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bebdf0837e
Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-ugly-plugins-docs.sgml: * docs/plugins/gst-plugins-ugly-plugins-sections.txt: * docs/plugins/inspect/plugin-a52dec.xml: * docs/plugins/inspect/plugin-amrnb.xml: * docs/plugins/inspect/plugin-asf.xml: * docs/plugins/inspect/plugin-dvdlpcmdec.xml: * docs/plugins/inspect/plugin-dvdsub.xml: * docs/plugins/inspect/plugin-iec958.xml: * docs/plugins/inspect/plugin-lame.xml: * docs/plugins/inspect/plugin-mad.xml: * docs/plugins/inspect/plugin-mpeg2dec.xml: * docs/plugins/inspect/plugin-mpegaudioparse.xml: * docs/plugins/inspect/plugin-mpegstream.xml: Add new docs. Scan c++ files too. * ext/amrnb/amrnbdec.c: (gst_amrnbdec_base_init), (gst_amrnbdec_event): * ext/amrnb/amrnbenc.c: (gst_amrnbenc_base_init): * ext/amrnb/amrnbparse.c: (gst_amrnbparse_base_init), (gst_amrnbparse_loop): Add documentation headers. * ext/mad/gstmad.c: * ext/mad/gstmad.h: Refactor for docs.
281 lines
8.3 KiB
C
281 lines
8.3 KiB
C
/* GStreamer Adaptive Multi-Rate Narrow-Band (AMR-NB) plugin
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* Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
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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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/**
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* SECTION:element-amrnbenc
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* @see_also: #GstAmrnbDec, #GstAmrnbParse
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*
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* <refsect2>
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* <para>
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* This is an AMR narrowband encoder based on the
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* <ulink url="http://www.penguin.cz/~utx/amr">reference codec implementation</ulink>.
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* </para>
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* <title>Example launch line</title>
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* <para>
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* <programlisting>
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* gst-launch filesrc location=abc.wav ! wavparse ! audioresample ! audioconvert ! amrnbenc ! filesink location=abc.amr
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* </programlisting>
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* </para>
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* Please not that the above stream misses the header, that is needed to play
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* the stream.
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* </refsect2>
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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 "amrnbenc.h"
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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-int, "
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"width = (int) 16, "
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"depth = (int) 16, "
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"signed = (boolean) TRUE, "
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"endianness = (int) BYTE_ORDER, "
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"rate = (int) 8000," "channels = (int) 1")
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);
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static GstStaticPadTemplate src_template = 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/AMR, " "rate = (int) 8000, " "channels = (int) 1")
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);
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GST_DEBUG_CATEGORY_STATIC (gst_amrnbenc_debug);
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#define GST_CAT_DEFAULT gst_amrnbenc_debug
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static void gst_amrnbenc_finalize (GObject * object);
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static GstFlowReturn gst_amrnbenc_chain (GstPad * pad, GstBuffer * buffer);
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static gboolean gst_amrnbenc_setcaps (GstPad * pad, GstCaps * caps);
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static GstStateChangeReturn gst_amrnbenc_state_change (GstElement * element,
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GstStateChange transition);
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#define _do_init(bla) \
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GST_DEBUG_CATEGORY_INIT (gst_amrnbenc_debug, "amrnbenc", 0, "AMR-NB audio encoder");
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GST_BOILERPLATE_FULL (GstAmrnbEnc, gst_amrnbenc, GstElement, GST_TYPE_ELEMENT,
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_do_init);
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static void
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gst_amrnbenc_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstElementDetails details = GST_ELEMENT_DETAILS ("AMR-NB audio encoder",
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"Codec/Encoder/Audio",
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"Adaptive Multi-Rate Narrow-Band audio encoder",
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"Ronald Bultje <rbultje@ronald.bitfreak.net>, "
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"Wim Taymans <wim@fluendo.com>");
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&src_template));
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gst_element_class_set_details (element_class, &details);
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}
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static void
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gst_amrnbenc_class_init (GstAmrnbEncClass * klass)
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{
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GObjectClass *object_class = G_OBJECT_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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object_class->finalize = gst_amrnbenc_finalize;
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element_class->change_state = GST_DEBUG_FUNCPTR (gst_amrnbenc_state_change);
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}
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static void
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gst_amrnbenc_init (GstAmrnbEnc * amrnbenc, GstAmrnbEncClass * klass)
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{
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/* create the sink pad */
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amrnbenc->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
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gst_pad_set_setcaps_function (amrnbenc->sinkpad, gst_amrnbenc_setcaps);
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gst_pad_set_chain_function (amrnbenc->sinkpad, gst_amrnbenc_chain);
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gst_element_add_pad (GST_ELEMENT (amrnbenc), amrnbenc->sinkpad);
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/* create the src pad */
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amrnbenc->srcpad = gst_pad_new_from_static_template (&src_template, "src");
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gst_pad_use_fixed_caps (amrnbenc->srcpad);
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gst_element_add_pad (GST_ELEMENT (amrnbenc), amrnbenc->srcpad);
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amrnbenc->adapter = gst_adapter_new ();
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/* init rest */
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amrnbenc->handle = NULL;
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}
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static void
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gst_amrnbenc_finalize (GObject * object)
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{
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GstAmrnbEnc *amrnbenc;
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amrnbenc = GST_AMRNBENC (object);
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g_object_unref (G_OBJECT (amrnbenc->adapter));
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amrnbenc->adapter = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gboolean
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gst_amrnbenc_setcaps (GstPad * pad, GstCaps * caps)
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{
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GstStructure *structure;
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GstAmrnbEnc *amrnbenc;
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GstCaps *copy;
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amrnbenc = GST_AMRNBENC (GST_PAD_PARENT (pad));
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structure = gst_caps_get_structure (caps, 0);
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/* get channel count */
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gst_structure_get_int (structure, "channels", &amrnbenc->channels);
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gst_structure_get_int (structure, "rate", &amrnbenc->rate);
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/* this is not wrong but will sound bad */
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if (amrnbenc->channels != 1) {
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g_warning ("amrnbdec is only optimized for mono channels");
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}
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if (amrnbenc->rate != 8000) {
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g_warning ("amrnbdec is only optimized for 8000 Hz samplerate");
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}
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/* create reverse caps */
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copy = gst_caps_new_simple ("audio/AMR",
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"channels", G_TYPE_INT, amrnbenc->channels,
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"rate", G_TYPE_INT, amrnbenc->rate, NULL);
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/* precalc duration as it's constant now */
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amrnbenc->duration = gst_util_uint64_scale_int (160, GST_SECOND,
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amrnbenc->rate * amrnbenc->channels);
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gst_pad_set_caps (amrnbenc->srcpad, copy);
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gst_caps_unref (copy);
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return TRUE;
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}
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static GstFlowReturn
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gst_amrnbenc_chain (GstPad * pad, GstBuffer * buffer)
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{
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GstAmrnbEnc *amrnbenc;
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GstFlowReturn ret;
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amrnbenc = GST_AMRNBENC (GST_PAD_PARENT (pad));
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g_return_val_if_fail (amrnbenc->handle, GST_FLOW_WRONG_STATE);
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if (amrnbenc->rate == 0 || amrnbenc->channels == 0)
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goto not_negotiated;
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/* discontinuity clears adapter, FIXME, maybe we can set some
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* encoder flag to mask the discont. */
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if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
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gst_adapter_clear (amrnbenc->adapter);
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amrnbenc->ts = 0;
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}
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/* take latest timestamp, FIXME timestamp is the one of the
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* first buffer in the adapter. */
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if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer))
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amrnbenc->ts = GST_BUFFER_TIMESTAMP (buffer);
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ret = GST_FLOW_OK;
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gst_adapter_push (amrnbenc->adapter, buffer);
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/* Collect samples until we have enough for an output frame */
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while (gst_adapter_available (amrnbenc->adapter) >= 320) {
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GstBuffer *out;
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guint8 *data;
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gint outsize;
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/* get output, max size is 32 */
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out = gst_buffer_new_and_alloc (32);
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GST_BUFFER_DURATION (out) = amrnbenc->duration;
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GST_BUFFER_TIMESTAMP (out) = amrnbenc->ts;
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if (amrnbenc->ts != -1)
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amrnbenc->ts += amrnbenc->duration;
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gst_buffer_set_caps (out, GST_PAD_CAPS (amrnbenc->srcpad));
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/* The AMR encoder actually writes into the source data buffers it gets */
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data = gst_adapter_take (amrnbenc->adapter, 320);
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/* encode */
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outsize = Encoder_Interface_Encode (amrnbenc->handle, MR122, (short *) data,
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(guint8 *) GST_BUFFER_DATA (out), 0);
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g_free (data);
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GST_BUFFER_SIZE (out) = outsize;
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/* play */
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if ((ret = gst_pad_push (amrnbenc->srcpad, out)) != GST_FLOW_OK)
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break;
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}
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return ret;
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/* ERRORS */
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not_negotiated:
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{
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GST_ELEMENT_ERROR (amrnbenc, STREAM, TYPE_NOT_FOUND,
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(NULL), ("unknown type"));
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return GST_FLOW_NOT_NEGOTIATED;
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}
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}
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static GstStateChangeReturn
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gst_amrnbenc_state_change (GstElement * element, GstStateChange transition)
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{
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GstAmrnbEnc *amrnbenc;
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GstStateChangeReturn ret;
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amrnbenc = GST_AMRNBENC (element);
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switch (transition) {
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case GST_STATE_CHANGE_NULL_TO_READY:
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if (!(amrnbenc->handle = Encoder_Interface_init (0)))
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return GST_STATE_CHANGE_FAILURE;
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break;
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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amrnbenc->rate = 0;
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amrnbenc->channels = 0;
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amrnbenc->ts = 0;
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gst_adapter_clear (amrnbenc->adapter);
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break;
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default:
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break;
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}
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ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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switch (transition) {
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case GST_STATE_CHANGE_READY_TO_NULL:
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Encoder_Interface_exit (amrnbenc->handle);
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break;
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default:
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break;
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}
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return ret;
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}
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