--- /dev/null
+/* 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>
+
+#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 <wim@fluendo.com>");
+
+ 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));
+
+ 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;
+ }
+}
projects / mafwsubrenderer / blobdiff