--- /dev/null Thu Jan 01 00:00:00 1970 +0000
+++ b/src/3rdparty/phonon/gstreamer/alsasink2.c Mon Jan 11 14:00:40 2010 +0000
@@ -0,0 +1,1756 @@
+/* GStreamer
+ * Copyright (C) 2001 CodeFactory AB
+ * Copyright (C) 2001 Thomas Nyberg <thomas@codefactory.se>
+ * Copyright (C) 2001-2002 Andy Wingo <apwingo@eos.ncsu.edu>
+ * Copyright (C) 2003 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ * Copyright (C) 2005 Wim Taymans <wim@fluendo.com>
+ * Copyright (C) 2005, 2006 Tim-Philipp Müller <tim centricular net>
+ * Copyright (C) 2008 Matthias Kretz <kretz@kde.org>
+ *
+ * gstalsasink2.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 Lesser General Public
+ * License along with this library. If not, see <http://www.gnu.org/licenses/>.
+ */
+
+/**
+ * SECTION:element-alsasink2
+ * @short_description: play audio to an ALSA device
+ * @see_also: alsasrc, alsamixer
+ *
+ * <refsect2>
+ * <para>
+ * This element renders raw audio samples using the ALSA api.
+ * </para>
+ * <title>Example pipelines</title>
+ * <para>
+ * Play an Ogg/Vorbis file.
+ * </para>
+ * <programlisting>
+ * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! alsasink2
+ * </programlisting>
+ * </refsect2>
+ *
+ * Last reviewed on 2006-03-01 (0.10.4)
+ */
+
+#define _XOPEN_SOURCE 600
+
+#include <sys/ioctl.h>
+#include <fcntl.h>
+#include <errno.h>
+#include <unistd.h>
+#include <string.h>
+#include <getopt.h>
+#include <alsa/asoundlib.h>
+
+#include "alsasink2.h"
+
+#include <gst/interfaces/propertyprobe.h>
+#include <gst/audio/multichannel.h>
+
+#define _(text) (text)
+
+#define GST_CHECK_ALSA_VERSION(major,minor,micro) \
+ (SND_LIB_MAJOR > (major) || \
+ (SND_LIB_MAJOR == (major) && SND_LIB_MINOR > (minor)) || \
+ (SND_LIB_MAJOR == (major) && SND_LIB_MINOR == (minor) && \
+ SND_LIB_SUBMINOR >= (micro)))
+
+static const GList *
+gst_alsa_device_property_probe_get_properties (GstPropertyProbe * probe)
+{
+ GObjectClass *klass = G_OBJECT_GET_CLASS (probe);
+ static GList *list = NULL;
+
+ /* well, not perfect, but better than no locking at all.
+ * In the worst case we leak a list node, so who cares? */
+ GST_CLASS_LOCK (GST_OBJECT_CLASS (klass));
+
+ if (!list) {
+ GParamSpec *pspec;
+
+ pspec = g_object_class_find_property (klass, "device");
+ list = g_list_append (NULL, pspec);
+ }
+
+ GST_CLASS_UNLOCK (GST_OBJECT_CLASS (klass));
+
+ return list;
+}
+
+static GList *
+gst_alsa_get_device_list (snd_pcm_stream_t stream)
+{
+ snd_ctl_t *handle;
+ int card, err, dev;
+ snd_ctl_card_info_t *info;
+ snd_pcm_info_t *pcminfo;
+ gboolean mixer = (stream == ~0u);
+ GList *list = NULL;
+
+ if (stream == ~0u)
+ stream = 0;
+
+ snd_ctl_card_info_malloc (&info);
+ snd_pcm_info_malloc (&pcminfo);
+ card = -1;
+
+ if (snd_card_next (&card) < 0 || card < 0) {
+ /* no soundcard found */
+ return NULL;
+ }
+
+ while (card >= 0) {
+ gchar name[32];
+
+ g_snprintf (name, sizeof (name), "hw:%d", card);
+ if ((err = snd_ctl_open (&handle, name, 0)) < 0) {
+ goto next_card;
+ }
+ if ((err = snd_ctl_card_info (handle, info)) < 0) {
+ snd_ctl_close (handle);
+ goto next_card;
+ }
+
+ if (mixer) {
+ list = g_list_append (list, g_strdup (name));
+ } else {
+ g_snprintf (name, sizeof (name), "default:CARD=%d", card);
+ list = g_list_append (list, g_strdup (name));
+ dev = -1;
+ while (1) {
+ gchar *gst_device;
+
+ snd_ctl_pcm_next_device (handle, &dev);
+
+ if (dev < 0)
+ break;
+ snd_pcm_info_set_device (pcminfo, dev);
+ snd_pcm_info_set_subdevice (pcminfo, 0);
+ snd_pcm_info_set_stream (pcminfo, stream);
+ if ((err = snd_ctl_pcm_info (handle, pcminfo)) < 0) {
+ continue;
+ }
+
+ gst_device = g_strdup_printf ("hw:%d,%d", card, dev);
+ list = g_list_append (list, gst_device);
+ }
+ }
+ snd_ctl_close (handle);
+ next_card:
+ if (snd_card_next (&card) < 0) {
+ break;
+ }
+ }
+
+ snd_ctl_card_info_free (info);
+ snd_pcm_info_free (pcminfo);
+
+ return list;
+}
+
+static void
+gst_alsa_device_property_probe_probe_property (GstPropertyProbe * probe,
+ guint prop_id, const GParamSpec * pspec)
+{
+ if (!g_str_equal (pspec->name, "device")) {
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (probe, prop_id, pspec);
+ }
+}
+
+static gboolean
+gst_alsa_device_property_probe_needs_probe (GstPropertyProbe * probe,
+ guint prop_id, const GParamSpec * pspec)
+{
+ /* don't cache probed data */
+ return TRUE;
+}
+
+static GValueArray *
+gst_alsa_device_property_probe_get_values (GstPropertyProbe * probe,
+ guint prop_id, const GParamSpec * pspec)
+{
+ GstElementClass *klass;
+ const GList *templates;
+ snd_pcm_stream_t mode = -1;
+ GValueArray *array;
+ GValue value = { 0, };
+ GList *l, *list;
+
+ if (!g_str_equal (pspec->name, "device")) {
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (probe, prop_id, pspec);
+ return NULL;
+ }
+
+ klass = GST_ELEMENT_GET_CLASS (GST_ELEMENT (probe));
+
+ /* I'm pretty sure ALSA has a good way to do this. However, their cool
+ * auto-generated documentation is pretty much useless if you try to
+ * do function-wise look-ups. */
+ /* we assume one pad template at max [zero=mixer] */
+ templates = gst_element_class_get_pad_template_list (klass);
+ if (templates) {
+ if (GST_PAD_TEMPLATE_DIRECTION (templates->data) == GST_PAD_SRC)
+ mode = SND_PCM_STREAM_CAPTURE;
+ else
+ mode = SND_PCM_STREAM_PLAYBACK;
+ }
+
+ list = gst_alsa_get_device_list (mode);
+
+ if (list == NULL) {
+ GST_LOG_OBJECT (probe, "No devices found");
+ return NULL;
+ }
+
+ array = g_value_array_new (g_list_length (list));
+ g_value_init (&value, G_TYPE_STRING);
+ for (l = list; l != NULL; l = l->next) {
+ GST_LOG_OBJECT (probe, "Found device: %s", (gchar *) l->data);
+ g_value_take_string (&value, (gchar *) l->data);
+ l->data = NULL;
+ g_value_array_append (array, &value);
+ }
+ g_value_unset (&value);
+ g_list_free (list);
+
+ return array;
+}
+
+static void
+gst_alsa_property_probe_interface_init (GstPropertyProbeInterface * iface)
+{
+ iface->get_properties = gst_alsa_device_property_probe_get_properties;
+ iface->probe_property = gst_alsa_device_property_probe_probe_property;
+ iface->needs_probe = gst_alsa_device_property_probe_needs_probe;
+ iface->get_values = gst_alsa_device_property_probe_get_values;
+}
+
+static void
+gst_alsa_type_add_device_property_probe_interface (GType type)
+{
+ static const GInterfaceInfo probe_iface_info = {
+ (GInterfaceInitFunc) gst_alsa_property_probe_interface_init,
+ NULL,
+ NULL,
+ };
+
+ g_type_add_interface_static (type, GST_TYPE_PROPERTY_PROBE,
+ &probe_iface_info);
+}
+
+static GstCaps *
+gst_alsa_detect_rates (GstObject * obj, snd_pcm_hw_params_t * hw_params,
+ GstCaps * in_caps)
+{
+ GstCaps *caps;
+ guint min, max;
+ gint err, dir, min_rate, max_rate;
+ guint i;
+
+ GST_LOG_OBJECT (obj, "probing sample rates ...");
+
+ if ((err = snd_pcm_hw_params_get_rate_min (hw_params, &min, &dir)) < 0)
+ goto min_rate_err;
+
+ if ((err = snd_pcm_hw_params_get_rate_max (hw_params, &max, &dir)) < 0)
+ goto max_rate_err;
+
+ min_rate = min;
+ max_rate = max;
+
+ if (min_rate < 4000)
+ min_rate = 4000; /* random 'sensible minimum' */
+
+ if (max_rate <= 0)
+ max_rate = G_MAXINT; /* or maybe just use 192400 or so? */
+ else if (max_rate > 0 && max_rate < 4000)
+ max_rate = MAX (4000, min_rate);
+
+ GST_DEBUG_OBJECT (obj, "Min. rate = %u (%d)", min_rate, min);
+ GST_DEBUG_OBJECT (obj, "Max. rate = %u (%d)", max_rate, max);
+
+ caps = gst_caps_make_writable (in_caps);
+
+ for (i = 0; i < gst_caps_get_size (caps); ++i) {
+ GstStructure *s;
+
+ s = gst_caps_get_structure (caps, i);
+ if (min_rate == max_rate) {
+ gst_structure_set (s, "rate", G_TYPE_INT, min_rate, NULL);
+ } else {
+ gst_structure_set (s, "rate", GST_TYPE_INT_RANGE,
+ min_rate, max_rate, NULL);
+ }
+ }
+
+ return caps;
+
+ /* ERRORS */
+min_rate_err:
+ {
+ GST_ERROR_OBJECT (obj, "failed to query minimum sample rate: %s",
+ snd_strerror (err));
+ gst_caps_unref (in_caps);
+ return NULL;
+ }
+max_rate_err:
+ {
+ GST_ERROR_OBJECT (obj, "failed to query maximum sample rate: %s",
+ snd_strerror (err));
+ gst_caps_unref (in_caps);
+ return NULL;
+ }
+}
+
+static const struct
+{
+ const int width;
+ const int depth;
+ const int sformat;
+ const int uformat;
+} pcmformats[] = {
+ {
+ 8, 8, SND_PCM_FORMAT_S8, SND_PCM_FORMAT_U8}, {
+ 16, 16, SND_PCM_FORMAT_S16, SND_PCM_FORMAT_U16}, {
+ 32, 24, SND_PCM_FORMAT_S24, SND_PCM_FORMAT_U24}, {
+#if (G_BYTE_ORDER == G_LITTLE_ENDIAN) /* no endian-unspecific enum available */
+ 24, 24, SND_PCM_FORMAT_S24_3LE, SND_PCM_FORMAT_U24_3LE}, {
+#else
+ 24, 24, SND_PCM_FORMAT_S24_3BE, SND_PCM_FORMAT_U24_3BE}, {
+#endif
+ 32, 32, SND_PCM_FORMAT_S32, SND_PCM_FORMAT_U32}
+};
+
+static GstCaps *
+gst_alsa_detect_formats (GstObject * obj, snd_pcm_hw_params_t * hw_params,
+ GstCaps * in_caps)
+{
+ snd_pcm_format_mask_t *mask;
+ GstStructure *s;
+ GstCaps *caps;
+ guint i;
+
+ snd_pcm_format_mask_malloc (&mask);
+ snd_pcm_hw_params_get_format_mask (hw_params, mask);
+
+ caps = gst_caps_new_empty ();
+
+ for (i = 0; i < gst_caps_get_size (in_caps); ++i) {
+ GstStructure *scopy;
+ guint w;
+ gint width = 0, depth = 0;
+
+ s = gst_caps_get_structure (in_caps, i);
+ if (!gst_structure_has_name (s, "audio/x-raw-int")) {
+ GST_WARNING_OBJECT (obj, "skipping non-int format");
+ continue;
+ }
+ if (!gst_structure_get_int (s, "width", &width) ||
+ !gst_structure_get_int (s, "depth", &depth))
+ continue;
+ if (width == 0 || (width % 8) != 0)
+ continue; /* Only full byte widths are valid */
+ for (w = 0; w < G_N_ELEMENTS (pcmformats); w++)
+ if (pcmformats[w].width == width && pcmformats[w].depth == depth)
+ break;
+ if (w == G_N_ELEMENTS (pcmformats))
+ continue; /* Unknown format */
+
+ if (snd_pcm_format_mask_test (mask, pcmformats[w].sformat) &&
+ snd_pcm_format_mask_test (mask, pcmformats[w].uformat)) {
+ /* template contains { true, false } or just one, leave it as it is */
+ scopy = gst_structure_copy (s);
+ } else if (snd_pcm_format_mask_test (mask, pcmformats[w].sformat)) {
+ scopy = gst_structure_copy (s);
+ gst_structure_set (scopy, "signed", G_TYPE_BOOLEAN, TRUE, NULL);
+ } else if (snd_pcm_format_mask_test (mask, pcmformats[w].uformat)) {
+ scopy = gst_structure_copy (s);
+ gst_structure_set (scopy, "signed", G_TYPE_BOOLEAN, FALSE, NULL);
+ } else {
+ scopy = NULL;
+ }
+ if (scopy) {
+ if (width > 8) {
+ /* TODO: proper endianness detection, for now it's CPU endianness only */
+ gst_structure_set (scopy, "endianness", G_TYPE_INT, G_BYTE_ORDER, NULL);
+ }
+ gst_caps_append_structure (caps, scopy);
+ }
+ }
+
+ snd_pcm_format_mask_free (mask);
+ gst_caps_unref (in_caps);
+ return caps;
+}
+
+/* we don't have channel mappings for more than this many channels */
+#define GST_ALSA_MAX_CHANNELS 8
+
+static GstStructure *
+get_channel_free_structure (const GstStructure * in_structure)
+{
+ GstStructure *s = gst_structure_copy (in_structure);
+
+ gst_structure_remove_field (s, "channels");
+ return s;
+}
+
+static void
+caps_add_channel_configuration (GstCaps * caps,
+ const GstStructure * in_structure, gint min_chans, gint max_chans)
+{
+ GstAudioChannelPosition pos[8] = {
+ GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+ GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
+ GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
+ GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
+ GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
+ GST_AUDIO_CHANNEL_POSITION_LFE,
+ GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
+ GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT
+ };
+ GstStructure *s = NULL;
+ gint c;
+
+ if (min_chans == max_chans && max_chans <= 2) {
+ s = get_channel_free_structure (in_structure);
+ gst_structure_set (s, "channels", G_TYPE_INT, max_chans, NULL);
+ gst_caps_append_structure (caps, s);
+ return;
+ }
+
+ g_assert (min_chans >= 1);
+
+ /* mono and stereo don't need channel configurations */
+ if (min_chans == 2) {
+ s = get_channel_free_structure (in_structure);
+ gst_structure_set (s, "channels", G_TYPE_INT, 2, NULL);
+ gst_caps_append_structure (caps, s);
+ } else if (min_chans == 1 && max_chans >= 2) {
+ s = get_channel_free_structure (in_structure);
+ gst_structure_set (s, "channels", GST_TYPE_INT_RANGE, 1, 2, NULL);
+ gst_caps_append_structure (caps, s);
+ }
+
+ /* don't know whether to use 2.1 or 3.0 here - but I suspect
+ * alsa might work around that/fix it somehow. Can we tell alsa
+ * what our channel layout is like? */
+ if (max_chans >= 3 && min_chans <= 3) {
+ GstAudioChannelPosition pos_21[3] = {
+ GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+ GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
+ GST_AUDIO_CHANNEL_POSITION_LFE
+ };
+
+ s = get_channel_free_structure (in_structure);
+ gst_structure_set (s, "channels", G_TYPE_INT, 3, NULL);
+ gst_audio_set_channel_positions (s, pos_21);
+ gst_caps_append_structure (caps, s);
+ }
+
+ /* everything else (4, 6, 8 channels) needs a channel layout */
+ for (c = MAX (4, min_chans); c <= 8; c += 2) {
+ if (max_chans >= c) {
+ s = get_channel_free_structure (in_structure);
+ gst_structure_set (s, "channels", G_TYPE_INT, c, NULL);
+ gst_audio_set_channel_positions (s, pos);
+ gst_caps_append_structure (caps, s);
+ }
+ }
+
+ for (c = MAX (9, min_chans); c <= max_chans; ++c) {
+ GstAudioChannelPosition *ch_layout;
+ gint i;
+
+ ch_layout = g_new (GstAudioChannelPosition, c);
+ for (i = 0; i < c; ++i) {
+ ch_layout[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
+ }
+ s = get_channel_free_structure (in_structure);
+ gst_structure_set (s, "channels", G_TYPE_INT, c, NULL);
+ gst_audio_set_channel_positions (s, ch_layout);
+ gst_caps_append_structure (caps, s);
+ g_free (ch_layout);
+ }
+}
+
+static GstCaps *
+gst_alsa_detect_channels (GstObject * obj, snd_pcm_hw_params_t * hw_params,
+ GstCaps * in_caps)
+{
+ GstCaps *caps;
+ guint min, max;
+ gint min_chans, max_chans;
+ gint err;
+ guint i;
+
+ GST_LOG_OBJECT (obj, "probing channels ...");
+
+ if ((err = snd_pcm_hw_params_get_channels_min (hw_params, &min)) < 0)
+ goto min_chan_error;
+
+ if ((err = snd_pcm_hw_params_get_channels_max (hw_params, &max)) < 0)
+ goto max_chan_error;
+
+ /* note: the above functions may return (guint) -1 */
+ min_chans = min;
+ max_chans = max;
+
+ if (min_chans < 0) {
+ min_chans = 1;
+ max_chans = GST_ALSA_MAX_CHANNELS;
+ } else if (max_chans < 0) {
+ max_chans = GST_ALSA_MAX_CHANNELS;
+ }
+
+ if (min_chans > max_chans) {
+ gint temp;
+
+ GST_WARNING_OBJECT (obj, "minimum channels > maximum channels (%d > %d), "
+ "please fix your soundcard drivers", min, max);
+ temp = min_chans;
+ min_chans = max_chans;
+ max_chans = temp;
+ }
+
+ /* pro cards seem to return large numbers for min_channels */
+ if (min_chans > GST_ALSA_MAX_CHANNELS) {
+ GST_DEBUG_OBJECT (obj, "min_chans = %u, looks like a pro card", min_chans);
+ if (max_chans < min_chans) {
+ max_chans = min_chans;
+ } else {
+ /* only support [max_chans; max_chans] for these cards for now
+ * to avoid inflating the source caps with loads of structures ... */
+ min_chans = max_chans;
+ }
+ } else {
+ min_chans = MAX (min_chans, 1);
+ max_chans = MIN (GST_ALSA_MAX_CHANNELS, max_chans);
+ }
+
+ GST_DEBUG_OBJECT (obj, "Min. channels = %d (%d)", min_chans, min);
+ GST_DEBUG_OBJECT (obj, "Max. channels = %d (%d)", max_chans, max);
+
+ caps = gst_caps_new_empty ();
+
+ for (i = 0; i < gst_caps_get_size (in_caps); ++i) {
+ GstStructure *s;
+ GType field_type;
+ gint c_min = min_chans;
+ gint c_max = max_chans;
+
+ s = gst_caps_get_structure (in_caps, i);
+ /* the template caps might limit the number of channels (like alsasrc),
+ * in which case we don't want to return a superset, so hack around this
+ * for the two common cases where the channels are either a fixed number
+ * or a min/max range). Example: alsasrc template has channels = [1,2] and
+ * the detection will claim to support 8 channels for device 'plughw:0' */
+ field_type = gst_structure_get_field_type (s, "channels");
+ if (field_type == G_TYPE_INT) {
+ gst_structure_get_int (s, "channels", &c_min);
+ gst_structure_get_int (s, "channels", &c_max);
+ } else if (field_type == GST_TYPE_INT_RANGE) {
+ const GValue *val;
+
+ val = gst_structure_get_value (s, "channels");
+ c_min = CLAMP (gst_value_get_int_range_min (val), min_chans, max_chans);
+ c_max = CLAMP (gst_value_get_int_range_max (val), min_chans, max_chans);
+ } else {
+ c_min = min_chans;
+ c_max = max_chans;
+ }
+
+ caps_add_channel_configuration (caps, s, c_min, c_max);
+ }
+
+ gst_caps_unref (in_caps);
+
+ return caps;
+
+ /* ERRORS */
+min_chan_error:
+ {
+ GST_ERROR_OBJECT (obj, "failed to query minimum channel count: %s",
+ snd_strerror (err));
+ return NULL;
+ }
+max_chan_error:
+ {
+ GST_ERROR_OBJECT (obj, "failed to query maximum channel count: %s",
+ snd_strerror (err));
+ return NULL;
+ }
+}
+
+#ifndef GST_CHECK_VERSION
+#define GST_CHECK_VERSION(major,minor,micro) \
+ (GST_VERSION_MAJOR > (major) || \
+ (GST_VERSION_MAJOR == (major) && GST_VERSION_MINOR > (minor)) || \
+ (GST_VERSION_MAJOR == (major) && GST_VERSION_MINOR == (minor) && GST_VERSION_MICRO >= (micro)))
+#endif
+
+#if GST_CHECK_VERSION(0, 10, 18)
+snd_pcm_t *
+gst_alsa_open_iec958_pcm (GstObject * obj)
+{
+ char *iec958_pcm_name = NULL;
+ snd_pcm_t *pcm = NULL;
+ int res;
+ char devstr[256]; /* Storage for local 'default' device string */
+
+ /*
+ * Try and open our default iec958 device. Fall back to searching on card x
+ * if this fails, which should only happen on older alsa setups
+ */
+
+ /* The string will be one of these:
+ * SPDIF_CON: Non-audio flag not set:
+ * spdif:{AES0 0x0 AES1 0x82 AES2 0x0 AES3 0x2}
+ * SPDIF_CON: Non-audio flag set:
+ * spdif:{AES0 0x2 AES1 0x82 AES2 0x0 AES3 0x2}
+ */
+ sprintf (devstr,
+ "iec958:{AES0 0x%02x AES1 0x%02x AES2 0x%02x AES3 0x%02x}",
+ IEC958_AES0_CON_EMPHASIS_NONE | IEC958_AES0_NONAUDIO,
+ IEC958_AES1_CON_ORIGINAL | IEC958_AES1_CON_PCM_CODER,
+ 0, IEC958_AES3_CON_FS_48000);
+
+ GST_DEBUG_OBJECT (obj, "Generated device string \"%s\"", devstr);
+ iec958_pcm_name = devstr;
+
+ res = snd_pcm_open (&pcm, iec958_pcm_name, SND_PCM_STREAM_PLAYBACK, 0);
+ if (G_UNLIKELY (res < 0)) {
+ GST_DEBUG_OBJECT (obj, "failed opening IEC958 device: %s",
+ snd_strerror (res));
+ pcm = NULL;
+ }
+
+ return pcm;
+}
+#endif
+
+
+/*
+ * gst_alsa_probe_supported_formats:
+ *
+ * Takes the template caps and returns the subset which is actually
+ * supported by this device.
+ *
+ */
+
+GstCaps *
+gst_alsa_probe_supported_formats (GstObject * obj, snd_pcm_t * handle,
+ const GstCaps * template_caps)
+{
+ snd_pcm_hw_params_t *hw_params;
+ snd_pcm_stream_t stream_type;
+ GstCaps *caps;
+ gint err;
+
+ snd_pcm_hw_params_malloc (&hw_params);
+ if ((err = snd_pcm_hw_params_any (handle, hw_params)) < 0)
+ goto error;
+
+ stream_type = snd_pcm_stream (handle);
+
+ caps = gst_caps_copy (template_caps);
+
+ if (!(caps = gst_alsa_detect_formats (obj, hw_params, caps)))
+ goto subroutine_error;
+
+ if (!(caps = gst_alsa_detect_rates (obj, hw_params, caps)))
+ goto subroutine_error;
+
+ if (!(caps = gst_alsa_detect_channels (obj, hw_params, caps)))
+ goto subroutine_error;
+
+#if GST_CHECK_VERSION(0, 10, 18)
+ /* Try opening IEC958 device to see if we can support that format (playback
+ * only for now but we could add SPDIF capture later) */
+ if (stream_type == SND_PCM_STREAM_PLAYBACK) {
+ snd_pcm_t *pcm = gst_alsa_open_iec958_pcm (obj);
+
+ if (G_LIKELY (pcm)) {
+ gst_caps_append (caps, gst_caps_new_simple ("audio/x-iec958", NULL));
+ snd_pcm_close (pcm);
+ }
+ }
+#endif
+
+ snd_pcm_hw_params_free (hw_params);
+ return caps;
+
+ /* ERRORS */
+error:
+ {
+ GST_ERROR_OBJECT (obj, "failed to query formats: %s", snd_strerror (err));
+ snd_pcm_hw_params_free (hw_params);
+ return NULL;
+ }
+subroutine_error:
+ {
+ GST_ERROR_OBJECT (obj, "failed to query formats");
+ snd_pcm_hw_params_free (hw_params);
+ return NULL;
+ }
+}
+
+static gchar *
+gst_alsa_find_device_name_no_handle (GstObject * obj, const gchar * devcard,
+ gint device_num, snd_pcm_stream_t stream)
+{
+ snd_ctl_card_info_t *info = NULL;
+ snd_ctl_t *ctl = NULL;
+ gchar *ret = NULL;
+ gint dev = -1;
+
+ GST_LOG_OBJECT (obj, "[%s] device=%d", devcard, device_num);
+
+ if (snd_ctl_open (&ctl, devcard, 0) < 0)
+ return NULL;
+
+ snd_ctl_card_info_malloc (&info);
+ if (snd_ctl_card_info (ctl, info) < 0)
+ goto done;
+
+ while (snd_ctl_pcm_next_device (ctl, &dev) == 0 && dev >= 0) {
+ if (dev == device_num) {
+ snd_pcm_info_t *pcminfo;
+
+ snd_pcm_info_malloc (&pcminfo);
+ snd_pcm_info_set_device (pcminfo, dev);
+ snd_pcm_info_set_subdevice (pcminfo, 0);
+ snd_pcm_info_set_stream (pcminfo, stream);
+ if (snd_ctl_pcm_info (ctl, pcminfo) < 0) {
+ snd_pcm_info_free (pcminfo);
+ break;
+ }
+
+ ret = g_strdup (snd_pcm_info_get_name (pcminfo));
+ snd_pcm_info_free (pcminfo);
+ GST_LOG_OBJECT (obj, "name from pcminfo: %s", GST_STR_NULL (ret));
+ }
+ }
+
+ if (ret == NULL) {
+ char *name = NULL;
+ gint card;
+
+ GST_LOG_OBJECT (obj, "no luck so far, trying backup");
+ card = snd_ctl_card_info_get_card (info);
+ snd_card_get_name (card, &name);
+ ret = g_strdup (name);
+ free (name);
+ }
+
+done:
+ snd_ctl_card_info_free (info);
+ snd_ctl_close (ctl);
+
+ return ret;
+}
+
+gchar *
+gst_alsa_find_device_name (GstObject * obj, const gchar * device,
+ snd_pcm_t * handle, snd_pcm_stream_t stream)
+{
+ gchar *ret = NULL;
+
+ if (device != NULL) {
+ gchar *dev, *comma;
+ gint devnum;
+
+ GST_LOG_OBJECT (obj, "Trying to get device name from string '%s'", device);
+
+ /* only want name:card bit, but not devices and subdevices */
+ dev = g_strdup (device);
+ if ((comma = strchr (dev, ','))) {
+ *comma = '\0';
+ devnum = atoi (comma + 1);
+ ret = gst_alsa_find_device_name_no_handle (obj, dev, devnum, stream);
+ }
+ g_free (dev);
+ }
+
+ if (ret == NULL && handle != NULL) {
+ snd_pcm_info_t *info;
+
+ GST_LOG_OBJECT (obj, "Trying to get device name from open handle");
+ snd_pcm_info_malloc (&info);
+ snd_pcm_info (handle, info);
+ ret = g_strdup (snd_pcm_info_get_name (info));
+ snd_pcm_info_free (info);
+ }
+
+ GST_LOG_OBJECT (obj, "Device name for device '%s': %s",
+ GST_STR_NULL (device), GST_STR_NULL (ret));
+
+ return ret;
+}
+
+/* elementfactory information */
+static const GstElementDetails gst_alsasink2_details =
+GST_ELEMENT_DETAILS ("Audio sink (ALSA)",
+ "Sink/Audio",
+ "Output to a sound card via ALSA",
+ "Wim Taymans <wim@fluendo.com>");
+
+#define DEFAULT_DEVICE "default"
+#define DEFAULT_DEVICE_NAME ""
+#define SPDIF_PERIOD_SIZE 1536
+#define SPDIF_BUFFER_SIZE 15360
+
+enum
+{
+ PROP_0,
+ PROP_DEVICE,
+ PROP_DEVICE_NAME
+};
+
+static void gst_alsasink2_init_interfaces (GType type);
+
+GST_BOILERPLATE_FULL (_k_GstAlsaSink, gst_alsasink2, GstAudioSink,
+ GST_TYPE_AUDIO_SINK, gst_alsasink2_init_interfaces);
+
+static void gst_alsasink2_finalise (GObject * object);
+static void gst_alsasink2_set_property (GObject * object,
+ guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_alsasink2_get_property (GObject * object,
+ guint prop_id, GValue * value, GParamSpec * pspec);
+
+static GstCaps *gst_alsasink2_getcaps (GstBaseSink * bsink);
+
+static gboolean gst_alsasink2_open (GstAudioSink * asink);
+static gboolean gst_alsasink2_prepare (GstAudioSink * asink,
+ GstRingBufferSpec * spec);
+static gboolean gst_alsasink2_unprepare (GstAudioSink * asink);
+static gboolean gst_alsasink2_close (GstAudioSink * asink);
+static guint gst_alsasink2_write (GstAudioSink * asink, gpointer data,
+ guint length);
+static guint gst_alsasink2_delay (GstAudioSink * asink);
+static void gst_alsasink2_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_SINK2_FACTORY_ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN"
+#else
+# define ALSA_SINK2_FACTORY_ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN"
+#endif
+
+static GstStaticPadTemplate alsasink2_sink_factory =
+ GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw-int, "
+ "endianness = (int) { " ALSA_SINK2_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_SINK2_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_SINK2_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_SINK2_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 ];"
+ "audio/x-iec958")
+ );
+
+static void
+gst_alsasink2_finalise (GObject * object)
+{
+ _k_GstAlsaSink *sink = GST_ALSA_SINK2 (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_alsasink2_init_interfaces (GType type)
+{
+ gst_alsa_type_add_device_property_probe_interface (type);
+}
+
+static void
+gst_alsasink2_base_init (gpointer g_class)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+
+ gst_element_class_set_details (element_class, &gst_alsasink2_details);
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&alsasink2_sink_factory));
+}
+static void
+gst_alsasink2_class_init (_k_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_alsasink2_finalise);
+ gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_alsasink2_get_property);
+ gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_alsasink2_set_property);
+
+ gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasink2_getcaps);
+
+ gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_alsasink2_open);
+ gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasink2_prepare);
+ gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasink2_unprepare);
+ gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_alsasink2_close);
+ gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_alsasink2_write);
+ gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_alsasink2_delay);
+ gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_alsasink2_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_alsasink2_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ _k_GstAlsaSink *sink;
+
+ sink = GST_ALSA_SINK2 (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_alsasink2_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ _k_GstAlsaSink *sink;
+
+ sink = GST_ALSA_SINK2 (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_alsasink2_init (_k_GstAlsaSink * alsasink2, _k_GstAlsaSinkClass * g_class)
+{
+ GST_DEBUG_OBJECT (alsasink2, "initializing alsasink2");
+
+ alsasink2->device = g_strdup (DEFAULT_DEVICE);
+ alsasink2->handle = NULL;
+ alsasink2->cached_caps = NULL;
+ alsasink2->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_alsasink2_getcaps (GstBaseSink * bsink)
+{
+ GstElementClass *element_class;
+ GstPadTemplate *pad_template;
+ _k_GstAlsaSink *sink = GST_ALSA_SINK2 (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 (_k_GstAlsaSink * alsa)
+{
+ guint rrate;
+ gint err, dir;
+ snd_pcm_hw_params_t *params;
+ guint period_time, buffer_time;
+
+ snd_pcm_hw_params_malloc (¶ms);
+
+ GST_DEBUG_OBJECT (alsa, "Negotiating to %d channels @ %d Hz (format = %s) "
+ "SPDIF (%d)", alsa->channels, alsa->rate,
+ snd_pcm_format_name (alsa->format), alsa->iec958);
+
+ /* 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 */
+#if GST_CHECK_VERSION(0, 10, 18)
+ if (alsa->iec958) {
+ /* Try to use big endian first else fallback to le and swap bytes */
+ if (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format) < 0) {
+ alsa->format = SND_PCM_FORMAT_S16_LE;
+ alsa->need_swap = TRUE;
+ GST_DEBUG_OBJECT (alsa, "falling back to little endian with swapping");
+ } else {
+ alsa->need_swap = FALSE;
+ }
+ }
+#endif
+ 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;
+
+ /* 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 != ~0u && !alsa->iec958) {
+ /* 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;
+ }
+ GST_DEBUG_OBJECT (alsa, "buffer time %u", buffer_time);
+ }
+ if (period_time != ~0u && !alsa->iec958) {
+ /* 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;
+ }
+ GST_DEBUG_OBJECT (alsa, "period time %u", period_time);
+ }
+
+ /* Set buffer size and period size manually for SPDIF */
+ if (G_UNLIKELY (alsa->iec958)) {
+ snd_pcm_uframes_t buffer_size = SPDIF_BUFFER_SIZE;
+ snd_pcm_uframes_t period_size = SPDIF_PERIOD_SIZE;
+
+ CHECK (snd_pcm_hw_params_set_buffer_size_near (alsa->handle, params,
+ &buffer_size), buffer_size);
+ CHECK (snd_pcm_hw_params_set_period_size_near (alsa->handle, params,
+ &period_size, NULL), period_size);
+ }
+
+ /* 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);
+
+ snd_pcm_hw_params_free (params);
+ return 0;
+
+ /* ERRORS */
+no_config:
+ {
+ GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
+ ("Broken configuration for playback: 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 playback: %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 playback: %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 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);
+ snd_pcm_hw_params_free (params);
+ 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));
+ snd_pcm_hw_params_free (params);
+ return -EINVAL;
+ }
+buffer_size:
+ {
+ GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
+ ("Unable to get buffer size for playback: %s", 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 playback: %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 playback: %s", snd_strerror (err)));
+ snd_pcm_hw_params_free (params);
+ return err;
+ }
+}
+
+static int
+set_swparams (_k_GstAlsaSink * 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);
+ /* 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);
+
+#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 playback 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
+alsasink2_parse_spec (_k_GstAlsaSink * alsa, GstRingBufferSpec * spec)
+{
+ /* Initialize our boolean */
+ alsa->iec958 = FALSE;
+
+ 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;
+#if GST_CHECK_VERSION(0, 10, 18)
+ case GST_BUFTYPE_IEC958:
+ alsa->format = SND_PCM_FORMAT_S16_BE;
+ alsa->iec958 = TRUE;
+ break;
+#endif
+ 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_alsasink2_open (GstAudioSink * asink)
+{
+ _k_GstAlsaSink *alsa;
+ gint err;
+
+ alsa = GST_ALSA_SINK2 (asink);
+
+ CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_PLAYBACK,
+ SND_PCM_NONBLOCK), open_error);
+ GST_LOG_OBJECT (alsa, "Opened device %s", alsa->device);
+
+ return TRUE;
+
+ /* ERRORS */
+open_error:
+ {
+ if (err == -EBUSY) {
+ GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY,
+ (_("Could not open audio device for playback. "
+ "Device is being used by another application.")),
+ ("Device '%s' is busy", alsa->device));
+ } else {
+ GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_WRITE,
+ (_("Could not open audio device for playback.")),
+ ("Playback open error on device '%s': %s", alsa->device,
+ snd_strerror (err)));
+ }
+ return FALSE;
+ }
+}
+
+static gboolean
+gst_alsasink2_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
+{
+ _k_GstAlsaSink *alsa;
+ gint err;
+
+ alsa = GST_ALSA_SINK2 (asink);
+
+#if GST_CHECK_VERSION(0, 10, 18)
+ if (spec->format == GST_IEC958) {
+ snd_pcm_close (alsa->handle);
+ alsa->handle = gst_alsa_open_iec958_pcm (GST_OBJECT (alsa));
+ if (G_UNLIKELY (!alsa->handle)) {
+ goto no_iec958;
+ }
+ }
+#endif
+
+ if (!alsasink2_parse_spec (alsa, spec))
+ goto spec_parse;
+
+ 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;
+
+ {
+ snd_output_t *out_buf = NULL;
+ char *msg = NULL;
+
+ snd_output_buffer_open (&out_buf);
+ snd_pcm_dump_hw_setup (alsa->handle, out_buf);
+ snd_output_buffer_string (out_buf, &msg);
+ GST_DEBUG_OBJECT (alsa, "Hardware setup: \n%s", msg);
+ snd_output_close (out_buf);
+ snd_output_buffer_open (&out_buf);
+ snd_pcm_dump_sw_setup (alsa->handle, out_buf);
+ snd_output_buffer_string (out_buf, &msg);
+ GST_DEBUG_OBJECT (alsa, "Software setup: \n%s", msg);
+ snd_output_close (out_buf);
+ }
+
+ return TRUE;
+
+ /* ERRORS */
+#if GST_CHECK_VERSION(0, 10, 18)
+no_iec958:
+ {
+ GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_WRITE, (NULL),
+ ("Could not open IEC958 (SPDIF) device for playback"));
+ return FALSE;
+ }
+#endif
+spec_parse:
+ {
+ GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
+ ("Error parsing spec"));
+ 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_alsasink2_unprepare (GstAudioSink * asink)
+{
+ _k_GstAlsaSink *alsa;
+ gint err;
+
+ alsa = GST_ALSA_SINK2 (asink);
+
+ CHECK (snd_pcm_drop (alsa->handle), drop);
+
+ CHECK (snd_pcm_hw_free (alsa->handle), hw_free);
+
+ 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;
+ }
+}
+
+static gboolean
+gst_alsasink2_close (GstAudioSink * asink)
+{
+ _k_GstAlsaSink *alsa = GST_ALSA_SINK2 (asink);
+ gint err;
+
+ if (alsa->handle) {
+ 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 (_k_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_alsasink2_write (GstAudioSink * asink, gpointer data, guint length)
+{
+ _k_GstAlsaSink *alsa;
+ gint err;
+ gint cptr;
+ gint16 *ptr = data;
+
+ alsa = GST_ALSA_SINK2 (asink);
+
+ if (alsa->iec958 && alsa->need_swap) {
+ guint i;
+
+ GST_DEBUG_OBJECT (asink, "swapping bytes");
+ for (i = 0; i < length / 2; i++) {
+ ptr[i] = GUINT16_SWAP_LE_BE (ptr[i]);
+ }
+ }
+
+ GST_LOG_OBJECT (asink, "received audio samples buffer of %u bytes", length);
+
+ cptr = length / alsa->bytes_per_sample;
+
+ GST_ALSA_SINK2_LOCK (asink);
+ while (cptr > 0) {
+ /* start by doing a blocking wait for free space. Set the timeout
+ * to 4 times the period time */
+ err = snd_pcm_wait (alsa->handle, (4 * alsa->period_time / 1000));
+ if (err < 0) {
+ GST_DEBUG_OBJECT (asink, "wait timeout, %d", err);
+ } else {
+ err = snd_pcm_writei (alsa->handle, ptr, cptr);
+ }
+
+ GST_DEBUG_OBJECT (asink, "written %d frames out of %d", err, cptr);
+ 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 += snd_pcm_frames_to_bytes (alsa->handle, err);
+ cptr -= err;
+ }
+ GST_ALSA_SINK2_UNLOCK (asink);
+
+ return length - (cptr * alsa->bytes_per_sample);
+
+write_error:
+ {
+ GST_ALSA_SINK2_UNLOCK (asink);
+ return length; /* skip one period */
+ }
+}
+
+static guint
+gst_alsasink2_delay (GstAudioSink * asink)
+{
+ _k_GstAlsaSink *alsa;
+ snd_pcm_sframes_t delay;
+ int res;
+
+ alsa = GST_ALSA_SINK2 (asink);
+
+ res = snd_pcm_delay (alsa->handle, &delay);
+ if (G_UNLIKELY (res < 0)) {
+ /* on errors, report 0 delay */
+ GST_DEBUG_OBJECT (alsa, "snd_pcm_delay returned %d", res);
+ delay = 0;
+ }
+ if (G_UNLIKELY (delay < 0)) {
+ /* make sure we never return a negative delay */
+ GST_WARNING_OBJECT (alsa, "snd_pcm_delay returned negative delay");
+ delay = 0;
+ }
+
+ return delay;
+}
+
+static void
+gst_alsasink2_reset (GstAudioSink * asink)
+{
+ _k_GstAlsaSink *alsa;
+ gint err;
+
+ alsa = GST_ALSA_SINK2 (asink);
+
+ GST_ALSA_SINK2_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_SINK2_UNLOCK (asink);
+
+ return;
+
+ /* ERRORS */
+drop_error:
+ {
+ GST_ERROR_OBJECT (alsa, "alsa-reset: pcm drop error: %s",
+ snd_strerror (err));
+ GST_ALSA_SINK2_UNLOCK (asink);
+ return;
+ }
+prepare_error:
+ {
+ GST_ERROR_OBJECT (alsa, "alsa-reset: pcm prepare error: %s",
+ snd_strerror (err));
+ GST_ALSA_SINK2_UNLOCK (asink);
+ return;
+ }
+}
+
+static void
+gst_alsa_error_wrapper (const char *file, int line, const char *function,
+ int err, const char *fmt, ...)
+{
+}
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+ int err;
+
+ if (!gst_element_register (plugin, "_k_alsasink", GST_RANK_PRIMARY,
+ GST_TYPE_ALSA_SINK2))
+ return FALSE;
+
+ err = snd_lib_error_set_handler (gst_alsa_error_wrapper);
+ if (err != 0)
+ GST_WARNING ("failed to set alsa error handler");
+
+ return TRUE;
+}
+
+#define PACKAGE ""
+GST_PLUGIN_DEFINE_STATIC (GST_VERSION_MAJOR,
+ GST_VERSION_MINOR,
+ "_k_alsa",
+ "ALSA plugin library (hotfixed)",
+ plugin_init, "0.1", "LGPL", "Phonon-GStreamer", "")
+#undef PACKAGE