/* GStreamer * Copyright (C) 1999,2000 Erik Walthinsen * 2000 Wim Taymans * 2005 Wim Taymans * 2007 Andy Wingo * 2008 Sebastian Dröge * * deinterleave.c: deinterleave samples * * 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. */ /* TODO: * - handle changes in number of channels * - handle changes in channel positions * - better capsnego by using a buffer alloc function * and passing downstream caps changes upstream there */ /** * SECTION:element-deinterleave * @title: deinterleave * @see_also: interleave * * Splits one interleaved multichannel audio stream into many mono audio streams. * * This element handles all raw audio formats and supports changing the input caps as long as * all downstream elements can handle the new caps and the number of channels and the channel * positions stay the same. This restriction will be removed in later versions by adding or * removing some source pads as required. * * In most cases a queue and an audioconvert element should be added after each source pad * before further processing of the audio data. * * ## Example launch line * |[ * gst-launch-1.0 filesrc location=/path/to/file.mp3 ! decodebin ! audioconvert ! "audio/x-raw,channels=2 ! deinterleave name=d d.src_0 ! queue ! audioconvert ! vorbisenc ! oggmux ! filesink location=channel1.ogg d.src_1 ! queue ! audioconvert ! vorbisenc ! oggmux ! filesink location=channel2.ogg * ]| Decodes an MP3 file and encodes the left and right channel into separate * Ogg Vorbis files. * |[ * gst-launch-1.0 filesrc location=file.mp3 ! decodebin ! audioconvert ! "audio/x-raw,channels=2" ! deinterleave name=d interleave name=i ! audioconvert ! wavenc ! filesink location=test.wav d.src_0 ! queue ! audioconvert ! i.sink_1 d.src_1 ! queue ! audioconvert ! i.sink_0 * ]| Decodes and deinterleaves a Stereo MP3 file into separate channels and * then interleaves the channels again to a WAV file with the channel with the * channels exchanged. * */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include #include #include "gstinterleaveelements.h" #include "deinterleave.h" GST_DEBUG_CATEGORY_STATIC (gst_deinterleave_debug); #define GST_CAT_DEFAULT gst_deinterleave_debug static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src_%u", GST_PAD_SRC, GST_PAD_SOMETIMES, GST_STATIC_CAPS ("audio/x-raw, " "format = (string) " GST_AUDIO_FORMATS_ALL ", " "rate = (int) [ 1, MAX ], " "channels = (int) 1, layout = (string) {non-interleaved, interleaved}")); static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw, " "format = (string) " GST_AUDIO_FORMATS_ALL ", " "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ], layout = (string) interleaved")); #define MAKE_FUNC(type) \ static void deinterleave_##type (guint##type *out, guint##type *in, \ guint stride, guint nframes) \ { \ gint i; \ \ for (i = 0; i < nframes; i++) { \ out[i] = *in; \ in += stride; \ } \ } MAKE_FUNC (8); MAKE_FUNC (16); MAKE_FUNC (32); MAKE_FUNC (64); static void deinterleave_24 (guint8 * out, guint8 * in, guint stride, guint nframes) { gint i; for (i = 0; i < nframes; i++) { memcpy (out, in, 3); out += 3; in += stride * 3; } } #define gst_deinterleave_parent_class parent_class G_DEFINE_TYPE (GstDeinterleave, gst_deinterleave, GST_TYPE_ELEMENT); GST_ELEMENT_REGISTER_DEFINE (deinterleave, "deinterleave", GST_RANK_NONE, gst_deinterleave_get_type ()); enum { PROP_0, PROP_KEEP_POSITIONS }; static GstFlowReturn gst_deinterleave_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer); static gboolean gst_deinterleave_sink_setcaps (GstDeinterleave * self, GstCaps * caps); static GstStateChangeReturn gst_deinterleave_change_state (GstElement * element, GstStateChange transition); static gboolean gst_deinterleave_sink_event (GstPad * pad, GstObject * parent, GstEvent * event); static gboolean gst_deinterleave_sink_query (GstPad * pad, GstObject * parent, GstQuery * query); static gboolean gst_deinterleave_src_query (GstPad * pad, GstObject * parent, GstQuery * query); static void gst_deinterleave_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_deinterleave_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static void gst_deinterleave_finalize (GObject * obj) { GstDeinterleave *self = GST_DEINTERLEAVE (obj); if (self->pending_events) { g_list_foreach (self->pending_events, (GFunc) gst_mini_object_unref, NULL); g_list_free (self->pending_events); self->pending_events = NULL; } G_OBJECT_CLASS (parent_class)->finalize (obj); } static void gst_deinterleave_class_init (GstDeinterleaveClass * klass) { GObjectClass *gobject_class = G_OBJECT_CLASS (klass); GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass); GST_DEBUG_CATEGORY_INIT (gst_deinterleave_debug, "deinterleave", 0, "deinterleave element"); gst_element_class_set_static_metadata (gstelement_class, "Audio deinterleaver", "Filter/Converter/Audio", "Splits one interleaved multichannel audio stream into many mono audio streams", "Andy Wingo , " "Iain , " "Sebastian Dröge "); gst_element_class_add_static_pad_template (gstelement_class, &sink_template); gst_element_class_add_static_pad_template (gstelement_class, &src_template); gstelement_class->change_state = gst_deinterleave_change_state; gobject_class->finalize = gst_deinterleave_finalize; gobject_class->set_property = gst_deinterleave_set_property; gobject_class->get_property = gst_deinterleave_get_property; /** * GstDeinterleave:keep-positions * * Keep positions: When enable the caps on the output buffers will * contain the original channel positions. This can be used to correctly * interleave the output again later but can also lead to unwanted effects * if the output should be handled as Mono. * */ g_object_class_install_property (gobject_class, PROP_KEEP_POSITIONS, g_param_spec_boolean ("keep-positions", "Keep positions", "Keep the original channel positions on the output buffers", FALSE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); } static void gst_deinterleave_init (GstDeinterleave * self) { self->keep_positions = FALSE; self->func = NULL; gst_audio_info_init (&self->audio_info); /* Add sink pad */ self->sink = gst_pad_new_from_static_template (&sink_template, "sink"); gst_pad_set_chain_function (self->sink, GST_DEBUG_FUNCPTR (gst_deinterleave_chain)); gst_pad_set_event_function (self->sink, GST_DEBUG_FUNCPTR (gst_deinterleave_sink_event)); gst_pad_set_query_function (self->sink, GST_DEBUG_FUNCPTR (gst_deinterleave_sink_query)); gst_element_add_pad (GST_ELEMENT (self), self->sink); } typedef struct { GstCaps *caps; GstPad *pad; } CopyStickyEventsData; static gboolean copy_sticky_events (GstPad * pad, GstEvent ** event, gpointer user_data) { CopyStickyEventsData *data = user_data; if (GST_EVENT_TYPE (*event) >= GST_EVENT_CAPS && data->caps) { gst_pad_set_caps (data->pad, data->caps); data->caps = NULL; } if (GST_EVENT_TYPE (*event) != GST_EVENT_CAPS) gst_pad_push_event (data->pad, gst_event_ref (*event)); return TRUE; } static void gst_deinterleave_add_new_pads (GstDeinterleave * self, GstCaps * caps) { GstPad *pad; guint i; for (i = 0; i < GST_AUDIO_INFO_CHANNELS (&self->audio_info); i++) { gchar *name = g_strdup_printf ("src_%u", i); GstCaps *srccaps; GstAudioInfo info; GstAudioFormat format = GST_AUDIO_INFO_FORMAT (&self->audio_info); gint rate = GST_AUDIO_INFO_RATE (&self->audio_info); GstAudioChannelPosition position = GST_AUDIO_CHANNEL_POSITION_MONO; CopyStickyEventsData data; /* Set channel position if we know it */ if (self->keep_positions) position = GST_AUDIO_INFO_POSITION (&self->audio_info, i); gst_audio_info_init (&info); gst_audio_info_set_format (&info, format, rate, 1, &position); srccaps = gst_audio_info_to_caps (&info); pad = gst_pad_new_from_static_template (&src_template, name); g_free (name); gst_pad_use_fixed_caps (pad); gst_pad_set_query_function (pad, GST_DEBUG_FUNCPTR (gst_deinterleave_src_query)); gst_pad_set_active (pad, TRUE); data.pad = pad; data.caps = srccaps; gst_pad_sticky_events_foreach (self->sink, copy_sticky_events, &data); if (data.caps) gst_pad_set_caps (pad, data.caps); gst_element_add_pad (GST_ELEMENT (self), pad); self->srcpads = g_list_prepend (self->srcpads, gst_object_ref (pad)); gst_caps_unref (srccaps); } gst_element_no_more_pads (GST_ELEMENT (self)); self->srcpads = g_list_reverse (self->srcpads); } static gboolean gst_deinterleave_set_pads_caps (GstDeinterleave * self, GstCaps * caps) { GList *l; gint i; gboolean ret = TRUE; for (l = self->srcpads, i = 0; l; l = l->next, i++) { GstPad *pad = GST_PAD (l->data); GstCaps *srccaps; GstAudioInfo info; if (!gst_audio_info_from_caps (&info, caps)) { ret = FALSE; continue; } if (self->keep_positions) GST_AUDIO_INFO_POSITION (&info, 0) = GST_AUDIO_INFO_POSITION (&self->audio_info, i); srccaps = gst_audio_info_to_caps (&info); gst_pad_set_caps (pad, srccaps); gst_caps_unref (srccaps); } return ret; } static void gst_deinterleave_remove_pads (GstDeinterleave * self) { GList *l; GST_INFO_OBJECT (self, "removing pads"); for (l = self->srcpads; l; l = l->next) { GstPad *pad = GST_PAD (l->data); gst_element_remove_pad (GST_ELEMENT_CAST (self), pad); gst_object_unref (pad); } g_list_free (self->srcpads); self->srcpads = NULL; gst_caps_replace (&self->sinkcaps, NULL); } static gboolean gst_deinterleave_set_process_function (GstDeinterleave * self) { switch (GST_AUDIO_INFO_WIDTH (&self->audio_info)) { case 8: self->func = (GstDeinterleaveFunc) deinterleave_8; break; case 16: self->func = (GstDeinterleaveFunc) deinterleave_16; break; case 24: self->func = (GstDeinterleaveFunc) deinterleave_24; break; case 32: self->func = (GstDeinterleaveFunc) deinterleave_32; break; case 64: self->func = (GstDeinterleaveFunc) deinterleave_64; break; default: return FALSE; } return TRUE; } static gboolean gst_deinterleave_check_caps_change (GstDeinterleave * self, GstAudioInfo * old_info, GstAudioInfo * new_info) { gint i; gboolean same_layout = TRUE; gboolean was_unpositioned; gboolean is_unpositioned; gint new_channels; gint old_channels; new_channels = GST_AUDIO_INFO_CHANNELS (new_info); old_channels = GST_AUDIO_INFO_CHANNELS (old_info); if (GST_AUDIO_INFO_IS_UNPOSITIONED (new_info) || new_channels == 1) is_unpositioned = TRUE; else is_unpositioned = FALSE; if (GST_AUDIO_INFO_IS_UNPOSITIONED (old_info) || old_channels == 1) was_unpositioned = TRUE; else was_unpositioned = FALSE; /* We allow caps changes as long as the number of channels doesn't change * and the channel positions stay the same. _getcaps() should've cared * for this already but better be safe. */ if (new_channels != old_channels) goto cannot_change_caps; /* Now check the channel positions. If we had no channel positions * and get them or the other way around things have changed. * If we had channel positions and get different ones things have * changed too of course */ if ((!was_unpositioned && is_unpositioned) || (was_unpositioned && !is_unpositioned)) goto cannot_change_caps; if (!is_unpositioned) { if (GST_AUDIO_INFO_CHANNELS (old_info) != GST_AUDIO_INFO_CHANNELS (new_info)) goto cannot_change_caps; for (i = 0; i < GST_AUDIO_INFO_CHANNELS (old_info); i++) { if (new_info->position[i] != old_info->position[i]) { same_layout = FALSE; break; } } if (!same_layout) goto cannot_change_caps; } return TRUE; cannot_change_caps: return FALSE; } static gboolean gst_deinterleave_sink_setcaps (GstDeinterleave * self, GstCaps * caps) { GstCaps *srccaps; GstStructure *s; GST_DEBUG_OBJECT (self, "got caps: %" GST_PTR_FORMAT, caps); if (!gst_audio_info_from_caps (&self->audio_info, caps)) goto invalid_caps; if (!gst_deinterleave_set_process_function (self)) goto unsupported_caps; if (self->sinkcaps && !gst_caps_is_equal (caps, self->sinkcaps)) { GstAudioInfo old_info; gst_audio_info_init (&old_info); if (!gst_audio_info_from_caps (&old_info, self->sinkcaps)) goto info_from_caps_failed; if (gst_deinterleave_check_caps_change (self, &old_info, &self->audio_info)) { if (!gst_deinterleave_set_process_function (self)) goto cannot_change_caps; } else goto cannot_change_caps; } gst_caps_replace (&self->sinkcaps, caps); /* Get srcpad caps */ srccaps = gst_caps_copy (caps); s = gst_caps_get_structure (srccaps, 0); gst_structure_set (s, "channels", G_TYPE_INT, 1, NULL); gst_structure_remove_field (s, "channel-mask"); /* If we already have pads, update the caps otherwise * add new pads */ if (self->srcpads) { if (!gst_deinterleave_set_pads_caps (self, srccaps)) goto set_caps_failed; } else { gst_deinterleave_add_new_pads (self, srccaps); } gst_caps_unref (srccaps); return TRUE; cannot_change_caps: { GST_WARNING_OBJECT (self, "caps change from %" GST_PTR_FORMAT " to %" GST_PTR_FORMAT " not supported: channel number or channel " "positions change", self->sinkcaps, caps); return FALSE; } unsupported_caps: { GST_ERROR_OBJECT (self, "caps not supported: %" GST_PTR_FORMAT, caps); return FALSE; } invalid_caps: { GST_ERROR_OBJECT (self, "invalid caps"); return FALSE; } set_caps_failed: { GST_ERROR_OBJECT (self, "set_caps failed"); gst_caps_unref (srccaps); return FALSE; } info_from_caps_failed: { GST_ERROR_OBJECT (self, "could not get info from caps"); return FALSE; } } static void __remove_channels (GstCaps * caps) { GstStructure *s; gint i, size; size = gst_caps_get_size (caps); for (i = 0; i < size; i++) { s = gst_caps_get_structure (caps, i); gst_structure_remove_field (s, "channel-mask"); gst_structure_remove_field (s, "channels"); } } static void __set_channels (GstCaps * caps, gint channels) { GstStructure *s; gint i, size; size = gst_caps_get_size (caps); for (i = 0; i < size; i++) { s = gst_caps_get_structure (caps, i); if (channels > 0) gst_structure_set (s, "channels", G_TYPE_INT, channels, NULL); else gst_structure_set (s, "channels", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL); } } static gboolean gst_deinterleave_sink_acceptcaps (GstPad * pad, GstObject * parent, GstCaps * caps) { GstDeinterleave *self = GST_DEINTERLEAVE (parent); GstCaps *templ_caps = gst_pad_get_pad_template_caps (pad); gboolean ret; ret = gst_caps_can_intersect (templ_caps, caps); gst_caps_unref (templ_caps); if (ret && self->sinkcaps) { GstAudioInfo new_info; gst_audio_info_init (&new_info); if (!gst_audio_info_from_caps (&new_info, caps)) goto info_from_caps_failed; ret = gst_deinterleave_check_caps_change (self, &self->audio_info, &new_info); } return ret; info_from_caps_failed: { GST_ERROR_OBJECT (self, "could not get info from caps"); return FALSE; } } static GstCaps * gst_deinterleave_getcaps (GstPad * pad, GstObject * parent, GstCaps * filter) { GstDeinterleave *self = GST_DEINTERLEAVE (parent); GstCaps *ret; GstIterator *it; GstIteratorResult res; GValue v = G_VALUE_INIT; if (pad != self->sink) { ret = gst_pad_get_current_caps (pad); if (ret) { if (filter) { GstCaps *tmp = gst_caps_intersect_full (filter, ret, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (ret); ret = tmp; } return ret; } } /* Intersect all of our pad template caps with the peer caps of the pad * to get all formats that are possible up- and downstream. * * For the pad for which the caps are requested we don't remove the channel * information as they must be in the returned caps and incompatibilities * will be detected here already */ ret = gst_caps_new_any (); it = gst_element_iterate_pads (GST_ELEMENT_CAST (self)); do { res = gst_iterator_next (it, &v); switch (res) { case GST_ITERATOR_OK:{ GstPad *ourpad = GST_PAD (g_value_get_object (&v)); GstCaps *peercaps = NULL, *ourcaps; GstCaps *templ_caps = gst_pad_get_pad_template_caps (ourpad); ourcaps = gst_caps_copy (templ_caps); gst_caps_unref (templ_caps); if (pad == ourpad) { if (GST_PAD_DIRECTION (pad) == GST_PAD_SINK) __set_channels (ourcaps, GST_AUDIO_INFO_CHANNELS (&self->audio_info)); else __set_channels (ourcaps, 1); } else { __remove_channels (ourcaps); /* Only ask for peer caps for other pads than pad * as otherwise gst_pad_peer_get_caps() might call * back into this function and deadlock */ peercaps = gst_pad_peer_query_caps (ourpad, NULL); peercaps = gst_caps_make_writable (peercaps); } /* If the peer exists and has caps add them to the intersection, * otherwise assume that the peer accepts everything */ if (peercaps) { GstCaps *intersection; GstCaps *oldret = ret; __remove_channels (peercaps); intersection = gst_caps_intersect (peercaps, ourcaps); ret = gst_caps_intersect (ret, intersection); gst_caps_unref (intersection); gst_caps_unref (peercaps); gst_caps_unref (oldret); } else { GstCaps *oldret = ret; ret = gst_caps_intersect (ret, ourcaps); gst_caps_unref (oldret); } gst_caps_unref (ourcaps); g_value_reset (&v); break; } case GST_ITERATOR_DONE: break; case GST_ITERATOR_ERROR: gst_caps_unref (ret); ret = gst_caps_new_empty (); break; case GST_ITERATOR_RESYNC: gst_caps_unref (ret); ret = gst_caps_new_any (); gst_iterator_resync (it); break; } } while (res != GST_ITERATOR_DONE && res != GST_ITERATOR_ERROR); g_value_unset (&v); gst_iterator_free (it); if (filter) { GstCaps *aux; aux = gst_caps_intersect_full (filter, ret, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (ret); ret = aux; } GST_DEBUG_OBJECT (pad, "Intersected caps to %" GST_PTR_FORMAT, ret); return ret; } static gboolean gst_deinterleave_sink_event (GstPad * pad, GstObject * parent, GstEvent * event) { GstDeinterleave *self = GST_DEINTERLEAVE (parent); gboolean ret; GST_DEBUG ("Got %s event on pad %s:%s", GST_EVENT_TYPE_NAME (event), GST_DEBUG_PAD_NAME (pad)); /* Send FLUSH_STOP, FLUSH_START and EOS immediately, no matter if * we have src pads already or not. Queue all other events and * push them after we have src pads */ switch (GST_EVENT_TYPE (event)) { case GST_EVENT_FLUSH_STOP: case GST_EVENT_FLUSH_START: case GST_EVENT_EOS: ret = gst_pad_event_default (pad, parent, event); break; case GST_EVENT_CAPS: { GstCaps *caps; gst_event_parse_caps (event, &caps); ret = gst_deinterleave_sink_setcaps (self, caps); gst_event_unref (event); break; } default: if (!self->srcpads && !GST_EVENT_IS_STICKY (event)) { /* Sticky events are copied when creating a new pad */ GST_OBJECT_LOCK (self); self->pending_events = g_list_append (self->pending_events, event); GST_OBJECT_UNLOCK (self); ret = TRUE; } else { ret = gst_pad_event_default (pad, parent, event); } break; } return ret; } static gboolean gst_deinterleave_sink_query (GstPad * pad, GstObject * parent, GstQuery * query) { gboolean res; switch (GST_QUERY_TYPE (query)) { case GST_QUERY_CAPS:{ GstCaps *filter; GstCaps *caps; gst_query_parse_caps (query, &filter); caps = gst_deinterleave_getcaps (pad, parent, filter); gst_query_set_caps_result (query, caps); gst_caps_unref (caps); res = TRUE; break; } case GST_QUERY_ACCEPT_CAPS:{ GstCaps *caps; gboolean ret; gst_query_parse_accept_caps (query, &caps); ret = gst_deinterleave_sink_acceptcaps (pad, parent, caps); gst_query_set_accept_caps_result (query, ret); res = TRUE; break; } default: res = gst_pad_query_default (pad, parent, query); break; } return res; } static gboolean gst_deinterleave_src_query (GstPad * pad, GstObject * parent, GstQuery * query) { GstDeinterleave *self = GST_DEINTERLEAVE (parent); gboolean res; res = gst_pad_query_default (pad, parent, query); if (res && GST_QUERY_TYPE (query) == GST_QUERY_DURATION) { GstFormat format; gint64 dur; gst_query_parse_duration (query, &format, &dur); /* Need to divide by the number of channels in byte format * to get the correct value. All other formats should be fine */ if (format == GST_FORMAT_BYTES && dur != -1) gst_query_set_duration (query, format, dur / GST_AUDIO_INFO_CHANNELS (&self->audio_info)); } else if (res && GST_QUERY_TYPE (query) == GST_QUERY_POSITION) { GstFormat format; gint64 pos; gst_query_parse_position (query, &format, &pos); /* Need to divide by the number of channels in byte format * to get the correct value. All other formats should be fine */ if (format == GST_FORMAT_BYTES && pos != -1) gst_query_set_position (query, format, pos / GST_AUDIO_INFO_CHANNELS (&self->audio_info)); } else if (res && GST_QUERY_TYPE (query) == GST_QUERY_CAPS) { GstCaps *filter, *caps; gst_query_parse_caps (query, &filter); caps = gst_deinterleave_getcaps (pad, parent, filter); gst_query_set_caps_result (query, caps); gst_caps_unref (caps); } return res; } static void gst_deinterleave_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstDeinterleave *self = GST_DEINTERLEAVE (object); switch (prop_id) { case PROP_KEEP_POSITIONS: self->keep_positions = g_value_get_boolean (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_deinterleave_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstDeinterleave *self = GST_DEINTERLEAVE (object); switch (prop_id) { case PROP_KEEP_POSITIONS: g_value_set_boolean (value, self->keep_positions); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static GstFlowReturn gst_deinterleave_process (GstDeinterleave * self, GstBuffer * buf) { GstFlowReturn ret = GST_FLOW_OK; guint channels = GST_AUDIO_INFO_CHANNELS (&self->audio_info); guint pads_pushed = 0, buffers_allocated = 0; guint nframes = gst_buffer_get_size (buf) / channels / (GST_AUDIO_INFO_WIDTH (&self->audio_info) / 8); guint bufsize = nframes * (GST_AUDIO_INFO_WIDTH (&self->audio_info) / 8); guint i; GList *srcs; GstBuffer **buffers_out = g_new0 (GstBuffer *, channels); guint8 *in, *out; GstMapInfo read_info; GList *pending_events, *l; /* Send any pending events to all src pads */ GST_OBJECT_LOCK (self); pending_events = self->pending_events; self->pending_events = NULL; GST_OBJECT_UNLOCK (self); if (pending_events) { GstEvent *event; GST_DEBUG_OBJECT (self, "Sending pending events to all src pads"); for (l = pending_events; l; l = l->next) { event = l->data; for (srcs = self->srcpads; srcs != NULL; srcs = srcs->next) gst_pad_push_event (GST_PAD (srcs->data), gst_event_ref (event)); gst_event_unref (event); } g_list_free (pending_events); } gst_buffer_map (buf, &read_info, GST_MAP_READ); /* Allocate buffers */ for (srcs = self->srcpads, i = 0; srcs; srcs = srcs->next, i++) { buffers_out[i] = gst_buffer_new_allocate (NULL, bufsize, NULL); /* Make sure we got a correct buffer. The only other case we allow * here is an unliked pad */ if (!buffers_out[i]) goto alloc_buffer_failed; else if (buffers_out[i] && gst_buffer_get_size (buffers_out[i]) != bufsize) goto alloc_buffer_bad_size; if (buffers_out[i]) { gst_buffer_copy_into (buffers_out[i], buf, GST_BUFFER_COPY_METADATA, 0, -1); buffers_allocated++; } } /* Return NOT_LINKED if no pad was linked */ if (!buffers_allocated) { GST_WARNING_OBJECT (self, "Couldn't allocate any buffers because no pad was linked"); ret = GST_FLOW_NOT_LINKED; goto done; } /* deinterleave */ for (srcs = self->srcpads, i = 0; srcs; srcs = srcs->next, i++) { GstPad *pad = (GstPad *) srcs->data; GstMapInfo write_info; in = (guint8 *) read_info.data; in += i * (GST_AUDIO_INFO_WIDTH (&self->audio_info) / 8); if (buffers_out[i]) { gst_buffer_map (buffers_out[i], &write_info, GST_MAP_WRITE); out = (guint8 *) write_info.data; self->func (out, in, channels, nframes); gst_buffer_unmap (buffers_out[i], &write_info); ret = gst_pad_push (pad, buffers_out[i]); buffers_out[i] = NULL; if (ret == GST_FLOW_OK) pads_pushed++; else if (ret == GST_FLOW_NOT_LINKED) ret = GST_FLOW_OK; else goto push_failed; } } /* Return NOT_LINKED if no pad was linked */ if (!pads_pushed) ret = GST_FLOW_NOT_LINKED; GST_DEBUG_OBJECT (self, "Pushed on %d pads", pads_pushed); done: gst_buffer_unmap (buf, &read_info); gst_buffer_unref (buf); g_free (buffers_out); return ret; alloc_buffer_failed: { GST_WARNING ("gst_pad_alloc_buffer() returned %s", gst_flow_get_name (ret)); goto clean_buffers; } alloc_buffer_bad_size: { GST_WARNING ("called alloc_buffer(), but didn't get requested bytes"); ret = GST_FLOW_NOT_NEGOTIATED; goto clean_buffers; } push_failed: { GST_DEBUG ("push() failed, flow = %s", gst_flow_get_name (ret)); goto clean_buffers; } clean_buffers: { gst_buffer_unmap (buf, &read_info); for (i = 0; i < channels; i++) { if (buffers_out[i]) gst_buffer_unref (buffers_out[i]); } gst_buffer_unref (buf); g_free (buffers_out); return ret; } } static GstFlowReturn gst_deinterleave_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer) { GstDeinterleave *self = GST_DEINTERLEAVE (parent); GstFlowReturn ret; g_return_val_if_fail (self->func != NULL, GST_FLOW_NOT_NEGOTIATED); g_return_val_if_fail (GST_AUDIO_INFO_WIDTH (&self->audio_info) > 0, GST_FLOW_NOT_NEGOTIATED); g_return_val_if_fail (GST_AUDIO_INFO_CHANNELS (&self->audio_info) > 0, GST_FLOW_NOT_NEGOTIATED); ret = gst_deinterleave_process (self, buffer); if (ret != GST_FLOW_OK) GST_DEBUG_OBJECT (self, "flow return: %s", gst_flow_get_name (ret)); return ret; } static GstStateChangeReturn gst_deinterleave_change_state (GstElement * element, GstStateChange transition) { GstStateChangeReturn ret; GstDeinterleave *self = GST_DEINTERLEAVE (element); switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: break; case GST_STATE_CHANGE_READY_TO_PAUSED: gst_deinterleave_remove_pads (self); self->func = NULL; if (self->pending_events) { g_list_foreach (self->pending_events, (GFunc) gst_mini_object_unref, NULL); g_list_free (self->pending_events); self->pending_events = NULL; } break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: break; default: break; } ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); switch (transition) { case GST_STATE_CHANGE_PLAYING_TO_PAUSED: break; case GST_STATE_CHANGE_PAUSED_TO_READY: gst_deinterleave_remove_pads (self); self->func = NULL; if (self->pending_events) { g_list_foreach (self->pending_events, (GFunc) gst_mini_object_unref, NULL); g_list_free (self->pending_events); self->pending_events = NULL; } break; case GST_STATE_CHANGE_READY_TO_NULL: break; default: break; } return ret; }