/* GStreamer * Copyright (C) 2005 Wim Taymans * Copyright (C) 2006 Tim-Philipp Müller * * gstalsasink.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-alsasink * @short_description: play audio to an ALSA device * @see_also: alsasrc, alsamixer * * * * This element renders raw audio samples using the ALSA api. * * Example pipelines * * Play an Ogg/Vorbis file. * * * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! alsasink * * * * 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 "gstalsa.h" #include "gstalsasink.h" #include "gstalsadeviceprobe.h" #include /* elementfactory information */ static const GstElementDetails gst_alsasink_details = GST_ELEMENT_DETAILS ("Audio sink (ALSA)", "Sink/Audio", "Output to a sound card via ALSA", "Wim Taymans "); #define DEFAULT_DEVICE "default" #define DEFAULT_DEVICE_NAME "" enum { PROP_0, PROP_DEVICE, PROP_DEVICE_NAME }; static void gst_alsasink_init_interfaces (GType type); GST_BOILERPLATE_FULL (GstAlsaSink, gst_alsasink, GstAudioSink, GST_TYPE_AUDIO_SINK, gst_alsasink_init_interfaces); static void gst_alsasink_finalise (GObject * object); static void gst_alsasink_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_alsasink_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static GstCaps *gst_alsasink_getcaps (GstBaseSink * bsink); static gboolean gst_alsasink_open (GstAudioSink * asink); static gboolean gst_alsasink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec); static gboolean gst_alsasink_unprepare (GstAudioSink * asink); static gboolean gst_alsasink_close (GstAudioSink * asink); static guint gst_alsasink_write (GstAudioSink * asink, gpointer data, guint length); static guint gst_alsasink_delay (GstAudioSink * asink); static void gst_alsasink_reset (GstAudioSink * asink); static gint output_ref; /* 0 */ static snd_output_t *output; /* NULL */ static GStaticMutex output_mutex = G_STATIC_MUTEX_INIT; #if (G_BYTE_ORDER == G_LITTLE_ENDIAN) # define ALSA_SINK_FACTORY_ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN" #else # define ALSA_SINK_FACTORY_ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN" #endif static GstStaticPadTemplate alsasink_sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-int, " "endianness = (int) { " ALSA_SINK_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_SINK_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_alsasink_finalise (GObject * object) { GstAlsaSink *sink = GST_ALSA_SINK (object); g_free (sink->device); g_mutex_free (sink->alsa_lock); g_static_mutex_lock (&output_mutex); --output_ref; if (output_ref == 0) { snd_output_close (output); output = NULL; } g_static_mutex_unlock (&output_mutex); G_OBJECT_CLASS (parent_class)->finalize (object); } static void gst_alsasink_init_interfaces (GType type) { gst_alsa_type_add_device_property_probe_interface (type); } static void gst_alsasink_base_init (gpointer g_class) { GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); gst_element_class_set_details (element_class, &gst_alsasink_details); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&alsasink_sink_factory)); } static void gst_alsasink_class_init (GstAlsaSinkClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstBaseSinkClass *gstbasesink_class; GstBaseAudioSinkClass *gstbaseaudiosink_class; GstAudioSinkClass *gstaudiosink_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstbasesink_class = (GstBaseSinkClass *) klass; gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass; gstaudiosink_class = (GstAudioSinkClass *) klass; parent_class = g_type_class_peek_parent (klass); gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_alsasink_finalise); gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_alsasink_get_property); gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_alsasink_set_property); gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasink_getcaps); gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_alsasink_open); gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasink_prepare); gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasink_unprepare); gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_alsasink_close); gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_alsasink_write); gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_alsasink_delay); gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_alsasink_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_DEVICE, G_PARAM_READWRITE)); 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_DEVICE_NAME, G_PARAM_READABLE)); } static void gst_alsasink_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAlsaSink *sink; sink = GST_ALSA_SINK (object); switch (prop_id) { case PROP_DEVICE: g_free (sink->device); sink->device = g_value_dup_string (value); /* setting NULL restores the default device */ if (sink->device == NULL) { sink->device = g_strdup (DEFAULT_DEVICE); } break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_alsasink_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAlsaSink *sink; sink = GST_ALSA_SINK (object); switch (prop_id) { case PROP_DEVICE: g_value_set_string (value, sink->device); break; case PROP_DEVICE_NAME: g_value_take_string (value, gst_alsa_find_device_name (GST_OBJECT_CAST (sink), sink->device, sink->handle, SND_PCM_STREAM_PLAYBACK)); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_alsasink_init (GstAlsaSink * alsasink, GstAlsaSinkClass * g_class) { GST_DEBUG_OBJECT (alsasink, "initializing alsasink"); alsasink->device = g_strdup (DEFAULT_DEVICE); alsasink->handle = NULL; alsasink->cached_caps = NULL; alsasink->alsa_lock = g_mutex_new (); g_static_mutex_lock (&output_mutex); if (output_ref == 0) { snd_output_stdio_attach (&output, stdout, 0); ++output_ref; } g_static_mutex_unlock (&output_mutex); } #define CHECK(call, error) \ G_STMT_START { \ if ((err = call) < 0) \ goto error; \ } G_STMT_END; static GstCaps * gst_alsasink_getcaps (GstBaseSink * bsink) { GstElementClass *element_class; GstPadTemplate *pad_template; GstAlsaSink *sink = GST_ALSA_SINK (bsink); GstCaps *caps; if (sink->handle == NULL) { GST_DEBUG_OBJECT (sink, "device not open, using template caps"); return NULL; /* base class will get template caps for us */ } if (sink->cached_caps) { GST_LOG_OBJECT (sink, "Returning cached caps"); return gst_caps_ref (sink->cached_caps); } element_class = GST_ELEMENT_GET_CLASS (sink); pad_template = gst_element_class_get_pad_template (element_class, "sink"); g_return_val_if_fail (pad_template != NULL, NULL); caps = gst_alsa_probe_supported_formats (GST_OBJECT (sink), sink->handle, gst_pad_template_get_caps (pad_template)); if (caps) { sink->cached_caps = gst_caps_ref (caps); } GST_INFO_OBJECT (sink, "returning caps %" GST_PTR_FORMAT, caps); return caps; } static int set_hwparams (GstAlsaSink * alsa) { guint rrate; gint err, dir; snd_pcm_hw_params_t *params; guint period_time, buffer_time; snd_pcm_hw_params_alloca (¶ms); GST_DEBUG_OBJECT (alsa, "Negotiating to %d channels @ %d Hz (format = %s)", alsa->channels, alsa->rate, snd_pcm_format_name (alsa->format)); /* start with requested values, if we cannot configure alsa for those values, * we set these values to -1, which will leave the default alsa values */ buffer_time = alsa->buffer_time; period_time = alsa->period_time; retry: /* 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, 0), no_rate); if (rrate != alsa->rate) goto rate_match; /* get and dump some limits */ { guint min, max; snd_pcm_hw_params_get_buffer_time_min (params, &min, &dir); snd_pcm_hw_params_get_buffer_time_max (params, &max, &dir); GST_DEBUG_OBJECT (alsa, "buffer time %u, min %u, max %u", alsa->buffer_time, min, max); snd_pcm_hw_params_get_period_time_min (params, &min, &dir); snd_pcm_hw_params_get_period_time_max (params, &max, &dir); GST_DEBUG_OBJECT (alsa, "period time %u, min %u, max %u", alsa->period_time, min, max); snd_pcm_hw_params_get_periods_min (params, &min, &dir); snd_pcm_hw_params_get_periods_max (params, &max, &dir); GST_DEBUG_OBJECT (alsa, "periods min %u, max %u", min, max); } /* now try to configure the buffer time and period time, if one * of those fail, we fall back to the defaults and emit a warning. */ if (buffer_time != -1) { /* set the buffer time */ if ((err = snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params, &buffer_time, &dir)) < 0) { GST_ELEMENT_WARNING (alsa, RESOURCE, SETTINGS, (NULL), ("Unable to set buffer time %i for playback: %s", buffer_time, snd_strerror (err))); /* disable buffer_time the next round */ buffer_time = -1; goto retry; } } if (period_time != -1) { /* set the period time */ if ((err = snd_pcm_hw_params_set_period_time_near (alsa->handle, params, &period_time, &dir)) < 0) { GST_ELEMENT_WARNING (alsa, RESOURCE, SETTINGS, (NULL), ("Unable to set period time %i for playback: %s", period_time, snd_strerror (err))); /* disable period_time the next round */ period_time = -1; goto retry; } } /* write the parameters to device */ CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params); /* now get the configured values */ 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, &dir), period_size); GST_DEBUG_OBJECT (alsa, "buffer size %lu, period size %lu", alsa->buffer_size, alsa->period_size); return 0; /* ERRORS */ no_config: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Broken configuration for playback: no configurations available: %s", snd_strerror (err))); return err; } wrong_access: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Access type not available for playback: %s", snd_strerror (err))); return err; } no_sample_format: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Sample format not available for playback: %s", snd_strerror (err))); return err; } no_channels: { gchar *msg = NULL; if ((alsa->channels) == 1) msg = g_strdup (_("Could not open device for playback in mono mode.")); if ((alsa->channels) == 2) msg = g_strdup (_("Could not open device for playback in stereo mode.")); if ((alsa->channels) > 2) msg = g_strdup_printf (_ ("Could not open device for playback in %d-channel mode."), alsa->channels); GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (msg), (snd_strerror (err))); g_free (msg); return err; } no_rate: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Rate %iHz not available for playback: %s", alsa->rate, snd_strerror (err))); return err; } rate_match: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Rate doesn't match (requested %iHz, get %iHz)", alsa->rate, err)); return -EINVAL; } buffer_size: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Unable to get buffer size for playback: %s", snd_strerror (err))); return err; } period_size: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Unable to get period size for playback: %s", snd_strerror (err))); return err; } set_hw_params: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Unable to set hw params for playback: %s", snd_strerror (err))); return err; } } static int set_swparams (GstAlsaSink * alsa) { int err; snd_pcm_sw_params_t *params; snd_pcm_sw_params_alloca (¶ms); /* get the current swparams */ CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config); /* start the transfer when the buffer is almost full: */ /* (buffer_size / avail_min) * avail_min */ CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params, (alsa->buffer_size / alsa->period_size) * alsa->period_size), start_threshold); /* 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); /* align all transfers to 1 sample */ CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align); /* write the parameters to the playback device */ CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params); return 0; /* ERRORS */ no_config: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Unable to determine current swparams for playback: %s", snd_strerror (err))); return err; } start_threshold: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Unable to set start threshold mode for playback: %s", snd_strerror (err))); return err; } set_avail: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Unable to set avail min for playback: %s", snd_strerror (err))); return err; } set_align: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Unable to set transfer align for playback: %s", snd_strerror (err))); return err; } set_sw_params: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Unable to set sw params for playback: %s", snd_strerror (err))); return err; } } static gboolean alsasink_parse_spec (GstAlsaSink * alsa, GstRingBufferSpec * spec) { switch (spec->type) { case GST_BUFTYPE_LINEAR: GST_DEBUG_OBJECT (alsa, "Linear format : depth=%d, width=%d, sign=%d, bigend=%d", spec->depth, spec->width, spec->sign, spec->bigend); 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_alsasink_open (GstAudioSink * asink) { GstAlsaSink *alsa; gint err; alsa = GST_ALSA_SINK (asink); CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK), open_error); return TRUE; /* ERRORS */ open_error: { if (err == -EBUSY) { GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY, (NULL), ("Device '%s' is busy", alsa->device)); } else { GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_WRITE, (NULL), ("Playback open error on device '%s': %s", alsa->device, snd_strerror (err))); } return FALSE; } } static gboolean gst_alsasink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec) { GstAlsaSink *alsa; gint err; alsa = GST_ALSA_SINK (asink); if (!alsasink_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); 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; 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; } } static gboolean gst_alsasink_unprepare (GstAudioSink * asink) { GstAlsaSink *alsa; gint err; alsa = GST_ALSA_SINK (asink); CHECK (snd_pcm_drop (alsa->handle), drop); CHECK (snd_pcm_hw_free (alsa->handle), hw_free); CHECK (snd_pcm_nonblock (alsa->handle, 1), non_block); return TRUE; /* ERRORS */ drop: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Could not drop samples: %s", snd_strerror (err))); return FALSE; } hw_free: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Could not free hw params: %s", snd_strerror (err))); return FALSE; } non_block: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Could not set device to nonblocking: %s", snd_strerror (err))); return FALSE; } } static gboolean gst_alsasink_close (GstAudioSink * asink) { GstAlsaSink *alsa = GST_ALSA_SINK (asink); gint err; CHECK (snd_pcm_close (alsa->handle), close_error); alsa->handle = NULL; gst_caps_replace (&alsa->cached_caps, NULL); return TRUE; /* ERRORS */ close_error: { GST_ELEMENT_ERROR (alsa, RESOURCE, CLOSE, (NULL), ("Playback close error: %s", snd_strerror (err))); return FALSE; } } /* * Underrun and suspend recovery */ static gint xrun_recovery (GstAlsaSink * 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_alsasink_write (GstAudioSink * asink, gpointer data, guint length) { GstAlsaSink *alsa; gint err; gint cptr; gint16 *ptr; alsa = GST_ALSA_SINK (asink); cptr = length / alsa->bytes_per_sample; ptr = data; GST_ALSA_SINK_LOCK (asink); while (cptr > 0) { err = snd_pcm_writei (alsa->handle, ptr, cptr); GST_DEBUG_OBJECT (asink, "written %d result %d", cptr, err); if (err < 0) { GST_DEBUG_OBJECT (asink, "Write error: %s", snd_strerror (err)); if (err == -EAGAIN) { continue; } else if (xrun_recovery (alsa, alsa->handle, err) < 0) { goto write_error; } continue; } ptr += err * alsa->channels; cptr -= err; } GST_ALSA_SINK_UNLOCK (asink); return length - cptr; write_error: { GST_ALSA_SINK_UNLOCK (asink); return length; /* skip one period */ } } static guint gst_alsasink_delay (GstAudioSink * asink) { GstAlsaSink *alsa; snd_pcm_sframes_t delay; alsa = GST_ALSA_SINK (asink); snd_pcm_delay (alsa->handle, &delay); return delay; } static void gst_alsasink_reset (GstAudioSink * asink) { GstAlsaSink *alsa; gint err; alsa = GST_ALSA_SINK (asink); GST_ALSA_SINK_LOCK (asink); 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_SINK_UNLOCK (asink); return; /* ERRORS */ drop_error: { GST_ERROR_OBJECT (alsa, "alsa-reset: pcm drop error: %s", snd_strerror (err)); GST_ALSA_SINK_UNLOCK (asink); return; } prepare_error: { GST_ERROR_OBJECT (alsa, "alsa-reset: pcm prepare error: %s", snd_strerror (err)); GST_ALSA_SINK_UNLOCK (asink); return; } }