/* GStreamer * Copyright (C) 2005 Wim Taymans * * gstalsasrc.c: * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library 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 GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ /** * SECTION:element-alsasrc * @see_also: alsasink, alsamixer * * This element reads data from an audio card using the ALSA API. * * * Example pipelines * |[ * gst-launch -v alsasrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=alsasrc.ogg * ]| Record from a sound card using ALSA and encode to Ogg/Vorbis. * * * Last reviewed on 2006-03-01 (0.10.4) */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include #include #include #include #include #include "gstalsasrc.h" #include "gstalsadeviceprobe.h" #include "gst/glib-compat-private.h" #include #define DEFAULT_PROP_DEVICE "default" #define DEFAULT_PROP_DEVICE_NAME "" #define DEFAULT_PROP_CARD_NAME "" enum { PROP_0, PROP_DEVICE, PROP_DEVICE_NAME, PROP_CARD_NAME, PROP_LAST }; static void gst_alsasrc_init_interfaces (GType type); GST_BOILERPLATE_FULL (GstAlsaSrc, gst_alsasrc, GstAudioSrc, GST_TYPE_AUDIO_SRC, gst_alsasrc_init_interfaces); GST_IMPLEMENT_ALSA_MIXER_METHODS (GstAlsaSrc, gst_alsasrc_mixer); static void gst_alsasrc_finalize (GObject * object); static void gst_alsasrc_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_alsasrc_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static GstCaps *gst_alsasrc_getcaps (GstBaseSrc * bsrc); static gboolean gst_alsasrc_open (GstAudioSrc * asrc); static gboolean gst_alsasrc_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec); static gboolean gst_alsasrc_unprepare (GstAudioSrc * asrc); static gboolean gst_alsasrc_close (GstAudioSrc * asrc); static guint gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length); static guint gst_alsasrc_delay (GstAudioSrc * asrc); static void gst_alsasrc_reset (GstAudioSrc * asrc); /* AlsaSrc signals and args */ enum { LAST_SIGNAL }; #if (G_BYTE_ORDER == G_LITTLE_ENDIAN) # define ALSA_SRC_FACTORY_ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN" #else # define ALSA_SRC_FACTORY_ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN" #endif static GstStaticPadTemplate alsasrc_src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-int, " "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, " "signed = (boolean) { TRUE, FALSE }, " "width = (int) 32, " "depth = (int) 32, " "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; " "audio/x-raw-int, " "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, " "signed = (boolean) { TRUE, FALSE }, " "width = (int) 32, " "depth = (int) 24, " "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; " "audio/x-raw-int, " "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, " "signed = (boolean) { TRUE, FALSE }, " "width = (int) 24, " "depth = (int) 24, " "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; " "audio/x-raw-int, " "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, " "signed = (boolean) { TRUE, FALSE }, " "width = (int) 16, " "depth = (int) 16, " "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; " "audio/x-raw-int, " "signed = (boolean) { TRUE, FALSE }, " "width = (int) 8, " "depth = (int) 8, " "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]") ); static void gst_alsasrc_finalize (GObject * object) { GstAlsaSrc *src = GST_ALSA_SRC (object); g_free (src->device); g_mutex_free (src->alsa_lock); G_OBJECT_CLASS (parent_class)->finalize (object); } static gboolean gst_alsasrc_interface_supported (GstAlsaSrc * this, GType interface_type) { /* only support this one interface (wrapped by GstImplementsInterface) */ g_assert (interface_type == GST_TYPE_MIXER); return gst_alsasrc_mixer_supported (this, interface_type); } static void gst_implements_interface_init (GstImplementsInterfaceClass * klass) { klass->supported = (gpointer) gst_alsasrc_interface_supported; } static void gst_alsasrc_init_interfaces (GType type) { static const GInterfaceInfo implements_iface_info = { (GInterfaceInitFunc) gst_implements_interface_init, NULL, NULL, }; static const GInterfaceInfo mixer_iface_info = { (GInterfaceInitFunc) gst_alsasrc_mixer_interface_init, NULL, NULL, }; g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE, &implements_iface_info); g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_iface_info); gst_alsa_type_add_device_property_probe_interface (type); } static void gst_alsasrc_base_init (gpointer g_class) { GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); gst_element_class_set_details_simple (element_class, "Audio source (ALSA)", "Source/Audio", "Read from a sound card via ALSA", "Wim Taymans "); gst_element_class_add_static_pad_template (element_class, &alsasrc_src_factory); } static void gst_alsasrc_class_init (GstAlsaSrcClass * klass) { GObjectClass *gobject_class; GstBaseSrcClass *gstbasesrc_class; GstAudioSrcClass *gstaudiosrc_class; gobject_class = (GObjectClass *) klass; gstbasesrc_class = (GstBaseSrcClass *) klass; gstaudiosrc_class = (GstAudioSrcClass *) klass; gobject_class->finalize = gst_alsasrc_finalize; gobject_class->get_property = gst_alsasrc_get_property; gobject_class->set_property = gst_alsasrc_set_property; gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasrc_getcaps); gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_alsasrc_open); gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasrc_prepare); gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasrc_unprepare); gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_alsasrc_close); gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_alsasrc_read); gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_alsasrc_delay); gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_alsasrc_reset); g_object_class_install_property (gobject_class, PROP_DEVICE, g_param_spec_string ("device", "Device", "ALSA device, as defined in an asound configuration file", DEFAULT_PROP_DEVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_DEVICE_NAME, g_param_spec_string ("device-name", "Device name", "Human-readable name of the sound device", DEFAULT_PROP_DEVICE_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_CARD_NAME, g_param_spec_string ("card-name", "Card name", "Human-readable name of the sound card", DEFAULT_PROP_CARD_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); } static void gst_alsasrc_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAlsaSrc *src; src = GST_ALSA_SRC (object); switch (prop_id) { case PROP_DEVICE: g_free (src->device); src->device = g_value_dup_string (value); if (src->device == NULL) { src->device = g_strdup (DEFAULT_PROP_DEVICE); } break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_alsasrc_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAlsaSrc *src; src = GST_ALSA_SRC (object); switch (prop_id) { case PROP_DEVICE: g_value_set_string (value, src->device); break; case PROP_DEVICE_NAME: g_value_take_string (value, gst_alsa_find_device_name (GST_OBJECT_CAST (src), src->device, src->handle, SND_PCM_STREAM_CAPTURE)); break; case PROP_CARD_NAME: g_value_take_string (value, gst_alsa_find_card_name (GST_OBJECT_CAST (src), src->device, SND_PCM_STREAM_CAPTURE)); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_alsasrc_init (GstAlsaSrc * alsasrc, GstAlsaSrcClass * g_class) { GST_DEBUG_OBJECT (alsasrc, "initializing"); alsasrc->device = g_strdup (DEFAULT_PROP_DEVICE); alsasrc->cached_caps = NULL; alsasrc->alsa_lock = g_mutex_new (); } #define CHECK(call, error) \ G_STMT_START { \ if ((err = call) < 0) \ goto error; \ } G_STMT_END; static GstCaps * gst_alsasrc_getcaps (GstBaseSrc * bsrc) { GstElementClass *element_class; GstPadTemplate *pad_template; GstAlsaSrc *src; GstCaps *caps; src = GST_ALSA_SRC (bsrc); if (src->handle == NULL) { GST_DEBUG_OBJECT (src, "device not open, using template caps"); return NULL; /* base class will get template caps for us */ } if (src->cached_caps) { GST_LOG_OBJECT (src, "Returning cached caps"); return gst_caps_ref (src->cached_caps); } element_class = GST_ELEMENT_GET_CLASS (src); pad_template = gst_element_class_get_pad_template (element_class, "src"); g_return_val_if_fail (pad_template != NULL, NULL); caps = gst_alsa_probe_supported_formats (GST_OBJECT (src), src->handle, gst_pad_template_get_caps (pad_template)); if (caps) { src->cached_caps = gst_caps_ref (caps); } GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, caps); return caps; } static int set_hwparams (GstAlsaSrc * alsa) { guint rrate; gint err; snd_pcm_hw_params_t *params; snd_pcm_hw_params_malloc (¶ms); /* choose all parameters */ CHECK (snd_pcm_hw_params_any (alsa->handle, params), no_config); /* set the interleaved read/write format */ CHECK (snd_pcm_hw_params_set_access (alsa->handle, params, alsa->access), wrong_access); /* set the sample format */ CHECK (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format), no_sample_format); /* set the count of channels */ CHECK (snd_pcm_hw_params_set_channels (alsa->handle, params, alsa->channels), no_channels); /* set the stream rate */ rrate = alsa->rate; CHECK (snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, NULL), no_rate); if (rrate != alsa->rate) goto rate_match; if (alsa->buffer_time != -1) { /* set the buffer time */ CHECK (snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params, &alsa->buffer_time, NULL), buffer_time); } if (alsa->period_time != -1) { /* set the period time */ CHECK (snd_pcm_hw_params_set_period_time_near (alsa->handle, params, &alsa->period_time, NULL), period_time); } /* write the parameters to device */ CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params); CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size), buffer_size); CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size, NULL), period_size); snd_pcm_hw_params_free (params); return 0; /* ERRORS */ no_config: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Broken configuration for recording: no configurations available: %s", snd_strerror (err))); snd_pcm_hw_params_free (params); return err; } wrong_access: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Access type not available for recording: %s", snd_strerror (err))); snd_pcm_hw_params_free (params); return err; } no_sample_format: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Sample format not available for recording: %s", snd_strerror (err))); snd_pcm_hw_params_free (params); return err; } no_channels: { gchar *msg = NULL; if ((alsa->channels) == 1) msg = g_strdup (_("Could not open device for recording in mono mode.")); if ((alsa->channels) == 2) msg = g_strdup (_("Could not open device for recording in stereo mode.")); if ((alsa->channels) > 2) msg = g_strdup_printf (_ ("Could not open device for recording in %d-channel mode"), alsa->channels); GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, ("%s", msg), ("%s", snd_strerror (err))); g_free (msg); snd_pcm_hw_params_free (params); return err; } no_rate: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Rate %iHz not available for recording: %s", alsa->rate, snd_strerror (err))); snd_pcm_hw_params_free (params); return err; } rate_match: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Rate doesn't match (requested %iHz, get %iHz)", alsa->rate, err)); snd_pcm_hw_params_free (params); return -EINVAL; } buffer_time: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Unable to set buffer time %i for recording: %s", alsa->buffer_time, snd_strerror (err))); snd_pcm_hw_params_free (params); return err; } buffer_size: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Unable to get buffer size for recording: %s", snd_strerror (err))); snd_pcm_hw_params_free (params); return err; } period_time: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Unable to set period time %i for recording: %s", alsa->period_time, snd_strerror (err))); snd_pcm_hw_params_free (params); return err; } period_size: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Unable to get period size for recording: %s", snd_strerror (err))); snd_pcm_hw_params_free (params); return err; } set_hw_params: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Unable to set hw params for recording: %s", snd_strerror (err))); snd_pcm_hw_params_free (params); return err; } } static int set_swparams (GstAlsaSrc * alsa) { int err; snd_pcm_sw_params_t *params; snd_pcm_sw_params_malloc (¶ms); /* get the current swparams */ CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config); /* allow the transfer when at least period_size samples can be processed */ CHECK (snd_pcm_sw_params_set_avail_min (alsa->handle, params, alsa->period_size), set_avail); /* start the transfer on first read */ CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params, 0), start_threshold); #if GST_CHECK_ALSA_VERSION(1,0,16) /* snd_pcm_sw_params_set_xfer_align() is deprecated, alignment is always 1 */ #else /* align all transfers to 1 sample */ CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align); #endif /* write the parameters to the recording device */ CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params); snd_pcm_sw_params_free (params); return 0; /* ERRORS */ no_config: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Unable to determine current swparams for playback: %s", snd_strerror (err))); snd_pcm_sw_params_free (params); return err; } start_threshold: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Unable to set start threshold mode for playback: %s", snd_strerror (err))); snd_pcm_sw_params_free (params); return err; } set_avail: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Unable to set avail min for playback: %s", snd_strerror (err))); snd_pcm_sw_params_free (params); return err; } #if !GST_CHECK_ALSA_VERSION(1,0,16) set_align: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Unable to set transfer align for playback: %s", snd_strerror (err))); snd_pcm_sw_params_free (params); return err; } #endif set_sw_params: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Unable to set sw params for playback: %s", snd_strerror (err))); snd_pcm_sw_params_free (params); return err; } } static gboolean alsasrc_parse_spec (GstAlsaSrc * alsa, GstRingBufferSpec * spec) { switch (spec->type) { case GST_BUFTYPE_LINEAR: alsa->format = snd_pcm_build_linear_format (spec->depth, spec->width, spec->sign ? 0 : 1, spec->bigend ? 1 : 0); break; case GST_BUFTYPE_FLOAT: switch (spec->format) { case GST_FLOAT32_LE: alsa->format = SND_PCM_FORMAT_FLOAT_LE; break; case GST_FLOAT32_BE: alsa->format = SND_PCM_FORMAT_FLOAT_BE; break; case GST_FLOAT64_LE: alsa->format = SND_PCM_FORMAT_FLOAT64_LE; break; case GST_FLOAT64_BE: alsa->format = SND_PCM_FORMAT_FLOAT64_BE; break; default: goto error; } break; case GST_BUFTYPE_A_LAW: alsa->format = SND_PCM_FORMAT_A_LAW; break; case GST_BUFTYPE_MU_LAW: alsa->format = SND_PCM_FORMAT_MU_LAW; break; default: goto error; } alsa->rate = spec->rate; alsa->channels = spec->channels; alsa->buffer_time = spec->buffer_time; alsa->period_time = spec->latency_time; alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED; return TRUE; /* ERRORS */ error: { return FALSE; } } static gboolean gst_alsasrc_open (GstAudioSrc * asrc) { GstAlsaSrc *alsa; gint err; alsa = GST_ALSA_SRC (asrc); CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK), open_error); if (!alsa->mixer) alsa->mixer = gst_alsa_mixer_new (alsa->device, GST_ALSA_MIXER_CAPTURE); return TRUE; /* ERRORS */ open_error: { if (err == -EBUSY) { GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY, (_("Could not open audio device for recording. " "Device is being used by another application.")), ("Device '%s' is busy", alsa->device)); } else { GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ, (_("Could not open audio device for recording.")), ("Recording open error on device '%s': %s", alsa->device, snd_strerror (err))); } return FALSE; } } static gboolean gst_alsasrc_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec) { GstAlsaSrc *alsa; gint err; alsa = GST_ALSA_SRC (asrc); if (!alsasrc_parse_spec (alsa, spec)) goto spec_parse; CHECK (snd_pcm_nonblock (alsa->handle, 0), non_block); CHECK (set_hwparams (alsa), hw_params_failed); CHECK (set_swparams (alsa), sw_params_failed); CHECK (snd_pcm_prepare (alsa->handle), prepare_failed); alsa->bytes_per_sample = spec->bytes_per_sample; spec->segsize = alsa->period_size * spec->bytes_per_sample; spec->segtotal = alsa->buffer_size / alsa->period_size; spec->silence_sample[0] = 0; spec->silence_sample[1] = 0; spec->silence_sample[2] = 0; spec->silence_sample[3] = 0; return TRUE; /* ERRORS */ spec_parse: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Error parsing spec")); return FALSE; } non_block: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Could not set device to blocking: %s", snd_strerror (err))); return FALSE; } hw_params_failed: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Setting of hwparams failed: %s", snd_strerror (err))); return FALSE; } sw_params_failed: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Setting of swparams failed: %s", snd_strerror (err))); return FALSE; } prepare_failed: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Prepare failed: %s", snd_strerror (err))); return FALSE; } } static gboolean gst_alsasrc_unprepare (GstAudioSrc * asrc) { GstAlsaSrc *alsa; alsa = GST_ALSA_SRC (asrc); snd_pcm_drop (alsa->handle); snd_pcm_hw_free (alsa->handle); snd_pcm_nonblock (alsa->handle, 1); return TRUE; } static gboolean gst_alsasrc_close (GstAudioSrc * asrc) { GstAlsaSrc *alsa = GST_ALSA_SRC (asrc); snd_pcm_close (alsa->handle); alsa->handle = NULL; if (alsa->mixer) { gst_alsa_mixer_free (alsa->mixer); alsa->mixer = NULL; } gst_caps_replace (&alsa->cached_caps, NULL); return TRUE; } /* * Underrun and suspend recovery */ static gint xrun_recovery (GstAlsaSrc * alsa, snd_pcm_t * handle, gint err) { GST_DEBUG_OBJECT (alsa, "xrun recovery %d", err); if (err == -EPIPE) { /* under-run */ err = snd_pcm_prepare (handle); if (err < 0) GST_WARNING_OBJECT (alsa, "Can't recovery from underrun, prepare failed: %s", snd_strerror (err)); return 0; } else if (err == -ESTRPIPE) { while ((err = snd_pcm_resume (handle)) == -EAGAIN) g_usleep (100); /* wait until the suspend flag is released */ if (err < 0) { err = snd_pcm_prepare (handle); if (err < 0) GST_WARNING_OBJECT (alsa, "Can't recovery from suspend, prepare failed: %s", snd_strerror (err)); } return 0; } return err; } static guint gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length) { GstAlsaSrc *alsa; gint err; gint cptr; gint16 *ptr; alsa = GST_ALSA_SRC (asrc); cptr = length / alsa->bytes_per_sample; ptr = data; GST_ALSA_SRC_LOCK (asrc); while (cptr > 0) { if ((err = snd_pcm_readi (alsa->handle, ptr, cptr)) < 0) { if (err == -EAGAIN) { GST_DEBUG_OBJECT (asrc, "Read error: %s", snd_strerror (err)); continue; } else if (xrun_recovery (alsa, alsa->handle, err) < 0) { goto read_error; } continue; } ptr += err * alsa->channels; cptr -= err; } GST_ALSA_SRC_UNLOCK (asrc); return length - (cptr * alsa->bytes_per_sample); read_error: { GST_ALSA_SRC_UNLOCK (asrc); return length; /* skip one period */ } } static guint gst_alsasrc_delay (GstAudioSrc * asrc) { GstAlsaSrc *alsa; snd_pcm_sframes_t delay; int res; alsa = GST_ALSA_SRC (asrc); res = snd_pcm_delay (alsa->handle, &delay); if (G_UNLIKELY (res < 0)) { GST_DEBUG_OBJECT (alsa, "snd_pcm_delay returned %d", res); delay = 0; } return CLAMP (delay, 0, alsa->buffer_size); } static void gst_alsasrc_reset (GstAudioSrc * asrc) { GstAlsaSrc *alsa; gint err; alsa = GST_ALSA_SRC (asrc); GST_ALSA_SRC_LOCK (asrc); GST_DEBUG_OBJECT (alsa, "drop"); CHECK (snd_pcm_drop (alsa->handle), drop_error); GST_DEBUG_OBJECT (alsa, "prepare"); CHECK (snd_pcm_prepare (alsa->handle), prepare_error); GST_DEBUG_OBJECT (alsa, "reset done"); GST_ALSA_SRC_UNLOCK (asrc); return; /* ERRORS */ drop_error: { GST_ERROR_OBJECT (alsa, "alsa-reset: pcm drop error: %s", snd_strerror (err)); GST_ALSA_SRC_UNLOCK (asrc); return; } prepare_error: { GST_ERROR_OBJECT (alsa, "alsa-reset: pcm prepare error: %s", snd_strerror (err)); GST_ALSA_SRC_UNLOCK (asrc); return; } }