/* GStreamer * Copyright (C) 2012 Fluendo S.A. * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ #ifdef HAVE_CONFIG_H # include #endif #include "openslessink.h" GST_DEBUG_CATEGORY_STATIC (opensles_sink_debug); #define GST_CAT_DEFAULT opensles_sink_debug enum { PROP_0, PROP_VOLUME, PROP_MUTE, PROP_LAST }; #define DEFAULT_VOLUME 1.0 #define DEFAULT_MUTE FALSE /* According to Android's NDK doc the following are the supported rates */ #define RATES "8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000" static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-int, " "endianness = (int) {" G_STRINGIFY (G_BYTE_ORDER) " }, " "signed = (boolean) { TRUE }, " "width = (int) 16, " "depth = (int) 16, " "rate = (int) { " RATES "}, " "channels = (int) [1, 2];" "audio/x-raw-int, " "endianness = (int) {" G_STRINGIFY (G_BYTE_ORDER) " }, " "signed = (boolean) { FALSE }, " "width = (int) 8, " "depth = (int) 8, " "rate = (int) { " RATES "}, " "channels = (int) [1, 2]") ); static void _do_init (GType type) { GST_DEBUG_CATEGORY_INIT (opensles_sink_debug, "opensles_sink", 0, "OpenSL ES Sink"); } GST_BOILERPLATE_FULL (GstOpenSLESSink, gst_opensles_sink, GstBaseAudioSink, GST_TYPE_BASE_AUDIO_SINK, _do_init); static void gst_opensles_sink_base_init (gpointer g_class) { GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); gst_element_class_add_static_pad_template (element_class, &sink_factory); gst_element_class_set_details_simple (element_class, "OpenSL ES Sink", "Sink/Audio", "Output sound using the OpenSL ES APIs", "Josep Torra "); } static GstRingBuffer * gst_opensles_sink_create_ringbuffer (GstBaseAudioSink * base) { GstOpenSLESSink *sink = GST_OPENSLES_SINK (base); GstRingBuffer *rb; rb = gst_opensles_ringbuffer_new (RB_MODE_SINK_PCM); gst_opensles_ringbuffer_set_volume (rb, sink->volume); gst_opensles_ringbuffer_set_mute (rb, sink->mute); return rb; } #define AUDIO_OUTPUT_DESC_FORMAT \ "deviceName: %s deviceConnection: %d deviceScope: %d deviceLocation: %d " \ "isForTelephony: %d minSampleRate: %d maxSampleRate: %d " \ "isFreqRangeContinuous: %d maxChannels: %d" #define AUDIO_OUTPUT_DESC_ARGS(aod) \ (gchar*) (aod)->pDeviceName, (gint) (aod)->deviceConnection, \ (gint) (aod)->deviceScope, (gint) (aod)->deviceLocation, \ (gint) (aod)->isForTelephony, (gint) (aod)->minSampleRate, \ (gint) (aod)->maxSampleRate, (gint) (aod)->isFreqRangeContinuous, \ (gint) (aod)->maxChannels /* Next it's not defined in Android */ #ifndef MAX_NUMBER_OUTPUT_DEVICES #define MAX_NUMBER_OUTPUT_DEVICES 16 #endif static gboolean _opensles_query_capabilities (GstOpenSLESSink * sink) { gboolean res = FALSE; SLresult result; SLObjectItf engineObject = NULL; SLAudioIODeviceCapabilitiesItf audioIODeviceCapabilities; SLint32 i, j, numOutputs = MAX_NUMBER_OUTPUT_DEVICES; SLuint32 outputDeviceIDs[MAX_NUMBER_OUTPUT_DEVICES]; SLAudioOutputDescriptor audioOutputDescriptor; /* Create engine */ result = slCreateEngine (&engineObject, 0, NULL, 0, NULL, NULL); if (result != SL_RESULT_SUCCESS) { GST_ERROR_OBJECT (sink, "slCreateEngine failed(0x%08x)", (guint32) result); goto beach; } /* Realize the engine */ result = (*engineObject)->Realize (engineObject, SL_BOOLEAN_FALSE); if (result != SL_RESULT_SUCCESS) { GST_ERROR_OBJECT (sink, "engine.Realize failed(0x%08x)", (guint32) result); goto beach; } /* Get the engine interface, which is needed in order to * create other objects */ result = (*engineObject)->GetInterface (engineObject, SL_IID_AUDIOIODEVICECAPABILITIES, &audioIODeviceCapabilities); if (result != SL_RESULT_SUCCESS) { GST_ERROR_OBJECT (sink, "engine.GetInterface(IODeviceCapabilities) failed(0x%08x)", (guint32) result); goto beach; } result = (*audioIODeviceCapabilities)->GetAvailableAudioOutputs (audioIODeviceCapabilities, &numOutputs, outputDeviceIDs); if (result != SL_RESULT_SUCCESS) { GST_ERROR_OBJECT (sink, "IODeviceCapabilities.GetAvailableAudioOutputs failed(0x%08x)", (guint32) result); goto beach; } GST_DEBUG_OBJECT (sink, "Found %d output devices", (gint32) numOutputs); for (i = 0; i < numOutputs; i++) { result = (*audioIODeviceCapabilities)->QueryAudioOutputCapabilities (audioIODeviceCapabilities, outputDeviceIDs[i], &audioOutputDescriptor); if (result != SL_RESULT_SUCCESS) { GST_ERROR_OBJECT (sink, "IODeviceCapabilities.QueryAudioOutputCapabilities failed(0x%08x)", (guint32) result); continue; } GST_DEBUG_OBJECT (sink, " ID: %08x " AUDIO_OUTPUT_DESC_FORMAT, (guint) outputDeviceIDs[i], AUDIO_OUTPUT_DESC_ARGS (&audioOutputDescriptor)); GST_DEBUG_OBJECT (sink, " Found %d supported sample rated", audioOutputDescriptor.numOfSamplingRatesSupported); for (j = 0; j < audioOutputDescriptor.numOfSamplingRatesSupported; j++) { GST_DEBUG_OBJECT (sink, " %d Hz", (gint) audioOutputDescriptor.samplingRatesSupported[j]); } } res = TRUE; beach: /* Destroy engine object */ if (engineObject) { (*engineObject)->Destroy (engineObject); } return res; } static void gst_opensles_sink_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstOpenSLESSink *sink = GST_OPENSLES_SINK (object); GstRingBuffer *rb = GST_BASE_AUDIO_SINK (sink)->ringbuffer; switch (prop_id) { case PROP_VOLUME: sink->volume = g_value_get_double (value); if (rb && GST_IS_OPENSLES_RING_BUFFER (rb)) { gst_opensles_ringbuffer_set_volume (rb, sink->volume); } break; case PROP_MUTE: sink->mute = g_value_get_boolean (value); if (rb && GST_IS_OPENSLES_RING_BUFFER (rb)) { gst_opensles_ringbuffer_set_mute (rb, sink->mute); } break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_opensles_sink_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstOpenSLESSink *sink = GST_OPENSLES_SINK (object); switch (prop_id) { case PROP_VOLUME: g_value_set_double (value, sink->volume); break; case PROP_MUTE: g_value_set_boolean (value, sink->mute); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_opensles_sink_class_init (GstOpenSLESSinkClass * klass) { GObjectClass *gobject_class; GstBaseAudioSinkClass *gstbaseaudiosink_class; gobject_class = (GObjectClass *) klass; gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass; parent_class = g_type_class_peek_parent (klass); gobject_class->set_property = gst_opensles_sink_set_property; gobject_class->get_property = gst_opensles_sink_get_property; g_object_class_install_property (gobject_class, PROP_VOLUME, g_param_spec_double ("volume", "Volume", "Volume of this stream", 0, 1.0, 1.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_MUTE, g_param_spec_boolean ("mute", "Mute", "Mute state of this stream", DEFAULT_MUTE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gstbaseaudiosink_class->create_ringbuffer = GST_DEBUG_FUNCPTR (gst_opensles_sink_create_ringbuffer); } static void gst_opensles_sink_init (GstOpenSLESSink * sink, GstOpenSLESSinkClass * gclass) { sink->volume = DEFAULT_VOLUME; sink->mute = DEFAULT_MUTE; _opensles_query_capabilities (sink); gst_base_audio_sink_set_provide_clock (GST_BASE_AUDIO_SINK (sink), TRUE); }