamplers[0]->input.i_rate -= 2; /* Hz */ } /* Check if everything is back to normal, in which case we can stop the * resampling */ - if( p_input->pp_resamplers[0]->input.i_rate == - p_input->input.i_rate ) + unsigned int i_nominal_rate = + (p_input->pp_resamplers[0] == p_input->p_playback_rate_filter) + ? INPUT_RATE_DEFAULT * p_input->input.i_rate / i_input_rate + : p_input->input.i_rate; + if( p_input->pp_resamplers[0]->input.i_rate == i_nominal_rate ) { p_input->i_resampling_type = AOUT_RESAMPLING_NONE; - msg_Warn( p_aout, "resampling stopped after "I64Fi" usec", - mdate() - p_input->i_resamp_start_date ); + msg_Warn( p_aout, "resampling stopped after %"PRIi64" usec " + "(drift: %"PRIi64")", + mdate() - p_input->i_resamp_start_date, + p_buffer->start_date - start_date); } else if( abs( (int)(p_buffer->start_date - start_date) ) < abs( p_input->i_resamp_start_drift ) / 2 ) @@ -269,29 +734,261 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, /* If the drift is increasing and not decreasing, than something * is bad. We'd better stop the resampling right now. */ msg_Warn( p_aout, "timing screwed, stopping resampling" ); - p_input->i_resampling_type = AOUT_RESAMPLING_NONE; - p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate; + inputResamplingStop( p_input ); } - } - /* Adding the start date will be managed by aout_FifoPush(). */ - p_buffer->start_date = start_date; - p_buffer->end_date = start_date + - (p_buffer->end_date - p_buffer->start_date); - +#ifndef AOUT_PROCESS_BEFORE_CHEKS /* Actually run the resampler now. */ - if ( p_aout->mixer.mixer.i_rate != - p_input->pp_resamplers[0]->input.i_rate ) + if ( p_input->i_nb_resamplers > 0 ) { aout_FiltersPlay( p_aout, p_input->pp_resamplers, p_input->i_nb_resamplers, &p_buffer ); } - vlc_mutex_lock( &p_aout->input_fifos_lock ); - aout_FifoPush( p_aout, &p_input->fifo, p_buffer ); - vlc_mutex_unlock( &p_aout->input_fifos_lock ); + if( p_buffer->i_nb_samples <= 0 ) + { + aout_BufferFree( p_buffer ); + return 0; + } +#endif + /* Adding the start date will be managed by aout_FifoPush(). */ + p_buffer->end_date = start_date + + (p_buffer->end_date - p_buffer->start_date); + p_buffer->start_date = start_date; + + aout_lock_input_fifos( p_aout ); + aout_FifoPush( p_aout, &p_input->fifo, p_buffer ); + aout_unlock_input_fifos( p_aout ); return 0; } + +/***************************************************************************** + * static functions + *****************************************************************************/ + +static void inputFailure( aout_instance_t * p_aout, aout_input_t * p_input, + const char * psz_error_message ) +{ + /* error message */ + msg_Err( p_aout, "%s", psz_error_message ); + + /* clean up */ + aout_FiltersDestroyPipeline( p_aout, p_input->pp_filters, + p_input->i_nb_filters ); + aout_FiltersDestroyPipeline( p_aout, p_input->pp_resamplers, + p_input->i_nb_resamplers ); + aout_FifoDestroy( p_aout, &p_input->fifo ); + var_Destroy( p_aout, "visual" ); + var_Destroy( p_aout, "equalizer" ); + var_Destroy( p_aout, "audio-filter" ); + var_Destroy( p_aout, "audio-visual" ); + + var_Destroy( p_aout, "audio-replay-gain-mode" ); + var_Destroy( p_aout, "audio-replay-gain-default" ); + var_Destroy( p_aout, "audio-replay-gain-preamp" ); + var_Destroy( p_aout, "audio-replay-gain-peak-protection" ); + + /* error flag */ + p_input->b_error = 1; +} + +static void inputDrop( aout_input_t *p_input, aout_buffer_t *p_buffer ) +{ + aout_BufferFree( p_buffer ); + + p_input->i_buffer_lost++; +} + +static void inputResamplingStop( aout_input_t *p_input ) +{ + p_input->i_resampling_type = AOUT_RESAMPLING_NONE; + if( p_input->i_nb_resamplers != 0 ) + { + p_input->pp_resamplers[0]->input.i_rate = + ( p_input->pp_resamplers[0] == p_input->p_playback_rate_filter ) + ? INPUT_RATE_DEFAULT * p_input->input.i_rate / p_input->i_last_input_rate + : p_input->input.i_rate; + p_input->pp_resamplers[0]->b_continuity = false; + } +} + +static vout_thread_t *RequestVout( void *p_private, + vout_thread_t *p_vout, video_format_t *p_fmt, bool b_recycle ) +{ + aout_instance_t *p_aout = p_private; + VLC_UNUSED(b_recycle); + return vout_Request( p_aout, p_vout, p_fmt ); +} + +static vout_thread_t *RequestVoutFromFilter( void *p_private, + vout_thread_t *p_vout, video_format_t *p_fmt, bool b_recycle ) +{ + aout_input_t *p_input = p_private; + aout_request_vout_t *p_request = &p_input->request_vout; + + return p_request->pf_request_vout( p_request->p_private, + p_vout, p_fmt, p_input->b_recycle_vout && b_recycle ); +} + +static int ChangeFiltersString( aout_instance_t * p_aout, const char* psz_variable, + const char *psz_name, bool b_add ) +{ + return AoutChangeFilterString( VLC_OBJECT(p_aout), p_aout, + psz_variable, psz_name, b_add ) ? 1 : 0; +} + +static int VisualizationCallback( vlc_object_t *p_this, char const *psz_cmd, + vlc_value_t oldval, vlc_value_t newval, void *p_data ) +{ + aout_instance_t *p_aout = (aout_instance_t *)p_this; + char *psz_mode = newval.psz_string; + (void)psz_cmd; (void)oldval; (void)p_data; + + if( !psz_mode || !*psz_mode ) + { + ChangeFiltersString( p_aout, "audio-visual", "goom", false ); + ChangeFiltersString( p_aout, "audio-visual", "visual", false ); + ChangeFiltersString( p_aout, "audio-visual", "galaktos", false ); + ChangeFiltersString( p_aout, "audio-visual", "projectm", false ); + } + else + { + if( !strcmp( "goom", psz_mode ) ) + { + ChangeFiltersString( p_aout, "audio-visual", "visual", false ); + ChangeFiltersString( p_aout, "audio-visual", "goom", true ); + ChangeFiltersString( p_aout, "audio-visual", "galaktos", false ); + ChangeFiltersString( p_aout, "audio-visual", "projectm", false ); + } + else if( !strcmp( "galaktos", psz_mode ) ) + { + ChangeFiltersString( p_aout, "audio-visual", "visual", false ); + ChangeFiltersString( p_aout, "audio-visual", "goom", false ); + ChangeFiltersString( p_aout, "audio-visual", "galaktos", true ); + } + else if( !strcmp( "projectm", psz_mode ) ) + { + ChangeFiltersString( p_aout, "audio-visual", "visual", false ); + ChangeFiltersString( p_aout, "audio-visual", "goom", false ); + ChangeFiltersString( p_aout, "audio-visual", "galaktos", false ); + ChangeFiltersString( p_aout, "audio-visual", "projectm", true ); + } + else + { + var_Create( p_aout, "effect-list", VLC_VAR_STRING ); + var_SetString( p_aout, "effect-list", psz_mode ); + + ChangeFiltersString( p_aout, "audio-visual", "goom", false ); + ChangeFiltersString( p_aout, "audio-visual", "visual", true ); + ChangeFiltersString( p_aout, "audio-visual", "galaktos", false ); + ChangeFiltersString( p_aout, "audio-visual", "projectm", false ); + } + } + + /* That sucks */ + AoutInputsMarkToRestart( p_aout ); + + return VLC_SUCCESS; +} + +static int EqualizerCallback( vlc_object_t *p_this, char const *psz_cmd, + vlc_value_t oldval, vlc_value_t newval, void *p_data ) +{ + aout_instance_t *p_aout = (aout_instance_t *)p_this; + char *psz_mode = newval.psz_string; + int i_ret; + (void)psz_cmd; (void)oldval; (void)p_data; + + if( !psz_mode || !*psz_mode ) + { + i_ret = ChangeFiltersString( p_aout, "audio-filter", "equalizer", + false ); + } + else + { + var_Create( p_aout, "equalizer-preset", VLC_VAR_STRING ); + var_SetString( p_aout, "equalizer-preset", psz_mode ); + i_ret = ChangeFiltersString( p_aout, "audio-filter", "equalizer", + true ); + } + + /* That sucks */ + if( i_ret == 1 ) + AoutInputsMarkToRestart( p_aout ); + return VLC_SUCCESS; +} + +static int ReplayGainCallback( vlc_object_t *p_this, char const *psz_cmd, + vlc_value_t oldval, vlc_value_t newval, void *p_data ) +{ + VLC_UNUSED(psz_cmd); VLC_UNUSED(oldval); + VLC_UNUSED(newval); VLC_UNUSED(p_data); + aout_instance_t *p_aout = (aout_instance_t *)p_this; + int i; + + aout_lock_mixer( p_aout ); + for( i = 0; i < p_aout->i_nb_inputs; i++ ) + ReplayGainSelect( p_aout, p_aout->pp_inputs[i] ); + + /* Restart the mixer (a trivial mixer may be in use) */ + aout_MixerMultiplierSet( p_aout, p_aout->mixer.f_multiplier ); + aout_unlock_mixer( p_aout ); + + return VLC_SUCCESS; +} + +static void ReplayGainSelect( aout_instance_t *p_aout, aout_input_t *p_input ) +{ + char *psz_replay_gain = var_GetNonEmptyString( p_aout, + "audio-replay-gain-mode" ); + int i_mode; + int i_use; + float f_gain; + + p_input->f_multiplier = 1.0; + + if( !psz_replay_gain ) + return; + + /* Find select mode */ + if( !strcmp( psz_replay_gain, "track" ) ) + i_mode = AUDIO_REPLAY_GAIN_TRACK; + else if( !strcmp( psz_replay_gain, "album" ) ) + i_mode = AUDIO_REPLAY_GAIN_ALBUM; + else + i_mode = AUDIO_REPLAY_GAIN_MAX; + + /* If the select mode is not available, prefer the other one */ + i_use = i_mode; + if( i_use != AUDIO_REPLAY_GAIN_MAX && !p_input->replay_gain.pb_gain[i_use] ) + { + for( i_use = 0; i_use < AUDIO_REPLAY_GAIN_MAX; i_use++ ) + { + if( p_input->replay_gain.pb_gain[i_use] ) + break; + } + } + + /* */ + if( i_use != AUDIO_REPLAY_GAIN_MAX ) + f_gain = p_input->replay_gain.pf_gain[i_use] + var_GetFloat( p_aout, "audio-replay-gain-preamp" ); + else if( i_mode != AUDIO_REPLAY_GAIN_MAX ) + f_gain = var_GetFloat( p_aout, "audio-replay-gain-default" ); + else + f_gain = 0.0; + p_input->f_multiplier = pow( 10.0, f_gain / 20.0 ); + + /* */ + if( p_input->replay_gain.pb_peak[i_use] && + var_GetBool( p_aout, "audio-replay-gain-peak-protection" ) && + p_input->replay_gain.pf_peak[i_use] * p_input->f_multiplier > 1.0 ) + { + p_input->f_multiplier = 1.0f / p_input->replay_gain.pf_peak[i_use]; + } + + free( psz_replay_gain ); +} +