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List: kde-multimedia
Subject: =?utf-8?q?=5Bphonon-gstreamer/plumbing=5D_gstreamer=3A_Remove_ou?=
From: Trever Fischer <tdfischer () fedoraproject ! org>
Date: 2011-06-05 9:18:28
Message-ID: 20110605091828.7C7F5A60BE () git ! kde ! org
[Download RAW message or body]
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 <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
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 <wim@fluendo.com>
- * Copyright (C) 2008 Matthias Kretz <kretz@kde.org>
- *
- * 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 <gst/audio/gstaudiosink.h>
-#include <alsa/asoundlib.h>
-
-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 <phonon/pulsesupport.h>
-#ifdef USE_ALSASINK2
-#include "alsasink2.h"
-#endif
-
#include <QtCore/QSettings>
/*
@@ -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))
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