mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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1886 lines
52 KiB
C
1886 lines
52 KiB
C
/*
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* GStreamer pulseaudio plugin
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*
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* Copyright (c) 2004-2008 Lennart Poettering
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*
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* gst-pulse is free software; you can redistribute it and/or modify
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* it under the terms of the GNU Lesser General Public License as
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* published by the Free Software Foundation; either version 2.1 of the
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* License, or (at your option) any later version.
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*
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* gst-pulse is distributed in the hope that it will be useful, but
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* WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with gst-pulse; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301
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* USA.
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*/
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/**
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* SECTION:element-pulsesrc
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* @title: pulsesrc
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* @see_also: pulsesink
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*
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* This element captures audio from a
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* [PulseAudio sound server](http://www.pulseaudio.org).
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*
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* ## Example pipelines
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* |[
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* gst-launch-1.0 -v pulsesrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=alsasrc.ogg
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* ]| Record from a sound card using pulseaudio and encode to Ogg/Vorbis.
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*
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include <stdio.h>
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#include <gst/base/gstbasesrc.h>
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#include <gst/gsttaglist.h>
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#include <gst/audio/audio.h>
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#include "gstpulseelements.h"
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#include "pulsesrc.h"
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#include "pulseutil.h"
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GST_DEBUG_CATEGORY_EXTERN (pulse_debug);
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#define GST_CAT_DEFAULT pulse_debug
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#define DEFAULT_SERVER NULL
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#define DEFAULT_DEVICE NULL
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#define DEFAULT_CURRENT_DEVICE NULL
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#define DEFAULT_DEVICE_NAME NULL
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#define DEFAULT_VOLUME 1.0
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#define DEFAULT_MUTE FALSE
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#define MAX_VOLUME 10.0
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/* See the pulsesink code for notes on how we interact with the PA mainloop
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* thread. */
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enum
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{
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PROP_0,
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PROP_SERVER,
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PROP_DEVICE,
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PROP_DEVICE_NAME,
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PROP_CURRENT_DEVICE,
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PROP_CLIENT_NAME,
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PROP_STREAM_PROPERTIES,
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PROP_SOURCE_OUTPUT_INDEX,
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PROP_VOLUME,
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PROP_MUTE,
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PROP_LAST
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};
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static void gst_pulsesrc_destroy_stream (GstPulseSrc * pulsesrc);
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static void gst_pulsesrc_destroy_context (GstPulseSrc * pulsesrc);
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static void gst_pulsesrc_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_pulsesrc_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void gst_pulsesrc_finalize (GObject * object);
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static gboolean gst_pulsesrc_set_corked (GstPulseSrc * psrc, gboolean corked,
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gboolean wait);
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static gboolean gst_pulsesrc_open (GstAudioSrc * asrc);
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static gboolean gst_pulsesrc_close (GstAudioSrc * asrc);
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static gboolean gst_pulsesrc_prepare (GstAudioSrc * asrc,
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GstAudioRingBufferSpec * spec);
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static gboolean gst_pulsesrc_unprepare (GstAudioSrc * asrc);
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static guint gst_pulsesrc_read (GstAudioSrc * asrc, gpointer data,
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guint length, GstClockTime * timestamp);
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static guint gst_pulsesrc_delay (GstAudioSrc * asrc);
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static void gst_pulsesrc_reset (GstAudioSrc * src);
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static gboolean gst_pulsesrc_negotiate (GstBaseSrc * basesrc);
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static gboolean gst_pulsesrc_event (GstBaseSrc * basesrc, GstEvent * event);
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static GstStateChangeReturn gst_pulsesrc_change_state (GstElement *
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element, GstStateChange transition);
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static GstClockTime gst_pulsesrc_get_time (GstClock * clock, GstPulseSrc * src);
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#define gst_pulsesrc_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstPulseSrc, gst_pulsesrc, GST_TYPE_AUDIO_SRC,
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G_IMPLEMENT_INTERFACE (GST_TYPE_STREAM_VOLUME, NULL));
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GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (pulsesrc, "pulsesrc",
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GST_RANK_PRIMARY + 10, GST_TYPE_PULSESRC, pulse_element_init (plugin));
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static void
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gst_pulsesrc_class_init (GstPulseSrcClass * klass)
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{
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstAudioSrcClass *gstaudiosrc_class = GST_AUDIO_SRC_CLASS (klass);
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GstBaseSrcClass *gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
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GstCaps *caps;
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gchar *clientname;
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gobject_class->finalize = gst_pulsesrc_finalize;
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gobject_class->set_property = gst_pulsesrc_set_property;
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gobject_class->get_property = gst_pulsesrc_get_property;
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_pulsesrc_change_state);
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gstbasesrc_class->event = GST_DEBUG_FUNCPTR (gst_pulsesrc_event);
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gstbasesrc_class->negotiate = GST_DEBUG_FUNCPTR (gst_pulsesrc_negotiate);
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gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_pulsesrc_open);
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gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_pulsesrc_close);
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gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_pulsesrc_prepare);
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gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_pulsesrc_unprepare);
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gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_pulsesrc_read);
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gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_pulsesrc_delay);
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gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_pulsesrc_reset);
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/* Overwrite GObject fields */
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g_object_class_install_property (gobject_class,
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PROP_SERVER,
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g_param_spec_string ("server", "Server",
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"The PulseAudio server to connect to", DEFAULT_SERVER,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_DEVICE,
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g_param_spec_string ("device", "Device",
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"The PulseAudio source device to connect to", DEFAULT_DEVICE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_CURRENT_DEVICE,
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g_param_spec_string ("current-device", "Current Device",
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"The current PulseAudio source device", DEFAULT_CURRENT_DEVICE,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_DEVICE_NAME,
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g_param_spec_string ("device-name", "Device name",
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"Human-readable name of the sound device", DEFAULT_DEVICE_NAME,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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clientname = gst_pulse_client_name ();
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/**
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* GstPulseSrc:client-name
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*
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* The PulseAudio client name to use.
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*/
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g_object_class_install_property (gobject_class,
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PROP_CLIENT_NAME,
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g_param_spec_string ("client-name", "Client Name",
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"The PulseAudio client_name_to_use", clientname,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
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GST_PARAM_MUTABLE_READY));
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g_free (clientname);
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/**
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* GstPulseSrc:stream-properties:
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*
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* List of pulseaudio stream properties. A list of defined properties can be
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* found in the [pulseaudio api docs](http://0pointer.de/lennart/projects/pulseaudio/doxygen/proplist_8h.html).
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*
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* Below is an example for registering as a music application to pulseaudio.
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* |[
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* GstStructure *props;
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*
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* props = gst_structure_from_string ("props,media.role=music", NULL);
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* g_object_set (pulse, "stream-properties", props, NULL);
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* gst_structure_free (props);
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* ]|
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*/
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g_object_class_install_property (gobject_class,
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PROP_STREAM_PROPERTIES,
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g_param_spec_boxed ("stream-properties", "stream properties",
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"list of pulseaudio stream properties",
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GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstPulseSrc:source-output-index:
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*
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* The index of the PulseAudio source output corresponding to this element.
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*/
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g_object_class_install_property (gobject_class,
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PROP_SOURCE_OUTPUT_INDEX,
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g_param_spec_uint ("source-output-index", "source output index",
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"The index of the PulseAudio source output corresponding to this "
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"record stream", 0, G_MAXUINT, PA_INVALID_INDEX,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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gst_element_class_set_static_metadata (gstelement_class,
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"PulseAudio Audio Source",
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"Source/Audio",
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"Captures audio from a PulseAudio server", "Lennart Poettering");
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caps = gst_pulse_fix_pcm_caps (gst_caps_from_string (_PULSE_CAPS_PCM));
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gst_element_class_add_pad_template (gstelement_class,
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gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS, caps));
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gst_caps_unref (caps);
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/**
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* GstPulseSrc:volume:
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*
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* The volume of the record stream.
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*/
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g_object_class_install_property (gobject_class,
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PROP_VOLUME, g_param_spec_double ("volume", "Volume",
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"Linear volume of this stream, 1.0=100%",
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0.0, MAX_VOLUME, DEFAULT_VOLUME,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstPulseSrc:mute:
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*
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* Whether the stream is muted or not.
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*/
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g_object_class_install_property (gobject_class,
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PROP_MUTE, g_param_spec_boolean ("mute", "Mute",
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"Mute state of this stream",
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DEFAULT_MUTE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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}
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static void
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gst_pulsesrc_init (GstPulseSrc * pulsesrc)
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{
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pulsesrc->server = NULL;
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pulsesrc->device = NULL;
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pulsesrc->client_name = gst_pulse_client_name ();
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pulsesrc->device_description = NULL;
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pulsesrc->context = NULL;
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pulsesrc->stream = NULL;
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pulsesrc->stream_connected = FALSE;
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pulsesrc->source_output_idx = PA_INVALID_INDEX;
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pulsesrc->read_buffer = NULL;
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pulsesrc->read_buffer_length = 0;
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pa_sample_spec_init (&pulsesrc->sample_spec);
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pulsesrc->operation_success = FALSE;
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pulsesrc->paused = TRUE;
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pulsesrc->in_read = FALSE;
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pulsesrc->volume = DEFAULT_VOLUME;
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pulsesrc->volume_set = FALSE;
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pulsesrc->mute = DEFAULT_MUTE;
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pulsesrc->mute_set = FALSE;
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pulsesrc->notify = 0;
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pulsesrc->properties = NULL;
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pulsesrc->proplist = NULL;
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/* this should be the default but it isn't yet */
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gst_audio_base_src_set_slave_method (GST_AUDIO_BASE_SRC (pulsesrc),
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GST_AUDIO_BASE_SRC_SLAVE_SKEW);
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/* override with a custom clock */
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if (GST_AUDIO_BASE_SRC (pulsesrc)->clock)
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gst_object_unref (GST_AUDIO_BASE_SRC (pulsesrc)->clock);
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GST_AUDIO_BASE_SRC (pulsesrc)->clock =
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gst_audio_clock_new ("GstPulseSrcClock",
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(GstAudioClockGetTimeFunc) gst_pulsesrc_get_time, pulsesrc, NULL);
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}
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static void
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gst_pulsesrc_destroy_stream (GstPulseSrc * pulsesrc)
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{
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if (pulsesrc->stream) {
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pa_stream_disconnect (pulsesrc->stream);
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pa_stream_unref (pulsesrc->stream);
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pulsesrc->stream = NULL;
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pulsesrc->stream_connected = FALSE;
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pulsesrc->source_output_idx = PA_INVALID_INDEX;
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g_object_notify (G_OBJECT (pulsesrc), "source-output-index");
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}
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g_free (pulsesrc->device_description);
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pulsesrc->device_description = NULL;
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}
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static void
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gst_pulsesrc_destroy_context (GstPulseSrc * pulsesrc)
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{
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gst_pulsesrc_destroy_stream (pulsesrc);
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if (pulsesrc->context) {
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pa_context_disconnect (pulsesrc->context);
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/* Make sure we don't get any further callbacks */
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pa_context_set_state_callback (pulsesrc->context, NULL, NULL);
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pa_context_set_subscribe_callback (pulsesrc->context, NULL, NULL);
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pa_context_unref (pulsesrc->context);
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pulsesrc->context = NULL;
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}
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}
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static void
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gst_pulsesrc_finalize (GObject * object)
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{
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GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (object);
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g_free (pulsesrc->server);
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g_free (pulsesrc->device);
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g_free (pulsesrc->client_name);
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g_free (pulsesrc->current_source_name);
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if (pulsesrc->properties)
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gst_structure_free (pulsesrc->properties);
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if (pulsesrc->proplist)
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pa_proplist_free (pulsesrc->proplist);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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#define CONTEXT_OK(c) ((c) && PA_CONTEXT_IS_GOOD (pa_context_get_state ((c))))
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#define STREAM_OK(s) ((s) && PA_STREAM_IS_GOOD (pa_stream_get_state ((s))))
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static gboolean
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gst_pulsesrc_is_dead (GstPulseSrc * pulsesrc, gboolean check_stream)
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{
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if (!pulsesrc->stream_connected)
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return TRUE;
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if (!CONTEXT_OK (pulsesrc->context))
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goto error;
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if (check_stream && !STREAM_OK (pulsesrc->stream))
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goto error;
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return FALSE;
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error:
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{
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const gchar *err_str = pulsesrc->context ?
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pa_strerror (pa_context_errno (pulsesrc->context)) : NULL;
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GST_ELEMENT_ERROR ((pulsesrc), RESOURCE, FAILED, ("Disconnected: %s",
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err_str), (NULL));
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return TRUE;
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}
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}
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static void
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gst_pulsesrc_source_info_cb (pa_context * c, const pa_source_info * i, int eol,
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void *userdata)
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{
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GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (userdata);
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if (!i)
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goto done;
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g_free (pulsesrc->device_description);
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pulsesrc->device_description = g_strdup (i->description);
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done:
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pa_threaded_mainloop_signal (pulsesrc->mainloop, 0);
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}
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|
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static gchar *
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gst_pulsesrc_device_description (GstPulseSrc * pulsesrc)
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{
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pa_operation *o = NULL;
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gchar *t;
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|
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if (!pulsesrc->mainloop)
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goto no_mainloop;
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pa_threaded_mainloop_lock (pulsesrc->mainloop);
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|
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if (!(o = pa_context_get_source_info_by_name (pulsesrc->context,
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pulsesrc->device, gst_pulsesrc_source_info_cb, pulsesrc))) {
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GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
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("pa_stream_get_source_info() failed: %s",
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pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
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goto unlock;
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}
|
|
|
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while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
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|
|
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if (gst_pulsesrc_is_dead (pulsesrc, FALSE))
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goto unlock;
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pa_threaded_mainloop_wait (pulsesrc->mainloop);
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}
|
|
|
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unlock:
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|
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if (o)
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pa_operation_unref (o);
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|
|
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t = g_strdup (pulsesrc->device_description);
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|
|
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pa_threaded_mainloop_unlock (pulsesrc->mainloop);
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|
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return t;
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|
|
|
no_mainloop:
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{
|
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GST_DEBUG_OBJECT (pulsesrc, "have no mainloop");
|
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return NULL;
|
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}
|
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}
|
|
|
|
static void
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gst_pulsesrc_source_output_info_cb (pa_context * c,
|
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const pa_source_output_info * i, int eol, void *userdata)
|
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{
|
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GstPulseSrc *psrc;
|
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|
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psrc = GST_PULSESRC_CAST (userdata);
|
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|
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if (!i)
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goto done;
|
|
|
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/* If the index doesn't match our current stream,
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* it implies we just recreated the stream (caps change)
|
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*/
|
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if (i->index == psrc->source_output_idx) {
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psrc->volume = pa_sw_volume_to_linear (pa_cvolume_max (&i->volume));
|
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psrc->mute = i->mute;
|
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psrc->current_source_idx = i->source;
|
|
|
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if (G_UNLIKELY (psrc->volume > MAX_VOLUME)) {
|
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GST_WARNING_OBJECT (psrc, "Clipped volume from %f to %f",
|
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psrc->volume, MAX_VOLUME);
|
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psrc->volume = MAX_VOLUME;
|
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}
|
|
}
|
|
|
|
done:
|
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pa_threaded_mainloop_signal (psrc->mainloop, 0);
|
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}
|
|
|
|
static void
|
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gst_pulsesrc_get_source_output_info (GstPulseSrc * pulsesrc, gdouble * volume,
|
|
gboolean * mute)
|
|
{
|
|
pa_operation *o = NULL;
|
|
|
|
if (!pulsesrc->mainloop)
|
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goto no_mainloop;
|
|
|
|
if (pulsesrc->source_output_idx == PA_INVALID_INDEX)
|
|
goto no_index;
|
|
|
|
pa_threaded_mainloop_lock (pulsesrc->mainloop);
|
|
|
|
if (!(o = pa_context_get_source_output_info (pulsesrc->context,
|
|
pulsesrc->source_output_idx, gst_pulsesrc_source_output_info_cb,
|
|
pulsesrc)))
|
|
goto info_failed;
|
|
|
|
while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
|
|
pa_threaded_mainloop_wait (pulsesrc->mainloop);
|
|
if (gst_pulsesrc_is_dead (pulsesrc, TRUE))
|
|
goto unlock;
|
|
}
|
|
|
|
unlock:
|
|
|
|
if (volume)
|
|
*volume = pulsesrc->volume;
|
|
if (mute)
|
|
*mute = pulsesrc->mute;
|
|
|
|
if (o)
|
|
pa_operation_unref (o);
|
|
|
|
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
no_mainloop:
|
|
{
|
|
GST_DEBUG_OBJECT (pulsesrc, "we have no mainloop");
|
|
if (volume)
|
|
*volume = pulsesrc->volume;
|
|
if (mute)
|
|
*mute = pulsesrc->mute;
|
|
return;
|
|
}
|
|
no_index:
|
|
{
|
|
GST_DEBUG_OBJECT (pulsesrc, "we don't have a stream index");
|
|
if (volume)
|
|
*volume = pulsesrc->volume;
|
|
if (mute)
|
|
*mute = pulsesrc->mute;
|
|
return;
|
|
}
|
|
info_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
|
|
("pa_context_get_source_output_info() failed: %s",
|
|
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
|
|
goto unlock;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_pulsesrc_current_source_info_cb (pa_context * c, const pa_source_info * i,
|
|
int eol, void *userdata)
|
|
{
|
|
GstPulseSrc *psrc;
|
|
|
|
psrc = GST_PULSESRC_CAST (userdata);
|
|
|
|
if (!i)
|
|
goto done;
|
|
|
|
/* If the index doesn't match our current stream,
|
|
* it implies we just recreated the stream (caps change)
|
|
*/
|
|
if (i->index == psrc->current_source_idx) {
|
|
g_free (psrc->current_source_name);
|
|
psrc->current_source_name = g_strdup (i->name);
|
|
}
|
|
|
|
done:
|
|
pa_threaded_mainloop_signal (psrc->mainloop, 0);
|
|
}
|
|
|
|
static gchar *
|
|
gst_pulsesrc_get_current_device (GstPulseSrc * pulsesrc)
|
|
{
|
|
pa_operation *o = NULL;
|
|
gchar *current_src;
|
|
|
|
if (!pulsesrc->mainloop)
|
|
goto no_mainloop;
|
|
|
|
if (pulsesrc->source_output_idx == PA_INVALID_INDEX)
|
|
goto no_index;
|
|
|
|
gst_pulsesrc_get_source_output_info (pulsesrc, NULL, NULL);
|
|
|
|
pa_threaded_mainloop_lock (pulsesrc->mainloop);
|
|
|
|
|
|
if (!(o = pa_context_get_source_info_by_index (pulsesrc->context,
|
|
pulsesrc->current_source_idx, gst_pulsesrc_current_source_info_cb,
|
|
pulsesrc)))
|
|
goto info_failed;
|
|
|
|
while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
|
|
pa_threaded_mainloop_wait (pulsesrc->mainloop);
|
|
if (gst_pulsesrc_is_dead (pulsesrc, TRUE))
|
|
goto unlock;
|
|
}
|
|
|
|
unlock:
|
|
|
|
current_src = g_strdup (pulsesrc->current_source_name);
|
|
|
|
if (o)
|
|
pa_operation_unref (o);
|
|
|
|
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
|
|
|
|
return current_src;
|
|
|
|
/* ERRORS */
|
|
no_mainloop:
|
|
{
|
|
GST_DEBUG_OBJECT (pulsesrc, "we have no mainloop");
|
|
return NULL;
|
|
}
|
|
no_index:
|
|
{
|
|
GST_DEBUG_OBJECT (pulsesrc, "we don't have a stream index");
|
|
return NULL;
|
|
}
|
|
info_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
|
|
("pa_context_get_source_output_info() failed: %s",
|
|
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
|
|
goto unlock;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_pulsesrc_set_stream_volume (GstPulseSrc * pulsesrc, gdouble volume)
|
|
{
|
|
pa_cvolume v;
|
|
pa_operation *o = NULL;
|
|
|
|
if (!pulsesrc->mainloop)
|
|
goto no_mainloop;
|
|
|
|
if (pulsesrc->source_output_idx == PA_INVALID_INDEX)
|
|
goto no_index;
|
|
|
|
pa_threaded_mainloop_lock (pulsesrc->mainloop);
|
|
|
|
GST_DEBUG_OBJECT (pulsesrc, "setting volume to %f", volume);
|
|
|
|
gst_pulse_cvolume_from_linear (&v, pulsesrc->sample_spec.channels, volume);
|
|
|
|
if (!(o = pa_context_set_source_output_volume (pulsesrc->context,
|
|
pulsesrc->source_output_idx, &v, NULL, NULL)))
|
|
goto volume_failed;
|
|
|
|
/* We don't really care about the result of this call */
|
|
unlock:
|
|
|
|
if (o)
|
|
pa_operation_unref (o);
|
|
|
|
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
no_mainloop:
|
|
{
|
|
pulsesrc->volume = volume;
|
|
pulsesrc->volume_set = TRUE;
|
|
GST_DEBUG_OBJECT (pulsesrc, "we have no mainloop");
|
|
return;
|
|
}
|
|
no_index:
|
|
{
|
|
pulsesrc->volume = volume;
|
|
pulsesrc->volume_set = TRUE;
|
|
GST_DEBUG_OBJECT (pulsesrc, "we don't have a stream index");
|
|
return;
|
|
}
|
|
volume_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
|
|
("pa_stream_set_source_output_volume() failed: %s",
|
|
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
|
|
goto unlock;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_pulsesrc_set_stream_mute (GstPulseSrc * pulsesrc, gboolean mute)
|
|
{
|
|
pa_operation *o = NULL;
|
|
|
|
if (!pulsesrc->mainloop)
|
|
goto no_mainloop;
|
|
|
|
if (pulsesrc->source_output_idx == PA_INVALID_INDEX)
|
|
goto no_index;
|
|
|
|
pa_threaded_mainloop_lock (pulsesrc->mainloop);
|
|
|
|
GST_DEBUG_OBJECT (pulsesrc, "setting mute state to %d", mute);
|
|
|
|
if (!(o = pa_context_set_source_output_mute (pulsesrc->context,
|
|
pulsesrc->source_output_idx, mute, NULL, NULL)))
|
|
goto mute_failed;
|
|
|
|
/* We don't really care about the result of this call */
|
|
unlock:
|
|
|
|
if (o)
|
|
pa_operation_unref (o);
|
|
|
|
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
no_mainloop:
|
|
{
|
|
pulsesrc->mute = mute;
|
|
pulsesrc->mute_set = TRUE;
|
|
GST_DEBUG_OBJECT (pulsesrc, "we have no mainloop");
|
|
return;
|
|
}
|
|
no_index:
|
|
{
|
|
pulsesrc->mute = mute;
|
|
pulsesrc->mute_set = TRUE;
|
|
GST_DEBUG_OBJECT (pulsesrc, "we don't have a stream index");
|
|
return;
|
|
}
|
|
mute_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
|
|
("pa_stream_set_source_output_mute() failed: %s",
|
|
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
|
|
goto unlock;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_pulsesrc_set_stream_device (GstPulseSrc * pulsesrc, const gchar * device)
|
|
{
|
|
pa_operation *o = NULL;
|
|
|
|
if (!pulsesrc->mainloop)
|
|
goto no_mainloop;
|
|
|
|
if (pulsesrc->source_output_idx == PA_INVALID_INDEX)
|
|
goto no_index;
|
|
|
|
pa_threaded_mainloop_lock (pulsesrc->mainloop);
|
|
|
|
GST_DEBUG_OBJECT (pulsesrc, "setting stream device to %s", device);
|
|
|
|
if (!(o = pa_context_move_source_output_by_name (pulsesrc->context,
|
|
pulsesrc->source_output_idx, device, NULL, NULL)))
|
|
goto move_failed;
|
|
|
|
unlock:
|
|
|
|
if (o)
|
|
pa_operation_unref (o);
|
|
|
|
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
no_mainloop:
|
|
{
|
|
GST_DEBUG_OBJECT (pulsesrc, "we have no mainloop");
|
|
return;
|
|
}
|
|
no_index:
|
|
{
|
|
GST_DEBUG_OBJECT (pulsesrc, "we don't have a stream index");
|
|
return;
|
|
}
|
|
move_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
|
|
("pa_context_move_source_output_by_name(%s) failed: %s",
|
|
device, pa_strerror (pa_context_errno (pulsesrc->context))),
|
|
(NULL));
|
|
goto unlock;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_pulsesrc_set_property (GObject * object,
|
|
guint prop_id, const GValue * value, GParamSpec * pspec)
|
|
{
|
|
|
|
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_SERVER:
|
|
g_free (pulsesrc->server);
|
|
pulsesrc->server = g_value_dup_string (value);
|
|
break;
|
|
case PROP_DEVICE:
|
|
g_free (pulsesrc->device);
|
|
pulsesrc->device = g_value_dup_string (value);
|
|
gst_pulsesrc_set_stream_device (pulsesrc, pulsesrc->device);
|
|
break;
|
|
case PROP_CLIENT_NAME:
|
|
g_free (pulsesrc->client_name);
|
|
if (!g_value_get_string (value)) {
|
|
GST_WARNING_OBJECT (pulsesrc,
|
|
"Empty PulseAudio client name not allowed. Resetting to default value");
|
|
pulsesrc->client_name = gst_pulse_client_name ();
|
|
} else
|
|
pulsesrc->client_name = g_value_dup_string (value);
|
|
break;
|
|
case PROP_STREAM_PROPERTIES:
|
|
if (pulsesrc->properties)
|
|
gst_structure_free (pulsesrc->properties);
|
|
pulsesrc->properties =
|
|
gst_structure_copy (gst_value_get_structure (value));
|
|
if (pulsesrc->proplist)
|
|
pa_proplist_free (pulsesrc->proplist);
|
|
pulsesrc->proplist = gst_pulse_make_proplist (pulsesrc->properties);
|
|
break;
|
|
case PROP_VOLUME:
|
|
gst_pulsesrc_set_stream_volume (pulsesrc, g_value_get_double (value));
|
|
break;
|
|
case PROP_MUTE:
|
|
gst_pulsesrc_set_stream_mute (pulsesrc, g_value_get_boolean (value));
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_pulsesrc_get_property (GObject * object,
|
|
guint prop_id, GValue * value, GParamSpec * pspec)
|
|
{
|
|
|
|
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_SERVER:
|
|
g_value_set_string (value, pulsesrc->server);
|
|
break;
|
|
case PROP_DEVICE:
|
|
g_value_set_string (value, pulsesrc->device);
|
|
break;
|
|
case PROP_CURRENT_DEVICE:
|
|
{
|
|
gchar *current_device = gst_pulsesrc_get_current_device (pulsesrc);
|
|
if (current_device)
|
|
g_value_take_string (value, current_device);
|
|
else
|
|
g_value_set_string (value, "");
|
|
break;
|
|
}
|
|
case PROP_DEVICE_NAME:
|
|
g_value_take_string (value, gst_pulsesrc_device_description (pulsesrc));
|
|
break;
|
|
case PROP_CLIENT_NAME:
|
|
g_value_set_string (value, pulsesrc->client_name);
|
|
break;
|
|
case PROP_STREAM_PROPERTIES:
|
|
gst_value_set_structure (value, pulsesrc->properties);
|
|
break;
|
|
case PROP_SOURCE_OUTPUT_INDEX:
|
|
g_value_set_uint (value, pulsesrc->source_output_idx);
|
|
break;
|
|
case PROP_VOLUME:
|
|
{
|
|
gdouble volume;
|
|
gst_pulsesrc_get_source_output_info (pulsesrc, &volume, NULL);
|
|
g_value_set_double (value, volume);
|
|
break;
|
|
}
|
|
case PROP_MUTE:
|
|
{
|
|
gboolean mute;
|
|
gst_pulsesrc_get_source_output_info (pulsesrc, NULL, &mute);
|
|
g_value_set_boolean (value, mute);
|
|
break;
|
|
}
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_pulsesrc_context_state_cb (pa_context * c, void *userdata)
|
|
{
|
|
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (userdata);
|
|
|
|
switch (pa_context_get_state (c)) {
|
|
case PA_CONTEXT_READY:
|
|
case PA_CONTEXT_TERMINATED:
|
|
case PA_CONTEXT_FAILED:
|
|
pa_threaded_mainloop_signal (pulsesrc->mainloop, 0);
|
|
break;
|
|
|
|
case PA_CONTEXT_UNCONNECTED:
|
|
case PA_CONTEXT_CONNECTING:
|
|
case PA_CONTEXT_AUTHORIZING:
|
|
case PA_CONTEXT_SETTING_NAME:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_pulsesrc_stream_state_cb (pa_stream * s, void *userdata)
|
|
{
|
|
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (userdata);
|
|
|
|
switch (pa_stream_get_state (s)) {
|
|
|
|
case PA_STREAM_READY:
|
|
case PA_STREAM_FAILED:
|
|
case PA_STREAM_TERMINATED:
|
|
pa_threaded_mainloop_signal (pulsesrc->mainloop, 0);
|
|
break;
|
|
|
|
case PA_STREAM_UNCONNECTED:
|
|
case PA_STREAM_CREATING:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_pulsesrc_stream_request_cb (pa_stream * s, size_t length, void *userdata)
|
|
{
|
|
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (userdata);
|
|
|
|
GST_LOG_OBJECT (pulsesrc, "got request for length %" G_GSIZE_FORMAT, length);
|
|
|
|
if (pulsesrc->in_read) {
|
|
/* only signal when reading */
|
|
pa_threaded_mainloop_signal (pulsesrc->mainloop, 0);
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_pulsesrc_stream_latency_update_cb (pa_stream * s, void *userdata)
|
|
{
|
|
const pa_timing_info *info;
|
|
pa_usec_t source_usec;
|
|
|
|
info = pa_stream_get_timing_info (s);
|
|
|
|
if (!info) {
|
|
GST_LOG_OBJECT (GST_PULSESRC_CAST (userdata),
|
|
"latency update (information unknown)");
|
|
return;
|
|
}
|
|
source_usec = info->configured_source_usec;
|
|
|
|
GST_LOG_OBJECT (GST_PULSESRC_CAST (userdata),
|
|
"latency_update, %" G_GUINT64_FORMAT ", %d:%" G_GINT64_FORMAT ", %d:%"
|
|
G_GUINT64_FORMAT ", %" G_GUINT64_FORMAT ", %" G_GUINT64_FORMAT,
|
|
GST_TIMEVAL_TO_TIME (info->timestamp), info->write_index_corrupt,
|
|
info->write_index, info->read_index_corrupt, info->read_index,
|
|
info->source_usec, source_usec);
|
|
}
|
|
|
|
static void
|
|
gst_pulsesrc_stream_underflow_cb (pa_stream * s, void *userdata)
|
|
{
|
|
GST_WARNING_OBJECT (GST_PULSESRC_CAST (userdata), "Got underflow");
|
|
}
|
|
|
|
static void
|
|
gst_pulsesrc_stream_overflow_cb (pa_stream * s, void *userdata)
|
|
{
|
|
GST_WARNING_OBJECT (GST_PULSESRC_CAST (userdata), "Got overflow");
|
|
}
|
|
|
|
static void
|
|
gst_pulsesrc_context_subscribe_cb (pa_context * c,
|
|
pa_subscription_event_type_t t, uint32_t idx, void *userdata)
|
|
{
|
|
GstPulseSrc *psrc = GST_PULSESRC (userdata);
|
|
|
|
if (t != (PA_SUBSCRIPTION_EVENT_SOURCE_OUTPUT | PA_SUBSCRIPTION_EVENT_CHANGE)
|
|
&& t != (PA_SUBSCRIPTION_EVENT_SOURCE_OUTPUT | PA_SUBSCRIPTION_EVENT_NEW))
|
|
return;
|
|
|
|
if (idx != psrc->source_output_idx)
|
|
return;
|
|
|
|
/* Actually this event is also triggered when other properties of the stream
|
|
* change that are unrelated to the volume. However it is probably cheaper to
|
|
* signal the change here and check for the volume when the GObject property
|
|
* is read instead of querying it always. */
|
|
|
|
/* inform streaming thread to notify */
|
|
g_atomic_int_compare_and_exchange (&psrc->notify, 0, 1);
|
|
}
|
|
|
|
static gboolean
|
|
gst_pulsesrc_open (GstAudioSrc * asrc)
|
|
{
|
|
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
|
|
|
|
pa_threaded_mainloop_lock (pulsesrc->mainloop);
|
|
|
|
g_assert (!pulsesrc->context);
|
|
g_assert (!pulsesrc->stream);
|
|
|
|
GST_DEBUG_OBJECT (pulsesrc, "opening device");
|
|
|
|
if (!(pulsesrc->context =
|
|
pa_context_new (pa_threaded_mainloop_get_api (pulsesrc->mainloop),
|
|
pulsesrc->client_name))) {
|
|
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Failed to create context"),
|
|
(NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
pa_context_set_state_callback (pulsesrc->context,
|
|
gst_pulsesrc_context_state_cb, pulsesrc);
|
|
pa_context_set_subscribe_callback (pulsesrc->context,
|
|
gst_pulsesrc_context_subscribe_cb, pulsesrc);
|
|
|
|
GST_DEBUG_OBJECT (pulsesrc, "connect to server %s",
|
|
GST_STR_NULL (pulsesrc->server));
|
|
|
|
if (pa_context_connect (pulsesrc->context, pulsesrc->server, 0, NULL) < 0) {
|
|
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Failed to connect: %s",
|
|
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
for (;;) {
|
|
pa_context_state_t state;
|
|
|
|
state = pa_context_get_state (pulsesrc->context);
|
|
|
|
if (!PA_CONTEXT_IS_GOOD (state)) {
|
|
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Failed to connect: %s",
|
|
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
if (state == PA_CONTEXT_READY)
|
|
break;
|
|
|
|
/* Wait until the context is ready */
|
|
pa_threaded_mainloop_wait (pulsesrc->mainloop);
|
|
}
|
|
GST_DEBUG_OBJECT (pulsesrc, "connected");
|
|
|
|
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
unlock_and_fail:
|
|
{
|
|
gst_pulsesrc_destroy_context (pulsesrc);
|
|
|
|
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
|
|
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_pulsesrc_close (GstAudioSrc * asrc)
|
|
{
|
|
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
|
|
|
|
pa_threaded_mainloop_lock (pulsesrc->mainloop);
|
|
gst_pulsesrc_destroy_context (pulsesrc);
|
|
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_pulsesrc_unprepare (GstAudioSrc * asrc)
|
|
{
|
|
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
|
|
|
|
pa_threaded_mainloop_lock (pulsesrc->mainloop);
|
|
gst_pulsesrc_destroy_stream (pulsesrc);
|
|
|
|
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
|
|
|
|
pulsesrc->read_buffer = NULL;
|
|
pulsesrc->read_buffer_length = 0;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static guint
|
|
gst_pulsesrc_read (GstAudioSrc * asrc, gpointer data, guint length,
|
|
GstClockTime * timestamp)
|
|
{
|
|
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
|
|
size_t sum = 0;
|
|
|
|
if (g_atomic_int_compare_and_exchange (&pulsesrc->notify, 1, 0)) {
|
|
g_object_notify (G_OBJECT (pulsesrc), "volume");
|
|
g_object_notify (G_OBJECT (pulsesrc), "mute");
|
|
g_object_notify (G_OBJECT (pulsesrc), "current-device");
|
|
}
|
|
|
|
pa_threaded_mainloop_lock (pulsesrc->mainloop);
|
|
pulsesrc->in_read = TRUE;
|
|
|
|
if (!pulsesrc->stream_connected)
|
|
goto not_connected;
|
|
|
|
if (pulsesrc->paused)
|
|
goto was_paused;
|
|
|
|
while (length > 0) {
|
|
size_t l;
|
|
|
|
GST_LOG_OBJECT (pulsesrc, "reading %u bytes", length);
|
|
|
|
/*check if we have a leftover buffer */
|
|
if (!pulsesrc->read_buffer) {
|
|
for (;;) {
|
|
if (gst_pulsesrc_is_dead (pulsesrc, TRUE))
|
|
goto unlock_and_fail;
|
|
|
|
/* read all available data, we keep a pointer to the data and the length
|
|
* and take from it what we need. */
|
|
if (pa_stream_peek (pulsesrc->stream, &pulsesrc->read_buffer,
|
|
&pulsesrc->read_buffer_length) < 0)
|
|
goto peek_failed;
|
|
|
|
GST_LOG_OBJECT (pulsesrc, "have data of %" G_GSIZE_FORMAT " bytes",
|
|
pulsesrc->read_buffer_length);
|
|
|
|
/* if we have data, process if */
|
|
if (pulsesrc->read_buffer && pulsesrc->read_buffer_length)
|
|
break;
|
|
|
|
/* now wait for more data to become available */
|
|
GST_LOG_OBJECT (pulsesrc, "waiting for data");
|
|
pa_threaded_mainloop_wait (pulsesrc->mainloop);
|
|
|
|
if (pulsesrc->paused)
|
|
goto was_paused;
|
|
}
|
|
}
|
|
|
|
l = pulsesrc->read_buffer_length >
|
|
length ? length : pulsesrc->read_buffer_length;
|
|
|
|
memcpy (data, pulsesrc->read_buffer, l);
|
|
|
|
pulsesrc->read_buffer = (const guint8 *) pulsesrc->read_buffer + l;
|
|
pulsesrc->read_buffer_length -= l;
|
|
|
|
data = (guint8 *) data + l;
|
|
length -= l;
|
|
sum += l;
|
|
|
|
if (pulsesrc->read_buffer_length <= 0) {
|
|
/* we copied all of the data, drop it now */
|
|
if (pa_stream_drop (pulsesrc->stream) < 0)
|
|
goto drop_failed;
|
|
|
|
/* reset pointer to data */
|
|
pulsesrc->read_buffer = NULL;
|
|
pulsesrc->read_buffer_length = 0;
|
|
}
|
|
}
|
|
|
|
pulsesrc->in_read = FALSE;
|
|
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
|
|
|
|
return sum;
|
|
|
|
/* ERRORS */
|
|
not_connected:
|
|
{
|
|
GST_LOG_OBJECT (pulsesrc, "we are not connected");
|
|
goto unlock_and_fail;
|
|
}
|
|
was_paused:
|
|
{
|
|
GST_LOG_OBJECT (pulsesrc, "we are paused");
|
|
goto unlock_and_fail;
|
|
}
|
|
peek_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
|
|
("pa_stream_peek() failed: %s",
|
|
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
drop_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
|
|
("pa_stream_drop() failed: %s",
|
|
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
unlock_and_fail:
|
|
{
|
|
pulsesrc->in_read = FALSE;
|
|
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
|
|
|
|
return (guint) - 1;
|
|
}
|
|
}
|
|
|
|
/* return the delay in samples */
|
|
static guint
|
|
gst_pulsesrc_delay (GstAudioSrc * asrc)
|
|
{
|
|
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
|
|
pa_usec_t t;
|
|
int negative, res;
|
|
guint result;
|
|
|
|
pa_threaded_mainloop_lock (pulsesrc->mainloop);
|
|
if (gst_pulsesrc_is_dead (pulsesrc, TRUE))
|
|
goto server_dead;
|
|
|
|
/* get the latency, this can fail when we don't have a latency update yet.
|
|
* We don't want to wait for latency updates here but we just return 0. */
|
|
res = pa_stream_get_latency (pulsesrc->stream, &t, &negative);
|
|
|
|
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
|
|
|
|
if (res < 0) {
|
|
GST_DEBUG_OBJECT (pulsesrc, "could not get latency");
|
|
result = 0;
|
|
} else {
|
|
if (negative)
|
|
result = 0;
|
|
else
|
|
result = (guint) ((t * pulsesrc->sample_spec.rate) / 1000000LL);
|
|
}
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
server_dead:
|
|
{
|
|
GST_DEBUG_OBJECT (pulsesrc, "the server is dead");
|
|
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
|
|
return 0;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_pulsesrc_create_stream (GstPulseSrc * pulsesrc, GstCaps ** caps,
|
|
GstAudioRingBufferSpec * rspec)
|
|
{
|
|
pa_channel_map channel_map;
|
|
const pa_channel_map *m;
|
|
GstStructure *s;
|
|
gboolean need_channel_layout = FALSE;
|
|
GstAudioRingBufferSpec new_spec, *spec = NULL;
|
|
const gchar *name;
|
|
int i;
|
|
|
|
/* If we already have a stream (renegotiation), free it first */
|
|
if (pulsesrc->stream)
|
|
gst_pulsesrc_destroy_stream (pulsesrc);
|
|
|
|
if (rspec) {
|
|
/* Post-negotiation, we already have a ringbuffer spec, so we just need to
|
|
* use it to create a stream. */
|
|
spec = rspec;
|
|
|
|
/* At this point, we expect the channel-mask to be set in caps, so we just
|
|
* use that */
|
|
if (!gst_pulse_gst_to_channel_map (&channel_map, spec))
|
|
goto invalid_spec;
|
|
|
|
} else if (caps) {
|
|
/* At negotiation time, we get a fixed caps and use it to set up a stream */
|
|
s = gst_caps_get_structure (*caps, 0);
|
|
gst_structure_get_int (s, "channels", &new_spec.info.channels);
|
|
if (!gst_structure_has_field (s, "channel-mask")) {
|
|
if (new_spec.info.channels == 1) {
|
|
pa_channel_map_init_mono (&channel_map);
|
|
} else if (new_spec.info.channels == 2) {
|
|
pa_channel_map_init_stereo (&channel_map);
|
|
} else {
|
|
need_channel_layout = TRUE;
|
|
gst_structure_set (s, "channel-mask", GST_TYPE_BITMASK,
|
|
G_GUINT64_CONSTANT (0), NULL);
|
|
}
|
|
}
|
|
|
|
memset (&new_spec, 0, sizeof (GstAudioRingBufferSpec));
|
|
new_spec.latency_time = GST_SECOND;
|
|
if (!gst_audio_ring_buffer_parse_caps (&new_spec, *caps))
|
|
goto invalid_caps;
|
|
|
|
/* Keep the refcount of the caps at 1 to make them writable */
|
|
gst_caps_unref (new_spec.caps);
|
|
|
|
if (!need_channel_layout
|
|
&& !gst_pulse_gst_to_channel_map (&channel_map, &new_spec)) {
|
|
need_channel_layout = TRUE;
|
|
gst_structure_set (s, "channel-mask", GST_TYPE_BITMASK,
|
|
G_GUINT64_CONSTANT (0), NULL);
|
|
for (i = 0; i < G_N_ELEMENTS (new_spec.info.position); i++)
|
|
new_spec.info.position[i] = GST_AUDIO_CHANNEL_POSITION_INVALID;
|
|
}
|
|
|
|
spec = &new_spec;
|
|
} else {
|
|
/* !rspec && !caps */
|
|
g_assert_not_reached ();
|
|
}
|
|
|
|
if (!gst_pulse_fill_sample_spec (spec, &pulsesrc->sample_spec))
|
|
goto invalid_spec;
|
|
|
|
pa_threaded_mainloop_lock (pulsesrc->mainloop);
|
|
|
|
if (!pulsesrc->context)
|
|
goto bad_context;
|
|
|
|
name = "Record Stream";
|
|
if (pulsesrc->proplist) {
|
|
if (!(pulsesrc->stream = pa_stream_new_with_proplist (pulsesrc->context,
|
|
name, &pulsesrc->sample_spec,
|
|
(need_channel_layout) ? NULL : &channel_map,
|
|
pulsesrc->proplist)))
|
|
goto create_failed;
|
|
|
|
} else if (!(pulsesrc->stream = pa_stream_new (pulsesrc->context,
|
|
name, &pulsesrc->sample_spec,
|
|
(need_channel_layout) ? NULL : &channel_map)))
|
|
goto create_failed;
|
|
|
|
if (caps) {
|
|
m = pa_stream_get_channel_map (pulsesrc->stream);
|
|
gst_pulse_channel_map_to_gst (m, &new_spec);
|
|
gst_audio_channel_positions_to_valid_order (new_spec.info.position,
|
|
new_spec.info.channels);
|
|
gst_caps_unref (*caps);
|
|
*caps = gst_audio_info_to_caps (&new_spec.info);
|
|
|
|
GST_DEBUG_OBJECT (pulsesrc, "Caps are %" GST_PTR_FORMAT, *caps);
|
|
}
|
|
|
|
|
|
pa_stream_set_state_callback (pulsesrc->stream, gst_pulsesrc_stream_state_cb,
|
|
pulsesrc);
|
|
pa_stream_set_read_callback (pulsesrc->stream, gst_pulsesrc_stream_request_cb,
|
|
pulsesrc);
|
|
pa_stream_set_underflow_callback (pulsesrc->stream,
|
|
gst_pulsesrc_stream_underflow_cb, pulsesrc);
|
|
pa_stream_set_overflow_callback (pulsesrc->stream,
|
|
gst_pulsesrc_stream_overflow_cb, pulsesrc);
|
|
pa_stream_set_latency_update_callback (pulsesrc->stream,
|
|
gst_pulsesrc_stream_latency_update_cb, pulsesrc);
|
|
|
|
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
invalid_caps:
|
|
{
|
|
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, SETTINGS,
|
|
("Can't parse caps."), (NULL));
|
|
goto fail;
|
|
}
|
|
invalid_spec:
|
|
{
|
|
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, SETTINGS,
|
|
("Invalid sample specification."), (NULL));
|
|
goto fail;
|
|
}
|
|
bad_context:
|
|
{
|
|
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Bad context"), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
create_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
|
|
("Failed to create stream: %s",
|
|
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
unlock_and_fail:
|
|
{
|
|
gst_pulsesrc_destroy_stream (pulsesrc);
|
|
|
|
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
|
|
|
|
fail:
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_pulsesrc_event (GstBaseSrc * basesrc, GstEvent * event)
|
|
{
|
|
GST_DEBUG_OBJECT (basesrc, "handle event %" GST_PTR_FORMAT, event);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_RECONFIGURE:
|
|
gst_pad_check_reconfigure (GST_BASE_SRC_PAD (basesrc));
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
return GST_BASE_SRC_CLASS (parent_class)->event (basesrc, event);
|
|
}
|
|
|
|
/* This is essentially gst_base_src_negotiate_default() but the caps
|
|
* are guaranteed to have a channel layout for > 2 channels
|
|
*/
|
|
static gboolean
|
|
gst_pulsesrc_negotiate (GstBaseSrc * basesrc)
|
|
{
|
|
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (basesrc);
|
|
GstCaps *thiscaps;
|
|
GstCaps *caps = NULL;
|
|
GstCaps *peercaps = NULL;
|
|
gboolean result = FALSE;
|
|
|
|
/* first see what is possible on our source pad */
|
|
thiscaps = gst_pad_query_caps (GST_BASE_SRC_PAD (basesrc), NULL);
|
|
GST_DEBUG_OBJECT (basesrc, "caps of src: %" GST_PTR_FORMAT, thiscaps);
|
|
/* nothing or anything is allowed, we're done */
|
|
if (thiscaps == NULL || gst_caps_is_any (thiscaps))
|
|
goto no_nego_needed;
|
|
|
|
/* get the peer caps */
|
|
peercaps = gst_pad_peer_query_caps (GST_BASE_SRC_PAD (basesrc), NULL);
|
|
GST_DEBUG_OBJECT (basesrc, "caps of peer: %" GST_PTR_FORMAT, peercaps);
|
|
if (peercaps) {
|
|
/* get intersection */
|
|
caps = gst_caps_intersect (thiscaps, peercaps);
|
|
GST_DEBUG_OBJECT (basesrc, "intersect: %" GST_PTR_FORMAT, caps);
|
|
gst_caps_unref (thiscaps);
|
|
gst_caps_unref (peercaps);
|
|
} else {
|
|
/* no peer, work with our own caps then */
|
|
caps = thiscaps;
|
|
}
|
|
if (caps) {
|
|
/* take first (and best, since they are sorted) possibility */
|
|
caps = gst_caps_truncate (caps);
|
|
|
|
/* now fixate */
|
|
if (!gst_caps_is_empty (caps)) {
|
|
caps = GST_BASE_SRC_CLASS (parent_class)->fixate (basesrc, caps);
|
|
GST_DEBUG_OBJECT (basesrc, "fixated to: %" GST_PTR_FORMAT, caps);
|
|
|
|
if (gst_caps_is_any (caps)) {
|
|
/* hmm, still anything, so element can do anything and
|
|
* nego is not needed */
|
|
result = TRUE;
|
|
} else if (gst_caps_is_fixed (caps)) {
|
|
/* yay, fixed caps, use those then */
|
|
result = gst_pulsesrc_create_stream (pulsesrc, &caps, NULL);
|
|
if (result)
|
|
result = gst_base_src_set_caps (basesrc, caps);
|
|
}
|
|
}
|
|
gst_caps_unref (caps);
|
|
}
|
|
return result;
|
|
|
|
no_nego_needed:
|
|
{
|
|
GST_DEBUG_OBJECT (basesrc, "no negotiation needed");
|
|
if (thiscaps)
|
|
gst_caps_unref (thiscaps);
|
|
return TRUE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_pulsesrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
|
|
{
|
|
pa_buffer_attr wanted;
|
|
const pa_buffer_attr *actual;
|
|
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
|
|
pa_stream_flags_t flags;
|
|
pa_operation *o;
|
|
GstAudioClock *clock;
|
|
|
|
pa_threaded_mainloop_lock (pulsesrc->mainloop);
|
|
|
|
if (!pulsesrc->stream)
|
|
gst_pulsesrc_create_stream (pulsesrc, NULL, spec);
|
|
|
|
{
|
|
GstAudioRingBufferSpec s = *spec;
|
|
const pa_channel_map *m;
|
|
|
|
m = pa_stream_get_channel_map (pulsesrc->stream);
|
|
gst_pulse_channel_map_to_gst (m, &s);
|
|
gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SRC
|
|
(pulsesrc)->ringbuffer, s.info.position);
|
|
}
|
|
|
|
/* enable event notifications */
|
|
GST_LOG_OBJECT (pulsesrc, "subscribing to context events");
|
|
if (!(o = pa_context_subscribe (pulsesrc->context,
|
|
PA_SUBSCRIPTION_MASK_SOURCE_OUTPUT, NULL, NULL))) {
|
|
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
|
|
("pa_context_subscribe() failed: %s",
|
|
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
pa_operation_unref (o);
|
|
|
|
/* There's a bit of a disconnect here between the audio ringbuffer and what
|
|
* PulseAudio provides. The audio ringbuffer provide a total of buffer_time
|
|
* worth of buffering, divided into segments of latency_time size. We're
|
|
* asking PulseAudio to provide a total latency of latency_time, which, with
|
|
* PA_STREAM_ADJUST_LATENCY, effectively sets itself up as a ringbuffer with
|
|
* one segment being the hardware buffer, and the other the software buffer.
|
|
* This segment size is returned as the fragsize.
|
|
*
|
|
* Since the two concepts don't map very well, what we do is keep segsize as
|
|
* it is (unless fragsize is even larger, in which case we use that). We'll
|
|
* get data from PulseAudio in smaller chunks than we want to pass on as an
|
|
* element, and we coalesce those chunks in the ringbuffer memory and pass it
|
|
* on in the expected chunk size. */
|
|
wanted.maxlength = spec->segsize * spec->segtotal;
|
|
wanted.tlength = -1;
|
|
wanted.prebuf = 0;
|
|
wanted.minreq = -1;
|
|
wanted.fragsize = spec->segsize;
|
|
|
|
GST_INFO_OBJECT (pulsesrc, "maxlength: %d", wanted.maxlength);
|
|
GST_INFO_OBJECT (pulsesrc, "tlength: %d", wanted.tlength);
|
|
GST_INFO_OBJECT (pulsesrc, "prebuf: %d", wanted.prebuf);
|
|
GST_INFO_OBJECT (pulsesrc, "minreq: %d", wanted.minreq);
|
|
GST_INFO_OBJECT (pulsesrc, "fragsize: %d", wanted.fragsize);
|
|
|
|
flags = PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
|
|
PA_STREAM_NOT_MONOTONIC | PA_STREAM_ADJUST_LATENCY |
|
|
PA_STREAM_START_CORKED;
|
|
|
|
if (pa_stream_connect_record (pulsesrc->stream, pulsesrc->device, &wanted,
|
|
flags) < 0) {
|
|
goto connect_failed;
|
|
}
|
|
|
|
/* our clock will now start from 0 again */
|
|
clock = GST_AUDIO_CLOCK (GST_AUDIO_BASE_SRC (pulsesrc)->clock);
|
|
gst_audio_clock_reset (clock, 0);
|
|
|
|
pulsesrc->corked = TRUE;
|
|
|
|
for (;;) {
|
|
pa_stream_state_t state;
|
|
|
|
state = pa_stream_get_state (pulsesrc->stream);
|
|
|
|
if (!PA_STREAM_IS_GOOD (state))
|
|
goto stream_is_bad;
|
|
|
|
if (state == PA_STREAM_READY)
|
|
break;
|
|
|
|
/* Wait until the stream is ready */
|
|
pa_threaded_mainloop_wait (pulsesrc->mainloop);
|
|
}
|
|
pulsesrc->stream_connected = TRUE;
|
|
|
|
/* store the source output index so it can be accessed via a property */
|
|
pulsesrc->source_output_idx = pa_stream_get_index (pulsesrc->stream);
|
|
g_object_notify (G_OBJECT (pulsesrc), "source-output-index");
|
|
|
|
/* Although source output stream muting is supported, there is a bug in
|
|
* PulseAudio that doesn't allow us to do this at startup, so we mute
|
|
* manually post-connect. This should be moved back pre-connect once things
|
|
* are fixed on the PulseAudio side. */
|
|
if (pulsesrc->mute_set && pulsesrc->mute) {
|
|
gst_pulsesrc_set_stream_mute (pulsesrc, pulsesrc->mute);
|
|
pulsesrc->mute_set = FALSE;
|
|
}
|
|
|
|
if (pulsesrc->volume_set) {
|
|
gst_pulsesrc_set_stream_volume (pulsesrc, pulsesrc->volume);
|
|
pulsesrc->volume_set = FALSE;
|
|
}
|
|
|
|
/* get the actual buffering properties now */
|
|
actual = pa_stream_get_buffer_attr (pulsesrc->stream);
|
|
|
|
GST_INFO_OBJECT (pulsesrc, "maxlength: %d", actual->maxlength);
|
|
GST_INFO_OBJECT (pulsesrc, "tlength: %d (wanted: %d)",
|
|
actual->tlength, wanted.tlength);
|
|
GST_INFO_OBJECT (pulsesrc, "prebuf: %d", actual->prebuf);
|
|
GST_INFO_OBJECT (pulsesrc, "minreq: %d (wanted %d)", actual->minreq,
|
|
wanted.minreq);
|
|
GST_INFO_OBJECT (pulsesrc, "fragsize: %d (wanted %d)",
|
|
actual->fragsize, wanted.fragsize);
|
|
|
|
if (actual->fragsize >= spec->segsize) {
|
|
spec->segsize = actual->fragsize;
|
|
} else {
|
|
/* fragsize is smaller than what we wanted, so let the read function
|
|
* coalesce the smaller chunks as they come in */
|
|
}
|
|
|
|
/* Fix up the total ringbuffer size based on what we actually got */
|
|
spec->segtotal = actual->maxlength / spec->segsize;
|
|
/* Don't buffer less than 2 segments as the ringbuffer can't deal with it */
|
|
if (spec->segtotal < 2)
|
|
spec->segtotal = 2;
|
|
|
|
if (!pulsesrc->paused) {
|
|
GST_DEBUG_OBJECT (pulsesrc, "uncorking because we are playing");
|
|
gst_pulsesrc_set_corked (pulsesrc, FALSE, FALSE);
|
|
}
|
|
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
connect_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
|
|
("Failed to connect stream: %s",
|
|
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
stream_is_bad:
|
|
{
|
|
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
|
|
("Failed to connect stream: %s",
|
|
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
unlock_and_fail:
|
|
{
|
|
gst_pulsesrc_destroy_stream (pulsesrc);
|
|
|
|
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_pulsesrc_success_cb (pa_stream * s, int success, void *userdata)
|
|
{
|
|
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (userdata);
|
|
|
|
pulsesrc->operation_success = !!success;
|
|
pa_threaded_mainloop_signal (pulsesrc->mainloop, 0);
|
|
}
|
|
|
|
static void
|
|
gst_pulsesrc_reset (GstAudioSrc * asrc)
|
|
{
|
|
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
|
|
pa_operation *o = NULL;
|
|
|
|
pa_threaded_mainloop_lock (pulsesrc->mainloop);
|
|
GST_DEBUG_OBJECT (pulsesrc, "reset");
|
|
|
|
if (gst_pulsesrc_is_dead (pulsesrc, TRUE))
|
|
goto unlock_and_fail;
|
|
|
|
if (!(o =
|
|
pa_stream_flush (pulsesrc->stream, gst_pulsesrc_success_cb,
|
|
pulsesrc))) {
|
|
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
|
|
("pa_stream_flush() failed: %s",
|
|
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
pulsesrc->paused = TRUE;
|
|
/* Inform anyone waiting in _write() call that it shall wakeup */
|
|
if (pulsesrc->in_read) {
|
|
pa_threaded_mainloop_signal (pulsesrc->mainloop, 0);
|
|
}
|
|
|
|
pulsesrc->operation_success = FALSE;
|
|
while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
|
|
|
|
if (gst_pulsesrc_is_dead (pulsesrc, TRUE))
|
|
goto unlock_and_fail;
|
|
|
|
pa_threaded_mainloop_wait (pulsesrc->mainloop);
|
|
}
|
|
|
|
if (!pulsesrc->operation_success) {
|
|
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Flush failed: %s",
|
|
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
unlock_and_fail:
|
|
|
|
if (o) {
|
|
pa_operation_cancel (o);
|
|
pa_operation_unref (o);
|
|
}
|
|
|
|
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
|
|
}
|
|
|
|
/* update the corked state of a stream, must be called with the mainloop
|
|
* lock */
|
|
static gboolean
|
|
gst_pulsesrc_set_corked (GstPulseSrc * psrc, gboolean corked, gboolean wait)
|
|
{
|
|
pa_operation *o = NULL;
|
|
gboolean res = FALSE;
|
|
|
|
GST_DEBUG_OBJECT (psrc, "setting corked state to %d", corked);
|
|
if (!psrc->stream_connected)
|
|
return TRUE;
|
|
|
|
if (psrc->corked != corked) {
|
|
if (!(o = pa_stream_cork (psrc->stream, corked,
|
|
gst_pulsesrc_success_cb, psrc)))
|
|
goto cork_failed;
|
|
|
|
while (wait && pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
|
|
pa_threaded_mainloop_wait (psrc->mainloop);
|
|
if (gst_pulsesrc_is_dead (psrc, TRUE))
|
|
goto server_dead;
|
|
}
|
|
psrc->corked = corked;
|
|
} else {
|
|
GST_DEBUG_OBJECT (psrc, "skipping, already in requested state");
|
|
}
|
|
res = TRUE;
|
|
|
|
cleanup:
|
|
if (o)
|
|
pa_operation_unref (o);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
server_dead:
|
|
{
|
|
GST_DEBUG_OBJECT (psrc, "the server is dead");
|
|
goto cleanup;
|
|
}
|
|
cork_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (psrc, RESOURCE, FAILED,
|
|
("pa_stream_cork() failed: %s",
|
|
pa_strerror (pa_context_errno (psrc->context))), (NULL));
|
|
goto cleanup;
|
|
}
|
|
}
|
|
|
|
/* start/resume playback ASAP */
|
|
static gboolean
|
|
gst_pulsesrc_play (GstPulseSrc * psrc)
|
|
{
|
|
pa_threaded_mainloop_lock (psrc->mainloop);
|
|
GST_DEBUG_OBJECT (psrc, "playing");
|
|
psrc->paused = FALSE;
|
|
gst_pulsesrc_set_corked (psrc, FALSE, FALSE);
|
|
pa_threaded_mainloop_unlock (psrc->mainloop);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/* pause/stop playback ASAP */
|
|
static gboolean
|
|
gst_pulsesrc_pause (GstPulseSrc * psrc)
|
|
{
|
|
pa_threaded_mainloop_lock (psrc->mainloop);
|
|
GST_DEBUG_OBJECT (psrc, "pausing");
|
|
/* make sure the commit method stops writing */
|
|
psrc->paused = TRUE;
|
|
if (psrc->in_read) {
|
|
/* we are waiting in a read, signal */
|
|
GST_DEBUG_OBJECT (psrc, "signal read");
|
|
pa_threaded_mainloop_signal (psrc->mainloop, 0);
|
|
}
|
|
pa_threaded_mainloop_unlock (psrc->mainloop);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_pulsesrc_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstStateChangeReturn ret;
|
|
GstPulseSrc *this = GST_PULSESRC_CAST (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
if (!(this->mainloop = pa_threaded_mainloop_new ()))
|
|
goto mainloop_failed;
|
|
if (pa_threaded_mainloop_start (this->mainloop) < 0) {
|
|
pa_threaded_mainloop_free (this->mainloop);
|
|
this->mainloop = NULL;
|
|
goto mainloop_start_failed;
|
|
}
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
gst_element_post_message (element,
|
|
gst_message_new_clock_provide (GST_OBJECT_CAST (element),
|
|
GST_AUDIO_BASE_SRC (this)->clock, TRUE));
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
/* uncork and start recording */
|
|
gst_pulsesrc_play (this);
|
|
break;
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
/* stop recording ASAP by corking */
|
|
pa_threaded_mainloop_lock (this->mainloop);
|
|
GST_DEBUG_OBJECT (this, "corking");
|
|
gst_pulsesrc_set_corked (this, TRUE, FALSE);
|
|
pa_threaded_mainloop_unlock (this->mainloop);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
/* now make sure we get out of the _read method */
|
|
gst_pulsesrc_pause (this);
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
if (this->mainloop)
|
|
pa_threaded_mainloop_stop (this->mainloop);
|
|
|
|
gst_pulsesrc_destroy_context (this);
|
|
|
|
if (this->mainloop) {
|
|
pa_threaded_mainloop_free (this->mainloop);
|
|
this->mainloop = NULL;
|
|
}
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
/* format_lost is reset in release() in baseaudiosink */
|
|
gst_element_post_message (element,
|
|
gst_message_new_clock_lost (GST_OBJECT_CAST (element),
|
|
GST_AUDIO_BASE_SRC (this)->clock));
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
mainloop_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (this, RESOURCE, FAILED,
|
|
("pa_threaded_mainloop_new() failed"), (NULL));
|
|
return GST_STATE_CHANGE_FAILURE;
|
|
}
|
|
mainloop_start_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (this, RESOURCE, FAILED,
|
|
("pa_threaded_mainloop_start() failed"), (NULL));
|
|
return GST_STATE_CHANGE_FAILURE;
|
|
}
|
|
}
|
|
|
|
static GstClockTime
|
|
gst_pulsesrc_get_time (GstClock * clock, GstPulseSrc * src)
|
|
{
|
|
pa_usec_t time = 0;
|
|
|
|
if (src->mainloop == NULL)
|
|
goto out;
|
|
|
|
pa_threaded_mainloop_lock (src->mainloop);
|
|
if (!src->stream)
|
|
goto unlock_and_out;
|
|
|
|
if (gst_pulsesrc_is_dead (src, TRUE))
|
|
goto unlock_and_out;
|
|
|
|
if (pa_stream_get_time (src->stream, &time) < 0) {
|
|
GST_DEBUG_OBJECT (src, "could not get time");
|
|
time = GST_CLOCK_TIME_NONE;
|
|
} else {
|
|
time *= 1000;
|
|
}
|
|
|
|
|
|
unlock_and_out:
|
|
pa_threaded_mainloop_unlock (src->mainloop);
|
|
|
|
out:
|
|
return time;
|
|
}
|