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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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295 lines
8.7 KiB
C
295 lines
8.7 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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-amrnbenc
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* @title: 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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* [opencore codec implementation](http://sourceforge.net/projects/opencore-amr).
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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=abc.wav ! wavparse ! audioconvert ! audioresample ! 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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*
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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 (void)
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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, format = (string) " GST_AUDIO_NE (S16) ", "
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"layout = (string) interleaved, "
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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 gboolean gst_amrnbenc_start (GstAudioEncoder * enc);
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static gboolean gst_amrnbenc_stop (GstAudioEncoder * enc);
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static gboolean gst_amrnbenc_set_format (GstAudioEncoder * enc,
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GstAudioInfo * info);
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static GstFlowReturn gst_amrnbenc_handle_frame (GstAudioEncoder * enc,
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GstBuffer * in_buf);
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#define gst_amrnbenc_parent_class parent_class
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G_DEFINE_TYPE (GstAmrnbEnc, gst_amrnbenc, GST_TYPE_AUDIO_ENCODER);
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GST_ELEMENT_REGISTER_DEFINE (amrnbenc, "amrnbenc", GST_RANK_SECONDARY,
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GST_TYPE_AMRNBENC);
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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_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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GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_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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base_class->start = GST_DEBUG_FUNCPTR (gst_amrnbenc_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_amrnbenc_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_amrnbenc_set_format);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_amrnbenc_handle_frame);
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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,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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gst_element_class_add_static_pad_template (element_class, &sink_template);
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gst_element_class_add_static_pad_template (element_class, &src_template);
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gst_element_class_set_static_metadata (element_class, "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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"Wim Taymans <wim.taymans@gmail.com>");
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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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gst_type_mark_as_plugin_api (GST_AMRNBENC_BANDMODE_TYPE, 0);
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}
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static void
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gst_amrnbenc_init (GstAmrnbEnc * amrnbenc)
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{
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GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (amrnbenc));
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}
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static gboolean
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gst_amrnbenc_start (GstAudioEncoder * enc)
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{
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GstAmrnbEnc *amrnbenc = GST_AMRNBENC (enc);
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GST_DEBUG_OBJECT (amrnbenc, "start");
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if (!(amrnbenc->handle = Encoder_Interface_init (0)))
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return FALSE;
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return TRUE;
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}
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static gboolean
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gst_amrnbenc_stop (GstAudioEncoder * enc)
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{
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GstAmrnbEnc *amrnbenc = GST_AMRNBENC (enc);
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GST_DEBUG_OBJECT (amrnbenc, "stop");
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Encoder_Interface_exit (amrnbenc->handle);
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return TRUE;
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}
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static gboolean
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gst_amrnbenc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
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{
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GstAmrnbEnc *amrnbenc;
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GstCaps *copy;
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amrnbenc = GST_AMRNBENC (enc);
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/* parameters already parsed for us */
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amrnbenc->rate = GST_AUDIO_INFO_RATE (info);
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amrnbenc->channels = GST_AUDIO_INFO_CHANNELS (info);
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/* we do not really accept other input, but anyway ... */
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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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gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (amrnbenc), copy);
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gst_caps_unref (copy);
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/* report needs to base class: hand one frame at a time */
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gst_audio_encoder_set_frame_samples_min (enc, 160);
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gst_audio_encoder_set_frame_samples_max (enc, 160);
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gst_audio_encoder_set_frame_max (enc, 1);
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return TRUE;
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}
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static GstFlowReturn
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gst_amrnbenc_handle_frame (GstAudioEncoder * enc, GstBuffer * buffer)
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{
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GstAmrnbEnc *amrnbenc;
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GstFlowReturn ret;
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GstBuffer *out;
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GstMapInfo in_map, out_map;
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gsize out_size;
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amrnbenc = GST_AMRNBENC (enc);
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g_return_val_if_fail (amrnbenc->handle, GST_FLOW_FLUSHING);
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/* we don't deal with squeezing remnants, so simply discard those */
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if (G_UNLIKELY (buffer == NULL)) {
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GST_DEBUG_OBJECT (amrnbenc, "no data");
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return GST_FLOW_OK;
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}
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gst_buffer_map (buffer, &in_map, GST_MAP_READ);
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if (G_UNLIKELY (in_map.size < 320)) {
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gst_buffer_unmap (buffer, &in_map);
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GST_DEBUG_OBJECT (amrnbenc, "discarding trailing data of %" G_GSIZE_FORMAT
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" bytes", in_map.size);
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return gst_audio_encoder_finish_frame (enc, NULL, -1);
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}
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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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/* AMR encoder actually writes into the source data buffers it gets */
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/* should be able to handle that with what we are given */
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gst_buffer_map (out, &out_map, GST_MAP_WRITE);
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/* encode */
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out_size =
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Encoder_Interface_Encode (amrnbenc->handle, amrnbenc->bandmode,
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(short *) in_map.data, out_map.data, 0);
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gst_buffer_unmap (out, &out_map);
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gst_buffer_resize (out, 0, out_size);
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gst_buffer_unmap (buffer, &in_map);
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GST_LOG_OBJECT (amrnbenc, "output data size %" G_GSIZE_FORMAT, out_size);
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if (out_size) {
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ret = gst_audio_encoder_finish_frame (enc, out, 160);
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} else {
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/* should not happen (without dtx or so at least) */
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GST_WARNING_OBJECT (amrnbenc, "no encoded data; discarding input");
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gst_buffer_unref (out);
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ret = gst_audio_encoder_finish_frame (enc, NULL, -1);
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}
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return ret;
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}
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