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1029 lines
29 KiB
C
1029 lines
29 KiB
C
/* GStreamer
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* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
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* 2000 Wim Taymans <wtay@chello.be>
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* 2005 Wim Taymans <wim@fluendo.com>
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* 2007 Andy Wingo <wingo at pobox.com>
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* 2008 Sebastian Dröge <slomo@circular-chaos.org>
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*
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* deinterleave.c: deinterleave samples
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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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 GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/* TODO:
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* - handle changes in number of channels
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* - handle changes in channel positions
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* - better capsnego by using a buffer alloc function
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* and passing downstream caps changes upstream there
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*/
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/**
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* SECTION:element-deinterleave
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* @title: deinterleave
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* @see_also: interleave
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*
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* Splits one interleaved multichannel audio stream into many mono audio streams.
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*
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* This element handles all raw audio formats and supports changing the input caps as long as
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* all downstream elements can handle the new caps and the number of channels and the channel
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* positions stay the same. This restriction will be removed in later versions by adding or
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* removing some source pads as required.
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*
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* In most cases a queue and an audioconvert element should be added after each source pad
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* before further processing of the audio data.
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*
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* ## Example launch line
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* |[
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* 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
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* ]| Decodes an MP3 file and encodes the left and right channel into separate
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* Ogg Vorbis files.
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* |[
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* 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
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* ]| Decodes and deinterleaves a Stereo MP3 file into separate channels and
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* then interleaves the channels again to a WAV file with the channel with the
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* channels exchanged.
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*
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <gst/gst.h>
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#include <string.h>
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#include "deinterleave.h"
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GST_DEBUG_CATEGORY_STATIC (gst_deinterleave_debug);
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#define GST_CAT_DEFAULT gst_deinterleave_debug
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static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src_%u",
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GST_PAD_SRC,
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GST_PAD_SOMETIMES,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) " GST_AUDIO_FORMATS_ALL ", "
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"rate = (int) [ 1, MAX ], "
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"channels = (int) 1, layout = (string) {non-interleaved, interleaved}"));
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static GstStaticPadTemplate sink_template = 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 ("audio/x-raw, "
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"format = (string) " GST_AUDIO_FORMATS_ALL ", "
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"rate = (int) [ 1, MAX ], "
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"channels = (int) [ 1, MAX ], layout = (string) interleaved"));
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#define MAKE_FUNC(type) \
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static void deinterleave_##type (guint##type *out, guint##type *in, \
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guint stride, guint nframes) \
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{ \
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gint i; \
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\
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for (i = 0; i < nframes; i++) { \
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out[i] = *in; \
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in += stride; \
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} \
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}
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MAKE_FUNC (8);
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MAKE_FUNC (16);
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MAKE_FUNC (32);
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MAKE_FUNC (64);
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static void
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deinterleave_24 (guint8 * out, guint8 * in, guint stride, guint nframes)
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{
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gint i;
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for (i = 0; i < nframes; i++) {
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memcpy (out, in, 3);
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out += 3;
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in += stride * 3;
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}
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}
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#define gst_deinterleave_parent_class parent_class
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G_DEFINE_TYPE (GstDeinterleave, gst_deinterleave, GST_TYPE_ELEMENT);
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enum
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{
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PROP_0,
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PROP_KEEP_POSITIONS
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};
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static GstFlowReturn gst_deinterleave_chain (GstPad * pad, GstObject * parent,
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GstBuffer * buffer);
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static gboolean gst_deinterleave_sink_setcaps (GstDeinterleave * self,
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GstCaps * caps);
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static GstStateChangeReturn
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gst_deinterleave_change_state (GstElement * element, GstStateChange transition);
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static gboolean gst_deinterleave_sink_event (GstPad * pad, GstObject * parent,
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GstEvent * event);
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static gboolean gst_deinterleave_sink_query (GstPad * pad, GstObject * parent,
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GstQuery * query);
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static gboolean gst_deinterleave_src_query (GstPad * pad, GstObject * parent,
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GstQuery * query);
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static void gst_deinterleave_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_deinterleave_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static void
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gst_deinterleave_finalize (GObject * obj)
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{
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GstDeinterleave *self = GST_DEINTERLEAVE (obj);
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if (self->pending_events) {
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g_list_foreach (self->pending_events, (GFunc) gst_mini_object_unref, NULL);
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g_list_free (self->pending_events);
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self->pending_events = NULL;
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}
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G_OBJECT_CLASS (parent_class)->finalize (obj);
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}
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static void
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gst_deinterleave_class_init (GstDeinterleaveClass * klass)
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{
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
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GST_DEBUG_CATEGORY_INIT (gst_deinterleave_debug, "deinterleave", 0,
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"deinterleave element");
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gst_element_class_set_static_metadata (gstelement_class,
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"Audio deinterleaver", "Filter/Converter/Audio",
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"Splits one interleaved multichannel audio stream into many mono audio streams",
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"Andy Wingo <wingo at pobox.com>, " "Iain <iain@prettypeople.org>, "
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"Sebastian Dröge <slomo@circular-chaos.org>");
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gst_element_class_add_static_pad_template (gstelement_class, &sink_template);
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gst_element_class_add_static_pad_template (gstelement_class, &src_template);
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gstelement_class->change_state = gst_deinterleave_change_state;
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gobject_class->finalize = gst_deinterleave_finalize;
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gobject_class->set_property = gst_deinterleave_set_property;
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gobject_class->get_property = gst_deinterleave_get_property;
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/**
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* GstDeinterleave:keep-positions
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*
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* Keep positions: When enable the caps on the output buffers will
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* contain the original channel positions. This can be used to correctly
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* interleave the output again later but can also lead to unwanted effects
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* if the output should be handled as Mono.
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*
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*/
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g_object_class_install_property (gobject_class, PROP_KEEP_POSITIONS,
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g_param_spec_boolean ("keep-positions", "Keep positions",
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"Keep the original channel positions on the output buffers",
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FALSE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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}
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static void
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gst_deinterleave_init (GstDeinterleave * self)
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{
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self->keep_positions = FALSE;
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self->func = NULL;
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gst_audio_info_init (&self->audio_info);
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/* Add sink pad */
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self->sink = gst_pad_new_from_static_template (&sink_template, "sink");
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gst_pad_set_chain_function (self->sink,
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GST_DEBUG_FUNCPTR (gst_deinterleave_chain));
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gst_pad_set_event_function (self->sink,
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GST_DEBUG_FUNCPTR (gst_deinterleave_sink_event));
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gst_pad_set_query_function (self->sink,
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GST_DEBUG_FUNCPTR (gst_deinterleave_sink_query));
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gst_element_add_pad (GST_ELEMENT (self), self->sink);
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}
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typedef struct
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{
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GstCaps *caps;
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GstPad *pad;
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} CopyStickyEventsData;
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static gboolean
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copy_sticky_events (GstPad * pad, GstEvent ** event, gpointer user_data)
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{
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CopyStickyEventsData *data = user_data;
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if (GST_EVENT_TYPE (*event) >= GST_EVENT_CAPS && data->caps) {
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gst_pad_set_caps (data->pad, data->caps);
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data->caps = NULL;
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}
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if (GST_EVENT_TYPE (*event) != GST_EVENT_CAPS)
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gst_pad_push_event (data->pad, gst_event_ref (*event));
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return TRUE;
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}
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static void
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gst_deinterleave_add_new_pads (GstDeinterleave * self, GstCaps * caps)
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{
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GstPad *pad;
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guint i;
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for (i = 0; i < GST_AUDIO_INFO_CHANNELS (&self->audio_info); i++) {
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gchar *name = g_strdup_printf ("src_%u", i);
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GstCaps *srccaps;
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GstAudioInfo info;
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GstAudioFormat format = GST_AUDIO_INFO_FORMAT (&self->audio_info);
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gint rate = GST_AUDIO_INFO_RATE (&self->audio_info);
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GstAudioChannelPosition position = GST_AUDIO_CHANNEL_POSITION_MONO;
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CopyStickyEventsData data;
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/* Set channel position if we know it */
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if (self->keep_positions)
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position = GST_AUDIO_INFO_POSITION (&self->audio_info, i);
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gst_audio_info_init (&info);
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gst_audio_info_set_format (&info, format, rate, 1, &position);
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srccaps = gst_audio_info_to_caps (&info);
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pad = gst_pad_new_from_static_template (&src_template, name);
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g_free (name);
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gst_pad_use_fixed_caps (pad);
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gst_pad_set_query_function (pad,
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GST_DEBUG_FUNCPTR (gst_deinterleave_src_query));
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gst_pad_set_active (pad, TRUE);
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data.pad = pad;
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data.caps = srccaps;
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gst_pad_sticky_events_foreach (self->sink, copy_sticky_events, &data);
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if (data.caps)
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gst_pad_set_caps (pad, data.caps);
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gst_element_add_pad (GST_ELEMENT (self), pad);
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self->srcpads = g_list_prepend (self->srcpads, gst_object_ref (pad));
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gst_caps_unref (srccaps);
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}
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gst_element_no_more_pads (GST_ELEMENT (self));
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self->srcpads = g_list_reverse (self->srcpads);
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}
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static gboolean
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gst_deinterleave_set_pads_caps (GstDeinterleave * self, GstCaps * caps)
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{
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GList *l;
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gint i;
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gboolean ret = TRUE;
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for (l = self->srcpads, i = 0; l; l = l->next, i++) {
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GstPad *pad = GST_PAD (l->data);
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GstCaps *srccaps;
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GstAudioInfo info;
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if (!gst_audio_info_from_caps (&info, caps)) {
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ret = FALSE;
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continue;
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}
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if (self->keep_positions)
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GST_AUDIO_INFO_POSITION (&info, 0) =
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GST_AUDIO_INFO_POSITION (&self->audio_info, i);
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srccaps = gst_audio_info_to_caps (&info);
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gst_pad_set_caps (pad, srccaps);
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gst_caps_unref (srccaps);
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}
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return ret;
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}
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static void
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gst_deinterleave_remove_pads (GstDeinterleave * self)
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{
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GList *l;
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GST_INFO_OBJECT (self, "removing pads");
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for (l = self->srcpads; l; l = l->next) {
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GstPad *pad = GST_PAD (l->data);
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gst_element_remove_pad (GST_ELEMENT_CAST (self), pad);
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gst_object_unref (pad);
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}
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g_list_free (self->srcpads);
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self->srcpads = NULL;
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gst_caps_replace (&self->sinkcaps, NULL);
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}
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static gboolean
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gst_deinterleave_set_process_function (GstDeinterleave * self)
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{
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switch (GST_AUDIO_INFO_WIDTH (&self->audio_info)) {
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case 8:
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self->func = (GstDeinterleaveFunc) deinterleave_8;
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break;
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case 16:
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self->func = (GstDeinterleaveFunc) deinterleave_16;
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break;
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case 24:
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self->func = (GstDeinterleaveFunc) deinterleave_24;
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break;
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case 32:
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self->func = (GstDeinterleaveFunc) deinterleave_32;
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break;
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case 64:
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self->func = (GstDeinterleaveFunc) deinterleave_64;
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break;
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default:
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return FALSE;
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}
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return TRUE;
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}
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static gboolean
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gst_deinterleave_check_caps_change (GstDeinterleave * self,
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GstAudioInfo * old_info, GstAudioInfo * new_info)
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{
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gint i;
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gboolean same_layout = TRUE;
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gboolean was_unpositioned;
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gboolean is_unpositioned;
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gint new_channels;
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gint old_channels;
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new_channels = GST_AUDIO_INFO_CHANNELS (new_info);
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old_channels = GST_AUDIO_INFO_CHANNELS (old_info);
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if (GST_AUDIO_INFO_IS_UNPOSITIONED (new_info) || new_channels == 1)
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is_unpositioned = TRUE;
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else
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is_unpositioned = FALSE;
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if (GST_AUDIO_INFO_IS_UNPOSITIONED (old_info) || old_channels == 1)
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was_unpositioned = TRUE;
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else
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was_unpositioned = FALSE;
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|
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/* We allow caps changes as long as the number of channels doesn't change
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* and the channel positions stay the same. _getcaps() should've cared
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* for this already but better be safe.
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*/
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if (new_channels != old_channels)
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goto cannot_change_caps;
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|
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/* Now check the channel positions. If we had no channel positions
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* and get them or the other way around things have changed.
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* If we had channel positions and get different ones things have
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* changed too of course
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*/
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if ((!was_unpositioned && is_unpositioned) || (was_unpositioned
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&& !is_unpositioned))
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goto cannot_change_caps;
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if (!is_unpositioned) {
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if (GST_AUDIO_INFO_CHANNELS (old_info) !=
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GST_AUDIO_INFO_CHANNELS (new_info))
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goto cannot_change_caps;
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for (i = 0; i < GST_AUDIO_INFO_CHANNELS (old_info); i++) {
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if (new_info->position[i] != old_info->position[i]) {
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same_layout = FALSE;
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break;
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}
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}
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if (!same_layout)
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goto cannot_change_caps;
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}
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|
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return TRUE;
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|
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cannot_change_caps:
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return FALSE;
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}
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|
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static gboolean
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gst_deinterleave_sink_setcaps (GstDeinterleave * self, GstCaps * caps)
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{
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GstCaps *srccaps;
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GstStructure *s;
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GST_DEBUG_OBJECT (self, "got caps: %" GST_PTR_FORMAT, caps);
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if (!gst_audio_info_from_caps (&self->audio_info, caps))
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goto invalid_caps;
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if (!gst_deinterleave_set_process_function (self))
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goto unsupported_caps;
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|
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if (self->sinkcaps && !gst_caps_is_equal (caps, self->sinkcaps)) {
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GstAudioInfo old_info;
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gst_audio_info_init (&old_info);
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if (!gst_audio_info_from_caps (&old_info, self->sinkcaps))
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goto info_from_caps_failed;
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if (gst_deinterleave_check_caps_change (self, &old_info, &self->audio_info)) {
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if (!gst_deinterleave_set_process_function (self))
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goto cannot_change_caps;
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} else
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goto cannot_change_caps;
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}
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gst_caps_replace (&self->sinkcaps, caps);
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|
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/* Get srcpad caps */
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srccaps = gst_caps_copy (caps);
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s = gst_caps_get_structure (srccaps, 0);
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gst_structure_set (s, "channels", G_TYPE_INT, 1, NULL);
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gst_structure_remove_field (s, "channel-mask");
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|
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/* If we already have pads, update the caps otherwise
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* add new pads */
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if (self->srcpads) {
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if (!gst_deinterleave_set_pads_caps (self, srccaps))
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goto set_caps_failed;
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} else {
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gst_deinterleave_add_new_pads (self, srccaps);
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}
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|
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gst_caps_unref (srccaps);
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|
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return TRUE;
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|
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cannot_change_caps:
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{
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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);
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|
return FALSE;
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}
|
|
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;
|
|
}
|