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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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952 lines
29 KiB
C
952 lines
29 KiB
C
/*
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* GStreamer
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* Copyright (C) 2005 Wim Taymans <wim@fluendo.com>
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* Copyright (C) 2006 Tim-Philipp Müller <tim centricular net>
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* Copyright (C) 2009-2010 Chris Robinson <chris.kcat@gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but 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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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-openalsink
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*
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* This element renders raw audio samples using the OpenAL API
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*
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* <refsect2>
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* <title>Example pipelines</title>
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* |[
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* gst-launch -v audiotestsrc ! audioconvert ! volume volume=0.1 ! openalsink
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* ]| will output a sine wave (continuous beep sound) to your sound card (with
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* a very low volume as precaution).
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* |[
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* gst-launch -v filesrc location=music.ogg ! decodebin ! audioconvert ! audioresample ! openalsink
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* ]| will play an Ogg/Vorbis audio file and output it using OpenAL.
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* </refsect2>
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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 "gstopenalsink.h"
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GST_DEBUG_CATEGORY (openalsink_debug);
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static void gst_openal_sink_dispose (GObject * object);
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static void gst_openal_sink_finalize (GObject * object);
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static void gst_openal_sink_get_property (GObject * object, guint prop_id,
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GValue * val, GParamSpec * pspec);
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static void gst_openal_sink_set_property (GObject * object, guint prop_id,
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const GValue * val, GParamSpec * pspec);
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static GstCaps *gst_openal_sink_getcaps (GstBaseSink * bsink);
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static gboolean gst_openal_sink_open (GstAudioSink * asink);
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static gboolean gst_openal_sink_close (GstAudioSink * asink);
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static gboolean gst_openal_sink_prepare (GstAudioSink * asink,
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GstRingBufferSpec * spec);
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static gboolean gst_openal_sink_unprepare (GstAudioSink * asink);
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static guint gst_openal_sink_write (GstAudioSink * asink, gpointer data,
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guint length);
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static guint gst_openal_sink_delay (GstAudioSink * asink);
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static void gst_openal_sink_reset (GstAudioSink * asink);
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#define DEFAULT_DEVICE NULL
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enum
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{
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PROP_0,
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PROP_DEVICE,
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PROP_DEVICE_NAME,
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PROP_DEVICE_HDL,
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PROP_CONTEXT_HDL,
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PROP_SOURCE_ID
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};
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static GstStaticPadTemplate openalsink_sink_factory =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw-float, "
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"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", "
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"width = (int) 32, "
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"rate = (int) [ 1, MAX ], "
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"channels = (int) [ 1, MAX ]; "
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"audio/x-raw-int, "
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"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", "
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"signed = (boolean) TRUE, "
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"width = (int) 16, "
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"depth = (int) 16, "
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"rate = (int) [ 1, MAX ], "
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"channels = (int) [ 1, MAX ]; "
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"audio/x-raw-int, "
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"signed = (boolean) FALSE, "
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"width = (int) 8, "
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"depth = (int) 8, "
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"rate = (int) [ 1, MAX ], "
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"channels = (int) [ 1, MAX ]; "
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"audio/x-mulaw, "
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"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
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);
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static PFNALCSETTHREADCONTEXTPROC palcSetThreadContext;
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static PFNALCGETTHREADCONTEXTPROC palcGetThreadContext;
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static inline ALCcontext *
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pushContext (ALCcontext * ctx)
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{
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ALCcontext *old;
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if (!palcGetThreadContext || !palcSetThreadContext)
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return NULL;
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old = palcGetThreadContext ();
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if (old != ctx)
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palcSetThreadContext (ctx);
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return old;
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}
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static inline void
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popContext (ALCcontext * old, ALCcontext * ctx)
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{
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if (!palcGetThreadContext || !palcSetThreadContext)
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return;
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if (old != ctx)
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palcSetThreadContext (old);
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}
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static inline ALenum
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checkALError (const char *fname, unsigned int fline)
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{
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ALenum err = alGetError ();
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if (err != AL_NO_ERROR)
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g_warning ("%s:%u: context error: %s", fname, fline, alGetString (err));
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return err;
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}
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#define checkALError() checkALError(__FILE__, __LINE__)
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GST_BOILERPLATE (GstOpenALSink, gst_openal_sink, GstAudioSink,
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GST_TYPE_AUDIO_SINK);
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static void
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gst_openal_sink_dispose (GObject * object)
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{
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GstOpenALSink *sink = GST_OPENAL_SINK (object);
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if (sink->probed_caps)
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gst_caps_unref (sink->probed_caps);
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sink->probed_caps = NULL;
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G_OBJECT_CLASS (parent_class)->dispose (object);
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}
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/* GObject vmethod implementations */
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static void
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gst_openal_sink_base_init (gpointer gclass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (gclass);
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GstPadTemplate *pad_template;
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gst_element_class_set_details_simple (element_class, "Audio sink (OpenAL)",
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"Sink/Audio",
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"Output to a sound device via OpenAL",
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"Chris Robinson <chris.kcat@gmail.com>");
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pad_template = gst_static_pad_template_get (&openalsink_sink_factory);
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gst_element_class_add_pad_template (element_class, pad_template);
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}
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/* initialize the plugin's class */
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static void
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gst_openal_sink_class_init (GstOpenALSinkClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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GstBaseSinkClass *gstbasesink_class = (GstBaseSinkClass *) klass;
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GstAudioSinkClass *gstaudiosink_class = (GstAudioSinkClass *) klass;
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GParamSpec *spec;
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if (alcIsExtensionPresent (NULL, "ALC_EXT_thread_local_context")) {
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palcSetThreadContext = alcGetProcAddress (NULL, "alcSetThreadContext");
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palcGetThreadContext = alcGetProcAddress (NULL, "alcGetThreadContext");
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}
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GST_DEBUG_CATEGORY_INIT (openalsink_debug, "openalsink", 0, "OpenAL sink");
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gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_openal_sink_dispose);
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gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_openal_sink_finalize);
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gobject_class->set_property =
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GST_DEBUG_FUNCPTR (gst_openal_sink_set_property);
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gobject_class->get_property =
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GST_DEBUG_FUNCPTR (gst_openal_sink_get_property);
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spec = g_param_spec_string ("device-name", "Device name",
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"Opened OpenAL device name", "", G_PARAM_READABLE);
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g_object_class_install_property (gobject_class, PROP_DEVICE_NAME, spec);
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spec = g_param_spec_string ("device", "Device", "OpenAL device string",
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DEFAULT_DEVICE, G_PARAM_READWRITE);
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g_object_class_install_property (gobject_class, PROP_DEVICE, spec);
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spec = g_param_spec_pointer ("device-handle", "ALCdevice",
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"Custom playback device", G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS);
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g_object_class_install_property (gobject_class, PROP_DEVICE_HDL, spec);
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spec = g_param_spec_pointer ("context-handle", "ALCcontext",
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"Custom playback context", G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS);
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g_object_class_install_property (gobject_class, PROP_CONTEXT_HDL, spec);
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spec = g_param_spec_uint ("source-id", "Source ID", "Custom playback sID",
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0, UINT_MAX, 0, G_PARAM_READWRITE);
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g_object_class_install_property (gobject_class, PROP_SOURCE_ID, spec);
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parent_class = g_type_class_peek_parent (klass);
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gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_openal_sink_getcaps);
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gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_openal_sink_open);
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gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_openal_sink_close);
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gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_openal_sink_prepare);
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gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_openal_sink_unprepare);
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gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_openal_sink_write);
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gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_openal_sink_delay);
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gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_openal_sink_reset);
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}
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static void
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gst_openal_sink_init (GstOpenALSink * sink, GstOpenALSinkClass * klass)
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{
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GST_DEBUG_OBJECT (sink, "initializing openalsink");
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sink->devname = g_strdup (DEFAULT_DEVICE);
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sink->custom_dev = NULL;
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sink->custom_ctx = NULL;
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sink->custom_sID = 0;
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sink->device = NULL;
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sink->context = NULL;
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sink->sID = 0;
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sink->bID_idx = 0;
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sink->bID_count = 0;
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sink->bIDs = NULL;
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sink->bID_length = 0;
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sink->write_reset = AL_FALSE;
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sink->probed_caps = NULL;
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sink->openal_lock = g_mutex_new ();
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}
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static void
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gst_openal_sink_finalize (GObject * object)
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{
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GstOpenALSink *sink = GST_OPENAL_SINK (object);
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g_free (sink->devname);
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sink->devname = NULL;
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g_mutex_free (sink->openal_lock);
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sink->openal_lock = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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gst_openal_sink_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstOpenALSink *sink = GST_OPENAL_SINK (object);
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switch (prop_id) {
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case PROP_DEVICE:
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g_free (sink->devname);
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sink->devname = g_value_dup_string (value);
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if (sink->probed_caps)
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gst_caps_unref (sink->probed_caps);
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sink->probed_caps = NULL;
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break;
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case PROP_DEVICE_HDL:
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if (!sink->device)
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sink->custom_dev = g_value_get_pointer (value);
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break;
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case PROP_CONTEXT_HDL:
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if (!sink->device)
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sink->custom_ctx = g_value_get_pointer (value);
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break;
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case PROP_SOURCE_ID:
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if (!sink->device)
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sink->custom_sID = g_value_get_uint (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_openal_sink_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstOpenALSink *sink = GST_OPENAL_SINK (object);
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const ALCchar *name = sink->devname;
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ALCdevice *device = sink->device;
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ALCcontext *context = sink->context;
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ALuint sourceID = sink->sID;
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switch (prop_id) {
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case PROP_DEVICE_NAME:
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name = "";
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if (device)
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name = alcGetString (device, ALC_DEVICE_SPECIFIER);
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/* fall-through */
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case PROP_DEVICE:
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g_value_set_string (value, name);
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break;
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case PROP_DEVICE_HDL:
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if (!device)
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device = sink->custom_dev;
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g_value_set_pointer (value, device);
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break;
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case PROP_CONTEXT_HDL:
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if (!context)
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context = sink->custom_ctx;
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g_value_set_pointer (value, context);
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break;
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case PROP_SOURCE_ID:
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if (!sourceID)
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sourceID = sink->custom_sID;
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g_value_set_uint (value, sourceID);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static GstCaps *
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gst_openal_helper_probe_caps (ALCcontext * ctx)
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{
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static const struct
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{
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gint count;
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GstAudioChannelPosition pos[8];
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} chans[] = {
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{
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1, {
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GST_AUDIO_CHANNEL_POSITION_FRONT_MONO}}, {
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2, {
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}}, {
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4, {
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}}, {
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6, {
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_LFE,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}}, {
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7, {
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_LFE,
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GST_AUDIO_CHANNEL_POSITION_REAR_CENTER,
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GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
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GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}}, {
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8, {
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_LFE,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
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GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}},};
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GstStructure *structure;
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ALCcontext *old;
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GstCaps *caps;
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old = pushContext (ctx);
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caps = gst_caps_new_empty ();
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if (alIsExtensionPresent ("AL_EXT_MCFORMATS")) {
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const char *fmt32[] = {
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"AL_FORMAT_MONO_FLOAT32", "AL_FORMAT_STEREO_FLOAT32",
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"AL_FORMAT_QUAD32", "AL_FORMAT_51CHN32", "AL_FORMAT_61CHN32",
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"AL_FORMAT_71CHN32", NULL
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}, *fmt16[] = {
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"AL_FORMAT_MONO16", "AL_FORMAT_STEREO16", "AL_FORMAT_QUAD16",
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"AL_FORMAT_51CHN16", "AL_FORMAT_61CHN16", "AL_FORMAT_71CHN16", NULL},
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*fmt8[] = {
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"AL_FORMAT_MONO8", "AL_FORMAT_STEREO8", "AL_FORMAT_QUAD8",
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"AL_FORMAT_51CHN8", "AL_FORMAT_61CHN8", "AL_FORMAT_71CHN8", NULL};
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int i;
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if (alIsExtensionPresent ("AL_EXT_FLOAT32")) {
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for (i = 0; fmt32[i]; i++) {
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ALenum val = alGetEnumValue (fmt32[i]);
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if (checkALError () != AL_NO_ERROR || val == 0 || val == -1)
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continue;
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structure = gst_structure_new ("audio/x-raw-float",
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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"rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE,
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OPENAL_MAX_RATE, "width", G_TYPE_INT, 32, NULL);
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gst_structure_set (structure, "channels", G_TYPE_INT,
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chans[i].count, NULL);
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if (chans[i].count > 2)
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gst_audio_set_channel_positions (structure, chans[i].pos);
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gst_caps_append_structure (caps, structure);
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}
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}
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for (i = 0; fmt16[i]; i++) {
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ALenum val = alGetEnumValue (fmt16[i]);
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if (checkALError () != AL_NO_ERROR || val == 0 || val == -1)
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continue;
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structure = gst_structure_new ("audio/x-raw-int",
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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"rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE, OPENAL_MAX_RATE,
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"width", G_TYPE_INT, 16,
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"depth", G_TYPE_INT, 16, "signed", G_TYPE_BOOLEAN, TRUE, NULL);
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gst_structure_set (structure, "channels", G_TYPE_INT,
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chans[i].count, NULL);
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if (chans[i].count > 2)
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gst_audio_set_channel_positions (structure, chans[i].pos);
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gst_caps_append_structure (caps, structure);
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}
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for (i = 0; fmt8[i]; i++) {
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ALenum val = alGetEnumValue (fmt8[i]);
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if (checkALError () != AL_NO_ERROR || val == 0 || val == -1)
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continue;
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structure = gst_structure_new ("audio/x-raw-int",
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"rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE, OPENAL_MAX_RATE,
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"width", G_TYPE_INT, 8,
|
|
"depth", G_TYPE_INT, 8, "signed", G_TYPE_BOOLEAN, FALSE, NULL);
|
|
gst_structure_set (structure, "channels", G_TYPE_INT,
|
|
chans[i].count, NULL);
|
|
if (chans[i].count > 2)
|
|
gst_audio_set_channel_positions (structure, chans[i].pos);
|
|
gst_caps_append_structure (caps, structure);
|
|
}
|
|
} else {
|
|
if (alIsExtensionPresent ("AL_EXT_FLOAT32")) {
|
|
structure = gst_structure_new ("audio/x-raw-float",
|
|
"endianness", G_TYPE_INT, G_BYTE_ORDER,
|
|
"rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE, OPENAL_MAX_RATE,
|
|
"width", G_TYPE_INT, 32, "channels", GST_TYPE_INT_RANGE, 1, 2, NULL);
|
|
gst_caps_append_structure (caps, structure);
|
|
}
|
|
|
|
structure = gst_structure_new ("audio/x-raw-int",
|
|
"endianness", G_TYPE_INT, G_BYTE_ORDER,
|
|
"rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE, OPENAL_MAX_RATE,
|
|
"width", G_TYPE_INT, 16,
|
|
"depth", G_TYPE_INT, 16,
|
|
"signed", G_TYPE_BOOLEAN, TRUE,
|
|
"channels", GST_TYPE_INT_RANGE, 1, 2, NULL);
|
|
gst_caps_append_structure (caps, structure);
|
|
|
|
structure = gst_structure_new ("audio/x-raw-int",
|
|
"rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE, OPENAL_MAX_RATE,
|
|
"width", G_TYPE_INT, 8,
|
|
"depth", G_TYPE_INT, 8,
|
|
"signed", G_TYPE_BOOLEAN, FALSE,
|
|
"channels", GST_TYPE_INT_RANGE, 1, 2, NULL);
|
|
gst_caps_append_structure (caps, structure);
|
|
}
|
|
|
|
if (alIsExtensionPresent ("AL_EXT_MULAW_MCFORMATS")) {
|
|
const char *fmtmulaw[] = {
|
|
"AL_FORMAT_MONO_MULAW", "AL_FORMAT_STEREO_MULAW",
|
|
"AL_FORMAT_QUAD_MULAW", "AL_FORMAT_51CHN_MULAW",
|
|
"AL_FORMAT_61CHN_MULAW", "AL_FORMAT_71CHN_MULAW", NULL
|
|
};
|
|
int i;
|
|
|
|
for (i = 0; fmtmulaw[i]; i++) {
|
|
ALenum val = alGetEnumValue (fmtmulaw[i]);
|
|
if (checkALError () != AL_NO_ERROR || val == 0 || val == -1)
|
|
continue;
|
|
|
|
structure = gst_structure_new ("audio/x-mulaw",
|
|
"rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE, OPENAL_MAX_RATE, NULL);
|
|
gst_structure_set (structure, "channels", G_TYPE_INT,
|
|
chans[i].count, NULL);
|
|
if (chans[i].count > 2)
|
|
gst_audio_set_channel_positions (structure, chans[i].pos);
|
|
gst_caps_append_structure (caps, structure);
|
|
}
|
|
} else if (alIsExtensionPresent ("AL_EXT_MULAW")) {
|
|
structure = gst_structure_new ("audio/x-mulaw",
|
|
"rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE, OPENAL_MAX_RATE,
|
|
"channels", GST_TYPE_INT_RANGE, 1, 2, NULL);
|
|
gst_caps_append_structure (caps, structure);
|
|
}
|
|
|
|
popContext (old, ctx);
|
|
return caps;
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_openal_sink_getcaps (GstBaseSink * bsink)
|
|
{
|
|
GstOpenALSink *sink = GST_OPENAL_SINK (bsink);
|
|
GstCaps *caps;
|
|
|
|
if (sink->device == NULL) {
|
|
GstPad *pad = GST_BASE_SINK_PAD (bsink);
|
|
caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
|
|
} else if (sink->probed_caps)
|
|
caps = gst_caps_copy (sink->probed_caps);
|
|
else {
|
|
if (sink->context)
|
|
caps = gst_openal_helper_probe_caps (sink->context);
|
|
else if (sink->custom_ctx)
|
|
caps = gst_openal_helper_probe_caps (sink->custom_ctx);
|
|
else {
|
|
ALCcontext *ctx = alcCreateContext (sink->device, NULL);
|
|
if (ctx) {
|
|
caps = gst_openal_helper_probe_caps (ctx);
|
|
alcDestroyContext (ctx);
|
|
} else {
|
|
GST_ELEMENT_WARNING (sink, RESOURCE, FAILED,
|
|
("Could not create temporary context."),
|
|
GST_ALC_ERROR (sink->device));
|
|
caps = NULL;
|
|
}
|
|
}
|
|
|
|
if (caps && !gst_caps_is_empty (caps))
|
|
sink->probed_caps = gst_caps_copy (caps);
|
|
}
|
|
|
|
return caps;
|
|
}
|
|
|
|
static gboolean
|
|
gst_openal_sink_open (GstAudioSink * asink)
|
|
{
|
|
GstOpenALSink *openal = GST_OPENAL_SINK (asink);
|
|
|
|
if (openal->custom_dev) {
|
|
ALCint val = -1;
|
|
alcGetIntegerv (openal->custom_dev, ALC_ATTRIBUTES_SIZE, 1, &val);
|
|
if (val > 0) {
|
|
if (!openal->custom_ctx ||
|
|
alcGetContextsDevice (openal->custom_ctx) == openal->custom_dev)
|
|
openal->device = openal->custom_dev;
|
|
}
|
|
} else if (openal->custom_ctx)
|
|
openal->device = alcGetContextsDevice (openal->custom_ctx);
|
|
else
|
|
openal->device = alcOpenDevice (openal->devname);
|
|
if (!openal->device) {
|
|
GST_ELEMENT_ERROR (openal, RESOURCE, OPEN_WRITE,
|
|
("Could not open audio device for playback."),
|
|
GST_ALC_ERROR (openal->device));
|
|
return FALSE;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_openal_sink_close (GstAudioSink * asink)
|
|
{
|
|
GstOpenALSink *openal = GST_OPENAL_SINK (asink);
|
|
|
|
if (!openal->custom_dev && !openal->custom_ctx) {
|
|
if (alcCloseDevice (openal->device) == ALC_FALSE) {
|
|
GST_ELEMENT_ERROR (openal, RESOURCE, CLOSE,
|
|
("Could not close audio device."), GST_ALC_ERROR (openal->device));
|
|
return FALSE;
|
|
}
|
|
}
|
|
openal->device = NULL;
|
|
|
|
if (openal->probed_caps)
|
|
gst_caps_unref (openal->probed_caps);
|
|
openal->probed_caps = NULL;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_openal_sink_parse_spec (GstOpenALSink * openal,
|
|
const GstRingBufferSpec * spec)
|
|
{
|
|
ALuint format = AL_NONE;
|
|
|
|
GST_DEBUG_OBJECT (openal, "Looking up format for type %d, gst-format %d, "
|
|
"and %d channels", spec->type, spec->format, spec->channels);
|
|
|
|
/* Don't need to verify supported formats, since the probed caps will only
|
|
* report what was detected and we shouldn't get anything different */
|
|
switch (spec->type) {
|
|
case GST_BUFTYPE_LINEAR:
|
|
switch (spec->format) {
|
|
case GST_U8:
|
|
if (spec->channels == 1)
|
|
format = AL_FORMAT_MONO8;
|
|
if (spec->channels == 2)
|
|
format = AL_FORMAT_STEREO8;
|
|
if (spec->channels == 4)
|
|
format = AL_FORMAT_QUAD8;
|
|
if (spec->channels == 6)
|
|
format = AL_FORMAT_51CHN8;
|
|
if (spec->channels == 7)
|
|
format = AL_FORMAT_61CHN8;
|
|
if (spec->channels == 8)
|
|
format = AL_FORMAT_71CHN8;
|
|
break;
|
|
|
|
case GST_S16_NE:
|
|
if (spec->channels == 1)
|
|
format = AL_FORMAT_MONO16;
|
|
if (spec->channels == 2)
|
|
format = AL_FORMAT_STEREO16;
|
|
if (spec->channels == 4)
|
|
format = AL_FORMAT_QUAD16;
|
|
if (spec->channels == 6)
|
|
format = AL_FORMAT_51CHN16;
|
|
if (spec->channels == 7)
|
|
format = AL_FORMAT_61CHN16;
|
|
if (spec->channels == 8)
|
|
format = AL_FORMAT_71CHN16;
|
|
break;
|
|
|
|
default:
|
|
break;
|
|
}
|
|
break;
|
|
|
|
case GST_BUFTYPE_FLOAT:
|
|
switch (spec->format) {
|
|
case GST_FLOAT32_NE:
|
|
if (spec->channels == 1)
|
|
format = AL_FORMAT_MONO_FLOAT32;
|
|
if (spec->channels == 2)
|
|
format = AL_FORMAT_STEREO_FLOAT32;
|
|
if (spec->channels == 4)
|
|
format = AL_FORMAT_QUAD32;
|
|
if (spec->channels == 6)
|
|
format = AL_FORMAT_51CHN32;
|
|
if (spec->channels == 7)
|
|
format = AL_FORMAT_61CHN32;
|
|
if (spec->channels == 8)
|
|
format = AL_FORMAT_71CHN32;
|
|
break;
|
|
|
|
default:
|
|
break;
|
|
}
|
|
break;
|
|
|
|
case GST_BUFTYPE_MU_LAW:
|
|
switch (spec->format) {
|
|
case GST_MU_LAW:
|
|
if (spec->channels == 1)
|
|
format = AL_FORMAT_MONO_MULAW;
|
|
if (spec->channels == 2)
|
|
format = AL_FORMAT_STEREO_MULAW;
|
|
if (spec->channels == 4)
|
|
format = AL_FORMAT_QUAD_MULAW;
|
|
if (spec->channels == 6)
|
|
format = AL_FORMAT_51CHN_MULAW;
|
|
if (spec->channels == 7)
|
|
format = AL_FORMAT_61CHN_MULAW;
|
|
if (spec->channels == 8)
|
|
format = AL_FORMAT_71CHN_MULAW;
|
|
break;
|
|
|
|
default:
|
|
break;
|
|
}
|
|
break;
|
|
|
|
default:
|
|
break;
|
|
}
|
|
|
|
openal->bytes_per_sample = spec->bytes_per_sample;
|
|
openal->srate = spec->rate;
|
|
openal->bID_count = spec->segtotal;
|
|
openal->bID_length = spec->segsize;
|
|
openal->format = format;
|
|
}
|
|
|
|
static gboolean
|
|
gst_openal_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
|
|
{
|
|
GstOpenALSink *openal = GST_OPENAL_SINK (asink);
|
|
ALCcontext *ctx, *old;
|
|
|
|
if (openal->context && !gst_openal_sink_unprepare (asink))
|
|
return FALSE;
|
|
|
|
if (openal->custom_ctx)
|
|
ctx = openal->custom_ctx;
|
|
else {
|
|
ALCint attribs[3] = { 0, 0, 0 };
|
|
|
|
/* Don't try to change the playback frequency of an app's device */
|
|
if (!openal->custom_dev) {
|
|
attribs[0] = ALC_FREQUENCY;
|
|
attribs[1] = spec->rate;
|
|
attribs[2] = 0;
|
|
}
|
|
|
|
ctx = alcCreateContext (openal->device, attribs);
|
|
if (!ctx) {
|
|
GST_ELEMENT_ERROR (openal, RESOURCE, FAILED,
|
|
("Unable to prepare device."), GST_ALC_ERROR (openal->device));
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
old = pushContext (ctx);
|
|
|
|
if (openal->custom_sID) {
|
|
if (!openal->custom_ctx || !alIsSource (openal->custom_sID)) {
|
|
GST_ELEMENT_ERROR (openal, RESOURCE, NOT_FOUND, (NULL),
|
|
("Invalid source ID specified for context"));
|
|
goto fail;
|
|
}
|
|
openal->sID = openal->custom_sID;
|
|
} else {
|
|
ALuint sourceID;
|
|
|
|
alGenSources (1, &sourceID);
|
|
if (checkALError () != AL_NO_ERROR) {
|
|
GST_ELEMENT_ERROR (openal, RESOURCE, NO_SPACE_LEFT, (NULL),
|
|
("Unable to generate source"));
|
|
goto fail;
|
|
}
|
|
openal->sID = sourceID;
|
|
}
|
|
|
|
gst_openal_sink_parse_spec (openal, spec);
|
|
if (openal->format == AL_NONE) {
|
|
GST_ELEMENT_ERROR (openal, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to get type %d, format %d, and %d channels",
|
|
spec->type, spec->format, spec->channels));
|
|
goto fail;
|
|
}
|
|
|
|
openal->bIDs = g_malloc (openal->bID_count * sizeof (*openal->bIDs));
|
|
if (!openal->bIDs) {
|
|
GST_ELEMENT_ERROR (openal, RESOURCE, FAILED, ("Out of memory."),
|
|
("Unable to allocate buffer IDs"));
|
|
goto fail;
|
|
}
|
|
|
|
alGenBuffers (openal->bID_count, openal->bIDs);
|
|
if (checkALError () != AL_NO_ERROR) {
|
|
GST_ELEMENT_ERROR (openal, RESOURCE, NO_SPACE_LEFT, (NULL),
|
|
("Unable to generate %d buffers", openal->bID_count));
|
|
goto fail;
|
|
}
|
|
openal->bID_idx = 0;
|
|
|
|
popContext (old, ctx);
|
|
openal->context = ctx;
|
|
return TRUE;
|
|
|
|
fail:
|
|
if (!openal->custom_sID && openal->sID)
|
|
alDeleteSources (1, &openal->sID);
|
|
openal->sID = 0;
|
|
|
|
g_free (openal->bIDs);
|
|
openal->bIDs = NULL;
|
|
openal->bID_count = 0;
|
|
openal->bID_length = 0;
|
|
|
|
popContext (old, ctx);
|
|
if (!openal->custom_ctx)
|
|
alcDestroyContext (ctx);
|
|
return FALSE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_openal_sink_unprepare (GstAudioSink * asink)
|
|
{
|
|
GstOpenALSink *openal = GST_OPENAL_SINK (asink);
|
|
ALCcontext *old;
|
|
|
|
if (!openal->context)
|
|
return TRUE;
|
|
|
|
old = pushContext (openal->context);
|
|
|
|
alSourceStop (openal->sID);
|
|
alSourcei (openal->sID, AL_BUFFER, 0);
|
|
|
|
if (!openal->custom_sID)
|
|
alDeleteSources (1, &openal->sID);
|
|
openal->sID = 0;
|
|
|
|
alDeleteBuffers (openal->bID_count, openal->bIDs);
|
|
g_free (openal->bIDs);
|
|
openal->bIDs = NULL;
|
|
openal->bID_idx = 0;
|
|
openal->bID_count = 0;
|
|
openal->bID_length = 0;
|
|
|
|
checkALError ();
|
|
popContext (old, openal->context);
|
|
if (!openal->custom_ctx)
|
|
alcDestroyContext (openal->context);
|
|
openal->context = NULL;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static guint
|
|
gst_openal_sink_write (GstAudioSink * asink, gpointer data, guint length)
|
|
{
|
|
GstOpenALSink *openal = GST_OPENAL_SINK (asink);
|
|
ALint processed, queued, state;
|
|
ALCcontext *old;
|
|
gulong rest_us;
|
|
|
|
g_assert (length == openal->bID_length);
|
|
|
|
old = pushContext (openal->context);
|
|
|
|
rest_us = (guint64) (openal->bID_length / openal->bytes_per_sample) *
|
|
G_USEC_PER_SEC / openal->srate / 2;
|
|
do {
|
|
alGetSourcei (openal->sID, AL_SOURCE_STATE, &state);
|
|
alGetSourcei (openal->sID, AL_BUFFERS_QUEUED, &queued);
|
|
alGetSourcei (openal->sID, AL_BUFFERS_PROCESSED, &processed);
|
|
if (checkALError () != AL_NO_ERROR) {
|
|
GST_ELEMENT_ERROR (openal, RESOURCE, WRITE, (NULL),
|
|
("Source state error detected"));
|
|
length = 0;
|
|
goto out_nolock;
|
|
}
|
|
|
|
if (processed > 0 || queued < openal->bID_count)
|
|
break;
|
|
if (state != AL_PLAYING)
|
|
alSourcePlay (openal->sID);
|
|
g_usleep (rest_us);
|
|
} while (1);
|
|
|
|
GST_OPENAL_SINK_LOCK (openal);
|
|
if (openal->write_reset != AL_FALSE) {
|
|
openal->write_reset = AL_FALSE;
|
|
length = 0;
|
|
goto out;
|
|
}
|
|
|
|
queued -= processed;
|
|
while (processed-- > 0) {
|
|
ALuint bid;
|
|
alSourceUnqueueBuffers (openal->sID, 1, &bid);
|
|
}
|
|
if (state == AL_STOPPED) {
|
|
/* "Restore" from underruns (not actually needed, but it keeps delay
|
|
* calculations correct while rebuffering) */
|
|
alSourceRewind (openal->sID);
|
|
}
|
|
|
|
alBufferData (openal->bIDs[openal->bID_idx], openal->format,
|
|
data, openal->bID_length, openal->srate);
|
|
alSourceQueueBuffers (openal->sID, 1, &openal->bIDs[openal->bID_idx]);
|
|
openal->bID_idx = (openal->bID_idx + 1) % openal->bID_count;
|
|
queued++;
|
|
|
|
if (state != AL_PLAYING && queued == openal->bID_count)
|
|
alSourcePlay (openal->sID);
|
|
|
|
if (checkALError () != ALC_NO_ERROR) {
|
|
GST_ELEMENT_ERROR (openal, RESOURCE, WRITE, (NULL),
|
|
("Source queue error detected"));
|
|
goto out;
|
|
}
|
|
|
|
out:
|
|
GST_OPENAL_SINK_UNLOCK (openal);
|
|
out_nolock:
|
|
popContext (old, openal->context);
|
|
return length;
|
|
}
|
|
|
|
static guint
|
|
gst_openal_sink_delay (GstAudioSink * asink)
|
|
{
|
|
GstOpenALSink *openal = GST_OPENAL_SINK (asink);
|
|
ALint queued, state, offset, delay;
|
|
ALCcontext *old;
|
|
|
|
if (!openal->context)
|
|
return 0;
|
|
|
|
GST_OPENAL_SINK_LOCK (openal);
|
|
old = pushContext (openal->context);
|
|
|
|
delay = 0;
|
|
alGetSourcei (openal->sID, AL_BUFFERS_QUEUED, &queued);
|
|
/* Order here is important. If the offset is queried after the state and an
|
|
* underrun occurs in between the two calls, it can end up with a 0 offset
|
|
* in a playing state, incorrectly reporting a len*queued/bps delay. */
|
|
alGetSourcei (openal->sID, AL_BYTE_OFFSET, &offset);
|
|
alGetSourcei (openal->sID, AL_SOURCE_STATE, &state);
|
|
|
|
/* Note: state=stopped is an underrun, meaning all buffers are processed
|
|
* and there's no delay when writing the next buffer. Pre-buffering is
|
|
* state=initial, which will introduce a delay while writing. */
|
|
if (checkALError () == AL_NO_ERROR && state != AL_STOPPED)
|
|
delay = ((queued * openal->bID_length) - offset) / openal->bytes_per_sample;
|
|
|
|
popContext (old, openal->context);
|
|
GST_OPENAL_SINK_UNLOCK (openal);
|
|
|
|
return delay;
|
|
}
|
|
|
|
static void
|
|
gst_openal_sink_reset (GstAudioSink * asink)
|
|
{
|
|
GstOpenALSink *openal = GST_OPENAL_SINK (asink);
|
|
ALCcontext *old;
|
|
|
|
GST_OPENAL_SINK_LOCK (openal);
|
|
old = pushContext (openal->context);
|
|
|
|
openal->write_reset = AL_TRUE;
|
|
alSourceStop (openal->sID);
|
|
alSourceRewind (openal->sID);
|
|
alSourcei (openal->sID, AL_BUFFER, 0);
|
|
checkALError ();
|
|
|
|
popContext (old, openal->context);
|
|
GST_OPENAL_SINK_UNLOCK (openal);
|
|
}
|