/* * GStreamer * Copyright (C) 2005,2006 Zaheer Abbas Merali * Copyright (C) 2007,2008 Pioneers of the Inevitable * Copyright (C) 2012 Fluendo S.A. * * Permission is hereby granted, free of charge, to any person obtaining a * copy of this software and associated documentation files (the "Software"), * to deal in the Software without restriction, including without limitation * the rights to use, copy, modify, merge, publish, distribute, sublicense, * and/or sell copies of the Software, and to permit persons to whom the * Software is furnished to do so, subject to the following conditions: * * The above copyright notice and this permission notice shall be included in * all copies or substantial portions of the Software. * * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER * DEALINGS IN THE SOFTWARE. * * Alternatively, the contents of this file may be used under the * GNU Lesser General Public License Version 2.1 (the "LGPL"), in * which case the following provisions apply instead of the ones * mentioned above: * * 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. * * The development of this code was made possible due to the involvement of * Pioneers of the Inevitable, the creators of the Songbird Music player * */ /** * SECTION:element-osxaudiosink * * This element renders raw audio samples using the CoreAudio api. * * * Example pipelines * |[ * gst-launch filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! osxaudiosink * ]| Play an Ogg/Vorbis file. * * * Last reviewed on 2006-03-01 (0.10.4) */ #ifdef HAVE_CONFIG_H # include #endif #include #include #include #include "gstosxaudiosink.h" #include "gstosxaudioelement.h" GST_DEBUG_CATEGORY_STATIC (osx_audiosink_debug); #define GST_CAT_DEFAULT osx_audiosink_debug #include "gstosxcoreaudio.h" /* Filter signals and args */ enum { /* FILL ME */ LAST_SIGNAL }; enum { ARG_0, ARG_DEVICE, ARG_VOLUME }; #define DEFAULT_VOLUME 1.0 static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, " "endianness = (int) {" G_STRINGIFY (G_BYTE_ORDER) " }, " "signed = (boolean) { TRUE }, " "width = (int) 32, " "depth = (int) 32, " "rate = (int) [1, MAX], " "channels = (int) [1, 9];" "audio/x-raw-int, " "endianness = (int) {" G_STRINGIFY (G_BYTE_ORDER) " }, " "signed = (boolean) { TRUE }, " "width = (int) 32, " "depth = (int) 32, " "rate = (int) [1, MAX], " "channels = (int) [1, 9];" "audio/x-raw-int, " "endianness = (int) {" G_STRINGIFY (G_BYTE_ORDER) " }, " "signed = (boolean) { TRUE }, " "width = (int) 24, " "depth = (int) 24, " "rate = (int) [1, MAX], " "channels = (int) [1, 9];" "audio/x-raw-int, " "endianness = (int) {" G_STRINGIFY (G_BYTE_ORDER) " }, " "signed = (boolean) { TRUE }, " "width = (int) 16, " "depth = (int) 16, " "rate = (int) [1, MAX], " "channels = (int) [1, 9];" "audio/x-raw-int, " "endianness = (int) {" G_STRINGIFY (G_BYTE_ORDER) " }, " "signed = (boolean) { TRUE }, " "width = (int) 8, " "depth = (int) 8, " "rate = (int) [1, MAX], " "channels = (int) [1, MAX];" "audio/x-ac3, framed = (boolean) true;" "audio/x-dts, framed = (boolean) true") ); static void gst_osx_audio_sink_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_osx_audio_sink_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static gboolean gst_osx_audio_sink_stop (GstBaseSink * base); static GstCaps *gst_osx_audio_sink_getcaps (GstBaseSink * base); static gboolean gst_osx_audio_sink_acceptcaps (GstPad * pad, GstCaps * caps); static GstBuffer *gst_osx_audio_sink_sink_payload (GstBaseAudioSink * sink, GstBuffer * buf); static GstRingBuffer *gst_osx_audio_sink_create_ringbuffer (GstBaseAudioSink * sink); static void gst_osx_audio_sink_osxelement_init (gpointer g_iface, gpointer iface_data); static gboolean gst_osx_audio_sink_select_device (GstOsxAudioSink * osxsink); static void gst_osx_audio_sink_set_volume (GstOsxAudioSink * sink); static OSStatus gst_osx_audio_sink_io_proc (GstOsxRingBuffer * buf, AudioUnitRenderActionFlags * ioActionFlags, const AudioTimeStamp * inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList * bufferList); static void gst_osx_audio_sink_do_init (GType type) { static const GInterfaceInfo osxelement_info = { gst_osx_audio_sink_osxelement_init, NULL, NULL }; GST_DEBUG_CATEGORY_INIT (osx_audiosink_debug, "osxaudiosink", 0, "OSX Audio Sink"); gst_core_audio_init_debug (); GST_DEBUG ("Adding static interface"); g_type_add_interface_static (type, GST_OSX_AUDIO_ELEMENT_TYPE, &osxelement_info); } GST_BOILERPLATE_FULL (GstOsxAudioSink, gst_osx_audio_sink, GstBaseAudioSink, GST_TYPE_BASE_AUDIO_SINK, gst_osx_audio_sink_do_init); static void gst_osx_audio_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, "Audio Sink (OSX)", "Sink/Audio", "Output to a sound card in OS X", "Zaheer Abbas Merali "); } static void gst_osx_audio_sink_class_init (GstOsxAudioSinkClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstBaseSinkClass *gstbasesink_class; GstBaseAudioSinkClass *gstbaseaudiosink_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstbasesink_class = (GstBaseSinkClass *) klass; gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass; parent_class = g_type_class_peek_parent (klass); gobject_class->set_property = gst_osx_audio_sink_set_property; gobject_class->get_property = gst_osx_audio_sink_get_property; #ifndef HAVE_IOS g_object_class_install_property (gobject_class, ARG_DEVICE, g_param_spec_int ("device", "Device ID", "Device ID of output device", 0, G_MAXINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); #endif g_object_class_install_property (gobject_class, ARG_VOLUME, g_param_spec_double ("volume", "Volume", "Volume of this stream", 0, 1.0, 1.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_osx_audio_sink_getcaps); gstbasesink_class->stop = GST_DEBUG_FUNCPTR (gst_osx_audio_sink_stop); gstbaseaudiosink_class->create_ringbuffer = GST_DEBUG_FUNCPTR (gst_osx_audio_sink_create_ringbuffer); gstbaseaudiosink_class->payload = GST_DEBUG_FUNCPTR (gst_osx_audio_sink_sink_payload); } static void gst_osx_audio_sink_init (GstOsxAudioSink * sink, GstOsxAudioSinkClass * gclass) { GST_DEBUG ("Initialising object"); sink->device_id = kAudioDeviceUnknown; sink->cached_caps = NULL; sink->volume = DEFAULT_VOLUME; gst_pad_set_acceptcaps_function (GST_BASE_SINK (sink)->sinkpad, GST_DEBUG_FUNCPTR (gst_osx_audio_sink_acceptcaps)); } static void gst_osx_audio_sink_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstOsxAudioSink *sink = GST_OSX_AUDIO_SINK (object); switch (prop_id) { #ifndef HAVE_IOS case ARG_DEVICE: sink->device_id = g_value_get_int (value); break; #endif case ARG_VOLUME: sink->volume = g_value_get_double (value); gst_osx_audio_sink_set_volume (sink); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_osx_audio_sink_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstOsxAudioSink *sink = GST_OSX_AUDIO_SINK (object); switch (prop_id) { #ifndef HAVE_IOS case ARG_DEVICE: g_value_set_int (value, sink->device_id); break; #endif case ARG_VOLUME: g_value_set_double (value, sink->volume); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static gboolean gst_osx_audio_sink_stop (GstBaseSink * base) { GstOsxAudioSink *sink = GST_OSX_AUDIO_SINK (base); if (sink->cached_caps) { gst_caps_unref (sink->cached_caps); sink->cached_caps = NULL; } return GST_CALL_PARENT_WITH_DEFAULT (GST_BASE_SINK_CLASS, stop, (base), TRUE); } static GstCaps * gst_osx_audio_sink_getcaps (GstBaseSink * base) { GstOsxAudioSink *sink = GST_OSX_AUDIO_SINK (base); gchar *caps_string = NULL; if (sink->cached_caps) { caps_string = gst_caps_to_string (sink->cached_caps); GST_DEBUG_OBJECT (sink, "using cached caps: %s", caps_string); g_free (caps_string); return gst_caps_ref (sink->cached_caps); } GST_DEBUG_OBJECT (sink, "using template caps"); return NULL; } static gboolean gst_osx_audio_sink_acceptcaps (GstPad * pad, GstCaps * caps) { GstOsxAudioSink *sink = GST_OSX_AUDIO_SINK (gst_pad_get_parent_element (pad)); GstOsxRingBuffer *osxbuf; GstCaps *pad_caps; GstStructure *st; gboolean ret = FALSE; GstRingBufferSpec spec = { 0 }; gchar *caps_string = NULL; osxbuf = GST_OSX_RING_BUFFER (GST_BASE_AUDIO_SINK (sink)->ringbuffer); caps_string = gst_caps_to_string (caps); GST_DEBUG_OBJECT (sink, "acceptcaps called with %s", caps_string); g_free (caps_string); pad_caps = gst_pad_get_caps_reffed (pad); if (pad_caps) { gboolean cret = gst_caps_can_intersect (pad_caps, caps); gst_caps_unref (pad_caps); if (!cret) goto done; } /* If we've not got fixed caps, creating a stream might fail, * so let's just return from here with default acceptcaps * behaviour */ if (!gst_caps_is_fixed (caps)) goto done; /* parse helper expects this set, so avoid nasty warning * will be set properly later on anyway */ spec.latency_time = GST_SECOND; if (!gst_ring_buffer_parse_caps (&spec, caps)) goto done; /* Make sure input is framed and can be payloaded */ switch (spec.type) { case GST_BUFTYPE_AC3: { gboolean framed = FALSE; st = gst_caps_get_structure (caps, 0); gst_structure_get_boolean (st, "framed", &framed); if (!framed || gst_audio_iec61937_frame_size (&spec) <= 0) goto done; break; } case GST_BUFTYPE_DTS: { gboolean parsed = FALSE; st = gst_caps_get_structure (caps, 0); gst_structure_get_boolean (st, "parsed", &parsed); if (!parsed || gst_audio_iec61937_frame_size (&spec) <= 0) goto done; break; } default: break; } ret = TRUE; done: gst_object_unref (sink); return ret; } static GstBuffer * gst_osx_audio_sink_sink_payload (GstBaseAudioSink * sink, GstBuffer * buf) { GstOsxAudioSink *osxsink; osxsink = GST_OSX_AUDIO_SINK (sink); if (RINGBUFFER_IS_SPDIF (sink->ringbuffer->spec.type)) { gint framesize = gst_audio_iec61937_frame_size (&sink->ringbuffer->spec); GstBuffer *out; if (framesize <= 0) return NULL; out = gst_buffer_new_and_alloc (framesize); if (!gst_audio_iec61937_payload (GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf), GST_BUFFER_DATA (out), GST_BUFFER_SIZE (out), &sink->ringbuffer->spec)) { gst_buffer_unref (out); return NULL; } gst_buffer_copy_metadata (out, buf, GST_BUFFER_COPY_ALL); /* Fix endianness */ swab ((gchar *) GST_BUFFER_DATA (buf), (gchar *) GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf)); return out; } else { return gst_buffer_ref (buf); } } static GstRingBuffer * gst_osx_audio_sink_create_ringbuffer (GstBaseAudioSink * sink) { GstOsxAudioSink *osxsink; GstOsxRingBuffer *ringbuffer; osxsink = GST_OSX_AUDIO_SINK (sink); if (!gst_osx_audio_sink_select_device (osxsink)) { GST_ERROR_OBJECT (sink, "Could not select device"); return NULL; } GST_DEBUG_OBJECT (sink, "Creating ringbuffer"); ringbuffer = g_object_new (GST_TYPE_OSX_RING_BUFFER, NULL); GST_DEBUG_OBJECT (sink, "osx sink %p element %p ioproc %p", osxsink, GST_OSX_AUDIO_ELEMENT_GET_INTERFACE (osxsink), (void *) gst_osx_audio_sink_io_proc); gst_osx_audio_sink_set_volume (osxsink); ringbuffer->core_audio->element = GST_OSX_AUDIO_ELEMENT_GET_INTERFACE (osxsink); ringbuffer->core_audio->device_id = osxsink->device_id; ringbuffer->core_audio->is_src = FALSE; return GST_RING_BUFFER (ringbuffer); } /* HALOutput AudioUnit will request fairly arbitrarily-sized chunks * of data, not of a fixed size. So, we keep track of where in * the current ringbuffer segment we are, and only advance the segment * once we've read the whole thing */ static OSStatus gst_osx_audio_sink_io_proc (GstOsxRingBuffer * buf, AudioUnitRenderActionFlags * ioActionFlags, const AudioTimeStamp * inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList * bufferList) { guint8 *readptr; gint readseg; gint len; gint stream_idx = buf->core_audio->stream_idx; gint remaining = bufferList->mBuffers[stream_idx].mDataByteSize; gint offset = 0; while (remaining) { if (!gst_ring_buffer_prepare_read (GST_RING_BUFFER (buf), &readseg, &readptr, &len)) return 0; len -= buf->segoffset; if (len > remaining) len = remaining; memcpy ((char *) bufferList->mBuffers[stream_idx].mData + offset, readptr + buf->segoffset, len); buf->segoffset += len; offset += len; remaining -= len; if ((gint) buf->segoffset == GST_RING_BUFFER (buf)->spec.segsize) { /* clear written samples */ gst_ring_buffer_clear (GST_RING_BUFFER (buf), readseg); /* we wrote one segment */ gst_ring_buffer_advance (GST_RING_BUFFER (buf), 1); buf->segoffset = 0; } } return 0; } static void gst_osx_audio_sink_osxelement_init (gpointer g_iface, gpointer iface_data) { GstOsxAudioElementInterface *iface = (GstOsxAudioElementInterface *) g_iface; iface->io_proc = (AURenderCallback) gst_osx_audio_sink_io_proc; } static void gst_osx_audio_sink_set_volume (GstOsxAudioSink * sink) { GstOsxRingBuffer *osxbuf; osxbuf = GST_OSX_RING_BUFFER (GST_BASE_AUDIO_SINK (sink)->ringbuffer); if (!osxbuf) return; gst_core_audio_set_volume (osxbuf->core_audio, sink->volume); } static gboolean gst_osx_audio_sink_allowed_caps (GstOsxAudioSink * osxsink) { gint i, max_channels = 0; gboolean spdif_allowed, use_positions = FALSE; AudioChannelLayout *layout; GstElementClass *element_class; GstPadTemplate *pad_template; GstCaps *caps, *in_caps; GstAudioChannelPosition pos[9] = { GST_AUDIO_CHANNEL_POSITION_INVALID, GST_AUDIO_CHANNEL_POSITION_INVALID, GST_AUDIO_CHANNEL_POSITION_INVALID, GST_AUDIO_CHANNEL_POSITION_INVALID, GST_AUDIO_CHANNEL_POSITION_INVALID, GST_AUDIO_CHANNEL_POSITION_INVALID, GST_AUDIO_CHANNEL_POSITION_INVALID, GST_AUDIO_CHANNEL_POSITION_INVALID, GST_AUDIO_CHANNEL_POSITION_INVALID }; /* First collect info about the HW capabilites and preferences */ spdif_allowed = gst_core_audio_audio_device_is_spdif_avail (osxsink->device_id); layout = gst_core_audio_audio_device_get_channel_layout (osxsink->device_id); GST_DEBUG_OBJECT (osxsink, "Selected device ID: %u SPDIF allowed: %d", (unsigned) osxsink->device_id, spdif_allowed); if (layout) { max_channels = layout->mNumberChannelDescriptions; } else { GST_WARNING_OBJECT (osxsink, "This driver does not support " "kAudioDevicePropertyPreferredChannelLayout."); max_channels = 2; } if (max_channels > 2) { max_channels = MIN (max_channels, 9); use_positions = TRUE; for (i = 0; i < max_channels; i++) { switch (layout->mChannelDescriptions[i].mChannelLabel) { case kAudioChannelLabel_Left: pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; break; case kAudioChannelLabel_Right: pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; break; case kAudioChannelLabel_Center: pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER; break; case kAudioChannelLabel_LFEScreen: pos[i] = GST_AUDIO_CHANNEL_POSITION_LFE; break; case kAudioChannelLabel_LeftSurround: pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT; break; case kAudioChannelLabel_RightSurround: pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT; break; case kAudioChannelLabel_RearSurroundLeft: pos[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT; break; case kAudioChannelLabel_RearSurroundRight: pos[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT; break; case kAudioChannelLabel_CenterSurround: pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER; break; default: GST_WARNING_OBJECT (osxsink, "unrecognized channel: %d", (int) layout->mChannelDescriptions[i].mChannelLabel); use_positions = FALSE; max_channels = 2; break; } } } g_free (layout); /* Recover the template caps */ element_class = GST_ELEMENT_GET_CLASS (osxsink); pad_template = gst_element_class_get_pad_template (element_class, "sink"); in_caps = gst_pad_template_get_caps (pad_template); /* Create the allowed subset */ caps = gst_caps_new_empty (); for (i = 0; i < gst_caps_get_size (in_caps); i++) { GstStructure *in_s, *out_s; in_s = gst_caps_get_structure (in_caps, i); if (gst_structure_has_name (in_s, "audio/x-ac3") || gst_structure_has_name (in_s, "audio/x-dts")) { if (spdif_allowed) { gst_caps_append_structure (caps, gst_structure_copy (in_s)); } } else { if (max_channels > 2 && use_positions) { out_s = gst_structure_copy (in_s); gst_structure_remove_field (out_s, "channels"); gst_structure_set (out_s, "channels", G_TYPE_INT, max_channels, NULL); gst_audio_set_channel_positions (out_s, pos); gst_caps_append_structure (caps, out_s); } out_s = gst_structure_copy (in_s); gst_structure_remove_field (out_s, "channels"); gst_structure_set (out_s, "channels", GST_TYPE_INT_RANGE, 1, 2, NULL); gst_caps_append_structure (caps, out_s); } } if (osxsink->cached_caps) { gst_caps_unref (osxsink->cached_caps); } osxsink->cached_caps = caps; return TRUE; } static gboolean gst_osx_audio_sink_select_device (GstOsxAudioSink * osxsink) { gboolean res = FALSE; if (!gst_core_audio_select_device (&osxsink->device_id)) return FALSE; res = gst_osx_audio_sink_allowed_caps (osxsink); return res; }