diff -r 567bb019e3e3 -r 7e817e7e631c gst_plugins_base/ext/alsa/gstalsasrc.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/gst_plugins_base/ext/alsa/gstalsasrc.c Wed Sep 01 12:16:41 2010 +0100 @@ -0,0 +1,881 @@ +/* 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 + * @short_description: capture audio from an alsa device + * @see_also: alsasink, alsamixer + * + * + * + * This element reads data from an audio card using the ALSA API. + * + * Example pipelines + * + * Record from a sound card using ALSA and encode to Ogg/Vorbis. + * + * + * gst-launch -v alsasrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=alsasrc.ogg + * + * + * + * 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 + +/* elementfactory information */ +static const GstElementDetails gst_alsasrc_details = +GST_ELEMENT_DETAILS ("Audio source (ALSA)", + "Source/Audio", + "Read from a sound card via ALSA", + "Wim Taymans "); + +#define DEFAULT_PROP_DEVICE "default" +#define DEFAULT_PROP_DEVICE_NAME "" + +enum +{ + PROP_0, + PROP_DEVICE, + PROP_DEVICE_NAME, +}; + +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 (element_class, &gst_alsasrc_details); + + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&alsasrc_src_factory)); +} + +static void +gst_alsasrc_class_init (GstAlsaSrcClass * klass) +{ + GObjectClass *gobject_class; + GstElementClass *gstelement_class; + GstBaseSrcClass *gstbasesrc_class; + GstBaseAudioSrcClass *gstbaseaudiosrc_class; + GstAudioSrcClass *gstaudiosrc_class; + + gobject_class = (GObjectClass *) klass; + gstelement_class = (GstElementClass *) klass; + gstbasesrc_class = (GstBaseSrcClass *) klass; + gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass; + gstaudiosrc_class = (GstAudioSrcClass *) klass; + + gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_alsasrc_finalize); + gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_alsasrc_get_property); + gobject_class->set_property = GST_DEBUG_FUNCPTR (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_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)); +} + +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; + 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, dir; + 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, &dir), 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, &dir), 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, &dir), + 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, (msg), (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; + gint err; + + alsa = GST_ALSA_SRC (asrc); + + 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_alsasrc_close (GstAudioSrc * asrc) +{ + GstAlsaSrc *alsa = GST_ALSA_SRC (asrc); + + snd_pcm_close (alsa->handle); + + 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; + +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; + } +}