gstreamer/subprojects/gst-plugins-base/ext/alsa/gstalsasrc.c

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/* GStreamer
* Copyright (C) 2005 Wim Taymans <wim@fluendo.com>
*
* gstalsasrc.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 Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-alsasrc
* @title: alsasrc
* @see_also: alsasink
*
* This element reads data from an audio card using the ALSA API.
Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe... Original commit message from CVS: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-overrides.txt: * docs/plugins/gst-plugins-base-plugins-sections.txt: * docs/plugins/gst-plugins-base-plugins.args: * docs/plugins/gst-plugins-base-plugins.hierarchy: * docs/plugins/gst-plugins-base-plugins.interfaces: * docs/plugins/gst-plugins-base-plugins.prerequisites: * docs/plugins/gst-plugins-base-plugins.signals: * docs/plugins/inspect/plugin-adder.xml: * docs/plugins/inspect/plugin-alsa.xml: * docs/plugins/inspect/plugin-audioconvert.xml: * docs/plugins/inspect/plugin-audiorate.xml: * docs/plugins/inspect/plugin-audioresample.xml: * docs/plugins/inspect/plugin-audiotestsrc.xml: * docs/plugins/inspect/plugin-cdparanoia.xml: * docs/plugins/inspect/plugin-decodebin.xml: * docs/plugins/inspect/plugin-ffmpegcolorspace.xml: * docs/plugins/inspect/plugin-gdp.xml: * docs/plugins/inspect/plugin-gnomevfs.xml: * docs/plugins/inspect/plugin-libvisual.xml: * docs/plugins/inspect/plugin-ogg.xml: * docs/plugins/inspect/plugin-pango.xml: * docs/plugins/inspect/plugin-playback.xml: * docs/plugins/inspect/plugin-queue2.xml: * docs/plugins/inspect/plugin-subparse.xml: * docs/plugins/inspect/plugin-tcp.xml: * docs/plugins/inspect/plugin-theora.xml: * docs/plugins/inspect/plugin-typefindfunctions.xml: * docs/plugins/inspect/plugin-uridecodebin.xml: * docs/plugins/inspect/plugin-video4linux.xml: * docs/plugins/inspect/plugin-videorate.xml: * docs/plugins/inspect/plugin-videoscale.xml: * docs/plugins/inspect/plugin-videotestsrc.xml: * docs/plugins/inspect/plugin-volume.xml: * docs/plugins/inspect/plugin-vorbis.xml: * docs/plugins/inspect/plugin-ximagesink.xml: * docs/plugins/inspect/plugin-xvimagesink.xml: * ext/alsa/gstalsamixer.c: * ext/alsa/gstalsasink.c: * ext/alsa/gstalsasrc.c: * ext/gio/gstgiosink.c: * ext/gio/gstgiosrc.c: * ext/gio/gstgiostreamsink.c: * ext/gio/gstgiostreamsrc.c: * ext/gnomevfs/gstgnomevfssink.c: * ext/gnomevfs/gstgnomevfssrc.c: * ext/ogg/gstoggdemux.c: * ext/ogg/gstoggmux.c: * ext/pango/gstclockoverlay.c: * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/pango/gsttimeoverlay.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/theora/theoraparse.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * ext/vorbis/vorbisparse.c: * ext/vorbis/vorbistag.c: * gst/adder/gstadder.c: * gst/audioconvert/gstaudioconvert.c: * gst/audioresample/gstaudioresample.c: * gst/audiotestsrc/gstaudiotestsrc.c: * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/gdp/gstgdpdepay.c: * gst/gdp/gstgdppay.c: * gst/playback/gstdecodebin2.c: * gst/playback/gstplaybin.c: * gst/playback/gstplaybin2.c: * gst/playback/gstqueue2.c: * gst/playback/gsturidecodebin.c: * gst/tcp/gstmultifdsink.c: * gst/tcp/gsttcpserversink.c: * gst/videorate/gstvideorate.c: * gst/videoscale/gstvideoscale.c: * gst/videotestsrc/gstvideotestsrc.c: * gst/volume/gstvolume.c: * sys/ximage/ximagesink.c: * sys/xvimage/xvimagesink.c: Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipelines are "simple" pipelines.
2008-07-10 21:06:06 +00:00
*
* ## Example pipelines
Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe... Original commit message from CVS: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-overrides.txt: * docs/plugins/gst-plugins-base-plugins-sections.txt: * docs/plugins/gst-plugins-base-plugins.args: * docs/plugins/gst-plugins-base-plugins.hierarchy: * docs/plugins/gst-plugins-base-plugins.interfaces: * docs/plugins/gst-plugins-base-plugins.prerequisites: * docs/plugins/gst-plugins-base-plugins.signals: * docs/plugins/inspect/plugin-adder.xml: * docs/plugins/inspect/plugin-alsa.xml: * docs/plugins/inspect/plugin-audioconvert.xml: * docs/plugins/inspect/plugin-audiorate.xml: * docs/plugins/inspect/plugin-audioresample.xml: * docs/plugins/inspect/plugin-audiotestsrc.xml: * docs/plugins/inspect/plugin-cdparanoia.xml: * docs/plugins/inspect/plugin-decodebin.xml: * docs/plugins/inspect/plugin-ffmpegcolorspace.xml: * docs/plugins/inspect/plugin-gdp.xml: * docs/plugins/inspect/plugin-gnomevfs.xml: * docs/plugins/inspect/plugin-libvisual.xml: * docs/plugins/inspect/plugin-ogg.xml: * docs/plugins/inspect/plugin-pango.xml: * docs/plugins/inspect/plugin-playback.xml: * docs/plugins/inspect/plugin-queue2.xml: * docs/plugins/inspect/plugin-subparse.xml: * docs/plugins/inspect/plugin-tcp.xml: * docs/plugins/inspect/plugin-theora.xml: * docs/plugins/inspect/plugin-typefindfunctions.xml: * docs/plugins/inspect/plugin-uridecodebin.xml: * docs/plugins/inspect/plugin-video4linux.xml: * docs/plugins/inspect/plugin-videorate.xml: * docs/plugins/inspect/plugin-videoscale.xml: * docs/plugins/inspect/plugin-videotestsrc.xml: * docs/plugins/inspect/plugin-volume.xml: * docs/plugins/inspect/plugin-vorbis.xml: * docs/plugins/inspect/plugin-ximagesink.xml: * docs/plugins/inspect/plugin-xvimagesink.xml: * ext/alsa/gstalsamixer.c: * ext/alsa/gstalsasink.c: * ext/alsa/gstalsasrc.c: * ext/gio/gstgiosink.c: * ext/gio/gstgiosrc.c: * ext/gio/gstgiostreamsink.c: * ext/gio/gstgiostreamsrc.c: * ext/gnomevfs/gstgnomevfssink.c: * ext/gnomevfs/gstgnomevfssrc.c: * ext/ogg/gstoggdemux.c: * ext/ogg/gstoggmux.c: * ext/pango/gstclockoverlay.c: * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/pango/gsttimeoverlay.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/theora/theoraparse.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * ext/vorbis/vorbisparse.c: * ext/vorbis/vorbistag.c: * gst/adder/gstadder.c: * gst/audioconvert/gstaudioconvert.c: * gst/audioresample/gstaudioresample.c: * gst/audiotestsrc/gstaudiotestsrc.c: * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/gdp/gstgdpdepay.c: * gst/gdp/gstgdppay.c: * gst/playback/gstdecodebin2.c: * gst/playback/gstplaybin.c: * gst/playback/gstplaybin2.c: * gst/playback/gstqueue2.c: * gst/playback/gsturidecodebin.c: * gst/tcp/gstmultifdsink.c: * gst/tcp/gsttcpserversink.c: * gst/videorate/gstvideorate.c: * gst/videoscale/gstvideoscale.c: * gst/videotestsrc/gstvideotestsrc.c: * gst/volume/gstvolume.c: * sys/ximage/ximagesink.c: * sys/xvimage/xvimagesink.c: Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipelines are "simple" pipelines.
2008-07-10 21:06:06 +00:00
* |[
* gst-launch-1.0 -v alsasrc ! queue ! audioconvert ! vorbisenc ! oggmux ! filesink location=alsasrc.ogg
* ]|
* Record from a sound card using ALSA and encode to Ogg/Vorbis.
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <sys/ioctl.h>
#include <fcntl.h>
#include <errno.h>
#include <unistd.h>
#include <string.h>
#include <getopt.h>
#include <alsa/asoundlib.h>
#include "gstalsaelements.h"
#include "gstalsasrc.h"
#include <glib/gi18n-lib.h>
#ifndef ESTRPIPE
#define ESTRPIPE EPIPE
#endif
#define DEFAULT_PROP_DEVICE "default"
#define DEFAULT_PROP_DEVICE_NAME ""
#define DEFAULT_PROP_CARD_NAME ""
#define DEFAULT_PROP_USE_DRIVER_TIMESTAMP TRUE
enum
{
PROP_0,
PROP_DEVICE,
PROP_DEVICE_NAME,
PROP_CARD_NAME,
PROP_USE_DRIVER_TIMESTAMP,
PROP_LAST
};
#define gst_alsasrc_parent_class parent_class
G_DEFINE_TYPE (GstAlsaSrc, gst_alsasrc, GST_TYPE_AUDIO_SRC);
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (alsasrc, "alsasrc", GST_RANK_PRIMARY,
GST_TYPE_ALSA_SRC, alsa_element_init (plugin));
static void gst_alsasrc_finalize (GObject * object);
static void gst_alsasrc_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_alsasrc_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static GstStateChangeReturn gst_alsasrc_change_state (GstElement * element,
GstStateChange transition);
static GstCaps *gst_alsasrc_getcaps (GstBaseSrc * bsrc, GstCaps * filter);
static gboolean gst_alsasrc_open (GstAudioSrc * asrc);
static gboolean gst_alsasrc_prepare (GstAudioSrc * asrc,
GstAudioRingBufferSpec * spec);
static gboolean gst_alsasrc_unprepare (GstAudioSrc * asrc);
static gboolean gst_alsasrc_close (GstAudioSrc * asrc);
static guint gst_alsasrc_read
(GstAudioSrc * asrc, gpointer data, guint length, GstClockTime * timestamp);
static guint gst_alsasrc_delay (GstAudioSrc * asrc);
static void gst_alsasrc_reset (GstAudioSrc * asrc);
/* AlsaSrc signals and args */
enum
{
LAST_SIGNAL
};
#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
# define ALSA_SRC_FACTORY_ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN"
#else
# define ALSA_SRC_FACTORY_ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN"
#endif
static GstStaticPadTemplate alsasrc_src_factory =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_FORMATS_ALL ", "
"layout = (string) interleaved, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
);
static void
gst_alsasrc_finalize (GObject * object)
{
GstAlsaSrc *src = GST_ALSA_SRC (object);
g_free (src->device);
2012-09-10 00:06:51 +00:00
g_mutex_clear (&src->alsa_lock);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_alsasrc_class_init (GstAlsaSrcClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSrcClass *gstbasesrc_class;
GstAudioSrcClass *gstaudiosrc_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesrc_class = (GstBaseSrcClass *) klass;
gstaudiosrc_class = (GstAudioSrcClass *) klass;
gobject_class->finalize = gst_alsasrc_finalize;
gobject_class->get_property = gst_alsasrc_get_property;
gobject_class->set_property = gst_alsasrc_set_property;
gst_element_class_set_static_metadata (gstelement_class,
"Audio source (ALSA)", "Source/Audio",
"Read from a sound card via ALSA", "Wim Taymans <wim@fluendo.com>");
gst_element_class_add_static_pad_template (gstelement_class,
&alsasrc_src_factory);
gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasrc_getcaps);
gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_alsasrc_open);
gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasrc_prepare);
gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasrc_unprepare);
gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_alsasrc_close);
gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_alsasrc_read);
gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_alsasrc_delay);
gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_alsasrc_reset);
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_alsasrc_change_state);
g_object_class_install_property (gobject_class, PROP_DEVICE,
g_param_spec_string ("device", "Device",
"ALSA device, as defined in an asound configuration file",
Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u... Original commit message from CVS: * configure.ac: * ext/alsa/gstalsamixerelement.c: (gst_alsa_mixer_element_class_init): * ext/alsa/gstalsasink.c: (gst_alsasink_class_init): * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init): * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_class_init): * ext/gio/gstgiosink.c: (gst_gio_sink_class_init): * ext/gio/gstgiosrc.c: (gst_gio_src_class_init): * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init): * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init): * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init): * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init): * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init): * ext/pango/gsttextrender.c: (gst_text_render_class_init): * ext/theora/theoradec.c: (gst_theora_dec_class_init): * ext/theora/theoraenc.c: (gst_theora_enc_class_init): * ext/theora/theoraparse.c: (gst_theora_parse_class_init): * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_class_init): * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_class_init): * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_class_init): * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_class_init): * gst-libs/gst/interfaces/mixertrack.c: (gst_mixer_track_class_init): * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_class_init): * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_class_init): * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_class_init): * gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init): * gst/audioresample/gstaudioresample.c: (gst_audioresample_class_init): * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_class_init): * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init): * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init): * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init), (preroll_unlinked): * gst/playback/gstplaybin.c: (gst_play_bin_class_init): * gst/playback/gstplaybin2.c: (gst_play_bin_class_init): * gst/playback/gstplaysink.c: (gst_play_sink_class_init): * gst/playback/gstqueue2.c: (gst_queue_class_init): * gst/playback/gststreaminfo.c: (gst_stream_info_class_init): * gst/playback/gststreamselector.c: (gst_selector_pad_class_init), (gst_stream_selector_class_init): * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init): * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init): * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init): * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init): * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init): * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init): * gst/videorate/gstvideorate.c: (gst_video_rate_class_init): * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init): * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_class_init): * gst/volume/gstvolume.c: (gst_volume_class_init): * sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init): * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init): * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init): * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init): * sys/ximage/ximagesink.c: (gst_ximagesink_class_init): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init): Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory usage, fewer allocations and thus less memory defragmentation. Depend on core CVS for this. Fixes bug #523806.
2008-03-22 15:00:53 +00:00
DEFAULT_PROP_DEVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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",
Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u... Original commit message from CVS: * configure.ac: * ext/alsa/gstalsamixerelement.c: (gst_alsa_mixer_element_class_init): * ext/alsa/gstalsasink.c: (gst_alsasink_class_init): * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init): * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_class_init): * ext/gio/gstgiosink.c: (gst_gio_sink_class_init): * ext/gio/gstgiosrc.c: (gst_gio_src_class_init): * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init): * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init): * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init): * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init): * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init): * ext/pango/gsttextrender.c: (gst_text_render_class_init): * ext/theora/theoradec.c: (gst_theora_dec_class_init): * ext/theora/theoraenc.c: (gst_theora_enc_class_init): * ext/theora/theoraparse.c: (gst_theora_parse_class_init): * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_class_init): * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_class_init): * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_class_init): * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_class_init): * gst-libs/gst/interfaces/mixertrack.c: (gst_mixer_track_class_init): * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_class_init): * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_class_init): * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_class_init): * gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init): * gst/audioresample/gstaudioresample.c: (gst_audioresample_class_init): * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_class_init): * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init): * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init): * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init), (preroll_unlinked): * gst/playback/gstplaybin.c: (gst_play_bin_class_init): * gst/playback/gstplaybin2.c: (gst_play_bin_class_init): * gst/playback/gstplaysink.c: (gst_play_sink_class_init): * gst/playback/gstqueue2.c: (gst_queue_class_init): * gst/playback/gststreaminfo.c: (gst_stream_info_class_init): * gst/playback/gststreamselector.c: (gst_selector_pad_class_init), (gst_stream_selector_class_init): * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init): * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init): * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init): * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init): * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init): * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init): * gst/videorate/gstvideorate.c: (gst_video_rate_class_init): * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init): * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_class_init): * gst/volume/gstvolume.c: (gst_volume_class_init): * sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init): * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init): * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init): * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init): * sys/ximage/ximagesink.c: (gst_ximagesink_class_init): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init): Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory usage, fewer allocations and thus less memory defragmentation. Depend on core CVS for this. Fixes bug #523806.
2008-03-22 15:00:53 +00:00
DEFAULT_PROP_DEVICE_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_CARD_NAME,
g_param_spec_string ("card-name", "Card name",
"Human-readable name of the sound card",
DEFAULT_PROP_CARD_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS
| GST_PARAM_DOC_SHOW_DEFAULT));
g_object_class_install_property (gobject_class, PROP_USE_DRIVER_TIMESTAMP,
g_param_spec_boolean ("use-driver-timestamps", "Use driver timestamps",
"Use driver timestamps or the pipeline clock timestamps",
DEFAULT_PROP_USE_DRIVER_TIMESTAMP,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
static void
gst_alsasrc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAlsaSrc *src;
src = GST_ALSA_SRC (object);
switch (prop_id) {
case PROP_DEVICE:
g_free (src->device);
src->device = g_value_dup_string (value);
if (src->device == NULL) {
src->device = g_strdup (DEFAULT_PROP_DEVICE);
}
break;
case PROP_USE_DRIVER_TIMESTAMP:
GST_OBJECT_LOCK (src);
src->use_driver_timestamps = g_value_get_boolean (value);
GST_OBJECT_UNLOCK (src);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_alsasrc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAlsaSrc *src;
src = GST_ALSA_SRC (object);
switch (prop_id) {
case PROP_DEVICE:
g_value_set_string (value, src->device);
break;
case PROP_DEVICE_NAME:
g_value_take_string (value,
gst_alsa_find_device_name (GST_OBJECT_CAST (src),
src->device, src->handle, SND_PCM_STREAM_CAPTURE));
break;
case PROP_CARD_NAME:
g_value_take_string (value,
gst_alsa_find_card_name (GST_OBJECT_CAST (src),
src->device, SND_PCM_STREAM_CAPTURE));
break;
case PROP_USE_DRIVER_TIMESTAMP:
GST_OBJECT_LOCK (src);
g_value_set_boolean (value, src->use_driver_timestamps);
GST_OBJECT_UNLOCK (src);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStateChangeReturn
gst_alsasrc_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstAlsaSrc *alsa = GST_ALSA_SRC (element);
GstClock *clk;
switch (transition) {
/* show the compiler that we care */
case GST_STATE_CHANGE_NULL_TO_READY:
case GST_STATE_CHANGE_READY_TO_PAUSED:
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
case GST_STATE_CHANGE_PAUSED_TO_READY:
case GST_STATE_CHANGE_READY_TO_NULL:
case GST_STATE_CHANGE_NULL_TO_NULL:
case GST_STATE_CHANGE_READY_TO_READY:
case GST_STATE_CHANGE_PAUSED_TO_PAUSED:
case GST_STATE_CHANGE_PLAYING_TO_PLAYING:
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
alsa->driver_timestamps = FALSE;
clk = gst_element_get_clock (element);
if (clk != NULL) {
if (G_OBJECT_TYPE (clk) == GST_TYPE_SYSTEM_CLOCK) {
gint clocktype;
g_object_get (clk, "clock-type", &clocktype, NULL);
if (clocktype == GST_CLOCK_TYPE_MONOTONIC &&
alsa->use_driver_timestamps) {
GST_INFO ("Using driver timestamps !");
alsa->driver_timestamps = TRUE;
} else {
GST_INFO ("Not using driver timestamps !");
alsa->driver_timestamps = FALSE;
}
}
gst_object_unref (clk);
}
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
return ret;
}
static void
gst_alsasrc_init (GstAlsaSrc * alsasrc)
{
GST_DEBUG_OBJECT (alsasrc, "initializing");
alsasrc->device = g_strdup (DEFAULT_PROP_DEVICE);
alsasrc->cached_caps = NULL;
alsasrc->driver_timestamps = FALSE;
alsasrc->use_driver_timestamps = DEFAULT_PROP_USE_DRIVER_TIMESTAMP;
2012-09-10 00:06:51 +00:00
g_mutex_init (&alsasrc->alsa_lock);
}
#define CHECK(call, error) \
G_STMT_START { \
if ((err = call) < 0) \
goto error; \
} G_STMT_END;
static GstCaps *
gst_alsasrc_getcaps (GstBaseSrc * bsrc, GstCaps * filter)
{
GstElementClass *element_class;
GstPadTemplate *pad_template;
GstAlsaSrc *src;
GstCaps *caps, *templ_caps;
src = GST_ALSA_SRC (bsrc);
if (src->handle == NULL) {
GST_DEBUG_OBJECT (src, "device not open, using template caps");
return GST_BASE_SRC_CLASS (parent_class)->get_caps (bsrc, filter);
}
if (src->cached_caps) {
GST_LOG_OBJECT (src, "Returning cached caps");
if (filter)
return gst_caps_intersect_full (filter, src->cached_caps,
GST_CAPS_INTERSECT_FIRST);
else
return gst_caps_ref (src->cached_caps);
}
element_class = GST_ELEMENT_GET_CLASS (src);
pad_template = gst_element_class_get_pad_template (element_class, "src");
g_return_val_if_fail (pad_template != NULL, NULL);
templ_caps = gst_pad_template_get_caps (pad_template);
GST_INFO_OBJECT (src, "template caps %" GST_PTR_FORMAT, templ_caps);
caps = gst_alsa_probe_supported_formats (GST_OBJECT (src),
src->device, src->handle, templ_caps);
gst_caps_unref (templ_caps);
if (caps) {
src->cached_caps = gst_caps_ref (caps);
}
GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, caps);
if (filter) {
GstCaps *intersection;
intersection =
gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
return intersection;
} else {
return caps;
}
}
static int
set_hwparams (GstAlsaSrc * alsa)
{
guint rrate;
ext: alsa: Fix fallback paths for setting buffer and period times Below fallback paths were introduced in https://github.com/GStreamer/gst-plugins-base/commit/9759810d8206b5f1aa199f98599caec3630a1813 if setting period time after buffer time failed : 1) Set period time and then buffer time if it doesn't work 2) Set only buffer time 3) Set only period time These all were not functioning properly since they were using old copy of snd_pcm_hw_params_t which already had some fields set as per previous try and this was causing issues as driver was referring to that old value while trying to set them again in fallback paths. So now we always use the initial copy of snd_pcm_hw_params_t for every fallback and same is also being done at https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/557c4295107dc7374c850b0bd5331dd35e8fdd0f Also we change the sequence to set period time earlier than buffer time since period bytes being the smaller unit, most of the times if underlying alsa device has a dependency then it is of period bytes to be a multiple of some value (as per underlying DMA constraint) and rest of the parameters like buffer bytes need to be adjusted as per period bytes. The same sequence is also followed in alsa-utils at https://github.com/alsa-project/alsa-utils/commit/9b621eeac4d55c4e881f093be5b163ca07d01b63 Fix 2) and 3) scenarios by returning success if the exclusive setting is passed and not doing any further setting for buffer time or period time. Add new fallback path of not setting any buffer time and period time if all above fallback paths fail. The same is also being followed at aforementioned pulseaudio commit. In case of alsasink, remove the retry goto label, since it is not required anymore as fallback paths take care of setting default values if driver is not accepting any of the fallback paths. Use separate label for exit to free params structs and return err code. This also fixes leak in no_rate goto path in alsasink Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1212>
2021-06-10 07:55:23 +00:00
gint err = 0;
snd_pcm_hw_params_t *params, *params_copy;
snd_pcm_hw_params_malloc (&params);
ext: alsa: Fix fallback paths for setting buffer and period times Below fallback paths were introduced in https://github.com/GStreamer/gst-plugins-base/commit/9759810d8206b5f1aa199f98599caec3630a1813 if setting period time after buffer time failed : 1) Set period time and then buffer time if it doesn't work 2) Set only buffer time 3) Set only period time These all were not functioning properly since they were using old copy of snd_pcm_hw_params_t which already had some fields set as per previous try and this was causing issues as driver was referring to that old value while trying to set them again in fallback paths. So now we always use the initial copy of snd_pcm_hw_params_t for every fallback and same is also being done at https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/557c4295107dc7374c850b0bd5331dd35e8fdd0f Also we change the sequence to set period time earlier than buffer time since period bytes being the smaller unit, most of the times if underlying alsa device has a dependency then it is of period bytes to be a multiple of some value (as per underlying DMA constraint) and rest of the parameters like buffer bytes need to be adjusted as per period bytes. The same sequence is also followed in alsa-utils at https://github.com/alsa-project/alsa-utils/commit/9b621eeac4d55c4e881f093be5b163ca07d01b63 Fix 2) and 3) scenarios by returning success if the exclusive setting is passed and not doing any further setting for buffer time or period time. Add new fallback path of not setting any buffer time and period time if all above fallback paths fail. The same is also being followed at aforementioned pulseaudio commit. In case of alsasink, remove the retry goto label, since it is not required anymore as fallback paths take care of setting default values if driver is not accepting any of the fallback paths. Use separate label for exit to free params structs and return err code. This also fixes leak in no_rate goto path in alsasink Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1212>
2021-06-10 07:55:23 +00:00
snd_pcm_hw_params_malloc (&params_copy);
/* 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 */
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;
Correct all relevant warnings found by the sparse semantic code analyzer. This include marking several symbols static... Original commit message from CVS: * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_get_type): * ext/alsa/gstalsasink.c: (set_hwparams): * ext/alsa/gstalsasrc.c: (set_hwparams): * ext/gio/gstgio.c: (gst_gio_uri_handler_get_uri): * ext/ogg/gstoggmux.h: * ext/ogg/gstogmparse.c: * gst-libs/gst/audio/audio.c: * gst-libs/gst/fft/kiss_fft_f64.c: (kiss_fft_f64_alloc): * gst-libs/gst/pbutils/missing-plugins.c: (gst_missing_uri_sink_message_new), (gst_missing_element_message_new), (gst_missing_decoder_message_new), (gst_missing_encoder_message_new): * gst-libs/gst/rtp/gstbasertppayload.c: * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_bye_get_reason): * gst/audioconvert/gstaudioconvert.c: * gst/audioresample/gstaudioresample.c: * gst/ffmpegcolorspace/imgconvert.c: * gst/playback/test.c: (gen_video_element), (gen_audio_element): * gst/typefind/gsttypefindfunctions.c: * gst/videoscale/vs_4tap.c: * gst/videoscale/vs_4tap.h: * sys/v4l/gstv4lelement.c: * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get_any_caps): * sys/v4l/v4l_calls.c: * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_capture_init), (gst_v4lsrc_try_capture): * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls), (gst_ximagesink_ximage_new): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new): * tests/check/elements/audioconvert.c: * tests/check/elements/audioresample.c: (fail_unless_perfect_stream): * tests/check/elements/audiotestsrc.c: (setup_audiotestsrc): * tests/check/elements/decodebin.c: * tests/check/elements/gdpdepay.c: (setup_gdpdepay), (setup_gdpdepay_streamheader): * tests/check/elements/gdppay.c: (setup_gdppay), (GST_START_TEST), (setup_gdppay_streamheader): * tests/check/elements/gnomevfssink.c: (setup_gnomevfssink): * tests/check/elements/multifdsink.c: (setup_multifdsink): * tests/check/elements/textoverlay.c: * tests/check/elements/videorate.c: (setup_videorate): * tests/check/elements/videotestsrc.c: (setup_videotestsrc): * tests/check/elements/volume.c: (setup_volume): * tests/check/elements/vorbisdec.c: (setup_vorbisdec): * tests/check/elements/vorbistag.c: * tests/check/generic/clock-selection.c: * tests/check/generic/states.c: (setup), (teardown): * tests/check/libs/cddabasesrc.c: * tests/check/libs/video.c: * tests/check/pipelines/gio.c: * tests/check/pipelines/oggmux.c: * tests/check/pipelines/simple-launch-lines.c: (simple_launch_lines_suite): * tests/check/pipelines/streamheader.c: * tests/check/pipelines/theoraenc.c: * tests/check/pipelines/vorbisdec.c: * tests/check/pipelines/vorbisenc.c: * tests/examples/seek/scrubby.c: * tests/examples/seek/seek.c: (query_positions_elems), (query_positions_pads): * tests/icles/stress-xoverlay.c: (myclock): Correct all relevant warnings found by the sparse semantic code analyzer. This include marking several symbols static, using NULL instead of 0 for pointers and using "foo (void)" instead of "foo ()" for declarations. * win32/common/libgstrtp.def: Add gst_rtp_buffer_set_extension_data to the symbol definition file.
2008-03-03 06:04:31 +00:00
CHECK (snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, NULL),
no_rate);
if (rrate != alsa->rate)
goto rate_match;
#ifndef GST_DISABLE_GST_DEBUG
/* get and dump some limits */
{
guint min, max;
snd_pcm_hw_params_get_buffer_time_min (params, &min, NULL);
snd_pcm_hw_params_get_buffer_time_max (params, &max, NULL);
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, NULL);
snd_pcm_hw_params_get_period_time_max (params, &max, NULL);
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, NULL);
snd_pcm_hw_params_get_periods_max (params, &max, NULL);
GST_DEBUG_OBJECT (alsa, "periods min %u, max %u", min, max);
}
#endif
ext: alsa: Fix fallback paths for setting buffer and period times Below fallback paths were introduced in https://github.com/GStreamer/gst-plugins-base/commit/9759810d8206b5f1aa199f98599caec3630a1813 if setting period time after buffer time failed : 1) Set period time and then buffer time if it doesn't work 2) Set only buffer time 3) Set only period time These all were not functioning properly since they were using old copy of snd_pcm_hw_params_t which already had some fields set as per previous try and this was causing issues as driver was referring to that old value while trying to set them again in fallback paths. So now we always use the initial copy of snd_pcm_hw_params_t for every fallback and same is also being done at https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/557c4295107dc7374c850b0bd5331dd35e8fdd0f Also we change the sequence to set period time earlier than buffer time since period bytes being the smaller unit, most of the times if underlying alsa device has a dependency then it is of period bytes to be a multiple of some value (as per underlying DMA constraint) and rest of the parameters like buffer bytes need to be adjusted as per period bytes. The same sequence is also followed in alsa-utils at https://github.com/alsa-project/alsa-utils/commit/9b621eeac4d55c4e881f093be5b163ca07d01b63 Fix 2) and 3) scenarios by returning success if the exclusive setting is passed and not doing any further setting for buffer time or period time. Add new fallback path of not setting any buffer time and period time if all above fallback paths fail. The same is also being followed at aforementioned pulseaudio commit. In case of alsasink, remove the retry goto label, since it is not required anymore as fallback paths take care of setting default values if driver is not accepting any of the fallback paths. Use separate label for exit to free params structs and return err code. This also fixes leak in no_rate goto path in alsasink Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1212>
2021-06-10 07:55:23 +00:00
/* Keep a copy of initial params struct that can be used later */
snd_pcm_hw_params_copy (params_copy, params);
/* Following pulseaudio's approach in
* https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/557c4295107dc7374c850b0bd5331dd35e8fdd0f
* we'll try various configuration to set the buffer time and period time as some
* driver can be picky on the order of the calls.
*/
if (alsa->period_time != -1 && alsa->buffer_time != -1) {
ext: alsa: Fix fallback paths for setting buffer and period times Below fallback paths were introduced in https://github.com/GStreamer/gst-plugins-base/commit/9759810d8206b5f1aa199f98599caec3630a1813 if setting period time after buffer time failed : 1) Set period time and then buffer time if it doesn't work 2) Set only buffer time 3) Set only period time These all were not functioning properly since they were using old copy of snd_pcm_hw_params_t which already had some fields set as per previous try and this was causing issues as driver was referring to that old value while trying to set them again in fallback paths. So now we always use the initial copy of snd_pcm_hw_params_t for every fallback and same is also being done at https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/557c4295107dc7374c850b0bd5331dd35e8fdd0f Also we change the sequence to set period time earlier than buffer time since period bytes being the smaller unit, most of the times if underlying alsa device has a dependency then it is of period bytes to be a multiple of some value (as per underlying DMA constraint) and rest of the parameters like buffer bytes need to be adjusted as per period bytes. The same sequence is also followed in alsa-utils at https://github.com/alsa-project/alsa-utils/commit/9b621eeac4d55c4e881f093be5b163ca07d01b63 Fix 2) and 3) scenarios by returning success if the exclusive setting is passed and not doing any further setting for buffer time or period time. Add new fallback path of not setting any buffer time and period time if all above fallback paths fail. The same is also being followed at aforementioned pulseaudio commit. In case of alsasink, remove the retry goto label, since it is not required anymore as fallback paths take care of setting default values if driver is not accepting any of the fallback paths. Use separate label for exit to free params structs and return err code. This also fixes leak in no_rate goto path in alsasink Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1212>
2021-06-10 07:55:23 +00:00
if ((snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
&alsa->period_time, NULL) >= 0)
&& (snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
&alsa->buffer_time, NULL) >= 0)) {
GST_DEBUG_OBJECT (alsa, "period time %u buffer time %u set correctly",
alsa->period_time, alsa->buffer_time);
goto success;
}
/* Try the new order with previous params struct as current one might
have partial settings from the order that was tried unsuccessfully */
snd_pcm_hw_params_copy (params, params_copy);
if ((snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
&alsa->buffer_time, NULL) >= 0)
&& (snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
&alsa->period_time, NULL) >= 0)) {
GST_DEBUG_OBJECT (alsa, "buffer time %u period time %u set correctly",
alsa->buffer_time, alsa->period_time);
ext: alsa: Fix fallback paths for setting buffer and period times Below fallback paths were introduced in https://github.com/GStreamer/gst-plugins-base/commit/9759810d8206b5f1aa199f98599caec3630a1813 if setting period time after buffer time failed : 1) Set period time and then buffer time if it doesn't work 2) Set only buffer time 3) Set only period time These all were not functioning properly since they were using old copy of snd_pcm_hw_params_t which already had some fields set as per previous try and this was causing issues as driver was referring to that old value while trying to set them again in fallback paths. So now we always use the initial copy of snd_pcm_hw_params_t for every fallback and same is also being done at https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/557c4295107dc7374c850b0bd5331dd35e8fdd0f Also we change the sequence to set period time earlier than buffer time since period bytes being the smaller unit, most of the times if underlying alsa device has a dependency then it is of period bytes to be a multiple of some value (as per underlying DMA constraint) and rest of the parameters like buffer bytes need to be adjusted as per period bytes. The same sequence is also followed in alsa-utils at https://github.com/alsa-project/alsa-utils/commit/9b621eeac4d55c4e881f093be5b163ca07d01b63 Fix 2) and 3) scenarios by returning success if the exclusive setting is passed and not doing any further setting for buffer time or period time. Add new fallback path of not setting any buffer time and period time if all above fallback paths fail. The same is also being followed at aforementioned pulseaudio commit. In case of alsasink, remove the retry goto label, since it is not required anymore as fallback paths take care of setting default values if driver is not accepting any of the fallback paths. Use separate label for exit to free params structs and return err code. This also fixes leak in no_rate goto path in alsasink Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1212>
2021-06-10 07:55:23 +00:00
goto success;
}
ext: alsa: Fix fallback paths for setting buffer and period times Below fallback paths were introduced in https://github.com/GStreamer/gst-plugins-base/commit/9759810d8206b5f1aa199f98599caec3630a1813 if setting period time after buffer time failed : 1) Set period time and then buffer time if it doesn't work 2) Set only buffer time 3) Set only period time These all were not functioning properly since they were using old copy of snd_pcm_hw_params_t which already had some fields set as per previous try and this was causing issues as driver was referring to that old value while trying to set them again in fallback paths. So now we always use the initial copy of snd_pcm_hw_params_t for every fallback and same is also being done at https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/557c4295107dc7374c850b0bd5331dd35e8fdd0f Also we change the sequence to set period time earlier than buffer time since period bytes being the smaller unit, most of the times if underlying alsa device has a dependency then it is of period bytes to be a multiple of some value (as per underlying DMA constraint) and rest of the parameters like buffer bytes need to be adjusted as per period bytes. The same sequence is also followed in alsa-utils at https://github.com/alsa-project/alsa-utils/commit/9b621eeac4d55c4e881f093be5b163ca07d01b63 Fix 2) and 3) scenarios by returning success if the exclusive setting is passed and not doing any further setting for buffer time or period time. Add new fallback path of not setting any buffer time and period time if all above fallback paths fail. The same is also being followed at aforementioned pulseaudio commit. In case of alsasink, remove the retry goto label, since it is not required anymore as fallback paths take care of setting default values if driver is not accepting any of the fallback paths. Use separate label for exit to free params structs and return err code. This also fixes leak in no_rate goto path in alsasink Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1212>
2021-06-10 07:55:23 +00:00
}
if (alsa->period_time != -1) {
snd_pcm_hw_params_copy (params, params_copy);
/* set the period time only */
if ((snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
ext: alsa: Fix fallback paths for setting buffer and period times Below fallback paths were introduced in https://github.com/GStreamer/gst-plugins-base/commit/9759810d8206b5f1aa199f98599caec3630a1813 if setting period time after buffer time failed : 1) Set period time and then buffer time if it doesn't work 2) Set only buffer time 3) Set only period time These all were not functioning properly since they were using old copy of snd_pcm_hw_params_t which already had some fields set as per previous try and this was causing issues as driver was referring to that old value while trying to set them again in fallback paths. So now we always use the initial copy of snd_pcm_hw_params_t for every fallback and same is also being done at https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/557c4295107dc7374c850b0bd5331dd35e8fdd0f Also we change the sequence to set period time earlier than buffer time since period bytes being the smaller unit, most of the times if underlying alsa device has a dependency then it is of period bytes to be a multiple of some value (as per underlying DMA constraint) and rest of the parameters like buffer bytes need to be adjusted as per period bytes. The same sequence is also followed in alsa-utils at https://github.com/alsa-project/alsa-utils/commit/9b621eeac4d55c4e881f093be5b163ca07d01b63 Fix 2) and 3) scenarios by returning success if the exclusive setting is passed and not doing any further setting for buffer time or period time. Add new fallback path of not setting any buffer time and period time if all above fallback paths fail. The same is also being followed at aforementioned pulseaudio commit. In case of alsasink, remove the retry goto label, since it is not required anymore as fallback paths take care of setting default values if driver is not accepting any of the fallback paths. Use separate label for exit to free params structs and return err code. This also fixes leak in no_rate goto path in alsasink Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1212>
2021-06-10 07:55:23 +00:00
&alsa->period_time, NULL) >= 0)) {
GST_DEBUG_OBJECT (alsa, "period time %u set correctly",
alsa->period_time);
goto success;
}
}
if (alsa->buffer_time != -1) {
ext: alsa: Fix fallback paths for setting buffer and period times Below fallback paths were introduced in https://github.com/GStreamer/gst-plugins-base/commit/9759810d8206b5f1aa199f98599caec3630a1813 if setting period time after buffer time failed : 1) Set period time and then buffer time if it doesn't work 2) Set only buffer time 3) Set only period time These all were not functioning properly since they were using old copy of snd_pcm_hw_params_t which already had some fields set as per previous try and this was causing issues as driver was referring to that old value while trying to set them again in fallback paths. So now we always use the initial copy of snd_pcm_hw_params_t for every fallback and same is also being done at https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/557c4295107dc7374c850b0bd5331dd35e8fdd0f Also we change the sequence to set period time earlier than buffer time since period bytes being the smaller unit, most of the times if underlying alsa device has a dependency then it is of period bytes to be a multiple of some value (as per underlying DMA constraint) and rest of the parameters like buffer bytes need to be adjusted as per period bytes. The same sequence is also followed in alsa-utils at https://github.com/alsa-project/alsa-utils/commit/9b621eeac4d55c4e881f093be5b163ca07d01b63 Fix 2) and 3) scenarios by returning success if the exclusive setting is passed and not doing any further setting for buffer time or period time. Add new fallback path of not setting any buffer time and period time if all above fallback paths fail. The same is also being followed at aforementioned pulseaudio commit. In case of alsasink, remove the retry goto label, since it is not required anymore as fallback paths take care of setting default values if driver is not accepting any of the fallback paths. Use separate label for exit to free params structs and return err code. This also fixes leak in no_rate goto path in alsasink Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1212>
2021-06-10 07:55:23 +00:00
snd_pcm_hw_params_copy (params, params_copy);
/* set the buffer time only */
if ((snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
&alsa->buffer_time, NULL) >= 0)) {
GST_DEBUG_OBJECT (alsa, "buffer time %u set correctly",
alsa->buffer_time);
goto success;
}
}
ext: alsa: Fix fallback paths for setting buffer and period times Below fallback paths were introduced in https://github.com/GStreamer/gst-plugins-base/commit/9759810d8206b5f1aa199f98599caec3630a1813 if setting period time after buffer time failed : 1) Set period time and then buffer time if it doesn't work 2) Set only buffer time 3) Set only period time These all were not functioning properly since they were using old copy of snd_pcm_hw_params_t which already had some fields set as per previous try and this was causing issues as driver was referring to that old value while trying to set them again in fallback paths. So now we always use the initial copy of snd_pcm_hw_params_t for every fallback and same is also being done at https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/557c4295107dc7374c850b0bd5331dd35e8fdd0f Also we change the sequence to set period time earlier than buffer time since period bytes being the smaller unit, most of the times if underlying alsa device has a dependency then it is of period bytes to be a multiple of some value (as per underlying DMA constraint) and rest of the parameters like buffer bytes need to be adjusted as per period bytes. The same sequence is also followed in alsa-utils at https://github.com/alsa-project/alsa-utils/commit/9b621eeac4d55c4e881f093be5b163ca07d01b63 Fix 2) and 3) scenarios by returning success if the exclusive setting is passed and not doing any further setting for buffer time or period time. Add new fallback path of not setting any buffer time and period time if all above fallback paths fail. The same is also being followed at aforementioned pulseaudio commit. In case of alsasink, remove the retry goto label, since it is not required anymore as fallback paths take care of setting default values if driver is not accepting any of the fallback paths. Use separate label for exit to free params structs and return err code. This also fixes leak in no_rate goto path in alsasink Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1212>
2021-06-10 07:55:23 +00:00
/* Set nothing if all above failed */
snd_pcm_hw_params_copy (params, params_copy);
GST_DEBUG_OBJECT (alsa, "Not setting period time and buffer time");
ext: alsa: Fix fallback paths for setting buffer and period times Below fallback paths were introduced in https://github.com/GStreamer/gst-plugins-base/commit/9759810d8206b5f1aa199f98599caec3630a1813 if setting period time after buffer time failed : 1) Set period time and then buffer time if it doesn't work 2) Set only buffer time 3) Set only period time These all were not functioning properly since they were using old copy of snd_pcm_hw_params_t which already had some fields set as per previous try and this was causing issues as driver was referring to that old value while trying to set them again in fallback paths. So now we always use the initial copy of snd_pcm_hw_params_t for every fallback and same is also being done at https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/557c4295107dc7374c850b0bd5331dd35e8fdd0f Also we change the sequence to set period time earlier than buffer time since period bytes being the smaller unit, most of the times if underlying alsa device has a dependency then it is of period bytes to be a multiple of some value (as per underlying DMA constraint) and rest of the parameters like buffer bytes need to be adjusted as per period bytes. The same sequence is also followed in alsa-utils at https://github.com/alsa-project/alsa-utils/commit/9b621eeac4d55c4e881f093be5b163ca07d01b63 Fix 2) and 3) scenarios by returning success if the exclusive setting is passed and not doing any further setting for buffer time or period time. Add new fallback path of not setting any buffer time and period time if all above fallback paths fail. The same is also being followed at aforementioned pulseaudio commit. In case of alsasink, remove the retry goto label, since it is not required anymore as fallback paths take care of setting default values if driver is not accepting any of the fallback paths. Use separate label for exit to free params structs and return err code. This also fixes leak in no_rate goto path in alsasink Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1212>
2021-06-10 07:55:23 +00:00
success:
/* write the parameters to device */
CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);
CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size),
buffer_size);
ext: alsa: Fix fallback paths for setting buffer and period times Below fallback paths were introduced in https://github.com/GStreamer/gst-plugins-base/commit/9759810d8206b5f1aa199f98599caec3630a1813 if setting period time after buffer time failed : 1) Set period time and then buffer time if it doesn't work 2) Set only buffer time 3) Set only period time These all were not functioning properly since they were using old copy of snd_pcm_hw_params_t which already had some fields set as per previous try and this was causing issues as driver was referring to that old value while trying to set them again in fallback paths. So now we always use the initial copy of snd_pcm_hw_params_t for every fallback and same is also being done at https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/557c4295107dc7374c850b0bd5331dd35e8fdd0f Also we change the sequence to set period time earlier than buffer time since period bytes being the smaller unit, most of the times if underlying alsa device has a dependency then it is of period bytes to be a multiple of some value (as per underlying DMA constraint) and rest of the parameters like buffer bytes need to be adjusted as per period bytes. The same sequence is also followed in alsa-utils at https://github.com/alsa-project/alsa-utils/commit/9b621eeac4d55c4e881f093be5b163ca07d01b63 Fix 2) and 3) scenarios by returning success if the exclusive setting is passed and not doing any further setting for buffer time or period time. Add new fallback path of not setting any buffer time and period time if all above fallback paths fail. The same is also being followed at aforementioned pulseaudio commit. In case of alsasink, remove the retry goto label, since it is not required anymore as fallback paths take care of setting default values if driver is not accepting any of the fallback paths. Use separate label for exit to free params structs and return err code. This also fixes leak in no_rate goto path in alsasink Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1212>
2021-06-10 07:55:23 +00:00
GST_DEBUG_OBJECT (alsa, "buffer size : %lu", alsa->buffer_size);
CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size,
NULL), period_size);
GST_DEBUG_OBJECT (alsa, "period size : %lu", alsa->period_size);
ext: alsa: Fix fallback paths for setting buffer and period times Below fallback paths were introduced in https://github.com/GStreamer/gst-plugins-base/commit/9759810d8206b5f1aa199f98599caec3630a1813 if setting period time after buffer time failed : 1) Set period time and then buffer time if it doesn't work 2) Set only buffer time 3) Set only period time These all were not functioning properly since they were using old copy of snd_pcm_hw_params_t which already had some fields set as per previous try and this was causing issues as driver was referring to that old value while trying to set them again in fallback paths. So now we always use the initial copy of snd_pcm_hw_params_t for every fallback and same is also being done at https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/557c4295107dc7374c850b0bd5331dd35e8fdd0f Also we change the sequence to set period time earlier than buffer time since period bytes being the smaller unit, most of the times if underlying alsa device has a dependency then it is of period bytes to be a multiple of some value (as per underlying DMA constraint) and rest of the parameters like buffer bytes need to be adjusted as per period bytes. The same sequence is also followed in alsa-utils at https://github.com/alsa-project/alsa-utils/commit/9b621eeac4d55c4e881f093be5b163ca07d01b63 Fix 2) and 3) scenarios by returning success if the exclusive setting is passed and not doing any further setting for buffer time or period time. Add new fallback path of not setting any buffer time and period time if all above fallback paths fail. The same is also being followed at aforementioned pulseaudio commit. In case of alsasink, remove the retry goto label, since it is not required anymore as fallback paths take care of setting default values if driver is not accepting any of the fallback paths. Use separate label for exit to free params structs and return err code. This also fixes leak in no_rate goto path in alsasink Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1212>
2021-06-10 07:55:23 +00:00
goto exit;
/* ERRORS */
no_config:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Broken configuration for recording: no configurations available: %s",
snd_strerror (err)));
ext: alsa: Fix fallback paths for setting buffer and period times Below fallback paths were introduced in https://github.com/GStreamer/gst-plugins-base/commit/9759810d8206b5f1aa199f98599caec3630a1813 if setting period time after buffer time failed : 1) Set period time and then buffer time if it doesn't work 2) Set only buffer time 3) Set only period time These all were not functioning properly since they were using old copy of snd_pcm_hw_params_t which already had some fields set as per previous try and this was causing issues as driver was referring to that old value while trying to set them again in fallback paths. So now we always use the initial copy of snd_pcm_hw_params_t for every fallback and same is also being done at https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/557c4295107dc7374c850b0bd5331dd35e8fdd0f Also we change the sequence to set period time earlier than buffer time since period bytes being the smaller unit, most of the times if underlying alsa device has a dependency then it is of period bytes to be a multiple of some value (as per underlying DMA constraint) and rest of the parameters like buffer bytes need to be adjusted as per period bytes. The same sequence is also followed in alsa-utils at https://github.com/alsa-project/alsa-utils/commit/9b621eeac4d55c4e881f093be5b163ca07d01b63 Fix 2) and 3) scenarios by returning success if the exclusive setting is passed and not doing any further setting for buffer time or period time. Add new fallback path of not setting any buffer time and period time if all above fallback paths fail. The same is also being followed at aforementioned pulseaudio commit. In case of alsasink, remove the retry goto label, since it is not required anymore as fallback paths take care of setting default values if driver is not accepting any of the fallback paths. Use separate label for exit to free params structs and return err code. This also fixes leak in no_rate goto path in alsasink Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1212>
2021-06-10 07:55:23 +00:00
goto exit;
}
wrong_access:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Access type not available for recording: %s", snd_strerror (err)));
ext: alsa: Fix fallback paths for setting buffer and period times Below fallback paths were introduced in https://github.com/GStreamer/gst-plugins-base/commit/9759810d8206b5f1aa199f98599caec3630a1813 if setting period time after buffer time failed : 1) Set period time and then buffer time if it doesn't work 2) Set only buffer time 3) Set only period time These all were not functioning properly since they were using old copy of snd_pcm_hw_params_t which already had some fields set as per previous try and this was causing issues as driver was referring to that old value while trying to set them again in fallback paths. So now we always use the initial copy of snd_pcm_hw_params_t for every fallback and same is also being done at https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/557c4295107dc7374c850b0bd5331dd35e8fdd0f Also we change the sequence to set period time earlier than buffer time since period bytes being the smaller unit, most of the times if underlying alsa device has a dependency then it is of period bytes to be a multiple of some value (as per underlying DMA constraint) and rest of the parameters like buffer bytes need to be adjusted as per period bytes. The same sequence is also followed in alsa-utils at https://github.com/alsa-project/alsa-utils/commit/9b621eeac4d55c4e881f093be5b163ca07d01b63 Fix 2) and 3) scenarios by returning success if the exclusive setting is passed and not doing any further setting for buffer time or period time. Add new fallback path of not setting any buffer time and period time if all above fallback paths fail. The same is also being followed at aforementioned pulseaudio commit. In case of alsasink, remove the retry goto label, since it is not required anymore as fallback paths take care of setting default values if driver is not accepting any of the fallback paths. Use separate label for exit to free params structs and return err code. This also fixes leak in no_rate goto path in alsasink Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1212>
2021-06-10 07:55:23 +00:00
goto exit;
}
no_sample_format:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Sample format not available for recording: %s", snd_strerror (err)));
ext: alsa: Fix fallback paths for setting buffer and period times Below fallback paths were introduced in https://github.com/GStreamer/gst-plugins-base/commit/9759810d8206b5f1aa199f98599caec3630a1813 if setting period time after buffer time failed : 1) Set period time and then buffer time if it doesn't work 2) Set only buffer time 3) Set only period time These all were not functioning properly since they were using old copy of snd_pcm_hw_params_t which already had some fields set as per previous try and this was causing issues as driver was referring to that old value while trying to set them again in fallback paths. So now we always use the initial copy of snd_pcm_hw_params_t for every fallback and same is also being done at https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/557c4295107dc7374c850b0bd5331dd35e8fdd0f Also we change the sequence to set period time earlier than buffer time since period bytes being the smaller unit, most of the times if underlying alsa device has a dependency then it is of period bytes to be a multiple of some value (as per underlying DMA constraint) and rest of the parameters like buffer bytes need to be adjusted as per period bytes. The same sequence is also followed in alsa-utils at https://github.com/alsa-project/alsa-utils/commit/9b621eeac4d55c4e881f093be5b163ca07d01b63 Fix 2) and 3) scenarios by returning success if the exclusive setting is passed and not doing any further setting for buffer time or period time. Add new fallback path of not setting any buffer time and period time if all above fallback paths fail. The same is also being followed at aforementioned pulseaudio commit. In case of alsasink, remove the retry goto label, since it is not required anymore as fallback paths take care of setting default values if driver is not accepting any of the fallback paths. Use separate label for exit to free params structs and return err code. This also fixes leak in no_rate goto path in alsasink Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1212>
2021-06-10 07:55:23 +00:00
goto exit;
}
no_channels:
{
gchar *msg = NULL;
if ((alsa->channels) == 1)
msg = g_strdup (_("Could not open device for recording in mono mode."));
if ((alsa->channels) == 2)
msg = g_strdup (_("Could not open device for recording in stereo mode."));
if ((alsa->channels) > 2)
msg =
g_strdup_printf (_
("Could not open device for recording in %d-channel mode"),
alsa->channels);
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, ("%s", msg),
("%s", snd_strerror (err)));
g_free (msg);
ext: alsa: Fix fallback paths for setting buffer and period times Below fallback paths were introduced in https://github.com/GStreamer/gst-plugins-base/commit/9759810d8206b5f1aa199f98599caec3630a1813 if setting period time after buffer time failed : 1) Set period time and then buffer time if it doesn't work 2) Set only buffer time 3) Set only period time These all were not functioning properly since they were using old copy of snd_pcm_hw_params_t which already had some fields set as per previous try and this was causing issues as driver was referring to that old value while trying to set them again in fallback paths. So now we always use the initial copy of snd_pcm_hw_params_t for every fallback and same is also being done at https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/557c4295107dc7374c850b0bd5331dd35e8fdd0f Also we change the sequence to set period time earlier than buffer time since period bytes being the smaller unit, most of the times if underlying alsa device has a dependency then it is of period bytes to be a multiple of some value (as per underlying DMA constraint) and rest of the parameters like buffer bytes need to be adjusted as per period bytes. The same sequence is also followed in alsa-utils at https://github.com/alsa-project/alsa-utils/commit/9b621eeac4d55c4e881f093be5b163ca07d01b63 Fix 2) and 3) scenarios by returning success if the exclusive setting is passed and not doing any further setting for buffer time or period time. Add new fallback path of not setting any buffer time and period time if all above fallback paths fail. The same is also being followed at aforementioned pulseaudio commit. In case of alsasink, remove the retry goto label, since it is not required anymore as fallback paths take care of setting default values if driver is not accepting any of the fallback paths. Use separate label for exit to free params structs and return err code. This also fixes leak in no_rate goto path in alsasink Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1212>
2021-06-10 07:55:23 +00:00
goto exit;
}
no_rate:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Rate %iHz not available for recording: %s",
alsa->rate, snd_strerror (err)));
ext: alsa: Fix fallback paths for setting buffer and period times Below fallback paths were introduced in https://github.com/GStreamer/gst-plugins-base/commit/9759810d8206b5f1aa199f98599caec3630a1813 if setting period time after buffer time failed : 1) Set period time and then buffer time if it doesn't work 2) Set only buffer time 3) Set only period time These all were not functioning properly since they were using old copy of snd_pcm_hw_params_t which already had some fields set as per previous try and this was causing issues as driver was referring to that old value while trying to set them again in fallback paths. So now we always use the initial copy of snd_pcm_hw_params_t for every fallback and same is also being done at https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/557c4295107dc7374c850b0bd5331dd35e8fdd0f Also we change the sequence to set period time earlier than buffer time since period bytes being the smaller unit, most of the times if underlying alsa device has a dependency then it is of period bytes to be a multiple of some value (as per underlying DMA constraint) and rest of the parameters like buffer bytes need to be adjusted as per period bytes. The same sequence is also followed in alsa-utils at https://github.com/alsa-project/alsa-utils/commit/9b621eeac4d55c4e881f093be5b163ca07d01b63 Fix 2) and 3) scenarios by returning success if the exclusive setting is passed and not doing any further setting for buffer time or period time. Add new fallback path of not setting any buffer time and period time if all above fallback paths fail. The same is also being followed at aforementioned pulseaudio commit. In case of alsasink, remove the retry goto label, since it is not required anymore as fallback paths take care of setting default values if driver is not accepting any of the fallback paths. Use separate label for exit to free params structs and return err code. This also fixes leak in no_rate goto path in alsasink Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1212>
2021-06-10 07:55:23 +00:00
goto exit;
}
rate_match:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Rate doesn't match (requested %iHz, get %iHz)", alsa->rate, err));
ext: alsa: Fix fallback paths for setting buffer and period times Below fallback paths were introduced in https://github.com/GStreamer/gst-plugins-base/commit/9759810d8206b5f1aa199f98599caec3630a1813 if setting period time after buffer time failed : 1) Set period time and then buffer time if it doesn't work 2) Set only buffer time 3) Set only period time These all were not functioning properly since they were using old copy of snd_pcm_hw_params_t which already had some fields set as per previous try and this was causing issues as driver was referring to that old value while trying to set them again in fallback paths. So now we always use the initial copy of snd_pcm_hw_params_t for every fallback and same is also being done at https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/557c4295107dc7374c850b0bd5331dd35e8fdd0f Also we change the sequence to set period time earlier than buffer time since period bytes being the smaller unit, most of the times if underlying alsa device has a dependency then it is of period bytes to be a multiple of some value (as per underlying DMA constraint) and rest of the parameters like buffer bytes need to be adjusted as per period bytes. The same sequence is also followed in alsa-utils at https://github.com/alsa-project/alsa-utils/commit/9b621eeac4d55c4e881f093be5b163ca07d01b63 Fix 2) and 3) scenarios by returning success if the exclusive setting is passed and not doing any further setting for buffer time or period time. Add new fallback path of not setting any buffer time and period time if all above fallback paths fail. The same is also being followed at aforementioned pulseaudio commit. In case of alsasink, remove the retry goto label, since it is not required anymore as fallback paths take care of setting default values if driver is not accepting any of the fallback paths. Use separate label for exit to free params structs and return err code. This also fixes leak in no_rate goto path in alsasink Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1212>
2021-06-10 07:55:23 +00:00
err = -EINVAL;
goto exit;
}
buffer_size:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to get buffer size for recording: %s", snd_strerror (err)));
ext: alsa: Fix fallback paths for setting buffer and period times Below fallback paths were introduced in https://github.com/GStreamer/gst-plugins-base/commit/9759810d8206b5f1aa199f98599caec3630a1813 if setting period time after buffer time failed : 1) Set period time and then buffer time if it doesn't work 2) Set only buffer time 3) Set only period time These all were not functioning properly since they were using old copy of snd_pcm_hw_params_t which already had some fields set as per previous try and this was causing issues as driver was referring to that old value while trying to set them again in fallback paths. So now we always use the initial copy of snd_pcm_hw_params_t for every fallback and same is also being done at https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/557c4295107dc7374c850b0bd5331dd35e8fdd0f Also we change the sequence to set period time earlier than buffer time since period bytes being the smaller unit, most of the times if underlying alsa device has a dependency then it is of period bytes to be a multiple of some value (as per underlying DMA constraint) and rest of the parameters like buffer bytes need to be adjusted as per period bytes. The same sequence is also followed in alsa-utils at https://github.com/alsa-project/alsa-utils/commit/9b621eeac4d55c4e881f093be5b163ca07d01b63 Fix 2) and 3) scenarios by returning success if the exclusive setting is passed and not doing any further setting for buffer time or period time. Add new fallback path of not setting any buffer time and period time if all above fallback paths fail. The same is also being followed at aforementioned pulseaudio commit. In case of alsasink, remove the retry goto label, since it is not required anymore as fallback paths take care of setting default values if driver is not accepting any of the fallback paths. Use separate label for exit to free params structs and return err code. This also fixes leak in no_rate goto path in alsasink Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1212>
2021-06-10 07:55:23 +00:00
goto exit;
}
period_size:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to get period size for recording: %s", snd_strerror (err)));
ext: alsa: Fix fallback paths for setting buffer and period times Below fallback paths were introduced in https://github.com/GStreamer/gst-plugins-base/commit/9759810d8206b5f1aa199f98599caec3630a1813 if setting period time after buffer time failed : 1) Set period time and then buffer time if it doesn't work 2) Set only buffer time 3) Set only period time These all were not functioning properly since they were using old copy of snd_pcm_hw_params_t which already had some fields set as per previous try and this was causing issues as driver was referring to that old value while trying to set them again in fallback paths. So now we always use the initial copy of snd_pcm_hw_params_t for every fallback and same is also being done at https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/557c4295107dc7374c850b0bd5331dd35e8fdd0f Also we change the sequence to set period time earlier than buffer time since period bytes being the smaller unit, most of the times if underlying alsa device has a dependency then it is of period bytes to be a multiple of some value (as per underlying DMA constraint) and rest of the parameters like buffer bytes need to be adjusted as per period bytes. The same sequence is also followed in alsa-utils at https://github.com/alsa-project/alsa-utils/commit/9b621eeac4d55c4e881f093be5b163ca07d01b63 Fix 2) and 3) scenarios by returning success if the exclusive setting is passed and not doing any further setting for buffer time or period time. Add new fallback path of not setting any buffer time and period time if all above fallback paths fail. The same is also being followed at aforementioned pulseaudio commit. In case of alsasink, remove the retry goto label, since it is not required anymore as fallback paths take care of setting default values if driver is not accepting any of the fallback paths. Use separate label for exit to free params structs and return err code. This also fixes leak in no_rate goto path in alsasink Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1212>
2021-06-10 07:55:23 +00:00
goto exit;
}
set_hw_params:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set hw params for recording: %s", snd_strerror (err)));
ext: alsa: Fix fallback paths for setting buffer and period times Below fallback paths were introduced in https://github.com/GStreamer/gst-plugins-base/commit/9759810d8206b5f1aa199f98599caec3630a1813 if setting period time after buffer time failed : 1) Set period time and then buffer time if it doesn't work 2) Set only buffer time 3) Set only period time These all were not functioning properly since they were using old copy of snd_pcm_hw_params_t which already had some fields set as per previous try and this was causing issues as driver was referring to that old value while trying to set them again in fallback paths. So now we always use the initial copy of snd_pcm_hw_params_t for every fallback and same is also being done at https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/557c4295107dc7374c850b0bd5331dd35e8fdd0f Also we change the sequence to set period time earlier than buffer time since period bytes being the smaller unit, most of the times if underlying alsa device has a dependency then it is of period bytes to be a multiple of some value (as per underlying DMA constraint) and rest of the parameters like buffer bytes need to be adjusted as per period bytes. The same sequence is also followed in alsa-utils at https://github.com/alsa-project/alsa-utils/commit/9b621eeac4d55c4e881f093be5b163ca07d01b63 Fix 2) and 3) scenarios by returning success if the exclusive setting is passed and not doing any further setting for buffer time or period time. Add new fallback path of not setting any buffer time and period time if all above fallback paths fail. The same is also being followed at aforementioned pulseaudio commit. In case of alsasink, remove the retry goto label, since it is not required anymore as fallback paths take care of setting default values if driver is not accepting any of the fallback paths. Use separate label for exit to free params structs and return err code. This also fixes leak in no_rate goto path in alsasink Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1212>
2021-06-10 07:55:23 +00:00
}
exit:
{
snd_pcm_hw_params_free (params);
ext: alsa: Fix fallback paths for setting buffer and period times Below fallback paths were introduced in https://github.com/GStreamer/gst-plugins-base/commit/9759810d8206b5f1aa199f98599caec3630a1813 if setting period time after buffer time failed : 1) Set period time and then buffer time if it doesn't work 2) Set only buffer time 3) Set only period time These all were not functioning properly since they were using old copy of snd_pcm_hw_params_t which already had some fields set as per previous try and this was causing issues as driver was referring to that old value while trying to set them again in fallback paths. So now we always use the initial copy of snd_pcm_hw_params_t for every fallback and same is also being done at https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/557c4295107dc7374c850b0bd5331dd35e8fdd0f Also we change the sequence to set period time earlier than buffer time since period bytes being the smaller unit, most of the times if underlying alsa device has a dependency then it is of period bytes to be a multiple of some value (as per underlying DMA constraint) and rest of the parameters like buffer bytes need to be adjusted as per period bytes. The same sequence is also followed in alsa-utils at https://github.com/alsa-project/alsa-utils/commit/9b621eeac4d55c4e881f093be5b163ca07d01b63 Fix 2) and 3) scenarios by returning success if the exclusive setting is passed and not doing any further setting for buffer time or period time. Add new fallback path of not setting any buffer time and period time if all above fallback paths fail. The same is also being followed at aforementioned pulseaudio commit. In case of alsasink, remove the retry goto label, since it is not required anymore as fallback paths take care of setting default values if driver is not accepting any of the fallback paths. Use separate label for exit to free params structs and return err code. This also fixes leak in no_rate goto path in alsasink Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1212>
2021-06-10 07:55:23 +00:00
snd_pcm_hw_params_free (params_copy);
return err;
}
}
static int
set_swparams (GstAlsaSrc * alsa)
{
int err;
snd_pcm_sw_params_t *params;
snd_pcm_sw_params_malloc (&params);
/* get the current swparams */
CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config);
/* 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);
/* start the transfer on first read */
CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params,
0), start_threshold);
/* use monotonic timestamping */
CHECK (snd_pcm_sw_params_set_tstamp_mode (alsa->handle, params,
SND_PCM_TSTAMP_MMAP), tstamp_mode);
#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 recording 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;
}
tstamp_mode:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set tstamp mode 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
alsasrc_parse_spec (GstAlsaSrc * alsa, GstAudioRingBufferSpec * spec)
{
switch (spec->type) {
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW:
switch (GST_AUDIO_INFO_FORMAT (&spec->info)) {
case GST_AUDIO_FORMAT_U8:
alsa->format = SND_PCM_FORMAT_U8;
break;
case GST_AUDIO_FORMAT_S8:
alsa->format = SND_PCM_FORMAT_S8;
break;
case GST_AUDIO_FORMAT_S16LE:
alsa->format = SND_PCM_FORMAT_S16_LE;
break;
case GST_AUDIO_FORMAT_S16BE:
alsa->format = SND_PCM_FORMAT_S16_BE;
break;
case GST_AUDIO_FORMAT_U16LE:
alsa->format = SND_PCM_FORMAT_U16_LE;
break;
case GST_AUDIO_FORMAT_U16BE:
alsa->format = SND_PCM_FORMAT_U16_BE;
break;
case GST_AUDIO_FORMAT_S24_32LE:
alsa->format = SND_PCM_FORMAT_S24_LE;
break;
case GST_AUDIO_FORMAT_S24_32BE:
alsa->format = SND_PCM_FORMAT_S24_BE;
break;
case GST_AUDIO_FORMAT_U24_32LE:
alsa->format = SND_PCM_FORMAT_U24_LE;
break;
case GST_AUDIO_FORMAT_U24_32BE:
alsa->format = SND_PCM_FORMAT_U24_BE;
break;
case GST_AUDIO_FORMAT_S32LE:
alsa->format = SND_PCM_FORMAT_S32_LE;
break;
case GST_AUDIO_FORMAT_S32BE:
alsa->format = SND_PCM_FORMAT_S32_BE;
break;
case GST_AUDIO_FORMAT_U32LE:
alsa->format = SND_PCM_FORMAT_U32_LE;
break;
case GST_AUDIO_FORMAT_U32BE:
alsa->format = SND_PCM_FORMAT_U32_BE;
break;
case GST_AUDIO_FORMAT_S24LE:
alsa->format = SND_PCM_FORMAT_S24_3LE;
break;
case GST_AUDIO_FORMAT_S24BE:
alsa->format = SND_PCM_FORMAT_S24_3BE;
break;
case GST_AUDIO_FORMAT_U24LE:
alsa->format = SND_PCM_FORMAT_U24_3LE;
break;
case GST_AUDIO_FORMAT_U24BE:
alsa->format = SND_PCM_FORMAT_U24_3BE;
break;
case GST_AUDIO_FORMAT_S20LE:
alsa->format = SND_PCM_FORMAT_S20_3LE;
break;
case GST_AUDIO_FORMAT_S20BE:
alsa->format = SND_PCM_FORMAT_S20_3BE;
break;
case GST_AUDIO_FORMAT_U20LE:
alsa->format = SND_PCM_FORMAT_U20_3LE;
break;
case GST_AUDIO_FORMAT_U20BE:
alsa->format = SND_PCM_FORMAT_U20_3BE;
break;
case GST_AUDIO_FORMAT_S18LE:
alsa->format = SND_PCM_FORMAT_S18_3LE;
break;
case GST_AUDIO_FORMAT_S18BE:
alsa->format = SND_PCM_FORMAT_S18_3BE;
break;
case GST_AUDIO_FORMAT_U18LE:
alsa->format = SND_PCM_FORMAT_U18_3LE;
break;
case GST_AUDIO_FORMAT_U18BE:
alsa->format = SND_PCM_FORMAT_U18_3BE;
break;
case GST_AUDIO_FORMAT_F32LE:
alsa->format = SND_PCM_FORMAT_FLOAT_LE;
break;
case GST_AUDIO_FORMAT_F32BE:
alsa->format = SND_PCM_FORMAT_FLOAT_BE;
break;
case GST_AUDIO_FORMAT_F64LE:
alsa->format = SND_PCM_FORMAT_FLOAT64_LE;
break;
case GST_AUDIO_FORMAT_F64BE:
alsa->format = SND_PCM_FORMAT_FLOAT64_BE;
break;
default:
goto error;
}
break;
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW:
alsa->format = SND_PCM_FORMAT_A_LAW;
break;
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW:
alsa->format = SND_PCM_FORMAT_MU_LAW;
break;
default:
goto error;
}
alsa->rate = GST_AUDIO_INFO_RATE (&spec->info);
alsa->channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
alsa->buffer_time = spec->buffer_time;
alsa->period_time = spec->latency_time;
alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED;
if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW && alsa->channels < 9)
gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SRC
(alsa)->ringbuffer, alsa_position[alsa->channels - 1]);
return TRUE;
/* ERRORS */
error:
{
return FALSE;
}
}
static gboolean
gst_alsasrc_open (GstAudioSrc * asrc)
{
GstAlsaSrc *alsa;
gint err;
alsa = GST_ALSA_SRC (asrc);
CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_CAPTURE,
(alsa->driver_timestamps) ? 0 : SND_PCM_NONBLOCK), open_error);
return TRUE;
/* ERRORS */
open_error:
{
if (err == -EBUSY) {
GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY,
(_("Could not open audio device for recording. "
"Device is being used by another application.")),
("Device '%s' is busy", alsa->device));
} else {
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
(_("Could not open audio device for recording.")),
("Recording open error on device '%s': %s", alsa->device,
snd_strerror (err)));
}
return FALSE;
}
}
static gboolean
gst_alsasrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
{
GstAlsaSrc *alsa;
gint err;
alsa = GST_ALSA_SRC (asrc);
if (!alsasrc_parse_spec (alsa, spec))
goto spec_parse;
CHECK (snd_pcm_nonblock (alsa->handle, 0), non_block);
CHECK (set_hwparams (alsa), hw_params_failed);
CHECK (set_swparams (alsa), sw_params_failed);
CHECK (snd_pcm_prepare (alsa->handle), prepare_failed);
alsa->bpf = GST_AUDIO_INFO_BPF (&spec->info);
spec->segsize = alsa->period_size * alsa->bpf;
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);
}
#ifdef SND_CHMAP_API_VERSION
alsa_detect_channels_mapping (GST_OBJECT (alsa), alsa->handle, spec,
alsa->channels, GST_AUDIO_BASE_SRC (alsa)->ringbuffer);
#endif /* SND_CHMAP_API_VERSION */
return TRUE;
/* ERRORS */
spec_parse:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Error parsing spec"));
return FALSE;
}
non_block:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Could not set device to blocking: %s", snd_strerror (err)));
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;
}
prepare_failed:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Prepare failed: %s", snd_strerror (err)));
return FALSE;
}
}
static gboolean
gst_alsasrc_unprepare (GstAudioSrc * asrc)
{
GstAlsaSrc *alsa;
alsa = GST_ALSA_SRC (asrc);
snd_pcm_drop (alsa->handle);
snd_pcm_hw_free (alsa->handle);
snd_pcm_nonblock (alsa->handle, 1);
return TRUE;
}
static gboolean
gst_alsasrc_close (GstAudioSrc * asrc)
{
GstAlsaSrc *alsa = GST_ALSA_SRC (asrc);
snd_pcm_close (alsa->handle);
alsa->handle = NULL;
gst_caps_replace (&alsa->cached_caps, NULL);
return TRUE;
}
/*
* Underrun and suspend recovery
*/
static gint
xrun_recovery (GstAlsaSrc * alsa, snd_pcm_t * handle, gint err)
{
GST_WARNING_OBJECT (alsa, "xrun recovery %d: %s", err, g_strerror (-err));
if (err == -EPIPE) { /* under-run */
err = snd_pcm_prepare (handle);
if (err < 0)
GST_WARNING_OBJECT (alsa,
"Can't recover 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 recover from suspend, prepare failed: %s",
snd_strerror (err));
}
return 0;
}
return err;
}
static GstClockTime
gst_alsasrc_get_timestamp (GstAlsaSrc * asrc)
{
snd_pcm_status_t *status;
snd_htimestamp_t tstamp;
GstClockTime timestamp;
snd_pcm_uframes_t avail;
gint err = -EPIPE;
if (G_UNLIKELY (!asrc)) {
GST_ERROR_OBJECT (asrc, "No alsa handle created yet !");
return GST_CLOCK_TIME_NONE;
}
if (G_UNLIKELY (snd_pcm_status_malloc (&status) != 0)) {
GST_ERROR_OBJECT (asrc, "snd_pcm_status_malloc failed");
return GST_CLOCK_TIME_NONE;
}
if (G_UNLIKELY (snd_pcm_status (asrc->handle, status) != 0)) {
GST_ERROR_OBJECT (asrc, "snd_pcm_status failed");
return GST_CLOCK_TIME_NONE;
}
2019-08-29 17:42:39 +00:00
/* in case an xrun condition has occurred we need to handle this */
if (snd_pcm_status_get_state (status) != SND_PCM_STATE_RUNNING) {
if (xrun_recovery (asrc, asrc->handle, err) < 0) {
GST_WARNING_OBJECT (asrc, "Could not recover from xrun condition !");
}
/* reload the status alsa status object, since recovery made it invalid */
if (G_UNLIKELY (snd_pcm_status (asrc->handle, status) != 0)) {
GST_ERROR_OBJECT (asrc, "snd_pcm_status failed");
}
}
/* get high resolution time stamp from driver */
snd_pcm_status_get_htstamp (status, &tstamp);
if (tstamp.tv_sec == 0 && tstamp.tv_nsec == 0)
return GST_CLOCK_TIME_NONE;
timestamp = GST_TIMESPEC_TO_TIME (tstamp);
/* max available frames sets the depth of the buffer */
avail = snd_pcm_status_get_avail (status);
/* calculate the timestamp of the next sample to be read */
timestamp -= gst_util_uint64_scale_int (avail, GST_SECOND, asrc->rate);
/* compensate for the fact that we really need the timestamp of the
* previously read data segment */
timestamp -= asrc->period_time * 1000;
snd_pcm_status_free (status);
GST_LOG_OBJECT (asrc, "ALSA timestamp : %" GST_TIME_FORMAT
", delay %lu", GST_TIME_ARGS (timestamp), avail);
return timestamp;
}
static guint
gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length,
GstClockTime * timestamp)
{
GstAlsaSrc *alsa;
gint err;
gint cptr;
guint8 *ptr = data;
alsa = GST_ALSA_SRC (asrc);
cptr = length / alsa->bpf;
GST_ALSA_SRC_LOCK (asrc);
while (cptr > 0) {
if ((err = snd_pcm_readi (alsa->handle, ptr, cptr)) < 0) {
if (err == -EAGAIN) {
GST_DEBUG_OBJECT (asrc, "Read error: %s", snd_strerror (err));
continue;
} else if (err == -ENODEV) {
goto device_disappeared;
} else if (xrun_recovery (alsa, alsa->handle, err) < 0) {
goto read_error;
}
continue;
}
ptr += snd_pcm_frames_to_bytes (alsa->handle, err);
cptr -= err;
}
GST_ALSA_SRC_UNLOCK (asrc);
/* if driver timestamps are enabled we need to return this here */
if (alsa->driver_timestamps && timestamp)
*timestamp = gst_alsasrc_get_timestamp (alsa);
return length - (cptr * alsa->bpf);
read_error:
{
GST_ALSA_SRC_UNLOCK (asrc);
return length; /* skip one period */
}
device_disappeared:
{
GST_ELEMENT_ERROR (asrc, RESOURCE, READ,
(_("Error recording from audio device. "
"The device has been disconnected.")), (NULL));
GST_ALSA_SRC_UNLOCK (asrc);
return (guint) - 1;
}
}
static guint
gst_alsasrc_delay (GstAudioSrc * asrc)
{
GstAlsaSrc *alsa;
snd_pcm_sframes_t delay;
int res;
alsa = GST_ALSA_SRC (asrc);
res = snd_pcm_delay (alsa->handle, &delay);
if (G_UNLIKELY (res < 0)) {
GST_DEBUG_OBJECT (alsa, "snd_pcm_delay returned %d", res);
delay = 0;
}
return CLAMP (delay, 0, alsa->buffer_size);
}
static void
gst_alsasrc_reset (GstAudioSrc * asrc)
{
GstAlsaSrc *alsa;
gint err;
alsa = GST_ALSA_SRC (asrc);
GST_ALSA_SRC_LOCK (asrc);
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_SRC_UNLOCK (asrc);
return;
/* ERRORS */
drop_error:
{
GST_ERROR_OBJECT (alsa, "alsa-reset: pcm drop error: %s",
snd_strerror (err));
GST_ALSA_SRC_UNLOCK (asrc);
return;
}
prepare_error:
{
GST_ERROR_OBJECT (alsa, "alsa-reset: pcm prepare error: %s",
snd_strerror (err));
GST_ALSA_SRC_UNLOCK (asrc);
return;
}
}