/* 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