diff options
Diffstat (limited to 'ext/alsa/gstalsasrc.c')
| -rw-r--r-- | ext/alsa/gstalsasrc.c | 873 |
1 files changed, 0 insertions, 873 deletions
diff --git a/ext/alsa/gstalsasrc.c b/ext/alsa/gstalsasrc.c deleted file mode 100644 index db9c3a60..00000000 --- a/ext/alsa/gstalsasrc.c +++ /dev/null @@ -1,873 +0,0 @@ -/* GStreamer - * Copyright (C) 2005 Wim Taymans <wim@fluendo.com> - * - * 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. - * - * <refsect2> - * <title>Example pipelines</title> - * |[ - * gst-launch -v alsasrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=alsasrc.ogg - * ]| Record from a sound card using ALSA and encode to Ogg/Vorbis. - * </refsect2> - * - * Last reviewed on 2006-03-01 (0.10.4) - */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif -#include <sys/ioctl.h> -#include <fcntl.h> -#include <errno.h> -#include <unistd.h> -#include <string.h> -#include <getopt.h> -#include <alsa/asoundlib.h> - -#include "gstalsasrc.h" -#include "gstalsadeviceprobe.h" - -#include <gst/gst-i18n-plugin.h> - -/* 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 <wim@fluendo.com>"); - -#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; - 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)); -} - -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); - 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; - -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; - } -} |