author | Eckhart Koeppen <eckhart.koppen@nokia.com> |
Wed, 21 Apr 2010 11:15:19 +0300 | |
branch | RCL_3 |
changeset 11 | 25a739ee40f4 |
parent 3 | 41300fa6a67c |
permissions | -rw-r--r-- |
0 | 1 |
/* This file is part of the KDE project. |
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Copyright (C) 2009 Nokia Corporation and/or its subsidiary(-ies). |
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This library 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 published by |
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the Free Software Foundation, either version 2.1 or 3 of the License. |
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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 |
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GNU Lesser General Public License for more details. |
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You should have received a copy of the GNU Lesser General Public License |
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along with this library. If not, see <http://www.gnu.org/licenses/>. |
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*/ |
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/***************************************** |
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* |
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* This is an aRts plugin for GStreamer |
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* |
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****************************************/ |
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#include <gst/gst.h> |
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#include <gst/audio/audio.h> |
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#include <gst/audio/gstaudiosink.h> |
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#include "artssink.h" |
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QT_BEGIN_NAMESPACE |
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namespace Phonon |
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{ |
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namespace Gstreamer |
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{ |
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static GstStaticPadTemplate sinktemplate = |
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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 ( |
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"audio/x-raw-int, " |
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"width = (int) { 8, 16 }, " |
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"depth = (int) { 8, 16 }, " |
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"endianness = (int) BYTE_ORDER, " |
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"channels = (int) { 1, 2 }, " |
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"rate = (int) [ 8000, 96000 ]" |
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) |
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); |
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typedef int (*Ptr_arts_init)(); |
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typedef arts_stream_t (*Ptr_arts_play_stream)(int, int, int, const char*); |
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typedef int (*Ptr_arts_close_stream)(arts_stream_t); |
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typedef int (*Ptr_arts_stream_get)(arts_stream_t, arts_parameter_t_enum); |
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typedef int (*Ptr_arts_stream_set)(arts_stream_t, arts_parameter_t_enum, int value); |
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typedef int (*Ptr_arts_write)(arts_stream_t, const void *, int); |
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typedef int (*Ptr_arts_suspended)(); |
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typedef void (*Ptr_arts_free)(); |
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static Ptr_arts_init p_arts_init = 0; |
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static Ptr_arts_play_stream p_arts_play_stream = 0; |
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static Ptr_arts_close_stream p_arts_close_stream = 0; |
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static Ptr_arts_stream_get p_arts_stream_get= 0; |
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static Ptr_arts_stream_set p_arts_stream_set= 0; |
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static Ptr_arts_write p_arts_write = 0; |
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static Ptr_arts_suspended p_arts_suspended = 0; |
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static Ptr_arts_free p_arts_free = 0; |
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static void arts_sink_dispose (GObject * object); |
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static void arts_sink_reset (GstAudioSink * asink); |
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static void arts_sink_finalize (GObject * object); |
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static GstCaps *arts_sink_get_caps (GstBaseSink * bsink); |
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static gboolean arts_sink_open (GstAudioSink * asink); |
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static gboolean arts_sink_close (GstAudioSink * asink); |
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static gboolean arts_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec); |
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static gboolean arts_sink_unprepare (GstAudioSink * asink); |
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static guint arts_sink_write (GstAudioSink * asink, gpointer data, guint length); |
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static guint arts_sink_delay (GstAudioSink * asink); |
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static gboolean connected = false; |
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static gboolean init = false; |
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static int sinkCount; |
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GST_BOILERPLATE (ArtsSink, arts_sink, GstAudioSink, GST_TYPE_AUDIO_SINK) |
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// ArtsSink args |
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enum |
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{ |
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ARG_0, |
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ARG_ARTSSINK |
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}; |
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/* open the device with given specs */ |
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gboolean arts_sink_open(GstAudioSink *sink) |
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{ |
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Q_UNUSED(sink); |
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// We already have an open connection to this device |
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if (!init) { |
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GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE, (NULL), ("Could not connect to aRts", NULL)); |
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return false; |
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} else if (connected) { |
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GST_ELEMENT_ERROR (sink, RESOURCE, BUSY, (NULL), ("Device is busy", NULL)); |
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return false; |
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} |
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// Check if all symbols were resolved |
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if (!(p_arts_init && p_arts_play_stream && p_arts_close_stream |
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&& p_arts_stream_get && p_arts_stream_set && p_arts_write && p_arts_free)) |
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return FALSE; |
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// Check if arts_init succeeded |
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if (!init) |
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return false; |
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return true; |
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} |
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/* prepare resources and state to operate with the given specs */ |
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static gboolean arts_sink_prepare(GstAudioSink *sink, GstRingBufferSpec *spec) |
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{ |
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ArtsSink *asink = (ArtsSink*)sink; |
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if (!init) |
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return false; |
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asink->samplerate = spec->rate; |
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asink->samplebits = spec->depth; |
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asink->channels = spec->channels; |
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asink->bytes_per_sample = spec->bytes_per_sample; |
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static int id = 0; |
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asink->stream = p_arts_play_stream(spec->rate, spec->depth, spec->channels, |
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QString("gstreamer-%0").arg(id++).toLatin1().constData()); |
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if (asink->stream) |
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connected = true; |
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return connected; |
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} |
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/* undo anything that was done in prepare() */ |
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static gboolean arts_sink_unprepare(GstAudioSink *sink) |
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{ |
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Q_UNUSED(sink); |
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ArtsSink *asink = (ArtsSink*)sink; |
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if (init && connected) { |
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p_arts_close_stream(asink->stream); |
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connected = false; |
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} |
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return true; |
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} |
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/* close the device */ |
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static gboolean arts_sink_close(GstAudioSink *sink) |
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{ |
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Q_UNUSED(sink); |
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return true; |
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} |
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/* write samples to the device */ |
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static guint arts_sink_write(GstAudioSink *sink, gpointer data, guint length) |
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{ |
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ArtsSink *asink = (ArtsSink*)sink; |
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if (!init) |
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return 0; |
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int errorcode = p_arts_write(asink->stream, (char*)data, length); |
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if (errorcode < 0) |
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GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL), ("Could not write to device.", NULL)); |
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return errorcode > 0 ? errorcode : 0; |
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} |
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/* get number of samples queued in the device */ |
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static guint arts_sink_delay(GstAudioSink *sink) |
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{ |
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ArtsSink *asink = (ArtsSink*)sink; |
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if (!init) |
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return 0; |
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// We get results in millisecons so we have to caculate the approximate size in samples |
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guint delay = p_arts_stream_get(asink->stream, ARTS_P_SERVER_LATENCY) * (asink->samplerate / 1000); |
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return delay; |
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} |
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/* reset the audio device, unblock from a write */ |
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static void arts_sink_reset(GstAudioSink *sink) |
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{ |
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// ### We are currently unable to gracefully recover |
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// after artsd has been restarted or killed. |
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Q_UNUSED(sink); |
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} |
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// Register element details |
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static void arts_sink_base_init (gpointer g_class) { |
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class); |
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static gchar longname[] = "Experimental aRts sink", |
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klass[] = "Sink/Audio", |
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description[] = "aRts Audio Output Device", |
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author[] = "Nokia Corporation and/or its subsidiary(-ies) <qt-info@nokia.com>"; |
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GstElementDetails details = GST_ELEMENT_DETAILS (longname, |
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klass, |
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description, |
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author); |
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gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&sinktemplate)); |
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gst_element_class_set_details (gstelement_class, &details); |
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} |
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static void arts_sink_class_init (ArtsSinkClass * klass) |
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{ |
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parent_class = (GstAudioSinkClass*)g_type_class_peek_parent(klass); |
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass); |
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gobject_class->finalize = GST_DEBUG_FUNCPTR (arts_sink_finalize); |
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gobject_class->dispose = GST_DEBUG_FUNCPTR (arts_sink_dispose); |
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GstBaseSinkClass *gstbasesink_class = (GstBaseSinkClass *) klass; |
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gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (arts_sink_get_caps); |
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GstAudioSinkClass *gstaudiosink_class = (GstAudioSinkClass*)klass; |
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gstaudiosink_class->open = GST_DEBUG_FUNCPTR(arts_sink_open); |
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gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR(arts_sink_prepare); |
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gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR(arts_sink_unprepare); |
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gstaudiosink_class->close = GST_DEBUG_FUNCPTR(arts_sink_close); |
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gstaudiosink_class->write = GST_DEBUG_FUNCPTR(arts_sink_write); |
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gstaudiosink_class->delay = GST_DEBUG_FUNCPTR(arts_sink_delay); |
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gstaudiosink_class->reset = GST_DEBUG_FUNCPTR(arts_sink_reset); |
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} |
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static void arts_sink_init (ArtsSink * src, ArtsSinkClass * g_class) |
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{ |
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Q_UNUSED(g_class); |
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GST_DEBUG_OBJECT (src, "initializing artssink"); |
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src->stream = 0; |
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41300fa6a67c
Revision: 201003
Dremov Kirill (Nokia-D-MSW/Tampere) <kirill.dremov@nokia.com>
parents:
0
diff
changeset
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#ifndef QT_NO_LIBRARY |
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p_arts_init = (Ptr_arts_init)QLibrary::resolve(QLatin1String("artsc"), 0, "arts_init"); |
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p_arts_play_stream = (Ptr_arts_play_stream)QLibrary::resolve(QLatin1String("artsc"), 0, "arts_play_stream"); |
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p_arts_close_stream = (Ptr_arts_close_stream)QLibrary::resolve(QLatin1String("artsc"), 0, "arts_close_stream"); |
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p_arts_stream_get = (Ptr_arts_stream_get)QLibrary::resolve(QLatin1String("artsc"), 0, "arts_stream_get"); |
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p_arts_stream_set = (Ptr_arts_stream_set)QLibrary::resolve(QLatin1String("artsc"), 0, "arts_stream_set"); |
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p_arts_write = (Ptr_arts_write)QLibrary::resolve(QLatin1String("artsc"), 0, "arts_write"); |
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p_arts_suspended = (Ptr_arts_suspended)QLibrary::resolve(QLatin1String("artsc"), 0, "arts_suspended"); |
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p_arts_free = (Ptr_arts_free)QLibrary::resolve(QLatin1String("artsc"), 0, "arts_free"); |
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if (!sinkCount) { |
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int errorcode = p_arts_init(); |
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if (!errorcode) { |
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init = TRUE; |
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} |
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} |
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sinkCount ++; |
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41300fa6a67c
Revision: 201003
Dremov Kirill (Nokia-D-MSW/Tampere) <kirill.dremov@nokia.com>
parents:
0
diff
changeset
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#endif //QT_NO_LIBRARY |
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} |
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static void arts_sink_dispose (GObject * object) |
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{ |
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Q_UNUSED(object); |
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if (--sinkCount == 0) { |
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p_arts_free(); |
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} |
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} |
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static void arts_sink_finalize (GObject * object) |
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{ |
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G_OBJECT_CLASS (parent_class)->finalize (object); |
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} |
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static GstCaps *arts_sink_get_caps (GstBaseSink * bsink) |
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{ |
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Q_UNUSED(bsink); |
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return NULL; |
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} |
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} |
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} //namespace Phonon::Gstreamer |
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QT_END_NAMESPACE |