diff options
Diffstat (limited to 'src/3rdparty/phonon/gstreamer/alsasink2.c')
| -rw-r--r-- | src/3rdparty/phonon/gstreamer/alsasink2.c | 1756 |
1 files changed, 1756 insertions, 0 deletions
diff --git a/src/3rdparty/phonon/gstreamer/alsasink2.c b/src/3rdparty/phonon/gstreamer/alsasink2.c new file mode 100644 index 0000000..4dcb140 --- /dev/null +++ b/src/3rdparty/phonon/gstreamer/alsasink2.c @@ -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 |