X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=modules%2Faudio_output%2Falsa.c;h=bead5dbbbe4b0898b331703f973ce903aeb4db7d;hb=2cce4b1d5e6a1e4bc60849095f947bf0d15e90f3;hp=79e335c469ed9e3c28e3cea9ebcd3fd29b091452;hpb=d24f1af688790c05e9983e05033807fa87d45289;p=vlc diff --git a/modules/audio_output/alsa.c b/modules/audio_output/alsa.c index 79e335c469..bead5dbbbe 100644 --- a/modules/audio_output/alsa.c +++ b/modules/audio_output/alsa.c @@ -2,7 +2,7 @@ * alsa.c : alsa plugin for vlc ***************************************************************************** * Copyright (C) 2000-2001 VideoLAN - * $Id: alsa.c,v 1.2 2002/08/14 10:50:12 bozo Exp $ + * $Id: alsa.c,v 1.28 2003/05/26 19:06:47 gbazin Exp $ * * Authors: Henri Fallon - Original Author * Jeffrey Baker - Port to ALSA 1.0 API @@ -13,7 +13,7 @@ * 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 @@ -37,7 +37,10 @@ #include "aout_internal.h" -/* ALSA part */ +/* ALSA part + Note: we use the new API which is available since 0.9.0beta10a. */ +#define ALSA_PCM_NEW_HW_PARAMS_API +#define ALSA_PCM_NEW_SW_PARAMS_API #include /***************************************************************************** @@ -49,35 +52,35 @@ struct aout_sys_t { snd_pcm_t * p_snd_pcm; - snd_pcm_sframes_t i_buffer_size; int i_period_time; - volatile vlc_bool_t b_initialized; - #ifdef DEBUG snd_output_t * p_snd_stderr; #endif }; +#define A52_FRAME_NB 1536 + /* These values are in frames. - * To convert them to a numer of bytes you have to multiply them by the - * number of channel(s) (eg. 2 for stereo) and the size of a sample (eg. - * 2 for s16). */ + To convert them to a number of bytes you have to multiply them by the + number of channel(s) (eg. 2 for stereo) and the size of a sample (eg. + 2 for s16). */ #define ALSA_DEFAULT_PERIOD_SIZE 2048 #define ALSA_DEFAULT_BUFFER_SIZE ( ALSA_DEFAULT_PERIOD_SIZE << 4 ) -#define ALSA_SPDIF_PERIOD_SIZE 1536 +#define ALSA_SPDIF_PERIOD_SIZE A52_FRAME_NB #define ALSA_SPDIF_BUFFER_SIZE ( ALSA_SPDIF_PERIOD_SIZE << 4 ) +/* Why << 4 ? --Meuuh */ +/* Why not ? --Bozo */ +/* Right. --Meuuh */ + +#define DEFAULT_ALSA_DEVICE "default" /***************************************************************************** * Local prototypes *****************************************************************************/ static int Open ( vlc_object_t * ); static void Close ( vlc_object_t * ); - -static int SetFormat ( aout_instance_t * ); -static void Play ( aout_instance_t *, - aout_buffer_t * ); - +static void Play ( aout_instance_t * ); static int ALSAThread ( aout_instance_t * ); static void ALSAFill ( aout_instance_t * ); @@ -85,223 +88,445 @@ static void ALSAFill ( aout_instance_t * ); * Module descriptor *****************************************************************************/ vlc_module_begin(); - add_category_hint( N_("Audio"), NULL ); - add_string( "alsa-device", NULL, NULL, N_("Name"), NULL ); - set_description( _("ALSA audio module") ); + add_category_hint( N_("ALSA"), NULL, VLC_FALSE ); + add_string( "alsadev", DEFAULT_ALSA_DEVICE, aout_FindAndRestart, + N_("ALSA device name"), NULL, VLC_FALSE ); + set_description( _("ALSA audio output") ); set_capability( "audio output", 50 ); set_callbacks( Open, Close ); vlc_module_end(); +/***************************************************************************** + * Probe: probe the audio device for available formats and channels + *****************************************************************************/ +static void Probe( aout_instance_t * p_aout, + const char * psz_device, const char * psz_iec_device, + int i_snd_pcm_format ) +{ + struct aout_sys_t * p_sys = p_aout->output.p_sys; + vlc_value_t val, text; + + var_Create ( p_aout, "audio-device", VLC_VAR_INTEGER | VLC_VAR_HASCHOICE ); + text.psz_string = _("Audio device"); + var_Change( p_aout, "audio-device", VLC_VAR_SETTEXT, &text, NULL ); + + /* Now test linear PCM capabilities */ + if ( !snd_pcm_open( &p_sys->p_snd_pcm, psz_device, + SND_PCM_STREAM_PLAYBACK, 0 ) ) + { + int i_channels; + snd_pcm_hw_params_t * p_hw; + snd_pcm_hw_params_alloca (&p_hw); + + if ( snd_pcm_hw_params_any( p_sys->p_snd_pcm, p_hw ) < 0 ) + { + msg_Warn( p_aout, "unable to retrieve initial hardware parameters" + ", disabling linear PCM audio" ); + snd_pcm_close( p_sys->p_snd_pcm ); + var_Destroy( p_aout, "audio-device" ); + return; + } + + if ( snd_pcm_hw_params_set_format( p_sys->p_snd_pcm, p_hw, + i_snd_pcm_format ) < 0 ) + { + /* Assume a FPU enabled computer can handle float32 format. + If somebody tells us it's not always true then we'll have + to change this */ + msg_Warn( p_aout, "unable to set stream sample size and word order" + ", disabling linear PCM audio" ); + snd_pcm_close( p_sys->p_snd_pcm ); + var_Destroy( p_aout, "audio-device" ); + return; + } + + i_channels = aout_FormatNbChannels( &p_aout->output.output ); + + while ( i_channels > 0 ) + { + /* Here we have to probe multi-channel capabilities but I have + no idea (at the moment) of how its managed by the ALSA + library. + It seems that '6' channels aren't well handled on a stereo + sound card like my i810 but it requires some more + investigations. That's why '4' and '6' cases are disabled. + -- Bozo */ + if ( !snd_pcm_hw_params_test_channels( p_sys->p_snd_pcm, p_hw, + i_channels ) ) + { + switch ( i_channels ) + { + case 1: + val.i_int = AOUT_VAR_MONO; + text.psz_string = N_("Mono"); + var_Change( p_aout, "audio-device", + VLC_VAR_ADDCHOICE, &val, &text ); + break; + case 2: + val.i_int = AOUT_VAR_STEREO; + text.psz_string = N_("Stereo"); + var_Change( p_aout, "audio-device", + VLC_VAR_ADDCHOICE, &val, &text ); + break; +/* + case 4: + val.i_int = AOUT_VAR_2F2R; + text.psz_string = N_("2 Front 2 Rear"); + var_Change( p_aout, "audio-device", + VLC_VAR_ADDCHOICE, &val, &text ); + break; + case 6: + val.i_int = AOUT_VAR_5_1; + text.psz_string = N_("5.1"); + var_Change( p_aout, "audio-device", + VLC_VAR_ADDCHOICE, &val, &text ); + break; +*/ + } + } + + --i_channels; + } + + /* Close the previously opened device */ + snd_pcm_close( p_sys->p_snd_pcm ); + } + + /* Test for S/PDIF device if needed */ + if ( psz_iec_device ) + { + /* Opening the device should be enough */ + if ( !snd_pcm_open( &p_sys->p_snd_pcm, psz_iec_device, + SND_PCM_STREAM_PLAYBACK, 0 ) ) + { + val.i_int = AOUT_VAR_SPDIF; + text.psz_string = N_("A/52 over S/PDIF"); + var_Change( p_aout, "audio-device", + VLC_VAR_ADDCHOICE, &val, &text ); + if( config_GetInt( p_aout, "spdif" ) ) + var_Set( p_aout, "audio-device", val ); + + snd_pcm_close( p_sys->p_snd_pcm ); + } + } + + var_Change( p_aout, "audio-device", VLC_VAR_CHOICESCOUNT, &val, NULL ); + if( val.i_int <= 0 ) + { + /* Probe() has failed. */ + var_Destroy( p_aout, "audio-device" ); + return; + } + + /* Add final settings to the variable */ + var_AddCallback( p_aout, "audio-device", aout_ChannelsRestart, NULL ); + val.b_bool = VLC_TRUE; + var_Set( p_aout, "intf-change", val ); +} + /***************************************************************************** * Open: create a handle and open an alsa device ***************************************************************************** - * This function opens an alsa device, through the alsa API + * This function opens an alsa device, through the alsa API. + * + * Note: the only heap-allocated string is psz_device. All the other pointers + * are references to psz_device or to stack-allocated data. *****************************************************************************/ static int Open( vlc_object_t *p_this ) { aout_instance_t * p_aout = (aout_instance_t *)p_this; struct aout_sys_t * p_sys; - char * psz_device; + vlc_value_t val; + + char psz_default_iec_device[128]; /* Buffer used to store the default + S/PDIF device */ + char * psz_device, * psz_iec_device; /* device names for PCM and S/PDIF + output */ - /* Allows user to choose which ALSA device to use */ - char psz_alsadev[128]; - char * psz_userdev; - int i_snd_rc; + int i_vlc_pcm_format; /* Audio format for VLC's data */ + int i_snd_pcm_format; /* Audio format for ALSA's data */ + + snd_pcm_uframes_t i_buffer_size = 0; + snd_pcm_uframes_t i_period_size = 0; + int i_channels = 0; + + snd_pcm_hw_params_t *p_hw; + snd_pcm_sw_params_t *p_sw; + + int i_snd_rc = -1; /* Allocate structures */ p_aout->output.p_sys = p_sys = malloc( sizeof( aout_sys_t ) ); if( p_sys == NULL ) { msg_Err( p_aout, "out of memory" ); - return -1; + return VLC_ENOMEM; } -#ifdef DEBUG - snd_output_stdio_attach( &p_sys->p_snd_stderr, stderr, 0 ); -#endif + /* Get device name */ + if( (psz_device = config_GetPsz( p_aout, "alsadev" )) == NULL ) + { + msg_Err( p_aout, "no audio device given (maybe \"default\" ?)" ); + free( p_sys ); + return VLC_EGENERIC; + } - /* Read in ALSA device preferences from configuration */ - psz_userdev = config_GetPsz( p_aout, "alsa-device" ); + /* Choose the IEC device for S/PDIF output: + if the device is overriden by the user then it will be the one + otherwise we compute the default device based on the output format. */ + if( AOUT_FMT_NON_LINEAR( &p_aout->output.output ) + && !strcmp( psz_device, DEFAULT_ALSA_DEVICE ) ) + { + snprintf( psz_default_iec_device, sizeof(psz_default_iec_device), + "iec958:AES0=0x%x,AES1=0x%x,AES2=0x%x,AES3=0x%x", + IEC958_AES0_CON_EMPHASIS_NONE | IEC958_AES0_NONAUDIO, + IEC958_AES1_CON_ORIGINAL | IEC958_AES1_CON_PCM_CODER, + 0, + ( p_aout->output.output.i_rate == 48000 ? + IEC958_AES3_CON_FS_48000 : + ( p_aout->output.output.i_rate == 44100 ? + IEC958_AES3_CON_FS_44100 : IEC958_AES3_CON_FS_32000 ) ) ); + psz_iec_device = psz_default_iec_device; + } + else if( AOUT_FMT_NON_LINEAR( &p_aout->output.output ) ) + { + psz_iec_device = psz_device; + } + else + { + psz_iec_device = NULL; + } - if( psz_userdev ) + /* Choose the linear PCM format (read the comment above about FPU + and float32) */ + if( p_aout->p_libvlc->i_cpu & CPU_CAPABILITY_FPU ) { - psz_device = psz_userdev; + i_vlc_pcm_format = VLC_FOURCC('f','l','3','2'); + i_snd_pcm_format = SND_PCM_FORMAT_FLOAT; } else { - /* Use the internal logic to decide on the device name */ - if ( p_aout->output.output.i_format == AOUT_FMT_SPDIF ) - { - unsigned char s[4]; - s[0] = IEC958_AES0_CON_EMPHASIS_NONE | IEC958_AES0_NONAUDIO; - s[1] = IEC958_AES1_CON_ORIGINAL | IEC958_AES1_CON_PCM_CODER; - s[2] = 0; - s[3] = IEC958_AES3_CON_FS_48000; - sprintf( psz_alsadev, - "iec958:AES0=0x%x,AES1=0x%x,AES2=0x%x,AES3=0x%x", - s[0], s[1], s[2], s[3] ); - psz_device = psz_alsadev; - p_sys->i_buffer_size = ALSA_SPDIF_BUFFER_SIZE; - p_aout->output.i_nb_samples = ALSA_SPDIF_PERIOD_SIZE; - } - else - { - psz_device = "default"; - p_sys->i_buffer_size = ALSA_DEFAULT_BUFFER_SIZE; - p_aout->output.i_nb_samples = ALSA_DEFAULT_PERIOD_SIZE; - } + i_vlc_pcm_format = AOUT_FMT_S16_NE; + i_snd_pcm_format = SND_PCM_FORMAT_S16; } - /* Open device */ - i_snd_rc = snd_pcm_open( &p_sys->p_snd_pcm, psz_device, - SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); - if( i_snd_rc < 0 ) + /* If the variable doesn't exist then it's the first time we're called + and we have to probe the available audio formats and channels */ + if ( var_Type( p_aout, "audio-device" ) == 0 ) + { + Probe( p_aout, psz_device, psz_iec_device, i_snd_pcm_format ); + } + + if ( var_Get( p_aout, "audio-device", &val ) < 0 ) { - msg_Err( p_aout, "cannot open ALSA device `%s' (%s)", - psz_device, snd_strerror(i_snd_rc) ); - if( psz_userdev ) - free( psz_userdev ); free( p_sys ); - p_sys->p_snd_pcm = NULL; - return -1; + free( psz_device ); + return VLC_EGENERIC; } - if( psz_userdev ) + if ( val.i_int == AOUT_VAR_SPDIF ) + { + p_aout->output.output.i_format = VLC_FOURCC('s','p','d','i'); + } + else if ( val.i_int == AOUT_VAR_5_1 ) + { + p_aout->output.output.i_format = i_vlc_pcm_format; + p_aout->output.output.i_physical_channels + = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_CENTER + | AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT + | AOUT_CHAN_LFE; + } + else if ( val.i_int == AOUT_VAR_2F2R ) { - free( psz_userdev ); + p_aout->output.output.i_format = i_vlc_pcm_format; + p_aout->output.output.i_physical_channels + = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT + | AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT; + } + else if ( val.i_int == AOUT_VAR_STEREO ) + { + p_aout->output.output.i_format = i_vlc_pcm_format; + p_aout->output.output.i_physical_channels + = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT; + } + else if ( val.i_int == AOUT_VAR_MONO ) + { + p_aout->output.output.i_format = i_vlc_pcm_format; + p_aout->output.output.i_physical_channels = AOUT_CHAN_CENTER; } - /* Create ALSA thread and wait for its readiness. */ - p_sys->b_initialized = VLC_FALSE; - if( vlc_thread_create( p_aout, "aout", ALSAThread, VLC_FALSE ) ) + else { - msg_Err( p_aout, "cannot create ALSA thread (%s)", strerror(errno) ); - if( psz_userdev ) - free( psz_userdev ); + /* This should not happen ! */ + msg_Err( p_aout, "internal: can't find audio-device (%i)", val.i_int ); free( p_sys ); - return -1; + return VLC_EGENERIC; } - p_aout->output.pf_setformat = SetFormat; - p_aout->output.pf_play = Play; +#ifdef DEBUG + snd_output_stdio_attach( &p_sys->p_snd_stderr, stderr, 0 ); +#endif - return 0; -} + /* Open the device */ + if ( AOUT_FMT_NON_LINEAR( &p_aout->output.output ) ) + { + if ( ( i_snd_rc = snd_pcm_open( &p_sys->p_snd_pcm, psz_iec_device, + SND_PCM_STREAM_PLAYBACK, 0 ) ) < 0 ) + { + msg_Err( p_aout, "cannot open ALSA device `%s' (%s)", + psz_iec_device, snd_strerror( i_snd_rc ) ); + free( p_sys ); + free( psz_device ); + return VLC_EGENERIC; + } + i_buffer_size = ALSA_SPDIF_BUFFER_SIZE; + i_snd_pcm_format = SND_PCM_FORMAT_S16; + i_channels = 2; -/***************************************************************************** - * SetFormat : sets the alsa output format - ***************************************************************************** - * This function prepares the device, sets the rate, format, the mode - * ( "play as soon as you have data" ), and buffer information. - *****************************************************************************/ -static int SetFormat( aout_instance_t * p_aout ) -{ - struct aout_sys_t * p_sys = p_aout->output.p_sys; + p_aout->output.i_nb_samples = i_period_size = ALSA_SPDIF_PERIOD_SIZE; + p_aout->output.output.i_bytes_per_frame = AOUT_SPDIF_SIZE; + p_aout->output.output.i_frame_length = A52_FRAME_NB; - int i_snd_rc; - int i_format; + aout_VolumeNoneInit( p_aout ); + } + else + { + if ( ( i_snd_rc = snd_pcm_open( &p_sys->p_snd_pcm, psz_device, + SND_PCM_STREAM_PLAYBACK, 0 ) ) < 0 ) + { + msg_Err( p_aout, "cannot open ALSA device `%s' (%s)", + psz_device, snd_strerror( i_snd_rc ) ); + free( p_sys ); + free( psz_device ); + return VLC_EGENERIC; + } + i_buffer_size = ALSA_DEFAULT_BUFFER_SIZE; + i_channels = aout_FormatNbChannels( &p_aout->output.output ); - snd_pcm_hw_params_t *p_hw; - snd_pcm_sw_params_t *p_sw; + p_aout->output.i_nb_samples = i_period_size = ALSA_DEFAULT_PERIOD_SIZE; + + aout_VolumeSoftInit( p_aout ); + } + + /* Free psz_device so that all the remaining data is stack-allocated */ + free( psz_device ); + + p_aout->output.pf_play = Play; snd_pcm_hw_params_alloca(&p_hw); snd_pcm_sw_params_alloca(&p_sw); - switch (p_aout->output.output.i_format) - { - case AOUT_FMT_MU_LAW: i_format = SND_PCM_FORMAT_MU_LAW; break; - case AOUT_FMT_A_LAW: i_format = SND_PCM_FORMAT_A_LAW; break; - case AOUT_FMT_IMA_ADPCM: i_format = SND_PCM_FORMAT_IMA_ADPCM; break; - case AOUT_FMT_U8: i_format = SND_PCM_FORMAT_U8; break; - case AOUT_FMT_S16_LE: i_format = SND_PCM_FORMAT_S16_LE; break; - case AOUT_FMT_S16_BE: i_format = SND_PCM_FORMAT_S16_BE; break; - case AOUT_FMT_S8: i_format = SND_PCM_FORMAT_S8; break; - case AOUT_FMT_U16_LE: i_format = SND_PCM_FORMAT_U16_LE; break; - case AOUT_FMT_U16_BE: i_format = SND_PCM_FORMAT_U16_BE; break; - case AOUT_FMT_FLOAT32: i_format = SND_PCM_FORMAT_FLOAT; break; - case AOUT_FMT_FIXED32: - default: - msg_Err( p_aout, "audio output format 0x%x not supported", - p_aout->output.output.i_format ); - return -1; - break; - } - - i_snd_rc = snd_pcm_hw_params_any( p_sys->p_snd_pcm, p_hw ); - if( i_snd_rc < 0 ) + /* Get Initial hardware parameters */ + if ( ( i_snd_rc = snd_pcm_hw_params_any( p_sys->p_snd_pcm, p_hw ) ) < 0 ) { - msg_Err( p_aout, "unable to retrieve initial hardware parameters" ); - return -1; + msg_Err( p_aout, "unable to retrieve initial hardware parameters (%s)", + snd_strerror( i_snd_rc ) ); + goto error; } - i_snd_rc = snd_pcm_hw_params_set_access( p_sys->p_snd_pcm, p_hw, - SND_PCM_ACCESS_RW_INTERLEAVED ); - if( i_snd_rc < 0 ) + /* Set format. */ + if ( ( i_snd_rc = snd_pcm_hw_params_set_format( p_sys->p_snd_pcm, p_hw, + i_snd_pcm_format ) ) < 0 ) { - msg_Err( p_aout, "unable to set interleaved stream format" ); - return -1; + msg_Err( p_aout, "unable to set stream sample size and word order (%s)", + snd_strerror( i_snd_rc ) ); + goto error; } - i_snd_rc = snd_pcm_hw_params_set_format( p_sys->p_snd_pcm, p_hw, i_format ); - if( i_snd_rc < 0 ) + if ( ( i_snd_rc = snd_pcm_hw_params_set_access( p_sys->p_snd_pcm, p_hw, + SND_PCM_ACCESS_RW_INTERLEAVED ) ) < 0 ) { - msg_Err( p_aout, "unable to set stream sample size and word order" ); - return -1; + msg_Err( p_aout, "unable to set interleaved stream format (%s)", + snd_strerror( i_snd_rc ) ); + goto error; } - i_snd_rc = snd_pcm_hw_params_set_channels( p_sys->p_snd_pcm, p_hw, - p_aout->output.output.i_channels ); - if( i_snd_rc < 0 ) + /* Set channels. */ + if ( ( i_snd_rc = snd_pcm_hw_params_set_channels( p_sys->p_snd_pcm, p_hw, + i_channels ) ) < 0 ) { - msg_Err( p_aout, "unable to set number of output channels" ); - return -1; + msg_Err( p_aout, "unable to set number of output channels (%s)", + snd_strerror( i_snd_rc ) ); + goto error; } - i_snd_rc = snd_pcm_hw_params_set_rate( p_sys->p_snd_pcm, p_hw, - p_aout->output.output.i_rate, 0 ); - if( i_snd_rc < 0 ) + /* Set rate. */ +#ifdef HAVE_ALSA_NEW_API + if ( ( i_snd_rc = snd_pcm_hw_params_set_rate_near( p_sys->p_snd_pcm, p_hw, + &p_aout->output.output.i_rate, NULL ) ) < 0 ) +#else + if ( ( i_snd_rc = snd_pcm_hw_params_set_rate_near( p_sys->p_snd_pcm, p_hw, + p_aout->output.output.i_rate, NULL ) ) < 0 ) +#endif { - msg_Err( p_aout, "unable to set sample rate" ); - return -1; + msg_Err( p_aout, "unable to set sample rate (%s)", + snd_strerror( i_snd_rc ) ); + goto error; } - i_snd_rc = snd_pcm_hw_params_set_buffer_size_near( p_sys->p_snd_pcm, p_hw, - p_sys->i_buffer_size ); - if( i_snd_rc < 0 ) + /* Set buffer size. */ +#ifdef HAVE_ALSA_NEW_API + if ( ( i_snd_rc = snd_pcm_hw_params_set_buffer_size_near( p_sys->p_snd_pcm, + p_hw, &i_buffer_size ) ) < 0 ) +#else + if ( ( i_snd_rc = snd_pcm_hw_params_set_buffer_size_near( p_sys->p_snd_pcm, + p_hw, i_buffer_size ) ) < 0 ) +#endif { - msg_Err( p_aout, "unable to set buffer time" ); - return -1; + msg_Err( p_aout, "unable to set buffer size (%s)", + snd_strerror( i_snd_rc ) ); + goto error; } - p_sys->i_buffer_size = i_snd_rc; - i_snd_rc = snd_pcm_hw_params_set_period_size_near( - p_sys->p_snd_pcm, p_hw, p_aout->output.i_nb_samples, 0 ); - if( i_snd_rc < 0 ) + /* Set period size. */ +#ifdef HAVE_ALSA_NEW_API + if ( ( i_snd_rc = snd_pcm_hw_params_set_period_size_near( p_sys->p_snd_pcm, + p_hw, &i_period_size, NULL ) ) < 0 ) +#else + if ( ( i_snd_rc = snd_pcm_hw_params_set_period_size_near( p_sys->p_snd_pcm, + p_hw, i_period_size, NULL ) ) < 0 ) +#endif + { + msg_Err( p_aout, "unable to set period size (%s)", + snd_strerror( i_snd_rc ) ); + goto error; + } + p_aout->output.i_nb_samples = i_period_size; + + /* Commit hardware parameters. */ + if ( ( i_snd_rc = snd_pcm_hw_params( p_sys->p_snd_pcm, p_hw ) ) < 0 ) { - msg_Err( p_aout, "unable to set period size" ); - return -1; + msg_Err( p_aout, "unable to commit hardware configuration (%s)", + snd_strerror( i_snd_rc ) ); + goto error; } - p_aout->output.i_nb_samples = i_snd_rc; - p_sys->i_period_time = snd_pcm_hw_params_get_period_time( p_hw, 0 ); - i_snd_rc = snd_pcm_hw_params( p_sys->p_snd_pcm, p_hw ); - if (i_snd_rc < 0) +#ifdef HAVE_ALSA_NEW_API + if( ( i_snd_rc = snd_pcm_hw_params_get_period_time( p_hw, + &p_sys->i_period_time, NULL ) ) < 0 ) +#else + if( ( p_sys->i_period_time = + snd_pcm_hw_params_get_period_time( p_hw, NULL ) ) < 0 ) +#endif { - msg_Err( p_aout, "unable to set hardware configuration" ); - return -1; + msg_Err( p_aout, "unable to get period time (%s)", + snd_strerror( i_snd_rc ) ); + goto error; } + /* Get Initial software parameters */ snd_pcm_sw_params_current( p_sys->p_snd_pcm, p_sw ); + i_snd_rc = snd_pcm_sw_params_set_sleep_min( p_sys->p_snd_pcm, p_sw, 0 ); i_snd_rc = snd_pcm_sw_params_set_avail_min( p_sys->p_snd_pcm, p_sw, p_aout->output.i_nb_samples ); - i_snd_rc = snd_pcm_sw_params( p_sys->p_snd_pcm, p_sw ); - if( i_snd_rc < 0 ) + /* Commit software parameters. */ + if ( snd_pcm_sw_params( p_sys->p_snd_pcm, p_sw ) < 0 ) { msg_Err( p_aout, "unable to set software configuration" ); - return -1; + goto error; } #ifdef DEBUG @@ -312,21 +537,34 @@ static int SetFormat( aout_instance_t * p_aout ) snd_output_printf( p_sys->p_snd_stderr, "\n" ); #endif - p_sys->b_initialized = VLC_TRUE; + /* Create ALSA thread and wait for its readiness. */ + if( vlc_thread_create( p_aout, "aout", ALSAThread, + VLC_THREAD_PRIORITY_OUTPUT, VLC_FALSE ) ) + { + msg_Err( p_aout, "cannot create ALSA thread (%s)", strerror(errno) ); + goto error; + } return 0; + +error: + snd_pcm_close( p_sys->p_snd_pcm ); +#ifdef DEBUG + snd_output_close( p_sys->p_snd_stderr ); +#endif + free( p_sys ); + return VLC_EGENERIC; } /***************************************************************************** - * Play: queue a buffer for playing by ALSAThread + * Play: nothing to do *****************************************************************************/ -static void Play( aout_instance_t *p_aout, aout_buffer_t * p_buffer ) +static void Play( aout_instance_t *p_aout ) { - aout_FifoPush( p_aout, &p_aout->output.fifo, p_buffer ); } /***************************************************************************** - * Close: close the Alsa device + * Close: close the ALSA device *****************************************************************************/ static void Close( vlc_object_t *p_this ) { @@ -334,20 +572,22 @@ static void Close( vlc_object_t *p_this ) struct aout_sys_t * p_sys = p_aout->output.p_sys; int i_snd_rc; - p_aout->b_die = 1; + p_aout->b_die = VLC_TRUE; vlc_thread_join( p_aout ); + p_aout->b_die = VLC_FALSE; - if( p_sys->p_snd_pcm ) - { - i_snd_rc = snd_pcm_close( p_sys->p_snd_pcm ); + i_snd_rc = snd_pcm_close( p_sys->p_snd_pcm ); - if( i_snd_rc > 0 ) - { - msg_Err( p_aout, "failed closing ALSA device (%s)", - snd_strerror( i_snd_rc ) ); - } + if( i_snd_rc > 0 ) + { + msg_Err( p_aout, "failed closing ALSA device (%s)", + snd_strerror( i_snd_rc ) ); } +#ifdef DEBUG + snd_output_close( p_sys->p_snd_stderr ); +#endif + free( p_sys ); } @@ -360,15 +600,16 @@ static int ALSAThread( aout_instance_t * p_aout ) while ( !p_aout->b_die ) { - if( !p_sys->b_initialized ) - { - msleep( THREAD_SLEEP ); - continue; - } - ALSAFill( p_aout ); - msleep( p_sys->i_period_time ); + /* Sleep during less than one period to avoid a lot of buffer + underruns */ + + /* Why do we need to sleep ? --Meuuh */ + /* Maybe because I don't want to eat all the cpu by looping + all the time. --Bozo */ + /* Shouldn't snd_pcm_wait() make us wait ? --Meuuh */ + msleep( p_sys->i_period_time >> 1 ); } return 0; @@ -382,25 +623,27 @@ static void ALSAFill( aout_instance_t * p_aout ) struct aout_sys_t * p_sys = p_aout->output.p_sys; aout_buffer_t * p_buffer; - mtime_t next_date = 0; snd_pcm_status_t * p_status; snd_timestamp_t ts_next; - int i_snd_rc, i_size; - byte_t * p_bytes; - snd_pcm_uframes_t i_avail; + int i_snd_rc; + mtime_t next_date; snd_pcm_status_alloca( &p_status ); + /* Wait for the device's readiness (ie. there is enough space in the + buffer to write at least one complete chunk) */ i_snd_rc = snd_pcm_wait( p_sys->p_snd_pcm, THREAD_SLEEP ); if( i_snd_rc < 0 ) { - msg_Err( p_aout, "alsa device not ready !!! (%s)", + msg_Err( p_aout, "ALSA device not ready !!! (%s)", snd_strerror( i_snd_rc ) ); return; } - while( VLC_TRUE ) + /* Fill in the buffer until space or audio output buffer shortage */ + for ( ; ; ) { + /* Get the status */ i_snd_rc = snd_pcm_status( p_sys->p_snd_pcm, p_status ); if( i_snd_rc < 0 ) { @@ -409,13 +652,17 @@ static void ALSAFill( aout_instance_t * p_aout ) return; } + /* Handle buffer underruns and reget the status */ if( snd_pcm_status_get_state( p_status ) == SND_PCM_STATE_XRUN ) { + /* Prepare the device */ i_snd_rc = snd_pcm_prepare( p_sys->p_snd_pcm ); + if( i_snd_rc == 0 ) { - msg_Err( p_aout, "recovered from buffer underrun" ); - next_date = mdate(); + msg_Warn( p_aout, "recovered from buffer underrun" ); + + /* Reget the status */ i_snd_rc = snd_pcm_status( p_sys->p_snd_pcm, p_status ); if( i_snd_rc < 0 ) { @@ -432,32 +679,35 @@ static void ALSAFill( aout_instance_t * p_aout ) } } - i_avail = snd_pcm_status_get_avail( p_status ); - if( i_avail >= p_aout->output.i_nb_samples ) - { - snd_pcm_status_get_tstamp( p_status, &ts_next ); - next_date = (mtime_t)ts_next.tv_sec * 1000000 + ts_next.tv_usec; + /* Here the device should be either in the RUNNING state either in + the PREPARE state. p_status is valid. */ - p_buffer = aout_OutputNextBuffer( p_aout, next_date, 0 ); + snd_pcm_status_get_tstamp( p_status, &ts_next ); + next_date = (mtime_t)ts_next.tv_sec * 1000000 + ts_next.tv_usec; + next_date += (mtime_t)snd_pcm_status_get_delay(p_status) + * 1000000 / p_aout->output.output.i_rate; - if( p_buffer == NULL ) - return; + p_buffer = aout_OutputNextBuffer( p_aout, next_date, + (p_aout->output.output.i_format == + VLC_FOURCC('s','p','d','i')) ); - p_bytes = p_buffer->p_buffer; + /* Audio output buffer shortage -> stop the fill process and + wait in ALSAThread */ + if( p_buffer == NULL ) + return; - i_snd_rc = snd_pcm_writei( p_sys->p_snd_pcm, p_bytes, - p_buffer->i_nb_samples ); + i_snd_rc = snd_pcm_writei( p_sys->p_snd_pcm, p_buffer->p_buffer, + p_buffer->i_nb_samples ); - if( i_snd_rc < 0 ) - { - msg_Err( p_aout, "write failed (%s)", - snd_strerror( i_snd_rc ) ); - } - else - { - aout_BufferFree( p_buffer ); - } + if( i_snd_rc < 0 ) + { + msg_Err( p_aout, "write failed (%s)", + snd_strerror( i_snd_rc ) ); } + + aout_BufferFree( p_buffer ); + + msleep( p_sys->i_period_time >> 2 ); } }