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79d4b18044
These are based on the OpenCore codecs. Fixes bug #584890.
372 lines
11 KiB
C
372 lines
11 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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* AMR narrowband encoder based on the
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* <ulink url="http://sourceforge.net/projects/opencore-amr">opencore codec implementation</ulink>.
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*
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* <refsect2>
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* <title>Example launch line</title>
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* |[
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* gst-launch filesrc location=abc.wav ! wavparse ! audioresample ! audioconvert ! amrnbenc ! filesink location=abc.amr
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* ]|
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* Please note 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 GType
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gst_amrnbenc_bandmode_get_type ()
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{
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static GType gst_amrnbenc_bandmode_type = 0;
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static const GEnumValue gst_amrnbenc_bandmode[] = {
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{MR475, "MR475", "MR475"},
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{MR515, "MR515", "MR515"},
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{MR59, "MR59", "MR59"},
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{MR67, "MR67", "MR67"},
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{MR74, "MR74", "MR74"},
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{MR795, "MR795", "MR795"},
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{MR102, "MR102", "MR102"},
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{MR122, "MR122", "MR122"},
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{MRDTX, "MRDTX", "MRDTX"},
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{0, NULL, NULL},
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};
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if (!gst_amrnbenc_bandmode_type) {
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gst_amrnbenc_bandmode_type =
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g_enum_register_static ("GstAmrnbEncBandMode", gst_amrnbenc_bandmode);
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}
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return gst_amrnbenc_bandmode_type;
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}
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#define GST_AMRNBENC_BANDMODE_TYPE (gst_amrnbenc_bandmode_get_type())
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#define BANDMODE_DEFAULT MR122
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enum
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{
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PROP_0,
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PROP_BANDMODE
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};
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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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static void
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_do_init (GType object_type)
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{
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const GInterfaceInfo preset_interface_info = {
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NULL, /* interface init */
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NULL, /* interface finalize */
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NULL /* interface_data */
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};
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g_type_add_interface_static (object_type, GST_TYPE_PRESET,
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&preset_interface_info);
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GST_DEBUG_CATEGORY_INIT (gst_amrnbenc_debug, "amrnbenc", 0,
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"AMR-NB audio encoder");
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}
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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_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstAmrnbEnc *self = GST_AMRNBENC (object);
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switch (prop_id) {
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case PROP_BANDMODE:
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self->bandmode = g_value_get_enum (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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return;
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}
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static void
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gst_amrnbenc_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstAmrnbEnc *self = GST_AMRNBENC (object);
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switch (prop_id) {
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case PROP_BANDMODE:
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g_value_set_enum (value, self->bandmode);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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return;
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}
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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.taymans@gmail.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->set_property = gst_amrnbenc_set_property;
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object_class->get_property = gst_amrnbenc_get_property;
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object_class->finalize = gst_amrnbenc_finalize;
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g_object_class_install_property (object_class, PROP_BANDMODE,
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g_param_spec_enum ("band-mode", "Band Mode",
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"Encoding Band Mode (Kbps)", GST_AMRNBENC_BANDMODE_TYPE,
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BANDMODE_DEFAULT, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
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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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amrnbenc->discont = TRUE;
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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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}
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if (amrnbenc->discont) {
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GST_BUFFER_FLAG_SET (out, GST_BUFFER_FLAG_DISCONT);
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amrnbenc->discont = FALSE;
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}
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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 =
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Encoder_Interface_Encode (amrnbenc->handle, amrnbenc->bandmode,
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(short *) data, (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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amrnbenc->discont = FALSE;
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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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