X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=src%2Faudio_output%2Foutput.c;h=8271868079a6d1cbaff60fb606384b35b512c9c0;hb=09d375d4451c139097d95e152e3da09352ae381e;hp=b33a60adfed61d0dd8b5ab11baf4ae6388cdbc13;hpb=5ffbc97e759f4e44a9ac33567901de1f5ff5528a;p=vlc diff --git a/src/audio_output/output.c b/src/audio_output/output.c index b33a60adfe..8271868079 100644 --- a/src/audio_output/output.c +++ b/src/audio_output/output.c @@ -1,8 +1,8 @@ /***************************************************************************** * output.c : internal management of output streams for the audio output ***************************************************************************** - * Copyright (C) 2002 VideoLAN - * $Id: output.c,v 1.7 2002/08/19 21:31:11 massiot Exp $ + * Copyright (C) 2002-2004 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 @@ -18,72 +18,156 @@ * * 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() */ -#include - -#include +#ifdef HAVE_CONFIG_H +# include "config.h" +#endif -#include "audio_output.h" +#include +#include #include "aout_internal.h" /***************************************************************************** * aout_OutputNew : allocate a new output and rework the filter pipeline + ***************************************************************************** + * This function is entered with the mixer lock. *****************************************************************************/ int aout_OutputNew( aout_instance_t * p_aout, audio_sample_format_t * p_format ) { - char * psz_name = config_GetPsz( p_aout, "aout" ); + /* Retrieve user defaults. */ int i_rate = config_GetInt( p_aout, "aout-rate" ); - int i_channels = config_GetInt( p_aout, "aout-channels" ); + vlc_value_t val, text; + /* kludge to avoid a fpu error when rate is 0... */ + if( i_rate == 0 ) i_rate = -1; - /* Prepare FIFO. */ - vlc_mutex_init( p_aout, &p_aout->output.fifo.lock ); - p_aout->output.fifo.p_first = NULL; - p_aout->output.fifo.pp_last = &p_aout->output.fifo.p_first; + memcpy( &p_aout->output.output, p_format, sizeof(audio_sample_format_t) ); + if ( i_rate != -1 ) + p_aout->output.output.i_rate = i_rate; + aout_FormatPrepare( &p_aout->output.output ); - p_aout->output.p_module = module_Need( p_aout, "audio output", - psz_name ); - if ( psz_name != NULL ) free( psz_name ); + aout_lock_output_fifo( p_aout ); + + /* Find the best output plug-in. */ + p_aout->output.p_module = module_need( p_aout, "audio output", "$aout", false ); if ( p_aout->output.p_module == NULL ) { - msg_Err( p_aout, "no suitable aout module" ); + msg_Err( p_aout, "no suitable audio output module" ); + aout_unlock_output_fifo( p_aout ); return -1; } - /* Retrieve user defaults. */ - memcpy( &p_aout->output.output, p_format, sizeof(audio_sample_format_t) ); - if ( i_rate != -1 ) p_aout->output.output.i_rate = i_rate; - if ( i_channels != -1 ) p_aout->output.output.i_channels = i_channels; - if ( AOUT_FMT_NON_LINEAR(&p_aout->output.output) ) + if ( var_Type( p_aout, "audio-channels" ) == + (VLC_VAR_INTEGER | VLC_VAR_HASCHOICE) ) { - p_aout->output.output.i_format = AOUT_FMT_SPDIF; + /* The user may have selected a different channels configuration. */ + var_Get( p_aout, "audio-channels", &val ); + + if ( val.i_int == AOUT_VAR_CHAN_RSTEREO ) + { + p_aout->output.output.i_original_channels |= + AOUT_CHAN_REVERSESTEREO; + } + else if ( val.i_int == AOUT_VAR_CHAN_STEREO ) + { + p_aout->output.output.i_original_channels = + AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT; + } + else if ( val.i_int == AOUT_VAR_CHAN_LEFT ) + { + p_aout->output.output.i_original_channels = AOUT_CHAN_LEFT; + } + else if ( val.i_int == AOUT_VAR_CHAN_RIGHT ) + { + p_aout->output.output.i_original_channels = AOUT_CHAN_RIGHT; + } + else if ( val.i_int == AOUT_VAR_CHAN_DOLBYS ) + { + p_aout->output.output.i_original_channels + = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_DOLBYSTEREO; + } } - else + else if ( p_aout->output.output.i_physical_channels == AOUT_CHAN_CENTER + && (p_aout->output.output.i_original_channels + & AOUT_CHAN_PHYSMASK) == (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT) ) { - /* Non-S/PDIF mixer only deals with float32 or fixed32. */ - p_aout->output.output.i_format - = (p_aout->p_vlc->i_cpu & CPU_CAPABILITY_FPU) ? - AOUT_FMT_FLOAT32 : AOUT_FMT_FIXED32; - } + /* Mono - create the audio-channels variable. */ + var_Create( p_aout, "audio-channels", + VLC_VAR_INTEGER | VLC_VAR_HASCHOICE ); + text.psz_string = _("Audio Channels"); + var_Change( p_aout, "audio-channels", VLC_VAR_SETTEXT, &text, NULL ); - /* Find the best output format. */ - if ( p_aout->output.pf_setformat( p_aout ) != 0 ) + val.i_int = AOUT_VAR_CHAN_STEREO; text.psz_string = _("Stereo"); + var_Change( p_aout, "audio-channels", VLC_VAR_ADDCHOICE, &val, &text ); + val.i_int = AOUT_VAR_CHAN_LEFT; text.psz_string = _("Left"); + var_Change( p_aout, "audio-channels", VLC_VAR_ADDCHOICE, &val, &text ); + val.i_int = AOUT_VAR_CHAN_RIGHT; text.psz_string = _("Right"); + var_Change( p_aout, "audio-channels", VLC_VAR_ADDCHOICE, &val, &text ); + if ( p_aout->output.output.i_original_channels & AOUT_CHAN_DUALMONO ) + { + /* Go directly to the left channel. */ + p_aout->output.output.i_original_channels = AOUT_CHAN_LEFT; + val.i_int = AOUT_VAR_CHAN_LEFT; + var_Set( p_aout, "audio-channels", val ); + } + var_AddCallback( p_aout, "audio-channels", aout_ChannelsRestart, + NULL ); + } + else if ( p_aout->output.output.i_physical_channels == + (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT) + && (p_aout->output.output.i_original_channels & + (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT)) ) { - msg_Err( p_aout, "couldn't set an output format" ); - module_Unneed( p_aout, p_aout->output.p_module ); - return -1; + /* Stereo - create the audio-channels variable. */ + var_Create( p_aout, "audio-channels", + VLC_VAR_INTEGER | VLC_VAR_HASCHOICE ); + text.psz_string = _("Audio Channels"); + var_Change( p_aout, "audio-channels", VLC_VAR_SETTEXT, &text, NULL ); + + if ( p_aout->output.output.i_original_channels & AOUT_CHAN_DOLBYSTEREO ) + { + val.i_int = AOUT_VAR_CHAN_DOLBYS; + text.psz_string = _("Dolby Surround"); + } + else + { + val.i_int = AOUT_VAR_CHAN_STEREO; + text.psz_string = _("Stereo"); + } + var_Change( p_aout, "audio-channels", VLC_VAR_ADDCHOICE, &val, &text ); + val.i_int = AOUT_VAR_CHAN_LEFT; text.psz_string = _("Left"); + var_Change( p_aout, "audio-channels", VLC_VAR_ADDCHOICE, &val, &text ); + val.i_int = AOUT_VAR_CHAN_RIGHT; text.psz_string = _("Right"); + var_Change( p_aout, "audio-channels", VLC_VAR_ADDCHOICE, &val, &text ); + val.i_int = AOUT_VAR_CHAN_RSTEREO; text.psz_string=_("Reverse stereo"); + var_Change( p_aout, "audio-channels", VLC_VAR_ADDCHOICE, &val, &text ); + if ( p_aout->output.output.i_original_channels & AOUT_CHAN_DUALMONO ) + { + /* Go directly to the left channel. */ + p_aout->output.output.i_original_channels = AOUT_CHAN_LEFT; + val.i_int = AOUT_VAR_CHAN_LEFT; + var_Set( p_aout, "audio-channels", val ); + } + var_AddCallback( p_aout, "audio-channels", aout_ChannelsRestart, + NULL ); } + val.b_bool = true; + var_Set( p_aout, "intf-change", val ); + aout_FormatPrepare( &p_aout->output.output ); - msg_Dbg( p_aout, "output format=%d rate=%d channels=%d", - p_aout->output.output.i_format, p_aout->output.output.i_rate, - p_aout->output.output.i_channels ); + /* Prepare FIFO. */ + aout_FifoInit( p_aout, &p_aout->output.fifo, + p_aout->output.output.i_rate ); + + aout_unlock_output_fifo( p_aout ); + + aout_FormatPrint( p_aout, "output", &p_aout->output.output ); /* Calculate the resulting mixer output format. */ memcpy( &p_aout->mixer.mixer, &p_aout->output.output, @@ -92,8 +176,8 @@ int aout_OutputNew( aout_instance_t * p_aout, { /* Non-S/PDIF mixer only deals with float32 or fixed32. */ p_aout->mixer.mixer.i_format - = (p_aout->p_vlc->i_cpu & CPU_CAPABILITY_FPU) ? - AOUT_FMT_FLOAT32 : AOUT_FMT_FIXED32; + = (vlc_CPU() & CPU_CAPABILITY_FPU) ? + VLC_CODEC_FL32 : VLC_CODEC_FI32; aout_FormatPrepare( &p_aout->mixer.mixer ); } else @@ -101,18 +185,17 @@ int aout_OutputNew( aout_instance_t * p_aout, p_aout->mixer.mixer.i_format = p_format->i_format; } - msg_Dbg( p_aout, "mixer format=%d rate=%d channels=%d", - p_aout->mixer.mixer.i_format, p_aout->mixer.mixer.i_rate, - p_aout->mixer.mixer.i_channels ); + aout_FormatPrint( p_aout, "mixer", &p_aout->mixer.mixer ); /* Create filters. */ + p_aout->output.i_nb_filters = 0; if ( aout_FiltersCreatePipeline( p_aout, p_aout->output.pp_filters, &p_aout->output.i_nb_filters, &p_aout->mixer.mixer, &p_aout->output.output ) < 0 ) { - msg_Err( p_aout, "couldn't set an output pipeline" ); - module_Unneed( p_aout, p_aout->output.p_module ); + msg_Err( p_aout, "couldn't create audio output pipeline" ); + module_unneed( p_aout, p_aout->output.p_module ); return -1; } @@ -127,23 +210,38 @@ int aout_OutputNew( aout_instance_t * p_aout, p_aout->output.i_nb_filters, &p_aout->mixer.output_alloc ); + p_aout->output.b_error = 0; return 0; } /***************************************************************************** * aout_OutputDelete : delete the output + ***************************************************************************** + * This function is entered with the mixer lock. *****************************************************************************/ void aout_OutputDelete( aout_instance_t * p_aout ) { - module_Unneed( p_aout, p_aout->output.p_module ); + if ( p_aout->output.b_error ) + { + return; + } + + module_unneed( p_aout, p_aout->output.p_module ); aout_FiltersDestroyPipeline( p_aout, p_aout->output.pp_filters, p_aout->output.i_nb_filters ); + + aout_lock_output_fifo( p_aout ); aout_FifoDestroy( p_aout, &p_aout->output.fifo ); + aout_unlock_output_fifo( p_aout ); + + p_aout->output.b_error = true; } /***************************************************************************** * aout_OutputPlay : play a buffer + ***************************************************************************** + * This function is entered with the mixer lock. *****************************************************************************/ void aout_OutputPlay( aout_instance_t * p_aout, aout_buffer_t * p_buffer ) { @@ -151,9 +249,16 @@ void aout_OutputPlay( aout_instance_t * p_aout, aout_buffer_t * p_buffer ) p_aout->output.i_nb_filters, &p_buffer ); - aout_FifoPush( p_aout, &p_aout->output.fifo, p_buffer ); + if( p_buffer->i_nb_bytes == 0 ) + { + aout_BufferFree( p_buffer ); + return; + } + aout_lock_output_fifo( p_aout ); + aout_FifoPush( p_aout, &p_aout->output.fifo, p_buffer ); p_aout->output.pf_play( p_aout ); + aout_unlock_output_fifo( p_aout ); } /***************************************************************************** @@ -161,60 +266,98 @@ void aout_OutputPlay( aout_instance_t * p_aout, aout_buffer_t * p_buffer ) ***************************************************************************** * If b_can_sleek is 1, the aout core functions won't try to resample * new buffers to catch up - that is we suppose that the output plug-in can - * do it by itself. S/PDIF outputs should always set b_can_sleek = 1. + * compensate it by itself. S/PDIF outputs should always set b_can_sleek = 1. + * This function is entered with no lock at all :-). *****************************************************************************/ aout_buffer_t * aout_OutputNextBuffer( aout_instance_t * p_aout, - mtime_t start_date , - vlc_bool_t b_can_sleek ) + mtime_t start_date, + bool b_can_sleek ) { aout_buffer_t * p_buffer; - vlc_mutex_lock( &p_aout->output.fifo.lock ); + aout_lock_output_fifo( p_aout ); + p_buffer = p_aout->output.fifo.p_first; - while ( p_buffer != NULL && p_buffer->end_date < start_date ) + /* Drop the audio sample if the audio output is really late. + * In the case of b_can_sleek, we don't use a resampler so we need to be + * a lot more severe. */ + while ( p_buffer && p_buffer->start_date < + (b_can_sleek ? start_date : mdate()) - AOUT_PTS_TOLERANCE ) { - msg_Dbg( p_aout, "audio output is too slow (%lld)", - start_date - p_buffer->end_date ); + msg_Dbg( p_aout, "audio output is too slow (%"PRId64"), " + "trashing %"PRId64"us", mdate() - p_buffer->start_date, + p_buffer->end_date - p_buffer->start_date ); p_buffer = p_buffer->p_next; + aout_BufferFree( p_aout->output.fifo.p_first ); + p_aout->output.fifo.p_first = p_buffer; } - p_aout->output.fifo.p_first = p_buffer; if ( p_buffer == NULL ) { p_aout->output.fifo.pp_last = &p_aout->output.fifo.p_first; - vlc_mutex_unlock( &p_aout->output.fifo.lock ); - msg_Dbg( p_aout, "audio output is starving" ); + +#if 0 /* This is bad because the audio output might just be trying to fill + * in its internal buffers. And anyway, it's up to the audio output + * to deal with this kind of starvation. */ + + /* Set date to 0, to allow the mixer to send a new buffer ASAP */ + aout_FifoSet( p_aout, &p_aout->output.fifo, 0 ); + if ( !p_aout->output.b_starving ) + msg_Dbg( p_aout, + "audio output is starving (no input), playing silence" ); + p_aout->output.b_starving = 1; +#endif + + aout_unlock_output_fifo( p_aout ); return NULL; } /* Here we suppose that all buffers have the same duration - this is - * generally true, and anyway if it's wrong it won't be a disaster. */ + * generally true, and anyway if it's wrong it won't be a disaster. + */ if ( p_buffer->start_date > start_date + (p_buffer->end_date - p_buffer->start_date) ) + /* + * + AOUT_PTS_TOLERANCE ) + * There is no reason to want that, it just worsen the scheduling of + * an audio sample after an output starvation (ie. on start or on resume) + * --Gibalou + */ { - vlc_mutex_unlock( &p_aout->output.fifo.lock ); - msg_Dbg( p_aout, "audio output is starving (%lld)", - p_buffer->start_date - start_date ); + const mtime_t i_delta = p_buffer->start_date - start_date; + aout_unlock_output_fifo( p_aout ); + + if ( !p_aout->output.b_starving ) + msg_Dbg( p_aout, "audio output is starving (%"PRId64"), " + "playing silence", i_delta ); + p_aout->output.b_starving = 1; return NULL; } -#if 0 - if ( !b_can_sleek ) + p_aout->output.b_starving = 0; + + if ( !b_can_sleek && + ( (p_buffer->start_date - start_date > AOUT_PTS_TOLERANCE) + || (start_date - p_buffer->start_date > AOUT_PTS_TOLERANCE) ) ) { /* Try to compensate the drift by doing some resampling. */ int i; + mtime_t difference = start_date - p_buffer->start_date; + msg_Warn( p_aout, "output date isn't PTS date, requesting " + "resampling (%"PRId64")", difference ); - /* Take the mixer lock because no input can be removed when the - * the mixer lock is taken. */ - vlc_mutex_lock( &p_aout->output.fifo.lock ); - for ( i = 0; i < p_input->i_nb_inputs; i++ ) + aout_lock_input_fifos( p_aout ); + for ( i = 0; i < p_aout->i_nb_inputs; i++ ) { - aout_input_t * p_input = p_aout->pp_inputs[i]; + aout_fifo_t * p_fifo = &p_aout->pp_inputs[i]->fifo; + + aout_FifoMoveDates( p_aout, p_fifo, difference ); } - vlc_mutex_lock( &p_aout->output.fifo.lock ); + + aout_FifoMoveDates( p_aout, &p_aout->output.fifo, difference ); + aout_unlock_input_fifos( p_aout ); } -#endif p_aout->output.fifo.p_first = p_buffer->p_next; if ( p_buffer->p_next == NULL ) @@ -222,6 +365,6 @@ aout_buffer_t * aout_OutputNextBuffer( aout_instance_t * p_aout, p_aout->output.fifo.pp_last = &p_aout->output.fifo.p_first; } - vlc_mutex_unlock( &p_aout->output.fifo.lock ); + aout_unlock_output_fifo( p_aout ); return p_buffer; }