X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=modules%2Faudio_output%2Falsa.c;h=00f8f548436b1274c2a9c973641180c7b7b6bc59;hb=7eb551db1c68e537c6fab6d9c68956acccd1eddd;hp=060e79af7a27d0e7e6ea83fdf1b4e6e10182a2a2;hpb=88f10cc6cd4670a355b1cdc463e6d1303396a74c;p=vlc diff --git a/modules/audio_output/alsa.c b/modules/audio_output/alsa.c index 060e79af7a..00f8f54843 100644 --- a/modules/audio_output/alsa.c +++ b/modules/audio_output/alsa.c @@ -1,8 +1,8 @@ /***************************************************************************** * alsa.c : alsa plugin for vlc ***************************************************************************** - * Copyright (C) 2000-2001 the VideoLAN team - * $Id$ + * Copyright (C) 2000-2010 the VideoLAN team + * Copyright (C) 2009-2011 Rémi Denis-Courmont * * Authors: Henri Fallon - Original Author * Jeffrey Baker - Port to ALSA 1.0 API @@ -24,9 +24,6 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA. *****************************************************************************/ -/***************************************************************************** - * Preamble - *****************************************************************************/ #ifdef HAVE_CONFIG_H # include "config.h" #endif @@ -35,66 +32,26 @@ #include #include - -#include /* ENOMEM */ #include - #include #include -/* 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 #include -/*#define ALSA_DEBUG*/ - -/***************************************************************************** - * aout_sys_t: ALSA audio output method descriptor - ***************************************************************************** - * This structure is part of the audio output thread descriptor. - * It describes the ALSA specific properties of an audio device. - *****************************************************************************/ +/** Private data for an ALSA PCM playback stream */ struct aout_sys_t { - snd_pcm_t * p_snd_pcm; - unsigned int i_period_time; - -#ifdef ALSA_DEBUG - snd_output_t * p_snd_stderr; -#endif - - mtime_t start_date; - vlc_thread_t thread; - vlc_sem_t wait; + snd_pcm_t *pcm; }; #define A52_FRAME_NB 1536 -/* These values are in frames. - 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 int16_t). */ -#define ALSA_DEFAULT_PERIOD_SIZE 1024 -#define ALSA_DEFAULT_BUFFER_SIZE ( ALSA_DEFAULT_PERIOD_SIZE << 8 ) -#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 "plug:default" - /***************************************************************************** * Local prototypes *****************************************************************************/ -static int Open ( vlc_object_t * ); -static void Close ( vlc_object_t * ); -static void Play ( aout_instance_t * ); -static void* ALSAThread ( void * ); -static void ALSAFill ( aout_instance_t * ); +static int Open (vlc_object_t *); +static void Close (vlc_object_t *); static int FindDevicesCallback( vlc_object_t *p_this, char const *psz_name, vlc_value_t newval, vlc_value_t oldval, void *p_unused ); static void GetDevices( vlc_object_t *, module_config_t * ); @@ -102,16 +59,29 @@ static void GetDevices( vlc_object_t *, module_config_t * ); /***************************************************************************** * Module descriptor *****************************************************************************/ -static const char *const ppsz_devices[] = { "default" }; -static const char *const ppsz_devices_text[] = { N_("Default") }; +static const char *const ppsz_devices[] = { + "default", "plug:front", + "plug:side", "plug:rear", "plug:center_lfe", + "plug:surround40", "plug:surround41", + "plug:surround50", "plug:surround51", + "plug:surround71", + "hdmi", "iec958", +}; +static const char *const ppsz_devices_text[] = { + N_("Default"), N_("Front speakers"), + N_("Side speakers"), N_("Rear speakers"), N_("Center and subwoofer"), + N_("Surround 4.0"), N_("Surround 4.1"), + N_("Surround 5.0"), N_("Surround 5.1"), + N_("Surround 7.1"), + N_("HDMI"), N_("S/PDIF"), +}; + vlc_module_begin () set_shortname( "ALSA" ) set_description( N_("ALSA audio output") ) set_category( CAT_AUDIO ) set_subcategory( SUBCAT_AUDIO_AOUT ) - add_string( "alsa-audio-device", DEFAULT_ALSA_DEVICE, - N_("ALSA Device Name"), NULL, false ) - add_deprecated_alias( "alsadev" ) /* deprecated since 0.9.3 */ + add_string ("alsa-audio-device", "default", N_("ALSA device"), NULL, false) change_string_list( ppsz_devices, ppsz_devices_text, FindDevicesCallback ) change_action_add( FindDevicesCallback, N_("Refresh list") ) @@ -119,11 +89,57 @@ vlc_module_begin () set_callbacks( Open, Close ) vlc_module_end () -/* VLC will insert a resampling filter in any case, so it is best to turn off - * ALSA (plug) resampling. */ -static const int mode = SND_PCM_NO_AUTO_RESAMPLE -/* VLC is currently unable to leverage ALSA softvol. Disable it. */ - | SND_PCM_NO_SOFTVOL; + +/** Helper for ALSA -> VLC debugging output */ +static void Dump (vlc_object_t *obj, const char *msg, + int (*cb)(void *, snd_output_t *), void *p) +{ + snd_output_t *output; + char *str; + + if (unlikely(snd_output_buffer_open (&output))) + return; + + int val = cb (p, output); + if (val) + { + msg_Warn (obj, "cannot get info: %s", snd_strerror (val)); + return; + } + + size_t len = snd_output_buffer_string (output, &str); + if (len > 0 && str[len - 1]) + len--; /* strip trailing newline */ + msg_Dbg (obj, "%s%.*s", msg, (int)len, str); + snd_output_close (output); +} +#define Dump(o, m, cb, p) \ + Dump(VLC_OBJECT(o), m, (int (*)(void *, snd_output_t *))(cb), p) + +static void DumpDevice (vlc_object_t *obj, snd_pcm_t *pcm) +{ + snd_pcm_info_t *info; + + Dump (obj, " ", snd_pcm_dump, pcm); + snd_pcm_info_alloca (&info); + if (snd_pcm_info (pcm, info) == 0) + { + msg_Dbg (obj, " device name : %s", snd_pcm_info_get_name (info)); + msg_Dbg (obj, " device ID : %s", snd_pcm_info_get_id (info)); + msg_Dbg (obj, " subdevice name: %s", + snd_pcm_info_get_subdevice_name (info)); + } +} + +static void DumpDeviceStatus (vlc_object_t *obj, snd_pcm_t *pcm) +{ + snd_pcm_status_t *status; + + snd_pcm_status_alloca (&status); + snd_pcm_status (pcm, status); + Dump (obj, "current status:\n", snd_pcm_status_dump, status); +} +#define DumpDeviceStatus(o, p) DumpDeviceStatus(VLC_OBJECT(o), p) /** * Initializes list of devices. @@ -147,60 +163,31 @@ static void Probe (vlc_object_t *obj) var_TriggerCallback (obj, "intf-change"); } -/***************************************************************************** - * Open: create a handle and open an alsa device - ***************************************************************************** - * 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 void Play (audio_output_t *, block_t *); +static void Pause (audio_output_t *, bool, mtime_t); +static void PauseDummy (audio_output_t *, bool, mtime_t); +static void Flush (audio_output_t *, bool); + +/** Initializes an ALSA playback stream */ static int Open (vlc_object_t *obj) { - aout_instance_t * p_aout = (aout_instance_t *)obj; + audio_output_t *aout = (audio_output_t *)obj; /* Get device name */ - char *psz_device; + char *device; - if (var_Type (p_aout, "audio-device")) - psz_device = var_GetString (p_aout, "audio-device"); + if (var_Type (aout, "audio-device")) + device = var_GetString (aout, "audio-device"); else - psz_device = var_InheritString( p_aout, "alsa-audio-device" ); - if (unlikely(psz_device == NULL)) + device = var_InheritString (aout, "alsa-audio-device"); + if (unlikely(device == NULL)) return VLC_ENOMEM; - /* Choose the IEC device for S/PDIF output: - if the device is overridden 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)) - { - unsigned aes3; - - switch (p_aout->output.output.i_rate) - { - case 48000: - aes3 = IEC958_AES3_CON_FS_48000; - break; - case 44100: - aes3 = IEC958_AES3_CON_FS_44100; - break; - default: - aes3 = IEC958_AES3_CON_FS_32000; - break; - } - - free (psz_device); - if (asprintf (&psz_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, aes3) == -1) - return VLC_ENOMEM; - } - snd_pcm_format_t pcm_format; /* ALSA sample format */ - vlc_fourcc_t fourcc = p_aout->output.output.i_format; + vlc_fourcc_t fourcc = aout->format.i_format; + bool spdif = false; + switch (fourcc) { case VLC_CODEC_F64B: @@ -256,6 +243,8 @@ static int Open (vlc_object_t *obj) pcm_format = SND_PCM_FORMAT_U8; break; default: + if (AOUT_FMT_SPDIF(&aout->format)) + spdif = var_InheritBool (aout, "spdif"); if (HAVE_FPU) { fourcc = VLC_CODEC_FL32; @@ -268,435 +257,449 @@ static int Open (vlc_object_t *obj) } } + /* Choose the IEC device for S/PDIF output: + if the device is overridden by the user then it will be the one. + Otherwise we compute the default device based on the output format. */ + if (spdif && !strcmp (device, "default")) + { + unsigned aes3; + + switch (aout->format.i_rate) + { +#define FS(freq) \ + case freq: aes3 = IEC958_AES3_CON_FS_ ## freq; break; + FS( 44100) /* def. */ FS( 48000) FS( 32000) + FS( 22050) FS( 24000) + FS( 88200) FS(768000) FS( 96000) + FS(176400) FS(192000) +#undef FS + default: + aes3 = IEC958_AES3_CON_FS_NOTID; + break; + } + + free (device); + if (asprintf (&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, aes3) == -1) + return VLC_ENOMEM; + } + /* Allocate structures */ - aout_sys_t *p_sys = malloc (sizeof (*p_sys)); - if (unlikely(p_sys == NULL)) + aout_sys_t *sys = malloc (sizeof (*sys)); + if (unlikely(sys == NULL)) { - free (psz_device); + free (device); return VLC_ENOMEM; } - p_aout->output.p_sys = p_sys; - -#ifdef ALSA_DEBUG - snd_output_stdio_attach( &p_sys->p_snd_stderr, stderr, 0 ); -#endif + aout->sys = sys; /* Open the device */ - msg_Dbg( p_aout, "opening ALSA device `%s'", psz_device ); - int val = snd_pcm_open (&p_sys->p_snd_pcm, psz_device, - SND_PCM_STREAM_PLAYBACK, mode); + snd_pcm_t *pcm; + /* VLC always has a resampler. No need for ALSA's. */ + const int mode = SND_PCM_NO_AUTO_RESAMPLE + /* ALSA discards extra channels (by default). This is not good. */ + | SND_PCM_NO_AUTO_CHANNELS + /* VLC is currently unable to leverage ALSA softvol. No need for it. */ + | SND_PCM_NO_SOFTVOL; + + int val = snd_pcm_open (&pcm, device, SND_PCM_STREAM_PLAYBACK, mode); #if (SND_LIB_VERSION <= 0x010015) # warning Please update alsa-lib to version > 1.0.21a. - var_Create (p_aout->p_libvlc, "alsa-working", VLC_VAR_BOOL); - if (val != 0 && var_GetBool (p_aout->p_libvlc, "alsa-working")) - dialog_Fatal (p_aout, "ALSA version problem", + var_Create (aout->p_libvlc, "alsa-working", VLC_VAR_BOOL); + if (val != 0 && var_GetBool (aout->p_libvlc, "alsa-working")) + dialog_Fatal (aout, "ALSA version problem", "VLC failed to re-initialize your audio output device.\n" "Please update alsa-lib to version 1.0.22 or higher " "to fix this issue."); - var_SetBool (p_aout->p_libvlc, "alsa-working", !val); + var_SetBool (aout->p_libvlc, "alsa-working", !val); #endif if (val != 0) { #if (SND_LIB_VERSION <= 0x010017) # warning Please update alsa-lib to version > 1.0.23. - var_Create (p_aout->p_libvlc, "alsa-broken", VLC_VAR_BOOL); - if (!var_GetBool (p_aout->p_libvlc, "alsa-broken")) + var_Create (aout->p_libvlc, "alsa-broken", VLC_VAR_BOOL); + if (!var_GetBool (aout->p_libvlc, "alsa-broken")) { - var_SetBool (p_aout->p_libvlc, "alsa-broken", true); - dialog_Fatal (p_aout, "Potential ALSA version problem", + var_SetBool (aout->p_libvlc, "alsa-broken", true); + dialog_Fatal (aout, "Potential ALSA version problem", "VLC failed to initialize your audio output device (if any).\n" "Please update alsa-lib to version 1.0.24 or higher " "to try to fix this issue."); } #endif - msg_Err (p_aout, "cannot open ALSA device `%s' (%s)", - psz_device, snd_strerror (val)); - dialog_Fatal (p_aout, _("Audio output failed"), + msg_Err (aout, "cannot open ALSA device \"%s\": %s", device, + snd_strerror (val)); + dialog_Fatal (aout, _("Audio output failed"), _("The audio device \"%s\" could not be used:\n%s."), - psz_device, snd_strerror (val)); - free (psz_device); - free (p_sys); + device, snd_strerror (val)); + free (device); + free (sys); return VLC_EGENERIC; } - free( psz_device ); + sys->pcm = pcm; + + /* Print some potentially useful debug */ + msg_Dbg (aout, "using ALSA device: %s", device); + free (device); + DumpDevice (VLC_OBJECT(aout), pcm); - snd_pcm_uframes_t i_buffer_size; - snd_pcm_uframes_t i_period_size; - int i_channels; + /* Setup */ + unsigned channels = aout_FormatNbChannels (&aout->format); - if (var_InheritBool (obj, "spdif")) + if (spdif) { fourcc = VLC_CODEC_SPDIFL; - i_buffer_size = ALSA_SPDIF_BUFFER_SIZE; pcm_format = SND_PCM_FORMAT_S16; - i_channels = 2; - - 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; - - aout_VolumeNoneInit( p_aout ); + channels = 2; } - else - { - i_buffer_size = ALSA_DEFAULT_BUFFER_SIZE; - i_channels = aout_FormatNbChannels( &p_aout->output.output ); - - p_aout->output.i_nb_samples = i_period_size = ALSA_DEFAULT_PERIOD_SIZE; - - aout_VolumeSoftInit( p_aout ); - } - - p_aout->output.pf_play = Play; - - snd_pcm_hw_params_t *p_hw; - snd_pcm_sw_params_t *p_sw; - - snd_pcm_hw_params_alloca(&p_hw); - snd_pcm_sw_params_alloca(&p_sw); /* Get Initial hardware parameters */ - val = snd_pcm_hw_params_any( p_sys->p_snd_pcm, p_hw ); - if( val < 0 ) + snd_pcm_hw_params_t *hw; + unsigned param; + + snd_pcm_hw_params_alloca (&hw); + snd_pcm_hw_params_any (pcm, hw); + Dump (aout, "initial hardware setup:\n", snd_pcm_hw_params_dump, hw); + + /* Set sample format */ + val = snd_pcm_hw_params_set_format (pcm, hw, pcm_format); + if (val == 0) + ; + else if (pcm_format != SND_PCM_FORMAT_FLOAT + && snd_pcm_hw_params_set_format (pcm, hw, SND_PCM_FORMAT_FLOAT) == 0) + fourcc = VLC_CODEC_FL32; + else if (pcm_format != SND_PCM_FORMAT_S32 + && snd_pcm_hw_params_set_format (pcm, hw, SND_PCM_FORMAT_S32) == 0) + fourcc = VLC_CODEC_S32N; + else if (pcm_format != SND_PCM_FORMAT_S16 + && snd_pcm_hw_params_set_format (pcm, hw, SND_PCM_FORMAT_S16) == 0) + fourcc = VLC_CODEC_S16N; + else { - msg_Err( p_aout, "unable to retrieve hardware parameters (%s)", - snd_strerror( val ) ); + msg_Err (aout, "cannot set sample format: %s", snd_strerror (val)); goto error; } - /* Set format. */ - val = snd_pcm_hw_params_set_format (p_sys->p_snd_pcm, p_hw, pcm_format); - if( val < 0 ) + val = snd_pcm_hw_params_set_access (pcm, hw, + SND_PCM_ACCESS_RW_INTERLEAVED); + if (val) { - msg_Err (p_aout, "cannot set sample format: %s", snd_strerror (val)); + msg_Err (aout, "cannot set access mode: %s", snd_strerror (val)); goto error; } - p_aout->output.output.i_format = fourcc; - - val = snd_pcm_hw_params_set_access( p_sys->p_snd_pcm, p_hw, - SND_PCM_ACCESS_RW_INTERLEAVED ); - if( val < 0 ) + /* Set channels count */ + unsigned chans; + /* By default, ALSA plug will discard extra channels and zero missing ones + * instead of remixing, so remixing was disabled at snd_pcm_open() above. + * Then, the configuration space will contain only channels configurations + * supported by the audio device, but not necessarily functional (e.g. + * surround-capable card with stereo speakers). */ + /* If there is only channels configuration, use that one. + * This should deal with "surround40", "surround51" and "surround71". */ + if (snd_pcm_hw_params_get_channels (hw, &chans) == 0) + ; + /* Otherwise, if we have 5 channels and they are supported, use that. + * This should deal with "surround41" and "surround50" routers. + * This assumes that no real hardware supports exactly 5 channels. */ + else if (channels == 5 + && snd_pcm_hw_params_set_channels (pcm, hw, channels)) + chans = channels; + /* Otherwise, if stereo is supported, then use that. This deals with + * the "front" device that fails to enforce stereo on Surround cards. */ + else if (snd_pcm_hw_params_set_channels (pcm, hw, 2) == 0) + chans = 2; + /* Out of desperation, try the original channels count. */ + else if (snd_pcm_hw_params_set_channels (pcm, hw, channels) == 0) + ; + /* As last chance, try anything. */ + else { - msg_Err( p_aout, "unable to set interleaved stream format (%s)", - snd_strerror( val ) ); - goto error; + chans = channels; + val = snd_pcm_hw_params_set_channels_near (pcm, hw, &chans); + if (val) + { + msg_Err (aout, "cannot set channels count: %s", + snd_strerror (val)); + goto error; + } } - /* Set channels. */ - val = snd_pcm_hw_params_set_channels( p_sys->p_snd_pcm, p_hw, i_channels ); - if( val < 0 ) + if (channels != chans) + msg_Dbg (aout, "remixing from %u to %u channels", channels, chans); + + /* Set sample rate */ + unsigned rate = aout->format.i_rate; + val = snd_pcm_hw_params_set_rate_near (pcm, hw, &rate, NULL); + if (val) { - msg_Err( p_aout, "unable to set number of output channels (%s)", - snd_strerror( val ) ); + msg_Err (aout, "cannot set sample rate: %s", snd_strerror (val)); goto error; } - - /* Set rate. */ - unsigned old_rate = p_aout->output.output.i_rate; - val = snd_pcm_hw_params_set_rate_near (p_sys->p_snd_pcm, p_hw, - &p_aout->output.output.i_rate, - NULL); + if (aout->format.i_rate != rate) + msg_Dbg (aout, "resampling from %d Hz to %d Hz", + aout->format.i_rate, rate); + + /* Set buffer size */ + param = AOUT_MAX_ADVANCE_TIME; + val = snd_pcm_hw_params_set_buffer_time_near (pcm, hw, ¶m, NULL); + if (val) + msg_Warn (aout, "cannot set buffer duration near %u us: %s", + param, snd_strerror (val)); + val = snd_pcm_hw_params_set_buffer_time_last (pcm, hw, ¶m, NULL); + if (val) + msg_Warn (aout, "cannot set buffer duration: %s", snd_strerror (val)); + + /* Set number of periods (at least two) */ + param = 2; + val = snd_pcm_hw_params_set_periods_min (pcm, hw, ¶m, NULL); + if (val) + msg_Warn (aout, "cannot set minimum of %u periods: %s", param, + snd_strerror (val)); + val = snd_pcm_hw_params_set_periods_first (pcm, hw, ¶m, NULL); + if (val) + msg_Warn (aout, "cannot set periods count near %u: %s", param, + snd_strerror (val)); + + /* Commit hardware parameters */ + val = snd_pcm_hw_params (pcm, hw); if (val < 0) { - msg_Err (p_aout, "unable to set sampling rate (%s)", + msg_Err (aout, "cannot commit hardware parameters: %s", snd_strerror (val)); goto error; } - if (p_aout->output.output.i_rate != old_rate) - msg_Warn (p_aout, "resampling from %d Hz to %d Hz\n", old_rate, - p_aout->output.output.i_rate); + Dump (aout, "final HW setup:\n", snd_pcm_hw_params_dump, hw); - /* Set period size. */ - val = snd_pcm_hw_params_set_period_size_near( p_sys->p_snd_pcm, p_hw, - &i_period_size, NULL ); - if( val < 0 ) - { - msg_Err( p_aout, "unable to set period size (%s)", - snd_strerror( val ) ); - goto error; - } - p_aout->output.i_nb_samples = i_period_size; + /* Get Initial software parameters */ + snd_pcm_sw_params_t *sw; - /* Set buffer size. */ - val = snd_pcm_hw_params_set_buffer_size_near( p_sys->p_snd_pcm, p_hw, - &i_buffer_size ); - if( val ) - { - msg_Err( p_aout, "unable to set buffer size (%s)", - snd_strerror( val ) ); - goto error; - } + snd_pcm_sw_params_alloca (&sw); + snd_pcm_sw_params_current (pcm, sw); + Dump (aout, "initial software parameters:\n", snd_pcm_sw_params_dump, sw); - /* Commit hardware parameters. */ - val = snd_pcm_hw_params( p_sys->p_snd_pcm, p_hw ); + /* START REVISIT */ + //snd_pcm_sw_params_set_avail_min( pcm, sw, i_period_size ); + // FIXME: useful? + val = snd_pcm_sw_params_set_start_threshold (pcm, sw, 1); if( val < 0 ) { - msg_Err( p_aout, "unable to commit hardware configuration (%s)", + msg_Err( aout, "unable to set start threshold (%s)", snd_strerror( val ) ); goto error; } + /* END REVISIT */ - val = snd_pcm_hw_params_get_period_time( p_hw, &p_sys->i_period_time, - NULL ); - if( val < 0 ) + /* Commit software parameters. */ + val = snd_pcm_sw_params (pcm, sw); + if (val) { - msg_Err( p_aout, "unable to get period time (%s)", - snd_strerror( val ) ); + msg_Err (aout, "cannot commit software parameters: %s", + snd_strerror (val)); goto error; } + Dump (aout, "final software parameters:\n", snd_pcm_sw_params_dump, sw); - /* Get Initial software parameters */ - snd_pcm_sw_params_current( p_sys->p_snd_pcm, p_sw ); - - snd_pcm_sw_params_set_avail_min( p_sys->p_snd_pcm, p_sw, - p_aout->output.i_nb_samples ); - /* start playing when one period has been written */ - val = snd_pcm_sw_params_set_start_threshold( p_sys->p_snd_pcm, p_sw, - ALSA_DEFAULT_PERIOD_SIZE); - if( val < 0 ) + val = snd_pcm_prepare (pcm); + if (val) { - msg_Err( p_aout, "unable to set start threshold (%s)", - snd_strerror( val ) ); + msg_Err (aout, "cannot prepare device: %s", snd_strerror (val)); goto error; } - /* Commit software parameters. */ - if ( snd_pcm_sw_params( p_sys->p_snd_pcm, p_sw ) < 0 ) + /* Guess the channel map */ + switch (chans) { - msg_Err( p_aout, "unable to set software configuration" ); - goto error; - } - -#ifdef ALSA_DEBUG - snd_output_printf( p_sys->p_snd_stderr, "\nALSA hardware setup:\n\n" ); - snd_pcm_dump_hw_setup( p_sys->p_snd_pcm, p_sys->p_snd_stderr ); - snd_output_printf( p_sys->p_snd_stderr, "\nALSA software setup:\n\n" ); - snd_pcm_dump_sw_setup( p_sys->p_snd_pcm, p_sys->p_snd_stderr ); - snd_output_printf( p_sys->p_snd_stderr, "\n" ); + case 1: + chans = AOUT_CHAN_CENTER; + break; + case 2: /* front */ + chans = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT; + break; + case 4: /* surround40 */ + chans = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT + | AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT; + break; + case 5: /* surround50 ... or surround41! Uho! */ +#if 1 + chans = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT + | AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT + | AOUT_CHAN_LFE; +#else + chans = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT + | AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT + | AOUT_CHAN_CENTER; +#endif + break; + case 6: /* surround51 */ + chans = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT + | AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT + | AOUT_CHAN_CENTER | AOUT_CHAN_LFE; + break; +#if 0 /* FIXME reorder */ + case 8: /* surround71 */ + chans = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT + | AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT + | AOUT_CHAN_CENTER | AOUT_CHAN_LFE + | AOUT_CHAN_MIDDLELEFT | AOUT_CHAN_MIDDLERIGHT; + break; #endif + default: + msg_Err (aout, "unknown %u channels configuration", chans); + goto error; + } - p_sys->start_date = 0; - vlc_sem_init( &p_sys->wait, 0 ); + /* Setup audio_output_t */ + aout->format.i_format = fourcc; + aout->format.i_rate = rate; + aout->format.i_original_channels = + aout->format.i_physical_channels = chans; + if (spdif) + { + aout->format.i_bytes_per_frame = AOUT_SPDIF_SIZE; + aout->format.i_frame_length = A52_FRAME_NB; + aout_VolumeNoneInit (aout); + } + else + aout_VolumeSoftInit (aout); - /* Create ALSA thread and wait for its readiness. */ - if( vlc_clone( &p_sys->thread, ALSAThread, p_aout, - VLC_THREAD_PRIORITY_OUTPUT ) ) + aout->pf_play = Play; + if (snd_pcm_hw_params_can_pause (hw)) + aout->pf_pause = Pause; + else { - msg_Err( p_aout, "cannot create ALSA thread (%m)" ); - vlc_sem_destroy( &p_sys->wait ); - goto error; + aout->pf_pause = PauseDummy; + msg_Warn (aout, "device cannot be paused"); } + aout->pf_flush = Flush; Probe (obj); return 0; error: - snd_pcm_close( p_sys->p_snd_pcm ); -#ifdef ALSA_DEBUG - snd_output_close( p_sys->p_snd_stderr ); -#endif - free( p_sys ); + snd_pcm_close (pcm); + free (sys); return VLC_EGENERIC; } -static void PlayIgnore( aout_instance_t *p_aout ) -{ /* Already playing - nothing to do */ - (void) p_aout; -} - -/***************************************************************************** - * Play: start playback - *****************************************************************************/ -static void Play( aout_instance_t *p_aout ) -{ - p_aout->output.pf_play = PlayIgnore; - - /* get the playing date of the first aout buffer */ - p_aout->output.p_sys->start_date = - aout_FifoFirstDate( p_aout, &p_aout->output.fifo ); - - /* wake up the audio output thread */ - sem_post( &p_aout->output.p_sys->wait ); -} - -/***************************************************************************** - * Close: close the ALSA device - *****************************************************************************/ -static void Close (vlc_object_t *obj) -{ - aout_instance_t *p_aout = (aout_instance_t *)obj; - struct aout_sys_t * p_sys = p_aout->output.p_sys; - - /* Make sure that the thread will stop once it is waken up */ - vlc_cancel( p_sys->thread ); - vlc_join( p_sys->thread, NULL ); - vlc_sem_destroy( &p_sys->wait ); - - snd_pcm_drop( p_sys->p_snd_pcm ); - snd_pcm_close( p_sys->p_snd_pcm ); -#ifdef ALSA_DEBUG - snd_output_close( p_sys->p_snd_stderr ); -#endif - free( p_sys ); -} - -/***************************************************************************** - * ALSAThread: asynchronous thread used to DMA the data to the device - *****************************************************************************/ -static void* ALSAThread( void *data ) -{ - aout_instance_t * p_aout = data; - struct aout_sys_t * p_sys = p_aout->output.p_sys; - - /* Wait for the exact time to start playing (avoids resampling) */ - vlc_sem_wait( &p_sys->wait ); - mwait( p_sys->start_date - AOUT_PTS_TOLERANCE / 4 ); - - for(;;) - ALSAFill( p_aout ); - - assert(0); -} - -/***************************************************************************** - * ALSAFill: function used to fill the ALSA buffer as much as possible - *****************************************************************************/ -static void ALSAFill( aout_instance_t * p_aout ) +/** + * Queues one audio buffer to the hardware. + */ +static void Play (audio_output_t *aout, block_t *block) { - struct aout_sys_t * p_sys = p_aout->output.p_sys; - snd_pcm_t *p_pcm = p_sys->p_snd_pcm; - snd_pcm_status_t * p_status; - int i_snd_rc; - mtime_t next_date; - - int canc = vlc_savecancel(); - /* Fill in the buffer until space or audio output buffer shortage */ - - /* Get the status */ - snd_pcm_status_alloca(&p_status); - i_snd_rc = snd_pcm_status( p_pcm, p_status ); - if( i_snd_rc < 0 ) - { - msg_Err( p_aout, "cannot get device status" ); - goto error; - } + aout_sys_t *sys = aout->sys; + snd_pcm_t *pcm = sys->pcm; + snd_pcm_sframes_t frames; + snd_pcm_state_t state = snd_pcm_state (pcm); - /* Handle buffer underruns and get the status again */ - if( snd_pcm_status_get_state( p_status ) == SND_PCM_STATE_XRUN ) + if (snd_pcm_delay (pcm, &frames) == 0) { - /* Prepare the device */ - i_snd_rc = snd_pcm_prepare( p_pcm ); - if( i_snd_rc ) - { - msg_Err( p_aout, "cannot recover from buffer underrun" ); - goto error; - } - - msg_Dbg( p_aout, "recovered from buffer underrun" ); + mtime_t delay = frames * CLOCK_FREQ / aout->format.i_rate; - /* Get the new status */ - i_snd_rc = snd_pcm_status( p_pcm, p_status ); - if( i_snd_rc < 0 ) + if (state != SND_PCM_STATE_RUNNING) { - msg_Err( p_aout, "cannot get device status after recovery" ); - goto error; + delay = block->i_pts - (mdate () + delay); + if (delay > 0) + { + frames = (delay * aout->format.i_rate) / CLOCK_FREQ; + msg_Dbg (aout, "prepending %ld zeroes", frames); + + void *pad = calloc (frames, aout->format.i_bytes_per_frame); + if (likely(pad != NULL)) + { + snd_pcm_writei (pcm, pad, frames); + free (pad); + } + } } - - /* Underrun, try to recover as quickly as possible */ - next_date = mdate(); - } - else - { - /* Here the device should be in RUNNING state, p_status is valid. */ - snd_pcm_sframes_t delay = snd_pcm_status_get_delay( p_status ); - if( delay == 0 ) /* workaround buggy alsa drivers */ - if( snd_pcm_delay( p_pcm, &delay ) < 0 ) - delay = 0; /* FIXME: use a positive minimal delay */ - - size_t i_bytes = snd_pcm_frames_to_bytes( p_pcm, delay ); - mtime_t delay_us = CLOCK_FREQ * i_bytes - / p_aout->output.output.i_bytes_per_frame - / p_aout->output.output.i_rate - * p_aout->output.output.i_frame_length; - -#ifdef ALSA_DEBUG - snd_pcm_state_t state = snd_pcm_status_get_state( p_status ); - if( state != SND_PCM_STATE_RUNNING ) - msg_Err( p_aout, "pcm status (%d) != RUNNING", state ); - - msg_Dbg( p_aout, "Delay is %ld frames (%zu bytes)", delay, i_bytes ); - - msg_Dbg( p_aout, "Bytes per frame: %d", p_aout->output.output.i_bytes_per_frame ); - msg_Dbg( p_aout, "Rate: %d", p_aout->output.output.i_rate ); - msg_Dbg( p_aout, "Frame length: %d", p_aout->output.output.i_frame_length ); - msg_Dbg( p_aout, "Next date: in %"PRId64" microseconds", delay_us ); -#endif - next_date = mdate() + delay_us; + else + aout_TimeReport (aout, block->i_pts - delay); } - block_t *p_buffer = aout_OutputNextBuffer( p_aout, next_date, - (p_aout->output.output.i_format == VLC_CODEC_SPDIFL) ); - - /* Audio output buffer shortage -> stop the fill process and wait */ - if( p_buffer == NULL ) - goto error; + /* TODO: better overflow handling */ + /* TODO: no period wake ups */ - block_cleanup_push( p_buffer ); - for (;;) + while (block->i_nb_samples > 0) { - int n = snd_pcm_poll_descriptors_count(p_pcm); - struct pollfd ufd[n]; - unsigned short revents; - - snd_pcm_poll_descriptors(p_pcm, ufd, n); - do + frames = snd_pcm_writei (pcm, block->p_buffer, block->i_nb_samples); + if (frames >= 0) { - vlc_restorecancel(canc); - poll(ufd, n, -1); - canc = vlc_savecancel(); - snd_pcm_poll_descriptors_revents(p_pcm, ufd, n, &revents); + size_t bytes = snd_pcm_frames_to_bytes (pcm, frames); + block->i_nb_samples -= frames; + block->p_buffer += bytes; + block->i_buffer -= bytes; + // pts, length } - while(!revents); - - if(revents & POLLOUT) + else { - i_snd_rc = snd_pcm_writei( p_pcm, p_buffer->p_buffer, - p_buffer->i_nb_samples ); - if( i_snd_rc != -ESTRPIPE ) + int val = snd_pcm_recover (pcm, frames, 1); + if (val) + { + msg_Err (aout, "cannot recover playback stream: %s", + snd_strerror (val)); + DumpDeviceStatus (aout, pcm); break; + } + msg_Warn (aout, "cannot write samples: %s", snd_strerror (frames)); } + } + block_Release (block); +} - /* a suspend event occurred - * (stream is suspended and waiting for an application recovery) */ - msg_Dbg( p_aout, "entering in suspend mode, trying to resume..." ); +/** + * Pauses/resumes the audio playback. + */ +static void Pause (audio_output_t *aout, bool pause, mtime_t date) +{ + snd_pcm_t *pcm = aout->sys->pcm; - while( ( i_snd_rc = snd_pcm_resume( p_pcm ) ) == -EAGAIN ) - { - vlc_restorecancel(canc); - msleep(CLOCK_FREQ); /* device still suspended, wait... */ - canc = vlc_savecancel(); - } + int val = snd_pcm_pause (pcm, pause); + if (unlikely(val)) + PauseDummy (aout, pause, date); +} - if( i_snd_rc < 0 ) - /* Device does not support resuming, restart it */ - i_snd_rc = snd_pcm_prepare( p_pcm ); +static void PauseDummy (audio_output_t *aout, bool pause, mtime_t date) +{ + snd_pcm_t *pcm = aout->sys->pcm; - } + /* Stupid device cannot pause. Discard samples. */ + if (pause) + snd_pcm_drop (pcm); + else + snd_pcm_prepare (pcm); + (void) date; +} - if( i_snd_rc < 0 ) - msg_Err( p_aout, "cannot write: %s", snd_strerror( i_snd_rc ) ); +/** + * Flushes/drains the audio playback buffer. + */ +static void Flush (audio_output_t *aout, bool wait) +{ + snd_pcm_t *pcm = aout->sys->pcm; - vlc_restorecancel(canc); - vlc_cleanup_run(); - return; + if (wait) + snd_pcm_drain (pcm); + else + snd_pcm_drop (pcm); + snd_pcm_prepare (pcm); +} -error: - if( i_snd_rc < 0 ) - msg_Err( p_aout, "ALSA error: %s", snd_strerror( i_snd_rc ) ); - vlc_restorecancel(canc); - msleep(p_sys->i_period_time / 2); +/** + * Releases the audio output. + */ +static void Close (vlc_object_t *obj) +{ + audio_output_t *aout = (audio_output_t *)obj; + aout_sys_t *sys = aout->sys; + snd_pcm_t *pcm = aout->sys->pcm; + + snd_pcm_drop (pcm); + snd_pcm_close (pcm); + free (sys); } /***************************************************************************** @@ -766,7 +769,7 @@ static void GetDevices (vlc_object_t *obj, module_config_t *item) if (desc != NULL) for (char *lf = strchr(desc, '\n'); lf; lf = strchr(lf, '\n')) *lf = ' '; - msg_Dbg(obj, " %s (%s)", (desc != NULL) ? desc : name, name); + msg_Dbg(obj, "%s (%s)", (desc != NULL) ? desc : name, name); if (item != NULL) {