From kde-multimedia Sun Jun 05 09:18:28 2011 From: Trever Fischer Date: Sun, 05 Jun 2011 09:18:28 +0000 To: kde-multimedia Subject: =?utf-8?q?=5Bphonon-gstreamer/plumbing=5D_gstreamer=3A_Remove_ou?= Message-Id: <20110605091828.7C7F5A60BE () git ! kde ! org> X-MARC-Message: https://marc.info/?l=kde-multimedia&m=130779695126283 Git commit fbfca0c202b775fd9c00b41ac6ee3f80278cfbb7 by Trever Fischer. Committed on 01/06/2011 at 03:37. Pushed by tdfischer into branch 'plumbing'. Remove our pet copy of alsasink2. CCMAIL: kde-multimedia@kde.org M +0 -9 gstreamer/CMakeLists.txt D +0 -1756 gstreamer/alsasink2.c D +0 -86 gstreamer/alsasink2.h M +1 -1 gstreamer/audiooutput.cpp M +0 -16 gstreamer/devicemanager.cpp http://commits.kde.org/phonon-gstreamer/fbfca0c202b775fd9c00b41ac6ee3f80278cfbb7 diff --git a/gstreamer/CMakeLists.txt b/gstreamer/CMakeLists.txt index fc66e0b..6f49a19 100644 --- a/gstreamer/CMakeLists.txt +++ b/gstreamer/CMakeLists.txt @@ -73,16 +73,7 @@ if (BUILD_PHONON_GSTREAMER) ${phonon_gstreamer_SRCS} x11renderer.cpp) macro_optional_find_package(Alsa) - macro_ensure_version("0.10.22" ${GSTREAMER_VERSION} GSTREAMER_HAS_NONBLOCKING_ALSASINK) endif(NOT WIN32) - if(ALSA_FOUND AND NOT GSTREAMER_HAS_NONBLOCKING_ALSASINK) - add_definitions(-DUSE_ALSASINK2) - include_directories(${ALSA_INCLUDES}) - set(phonon_gstreamer_SRCS - ${phonon_gstreamer_SRCS} - alsasink2.c - ) - endif(ALSA_FOUND AND NOT GSTREAMER_HAS_NONBLOCKING_ALSASINK) automoc4_add_library(phonon_gstreamer MODULE ${phonon_gstreamer_SRCS}) set_target_properties(phonon_gstreamer PROPERTIES PREFIX "") diff --git a/gstreamer/alsasink2.c b/gstreamer/alsasink2.c deleted file mode 100644 index 4dcb140..0000000 --- a/gstreamer/alsasink2.c +++ /dev/null @@ -1,1756 +0,0 @@ -/* GStreamer - * Copyright (C) 2001 CodeFactory AB - * Copyright (C) 2001 Thomas Nyberg - * Copyright (C) 2001-2002 Andy Wingo - * Copyright (C) 2003 Benjamin Otte - * Copyright (C) 2005 Wim Taymans - * Copyright (C) 2005, 2006 Tim-Philipp Müller - * Copyright (C) 2008 Matthias Kretz - * - * 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 . - */ - -/** - * SECTION:element-alsasink2 - * @short_description: play audio to an ALSA device - * @see_also: alsasrc, alsamixer - * - * - * - * This element renders raw audio samples using the ALSA api. - * - * Example pipelines - * - * Play an Ogg/Vorbis file. - * - * - * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! alsasink2 - * - * - * - * Last reviewed on 2006-03-01 (0.10.4) - */ - -#define _XOPEN_SOURCE 600 - -#include -#include -#include -#include -#include -#include -#include - -#include "alsasink2.h" - -#include -#include - -#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 "); - -#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 diff --git a/gstreamer/alsasink2.h b/gstreamer/alsasink2.h deleted file mode 100644 index f9c73d6..0000000 --- a/gstreamer/alsasink2.h +++ /dev/null @@ -1,86 +0,0 @@ -/* GStreamer - * Copyright (C) 2005 Wim Taymans - * Copyright (C) 2008 Matthias Kretz - * - * gstalsasink2.h: - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Library General Public - * License as published by the Free Software Foundation; either - * version 2 of the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Library General Public License for more details. - * - * You should have received a copy of the GNU Library General Public - * License along with this library; if not, write to the - * Free Software Foundation, Inc., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - - -#ifndef ALSASINK2_H -#define ALSASINK2_H - -#include -#include - -G_BEGIN_DECLS - -#define GST_TYPE_ALSA_SINK2 (gst_alsasink2_get_type()) -#define GST_ALSA_SINK2(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_ALSA_SINK2,_k_GstAlsaSink)) -#define GST_ALSA_SINK2_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_ALSA_SINK2,_k_GstAlsaSinkClass)) -#define GST_IS_ALSA_SINK2(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_ALSA_SINK2)) -#define GST_IS_ALSA_SINK2_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_ALSA_SINK2)) -#define GST_ALSA_SINK2_CAST(obj) ((_k_GstAlsaSink *) (obj)) - -typedef struct _k_GstAlsaSink _k_GstAlsaSink; -typedef struct _k_GstAlsaSinkClass _k_GstAlsaSinkClass; - -#define GST_ALSA_SINK2_GET_LOCK(obj) (GST_ALSA_SINK2_CAST (obj)->alsa_lock) -#define GST_ALSA_SINK2_LOCK(obj) (g_mutex_lock (GST_ALSA_SINK2_GET_LOCK (obj))) -#define GST_ALSA_SINK2_UNLOCK(obj) (g_mutex_unlock (GST_ALSA_SINK2_GET_LOCK (obj))) - -/** - * _k_GstAlsaSink: - * - * Opaque data structure - */ -struct _k_GstAlsaSink { - GstAudioSink sink; - - gchar *device; - - snd_pcm_t *handle; - snd_pcm_hw_params_t *hwparams; - snd_pcm_sw_params_t *swparams; - - snd_pcm_access_t access; - snd_pcm_format_t format; - guint rate; - guint channels; - gint bytes_per_sample; - gboolean iec958; - gboolean need_swap; - - guint buffer_time; - guint period_time; - snd_pcm_uframes_t buffer_size; - snd_pcm_uframes_t period_size; - - GstCaps *cached_caps; - - GMutex *alsa_lock; -}; - -struct _k_GstAlsaSinkClass { - GstAudioSinkClass parent_class; -}; - -GType gst_alsasink2_get_type(void); - -G_END_DECLS - -#endif /* ALSASINK2_H */ diff --git a/gstreamer/audiooutput.cpp b/gstreamer/audiooutput.cpp index 91b91ba..260ef7e 100644 --- a/gstreamer/audiooutput.cpp +++ b/gstreamer/audiooutput.cpp @@ -203,7 +203,7 @@ bool AudioOutput::setOutputDevice(const AudioOutputDevice &newDevice) const QByteArray oldDeviceValue = GstHelper::property(m_audioSink, "device"); const QByteArray sinkName = GstHelper::property(m_audioSink, "name"); - if (sinkName == "alsasink" || sinkName == "alsasink2") { + if (sinkName == "alsasink") { if (driver.toByteArray() != "alsa") { return false; } diff --git a/gstreamer/devicemanager.cpp b/gstreamer/devicemanager.cpp index 76c56d7..e41dfdb 100644 --- a/gstreamer/devicemanager.cpp +++ b/gstreamer/devicemanager.cpp @@ -27,10 +27,6 @@ #include "x11renderer.h" #include -#ifdef USE_ALSASINK2 -#include "alsasink2.h" -#endif - #include /* @@ -224,18 +220,6 @@ GstElement *DeviceManager::createAudioSink(Category category) } } -#ifdef USE_ALSASINK2 - if (!sink) { - sink = gst_element_factory_make ("_k_alsasink", NULL); - if (canOpenDevice(sink)) - m_backend->logMessage("AudioOutput using alsa2 audio sink"); - else if (sink) { - gst_object_unref(sink); - sink = 0; - } - } -#endif - if (!sink) { sink = gst_element_factory_make ("alsasink", NULL); if (canOpenDevice(sink)) _______________________________________________ kde-multimedia mailing list kde-multimedia@kde.org https://mail.kde.org/mailman/listinfo/kde-multimedia