/* * GStreamer * Copyright (C) 2005,2006 Zaheer Abbas Merali <zaheerabbas at merali dot org> * Copyright (C) 2007,2008 Pioneers of the Inevitable <songbird@songbirdnest.com> * Copyright (C) 2012 Fluendo S.A. <support@fluendo.com> * * 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., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, 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. * * <refsect2> * <title>Example pipelines</title> * |[ * gst-launch-1.0 filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! osxaudiosink * ]| Play an Ogg/Vorbis file. * </refsect2> * * Last reviewed on 2006-03-01 (0.10.4) */ #ifdef HAVE_CONFIG_H # include <config.h> #endif #include <gst/gst.h> #include <gst/audio/audio.h> #include <gst/audio/audio-channels.h> #include <gst/audio/gstaudioiec61937.h> #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 #if (G_BYTE_ORDER == G_LITTLE_ENDIAN) # define FORMATS "{ S32LE, S24LE, S16LE, U8 }" #else # define FORMATS "{ S32BE, S24BE, S16BE, U8 }" #endif static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw, " "format = (string) " FORMATS ", " "layout = (string) interleaved, " "rate = (int) [1, MAX], " "channels = (int) [1, 9];" "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_query (GstBaseSink * base, GstQuery * query); static gboolean gst_osx_audio_sink_stop (GstBaseSink * base); static GstCaps *gst_osx_audio_sink_getcaps (GstBaseSink * base, GstCaps * filter); static gboolean gst_osx_audio_sink_acceptcaps (GstOsxAudioSink * sink, GstCaps * caps); static GstBuffer *gst_osx_audio_sink_sink_payload (GstAudioBaseSink * sink, GstBuffer * buf); static GstAudioRingBuffer * gst_osx_audio_sink_create_ringbuffer (GstAudioBaseSink * 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 (GstOsxAudioRingBuffer * 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); } #define gst_osx_audio_sink_parent_class parent_class G_DEFINE_TYPE_WITH_CODE (GstOsxAudioSink, gst_osx_audio_sink, GST_TYPE_AUDIO_BASE_SINK, gst_osx_audio_sink_do_init (g_define_type_id)); static void gst_osx_audio_sink_class_init (GstOsxAudioSinkClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstBaseSinkClass *gstbasesink_class; GstAudioBaseSinkClass *gstaudiobasesink_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstbasesink_class = (GstBaseSinkClass *) klass; gstaudiobasesink_class = (GstAudioBaseSinkClass *) 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 gstbasesink_class->query = GST_DEBUG_FUNCPTR (gst_osx_audio_sink_query); 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); gstaudiobasesink_class->create_ringbuffer = GST_DEBUG_FUNCPTR (gst_osx_audio_sink_create_ringbuffer); gstaudiobasesink_class->payload = GST_DEBUG_FUNCPTR (gst_osx_audio_sink_sink_payload); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&sink_factory)); gst_element_class_set_static_metadata (gstelement_class, "Audio Sink (OSX)", "Sink/Audio", "Output to a sound card in OS X", "Zaheer Abbas Merali <zaheerabbas at merali dot org>"); } static void gst_osx_audio_sink_init (GstOsxAudioSink * sink) { gint i; GST_DEBUG ("Initialising object"); sink->device_id = kAudioDeviceUnknown; sink->cached_caps = NULL; sink->volume = DEFAULT_VOLUME; sink->channels = 0; for (i = 0; i < GST_OSX_AUDIO_MAX_CHANNEL; i++) { sink->channel_positions[i] = GST_AUDIO_CHANNEL_POSITION_INVALID; } } 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_query (GstBaseSink * base, GstQuery * query) { GstOsxAudioSink *sink = GST_OSX_AUDIO_SINK (base); gboolean ret = FALSE; switch (GST_QUERY_TYPE (query)) { case GST_QUERY_ACCEPT_CAPS: { GstCaps *caps = NULL; gst_query_parse_accept_caps (query, &caps); ret = gst_osx_audio_sink_acceptcaps (sink, caps); gst_query_set_accept_caps_result (query, ret); ret = TRUE; break; } default: ret = GST_BASE_SINK_CLASS (parent_class)->query (base, query); break; } return ret; } 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, GstCaps * filter) { 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); if (filter) return gst_caps_intersect_full (sink->cached_caps, filter, GST_CAPS_INTERSECT_FIRST); return gst_caps_ref (sink->cached_caps); } GST_DEBUG_OBJECT (sink, "using template caps"); return NULL; } static gboolean gst_osx_audio_sink_acceptcaps (GstOsxAudioSink * sink, GstCaps * caps) { GstCaps *pad_caps; GstStructure *st; gboolean ret = FALSE; GstAudioRingBufferSpec spec = { 0 }; gchar *caps_string = NULL; 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_query_caps (GST_BASE_SINK_PAD (sink), caps); 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_audio_ring_buffer_parse_caps (&spec, caps)) goto done; /* Make sure input is framed and can be payloaded */ switch (spec.type) { case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_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_AUDIO_RING_BUFFER_FORMAT_TYPE_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: return ret; } static GstBuffer * gst_osx_audio_sink_sink_payload (GstAudioBaseSink * 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; GstMapInfo inmap, outmap; gboolean res; if (framesize <= 0) return NULL; out = gst_buffer_new_and_alloc (framesize); gst_buffer_map (buf, &inmap, GST_MAP_READ); gst_buffer_map (out, &outmap, GST_MAP_WRITE); /* FIXME: the endianness needs to be queried and then set */ res = gst_audio_iec61937_payload (inmap.data, inmap.size, outmap.data, outmap.size, &sink->ringbuffer->spec, G_BIG_ENDIAN); gst_buffer_unmap (buf, &inmap); gst_buffer_unmap (out, &outmap); if (!res) { gst_buffer_unref (out); return NULL; } gst_buffer_copy_into (out, buf, GST_BUFFER_COPY_METADATA, 0, -1); return out; } else { return gst_buffer_ref (buf); } } static GstAudioRingBuffer * gst_osx_audio_sink_create_ringbuffer (GstAudioBaseSink * sink) { GstOsxAudioSink *osxsink; GstOsxAudioRingBuffer *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_AUDIO_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_AUDIO_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 (GstOsxAudioRingBuffer * 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_audio_ring_buffer_prepare_read (GST_AUDIO_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_AUDIO_RING_BUFFER (buf)->spec.segsize) { /* clear written samples */ gst_audio_ring_buffer_clear (GST_AUDIO_RING_BUFFER (buf), readseg); /* we wrote one segment */ gst_audio_ring_buffer_advance (GST_AUDIO_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) { GstOsxAudioRingBuffer *osxbuf; osxbuf = GST_OSX_AUDIO_RING_BUFFER (GST_AUDIO_BASE_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, channels; gboolean spdif_allowed; AudioChannelLayout *layout; GstElementClass *element_class; GstPadTemplate *pad_template; GstCaps *caps, *in_caps; guint64 channel_mask = 0; GstAudioChannelPosition *pos = osxsink->channel_positions; /* 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) { channels = MIN (layout->mNumberChannelDescriptions, GST_OSX_AUDIO_MAX_CHANNEL); } else { GST_WARNING_OBJECT (osxsink, "This driver does not support " "kAudioDevicePropertyPreferredChannelLayout."); channels = 2; } switch (channels) { case 0: pos[0] = GST_AUDIO_CHANNEL_POSITION_NONE; break; case 1: pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO; break; case 2: pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; channel_mask |= GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_LEFT); channel_mask |= GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_RIGHT); break; default: channels = MIN (layout->mNumberChannelDescriptions, GST_OSX_AUDIO_MAX_CHANNEL); for (i = 0; i < 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_LFE1; 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); channel_mask = 0; 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)); } } gst_audio_channel_positions_to_mask (pos, channels, false, &channel_mask); out_s = gst_structure_copy (in_s); gst_structure_remove_fields (out_s, "channels", "channel-mask", NULL); gst_structure_set (out_s, "channels", G_TYPE_INT, channels, "channel-mask", GST_TYPE_BITMASK, channel_mask, NULL); gst_caps_append_structure (caps, out_s); } if (osxsink->cached_caps) { gst_caps_unref (osxsink->cached_caps); } osxsink->cached_caps = caps; osxsink->channels = channels; 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; }