X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;ds=sidebyside;f=src%2Faudio_output%2Foutput.c;h=8644b34f4c83b8a0b3e24e68a543d4f55b4830e4;hb=2516061e583b6789c403bb9c8d73e5197c76b0f0;hp=d15d9909af0d133acc04d7855c281394bd0fcc76;hpb=f76500382dc059091e0d811ed55a22337d2fe04f;p=vlc diff --git a/src/audio_output/output.c b/src/audio_output/output.c index d15d9909af..8644b34f4c 100644 --- a/src/audio_output/output.c +++ b/src/audio_output/output.c @@ -30,6 +30,10 @@ #include #include +#include +#include + +#include "libvlc.h" #include "aout_internal.h" /***************************************************************************** @@ -37,28 +41,23 @@ ***************************************************************************** * This function is entered with the mixer lock. *****************************************************************************/ -int aout_OutputNew( aout_instance_t * p_aout, - audio_sample_format_t * p_format ) +int aout_OutputNew( audio_output_t * p_aout, + const audio_sample_format_t * p_format ) { - /* Retrieve user defaults. */ - int i_rate = config_GetInt( p_aout, "aout-rate" ); - vlc_value_t val, text; - /* kludge to avoid a fpu error when rate is 0... */ - if( i_rate == 0 ) i_rate = -1; + vlc_assert_locked( &p_aout->lock ); + p_aout->format = *p_format; - 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 ); - - aout_lock_output_fifo( p_aout ); + /* Retrieve user defaults. */ + int i_rate = var_InheritInteger( p_aout, "aout-rate" ); + if ( i_rate != 0 ) + p_aout->format.i_rate = i_rate; + aout_FormatPrepare( &p_aout->format ); /* Find the best output plug-in. */ - p_aout->output.p_module = module_need( p_aout, "audio output", "$aout", false ); - if ( p_aout->output.p_module == NULL ) + p_aout->module = module_need( p_aout, "audio output", "$aout", false ); + if ( p_aout->module == NULL ) { msg_Err( p_aout, "no suitable audio output module" ); - aout_unlock_output_fifo( p_aout ); return -1; } @@ -66,36 +65,33 @@ int aout_OutputNew( aout_instance_t * p_aout, (VLC_VAR_INTEGER | VLC_VAR_HASCHOICE) ) { /* 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 ) + switch( var_InheritInteger( p_aout, "audio-channels" ) ) { - p_aout->output.output.i_original_channels - = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_DOLBYSTEREO; + case AOUT_VAR_CHAN_RSTEREO: + p_aout->format.i_original_channels |= AOUT_CHAN_REVERSESTEREO; + break; + case AOUT_VAR_CHAN_STEREO: + p_aout->format.i_original_channels = + AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT; + break; + case AOUT_VAR_CHAN_LEFT: + p_aout->format.i_original_channels = AOUT_CHAN_LEFT; + break; + case AOUT_VAR_CHAN_RIGHT: + p_aout->format.i_original_channels = AOUT_CHAN_RIGHT; + break; + case AOUT_VAR_CHAN_DOLBYS: + p_aout->format.i_original_channels = + AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_DOLBYSTEREO; + break; } } - else if ( p_aout->output.output.i_physical_channels == AOUT_CHAN_CENTER - && (p_aout->output.output.i_original_channels + else if ( p_aout->format.i_physical_channels == AOUT_CHAN_CENTER + && (p_aout->format.i_original_channels & AOUT_CHAN_PHYSMASK) == (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT) ) { + vlc_value_t val, text; + /* Mono - create the audio-channels variable. */ var_Create( p_aout, "audio-channels", VLC_VAR_INTEGER | VLC_VAR_HASCHOICE ); @@ -108,28 +104,29 @@ int aout_OutputNew( aout_instance_t * p_aout, 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 ) + if ( p_aout->format.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 ); + p_aout->format.i_original_channels = AOUT_CHAN_LEFT; + var_SetInteger( p_aout, "audio-channels", AOUT_VAR_CHAN_LEFT ); } var_AddCallback( p_aout, "audio-channels", aout_ChannelsRestart, NULL ); } - else if ( p_aout->output.output.i_physical_channels == + else if ( p_aout->format.i_physical_channels == (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT) - && (p_aout->output.output.i_original_channels & + && (p_aout->format.i_original_channels & (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT)) ) { + vlc_value_t val, text; + /* 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 ) + if ( p_aout->format.i_original_channels & AOUT_CHAN_DOLBYSTEREO ) { val.i_int = AOUT_VAR_CHAN_DOLBYS; text.psz_string = _("Dolby Surround"); @@ -146,70 +143,57 @@ int aout_OutputNew( aout_instance_t * p_aout, 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 ) + if ( p_aout->format.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 ); + p_aout->format.i_original_channels = AOUT_CHAN_LEFT; + var_SetInteger( p_aout, "audio-channels", AOUT_VAR_CHAN_LEFT ); } var_AddCallback( p_aout, "audio-channels", aout_ChannelsRestart, NULL ); } - val.b_bool = true; - var_Set( p_aout, "intf-change", val ); + var_TriggerCallback( p_aout, "intf-change" ); - aout_FormatPrepare( &p_aout->output.output ); + aout_FormatPrepare( &p_aout->format ); /* 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 ); + aout_FifoInit( p_aout, &p_aout->fifo, p_aout->format.i_rate ); + aout_FormatPrint( p_aout, "output", &p_aout->format ); - /* Calculate the resulting mixer output format. */ - p_aout->mixer_format = p_aout->output.output; - if ( !AOUT_FMT_NON_LINEAR(&p_aout->output.output) ) - { - /* Non-S/PDIF mixer only deals with float32 or fixed32. */ - p_aout->mixer_format.i_format - = (vlc_CPU() & CPU_CAPABILITY_FPU) ? - VLC_CODEC_FL32 : VLC_CODEC_FI32; - aout_FormatPrepare( &p_aout->mixer_format ); - } - else - { + /* Choose the mixer format. */ + p_aout->mixer_format = p_aout->format; + if ( AOUT_FMT_NON_LINEAR(&p_aout->format) ) p_aout->mixer_format.i_format = p_format->i_format; - } + else + /* Most audio filters can only deal with single-precision, + * so lets always use that when hardware supports floating point. */ + if( HAVE_FPU ) + p_aout->mixer_format.i_format = VLC_CODEC_FL32; + else + /* Otherwise, audio filters will not work. Use fixed-point if the input has + * more than 16-bits depth. */ + if( p_format->i_bitspersample > 16 ) + p_aout->mixer_format.i_format = VLC_CODEC_FI32; + else + /* Fallback to 16-bits. This avoids pointless conversion to and from + * 32-bits samples for the sole purpose of software mixing. */ + p_aout->mixer_format.i_format = VLC_CODEC_S16N; + aout_FormatPrepare( &p_aout->mixer_format ); aout_FormatPrint( p_aout, "mixer", &p_aout->mixer_format ); /* 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->i_nb_filters = 0; + if ( aout_FiltersCreatePipeline( p_aout, p_aout->pp_filters, + &p_aout->i_nb_filters, &p_aout->mixer_format, - &p_aout->output.output ) < 0 ) + &p_aout->format ) < 0 ) { msg_Err( p_aout, "couldn't create audio output pipeline" ); - module_unneed( p_aout, p_aout->output.p_module ); + module_unneed( p_aout, p_aout->module ); + p_aout->module = NULL; return -1; } - - /* Prepare hints for the buffer allocator. */ - p_aout->mixer_allocation.b_alloc = true; - p_aout->mixer_allocation.i_bytes_per_sec - = p_aout->mixer_format.i_bytes_per_frame - * p_aout->mixer_format.i_rate - / p_aout->mixer_format.i_frame_length; - - aout_FiltersHintBuffers( p_aout, p_aout->output.pp_filters, - p_aout->output.i_nb_filters, - &p_aout->mixer_allocation ); - - p_aout->output.b_error = 0; return 0; } @@ -218,23 +202,17 @@ int aout_OutputNew( aout_instance_t * p_aout, ***************************************************************************** * This function is entered with the mixer lock. *****************************************************************************/ -void aout_OutputDelete( aout_instance_t * p_aout ) +void aout_OutputDelete( audio_output_t * p_aout ) { - 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 ); + vlc_assert_locked( &p_aout->lock ); - aout_lock_output_fifo( p_aout ); - aout_FifoDestroy( p_aout, &p_aout->output.fifo ); - aout_unlock_output_fifo( p_aout ); + if( p_aout->module == NULL ) + return; - p_aout->output.b_error = true; + module_unneed( p_aout, p_aout->module ); + p_aout->module = NULL; + aout_FiltersDestroyPipeline( p_aout->pp_filters, p_aout->i_nb_filters ); + aout_FifoDestroy( &p_aout->fifo ); } /***************************************************************************** @@ -242,22 +220,34 @@ void aout_OutputDelete( aout_instance_t * p_aout ) ***************************************************************************** * This function is entered with the mixer lock. *****************************************************************************/ -void aout_OutputPlay( aout_instance_t * p_aout, aout_buffer_t * p_buffer ) +void aout_OutputPlay( audio_output_t * p_aout, aout_buffer_t * p_buffer ) { - aout_FiltersPlay( p_aout, p_aout->output.pp_filters, - p_aout->output.i_nb_filters, - &p_buffer ); + vlc_assert_locked( &p_aout->lock ); - if( p_buffer->i_nb_bytes == 0 ) + aout_FiltersPlay( p_aout->pp_filters, p_aout->i_nb_filters, &p_buffer ); + if( !p_buffer ) + return; + if( p_buffer->i_buffer == 0 ) { - aout_BufferFree( p_buffer ); + block_Release( 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 ); + aout_FifoPush( &p_aout->fifo, p_buffer ); + p_aout->pf_play( p_aout ); +} + +/** + * Notifies the audio output (if any) of pause/resume events. + * This enables the output to expedite pause, instead of waiting for its + * buffers to drain. + */ +void aout_OutputPause( audio_output_t *aout, bool pause, mtime_t date ) +{ + vlc_assert_locked( &aout->lock ); + + if( aout->pf_pause != NULL ) + aout->pf_pause( aout, pause, date ); } /***************************************************************************** @@ -268,102 +258,72 @@ void aout_OutputPlay( aout_instance_t * p_aout, aout_buffer_t * p_buffer ) * 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, +aout_buffer_t * aout_OutputNextBuffer( audio_output_t * p_aout, mtime_t start_date, bool b_can_sleek ) { + aout_fifo_t *p_fifo = &p_aout->fifo; aout_buffer_t * p_buffer; + mtime_t now = mdate(); - aout_lock_output_fifo( p_aout ); - - p_buffer = p_aout->output.fifo.p_first; + aout_lock( p_aout ); /* 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 ) + while( ((p_buffer = p_fifo->p_first) != NULL) + && p_buffer->i_pts < (b_can_sleek ? start_date : now) - AOUT_MAX_PTS_DELAY ) { 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; + "trashing %"PRId64"us", now - p_buffer->i_pts, + p_buffer->i_length ); + aout_BufferFree( aout_FifoPop( p_fifo ) ); } - if ( p_buffer == NULL ) + if( p_buffer == NULL ) { - p_aout->output.fifo.pp_last = &p_aout->output.fifo.p_first; - #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 ) + aout_FifoSet( &p_aout->fifo, 0 ); + if ( !p_aout->b_starving ) msg_Dbg( p_aout, "audio output is starving (no input), playing silence" ); - p_aout->output.b_starving = 1; + p_aout->b_starving = true; #endif - - aout_unlock_output_fifo( p_aout ); - return NULL; + goto out; } + mtime_t delta = start_date - p_buffer->i_pts; /* 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. */ - 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 - */ + if ( 0 > delta + p_buffer->i_length ) { - const mtime_t i_delta = p_buffer->start_date - start_date; - aout_unlock_output_fifo( p_aout ); - - if ( !p_aout->output.b_starving ) + if ( !p_aout->b_starving ) msg_Dbg( p_aout, "audio output is starving (%"PRId64"), " - "playing silence", i_delta ); - p_aout->output.b_starving = 1; - return NULL; + "playing silence", -delta ); + p_aout->b_starving = true; + p_buffer = NULL; + goto out; } - p_aout->output.b_starving = 0; + p_aout->b_starving = false; + p_buffer = aout_FifoPop( p_fifo ); - if ( !b_can_sleek && - ( (p_buffer->start_date - start_date > AOUT_PTS_TOLERANCE) - || (start_date - p_buffer->start_date > AOUT_PTS_TOLERANCE) ) ) + if( !b_can_sleek + && ( delta > AOUT_MAX_PTS_DELAY || delta < -AOUT_MAX_PTS_ADVANCE ) ) { /* 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 ); - - aout_lock_input_fifos( p_aout ); - for ( i = 0; i < p_aout->i_nb_inputs; i++ ) - { - aout_fifo_t * p_fifo = &p_aout->pp_inputs[i]->mixer.fifo; + "resampling (%"PRId64")", delta ); - aout_FifoMoveDates( p_aout, p_fifo, difference ); - } - - aout_FifoMoveDates( p_aout, &p_aout->output.fifo, difference ); - aout_unlock_input_fifos( p_aout ); - } - - p_aout->output.fifo.p_first = p_buffer->p_next; - if ( p_buffer->p_next == NULL ) - { - p_aout->output.fifo.pp_last = &p_aout->output.fifo.p_first; + aout_FifoMoveDates( &p_aout->p_input->mixer.fifo, delta ); + aout_FifoMoveDates( p_fifo, delta ); } - - aout_unlock_output_fifo( p_aout ); +out: + aout_unlock( p_aout ); return p_buffer; }