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59a213c6d9
Original commit message from CVS: * docs/plugins/Makefile.am: Simplify --extra-dir as gtkdoc scans recursively. * ext/amrwb/gstamrwbenc.c: Fix comment start.
264 lines
7.6 KiB
C
264 lines
7.6 KiB
C
/* GStreamer Adaptive Multi-Rate Wide-Band (AMR-WB) plugin
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* Copyright (C) 2006 Edgard Lima <edgard.lima@indt.org.br>
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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-amrwbenc
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* @see_also: #GstAmrwbDec, #GstAmrwbParse
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*
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* <refsect2>
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* <para>
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* This is an AMR wideband 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 ! amrwbenc ! 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 "gstamrwbenc.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) 16000, " "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-WB, "
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"rate = (int) 16000, " "channels = (int) 1")
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);
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GST_DEBUG_CATEGORY_STATIC (gst_amrwbenc_debug);
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#define GST_CAT_DEFAULT gst_amrwbenc_debug
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static void gst_amrwbenc_finalize (GObject * object);
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static GstFlowReturn gst_amrwbenc_chain (GstPad * pad, GstBuffer * buffer);
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static gboolean gst_amrwbenc_setcaps (GstPad * pad, GstCaps * caps);
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static GstStateChangeReturn gst_amrwbenc_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_amrwbenc_debug, "amrwbenc", 0, "AMR-WB audio encoder");
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GST_BOILERPLATE_FULL (GstAmrwbEnc, gst_amrwbenc, GstElement, GST_TYPE_ELEMENT,
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_do_init);
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static void
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gst_amrwbenc_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-WB audio encoder",
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"Codec/Encoder/Audio",
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"Adaptive Multi-Rate Wideband audio encoder",
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"Renato Araujo <renato.filho@indt.org.br>");
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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_amrwbenc_class_init (GstAmrwbEncClass * 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_amrwbenc_finalize;
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element_class->change_state = GST_DEBUG_FUNCPTR (gst_amrwbenc_state_change);
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}
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static void
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gst_amrwbenc_init (GstAmrwbEnc * amrwbenc, GstAmrwbEncClass * klass)
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{
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/* create the sink pad */
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amrwbenc->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
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gst_pad_set_setcaps_function (amrwbenc->sinkpad, gst_amrwbenc_setcaps);
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gst_pad_set_chain_function (amrwbenc->sinkpad, gst_amrwbenc_chain);
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gst_element_add_pad (GST_ELEMENT (amrwbenc), amrwbenc->sinkpad);
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/* create the src pad */
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amrwbenc->srcpad = gst_pad_new_from_static_template (&src_template, "src");
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gst_pad_use_fixed_caps (amrwbenc->srcpad);
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gst_element_add_pad (GST_ELEMENT (amrwbenc), amrwbenc->srcpad);
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amrwbenc->adapter = gst_adapter_new ();
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/* init rest */
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amrwbenc->handle = NULL;
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amrwbenc->channels = 0;
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amrwbenc->rate = 0;
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amrwbenc->ts = 0;
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}
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static void
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gst_amrwbenc_finalize (GObject * object)
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{
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GstAmrwbEnc *amrwbenc;
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amrwbenc = GST_AMRWBENC (object);
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g_object_unref (G_OBJECT (amrwbenc->adapter));
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amrwbenc->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_amrwbenc_setcaps (GstPad * pad, GstCaps * caps)
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{
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GstStructure *structure;
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GstAmrwbEnc *amrwbenc;
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GstCaps *copy;
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amrwbenc = GST_AMRWBENC (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", &amrwbenc->channels);
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gst_structure_get_int (structure, "rate", &amrwbenc->rate);
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/* this is not wrong but will sound bad */
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if (amrwbenc->channels != 1) {
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GST_WARNING ("amrwbdec is only optimized for mono channels");
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}
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if (amrwbenc->rate != 16000) {
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GST_WARNING ("amrwbdec is only optimized for 16000 Hz samplerate");
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}
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/* create reverse caps */
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copy = gst_caps_new_simple ("audio/AMR-WB",
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"channels", G_TYPE_INT, amrwbenc->channels,
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"rate", G_TYPE_INT, amrwbenc->rate, NULL);
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gst_pad_set_caps (amrwbenc->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_amrwbenc_chain (GstPad * pad, GstBuffer * buffer)
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{
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GstAmrwbEnc *amrwbenc;
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GstFlowReturn ret = GST_FLOW_OK;
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const int buffer_size = sizeof (Word16) * L_FRAME16k;
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amrwbenc = GST_AMRWBENC (gst_pad_get_parent (pad));
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g_return_val_if_fail (amrwbenc->handle, GST_FLOW_WRONG_STATE);
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if (amrwbenc->rate == 0 || amrwbenc->channels == 0) {
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ret = GST_FLOW_NOT_NEGOTIATED;
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goto done;
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}
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if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer))
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amrwbenc->ts = GST_BUFFER_TIMESTAMP (buffer);
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ret = GST_FLOW_OK;
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gst_adapter_push (amrwbenc->adapter, buffer);
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/* Collect samples until we have enough for an output frame */
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while (gst_adapter_available (amrwbenc->adapter) >= buffer_size) {
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GstBuffer *out;
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guint8 *data;
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gint outsize;
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out = gst_buffer_new_and_alloc (buffer_size);
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GST_BUFFER_DURATION (out) = GST_SECOND * L_FRAME16k /
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(amrwbenc->rate * amrwbenc->channels);
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GST_BUFFER_TIMESTAMP (out) = amrwbenc->ts;
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amrwbenc->ts += GST_BUFFER_DURATION (out);
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gst_buffer_set_caps (out, gst_pad_get_caps (amrwbenc->srcpad));
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data = (guint8 *) gst_adapter_peek (amrwbenc->adapter, buffer_size);
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/* encode */
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outsize = E_IF_encode (amrwbenc->handle, 0, (Word16 *) data,
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(UWord8 *) GST_BUFFER_DATA (out), 0);
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gst_adapter_flush (amrwbenc->adapter, buffer_size);
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GST_BUFFER_SIZE (out) = outsize;
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ret = gst_pad_push (amrwbenc->srcpad, out);
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}
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done:
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gst_object_unref (amrwbenc);
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return ret;
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}
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static GstStateChangeReturn
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gst_amrwbenc_state_change (GstElement * element, GstStateChange transition)
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{
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GstAmrwbEnc *amrwbenc;
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GstStateChangeReturn ret;
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amrwbenc = GST_AMRWBENC (element);
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switch (transition) {
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case GST_STATE_CHANGE_NULL_TO_READY:
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if (!(amrwbenc->handle = E_IF_init ()))
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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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amrwbenc->ts = 0;
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gst_adapter_clear (amrwbenc->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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E_IF_exit (amrwbenc->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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