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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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4b6bcfb3bf
Original commit message from CVS: Setting caps on the outgoing buffers.
243 lines
6.7 KiB
C
243 lines
6.7 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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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 <string.h>
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#include "gstgsmenc.h"
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/* elementfactory information */
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GstElementDetails gst_gsmenc_details = {
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"GSM audio encoder",
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"Codec/Encoder/Audio",
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"Encodes audio using GSM",
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"Wim Taymans <wim.taymans@chello.be>",
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};
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/* GSMEnc signals and args */
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enum
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{
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FRAME_ENCODED,
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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ARG_0
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/* FILL ME */
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};
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static void gst_gsmenc_base_init (gpointer g_class);
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static void gst_gsmenc_class_init (GstGSMEnc * klass);
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static void gst_gsmenc_init (GstGSMEnc * gsmenc);
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static GstFlowReturn gst_gsmenc_chain (GstPad * pad, GstBuffer * buf);
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static GstElementClass *parent_class = NULL;
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static guint gst_gsmenc_signals[LAST_SIGNAL] = { 0 };
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GType
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gst_gsmenc_get_type (void)
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{
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static GType gsmenc_type = 0;
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if (!gsmenc_type) {
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static const GTypeInfo gsmenc_info = {
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sizeof (GstGSMEncClass),
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gst_gsmenc_base_init,
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NULL,
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(GClassInitFunc) gst_gsmenc_class_init,
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NULL,
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NULL,
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sizeof (GstGSMEnc),
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0,
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(GInstanceInitFunc) gst_gsmenc_init,
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};
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gsmenc_type =
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g_type_register_static (GST_TYPE_ELEMENT, "GstGSMEnc", &gsmenc_info, 0);
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}
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return gsmenc_type;
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}
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static GstStaticPadTemplate gsmenc_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-gsm, " "rate = (int) 8000, " "channels = (int) 1")
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);
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static GstStaticPadTemplate gsmenc_sink_template =
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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-int, "
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"endianness = (int) BYTE_ORDER, "
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"signed = (boolean) true, "
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"width = (int) 16, "
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"depth = (int) 16, " "rate = (int) 8000, " "channels = (int) 1")
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);
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static void
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gst_gsmenc_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gsmenc_sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gsmenc_src_template));
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gst_element_class_set_details (element_class, &gst_gsmenc_details);
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}
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static void
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gst_gsmenc_class_init (GstGSMEnc * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
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gst_gsmenc_signals[FRAME_ENCODED] =
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g_signal_new ("frame-encoded", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstGSMEncClass, frame_encoded), NULL,
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NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0);
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}
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static void
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gst_gsmenc_init (GstGSMEnc * gsmenc)
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{
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/* create the sink and src pads */
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gsmenc->sinkpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&gsmenc_sink_template), "sink");
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gst_element_add_pad (GST_ELEMENT (gsmenc), gsmenc->sinkpad);
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gst_pad_set_chain_function (gsmenc->sinkpad, gst_gsmenc_chain);
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gsmenc->srcpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&gsmenc_src_template), "src");
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gst_element_add_pad (GST_ELEMENT (gsmenc), gsmenc->srcpad);
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gsmenc->state = gsm_create ();
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gsmenc->bufsize = 0;
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gsmenc->next_ts = 0;
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}
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static GstFlowReturn
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gst_gsmenc_chain (GstPad * pad, GstBuffer * buf)
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{
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GstGSMEnc *gsmenc;
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gsmenc = GST_GSMENC (GST_OBJECT_PARENT (pad));
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if (GST_IS_EVENT (buf)) {
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GstEvent *event = GST_EVENT (buf);
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_EOS:{
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gst_pad_push_event (gsmenc->srcpad, gst_event_new_eos ());
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gst_pad_push (gsmenc->srcpad, buf);
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break;
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}
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case GST_EVENT_NEWSEGMENT:{
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/* drop the discontinuity */
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break;
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}
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default:{
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gst_pad_event_default (pad, event);
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break;
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}
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}
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return GST_FLOW_OK;
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} else if (GST_IS_BUFFER (buf)) {
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gsm_signal *data;
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guint size;
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GstCaps *tempcaps = NULL;
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data = (gsm_signal *) GST_BUFFER_DATA (buf);
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size = GST_BUFFER_SIZE (buf) / sizeof (gsm_signal);
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if (gsmenc->bufsize && (gsmenc->bufsize + size >= 160)) {
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GstBuffer *outbuf;
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memcpy (gsmenc->buffer + gsmenc->bufsize, data,
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(160 - gsmenc->bufsize) * sizeof (gsm_signal));
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outbuf = gst_buffer_new_and_alloc (33 * sizeof (gsm_byte));
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GST_BUFFER_TIMESTAMP (outbuf) = gsmenc->next_ts;
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GST_BUFFER_DURATION (outbuf) = 20 * GST_MSECOND;
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gsmenc->next_ts += 20 * GST_MSECOND;
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gsm_encode (gsmenc->state, gsmenc->buffer,
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(gsm_byte *) GST_BUFFER_DATA (outbuf));
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tempcaps = gst_caps_new_simple ("audio/x-gsm",
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"rate", G_TYPE_INT, 8000, "channels", G_TYPE_INT, 1, NULL);
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gst_buffer_set_caps (outbuf, tempcaps);
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gst_pad_push (gsmenc->srcpad, outbuf);
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size -= (160 - gsmenc->bufsize);
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data += (160 - gsmenc->bufsize);
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gsmenc->bufsize = 0;
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}
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while (size >= 160) {
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GstBuffer *outbuf;
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outbuf = gst_buffer_new_and_alloc (33 * sizeof (gsm_byte));
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GST_BUFFER_TIMESTAMP (outbuf) = gsmenc->next_ts;
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GST_BUFFER_DURATION (outbuf) = 20 * GST_MSECOND;
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gsmenc->next_ts += 20 * GST_MSECOND;
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gsm_encode (gsmenc->state, data, (gsm_byte *) GST_BUFFER_DATA (outbuf));
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/* I was wondering that gst_buffer_set_caps (outbuf, GST_PAD_CAPS (gsmenc->srcpad));
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* could work, but it doens't work */
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tempcaps = gst_caps_new_simple ("audio/x-gsm",
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"rate", G_TYPE_INT, 8000, "channels", G_TYPE_INT, 1, NULL);
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gst_buffer_set_caps (outbuf, tempcaps);
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gst_pad_push (gsmenc->srcpad, outbuf);
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size -= 160;
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data += 160;
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}
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if (size) {
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memcpy (gsmenc->buffer + gsmenc->bufsize, data,
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size * sizeof (gsm_signal));
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gsmenc->bufsize += size;
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}
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/*gst_buffer_unref (buf); */
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return GST_FLOW_OK;
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}
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return GST_FLOW_OK;
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}
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