gstreamer/ext/amrnb/amrnbenc.c
2019-08-23 19:06:59 +02:00

291 lines
8.6 KiB
C

/* GStreamer Adaptive Multi-Rate Narrow-Band (AMR-NB) plugin
* Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-amrnbenc
* @title: amrnbenc
* @see_also: #GstAmrnbDec, #GstAmrnbParse
*
* AMR narrowband encoder based on the
* [opencore codec implementation](http://sourceforge.net/projects/opencore-amr).
*
* ## Example launch line
* |[
* gst-launch-1.0 filesrc location=abc.wav ! wavparse ! audioconvert ! audioresample ! amrnbenc ! filesink location=abc.amr
* ]|
* Please note that the above stream misses the header, that is needed to play
* the stream.
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "amrnbenc.h"
static GType
gst_amrnbenc_bandmode_get_type (void)
{
static GType gst_amrnbenc_bandmode_type = 0;
static const GEnumValue gst_amrnbenc_bandmode[] = {
{MR475, "MR475", "MR475"},
{MR515, "MR515", "MR515"},
{MR59, "MR59", "MR59"},
{MR67, "MR67", "MR67"},
{MR74, "MR74", "MR74"},
{MR795, "MR795", "MR795"},
{MR102, "MR102", "MR102"},
{MR122, "MR122", "MR122"},
{MRDTX, "MRDTX", "MRDTX"},
{0, NULL, NULL},
};
if (!gst_amrnbenc_bandmode_type) {
gst_amrnbenc_bandmode_type =
g_enum_register_static ("GstAmrnbEncBandMode", gst_amrnbenc_bandmode);
}
return gst_amrnbenc_bandmode_type;
}
#define GST_AMRNBENC_BANDMODE_TYPE (gst_amrnbenc_bandmode_get_type())
#define BANDMODE_DEFAULT MR122
enum
{
PROP_0,
PROP_BANDMODE
};
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, format = (string) " GST_AUDIO_NE (S16) ", "
"layout = (string) interleaved, "
"rate = (int) 8000," "channels = (int) 1")
);
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/AMR, " "rate = (int) 8000, " "channels = (int) 1")
);
GST_DEBUG_CATEGORY_STATIC (gst_amrnbenc_debug);
#define GST_CAT_DEFAULT gst_amrnbenc_debug
static gboolean gst_amrnbenc_start (GstAudioEncoder * enc);
static gboolean gst_amrnbenc_stop (GstAudioEncoder * enc);
static gboolean gst_amrnbenc_set_format (GstAudioEncoder * enc,
GstAudioInfo * info);
static GstFlowReturn gst_amrnbenc_handle_frame (GstAudioEncoder * enc,
GstBuffer * in_buf);
#define gst_amrnbenc_parent_class parent_class
G_DEFINE_TYPE (GstAmrnbEnc, gst_amrnbenc, GST_TYPE_AUDIO_ENCODER);
static void
gst_amrnbenc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAmrnbEnc *self = GST_AMRNBENC (object);
switch (prop_id) {
case PROP_BANDMODE:
self->bandmode = g_value_get_enum (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
return;
}
static void
gst_amrnbenc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAmrnbEnc *self = GST_AMRNBENC (object);
switch (prop_id) {
case PROP_BANDMODE:
g_value_set_enum (value, self->bandmode);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
return;
}
static void
gst_amrnbenc_class_init (GstAmrnbEncClass * klass)
{
GObjectClass *object_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass);
object_class->set_property = gst_amrnbenc_set_property;
object_class->get_property = gst_amrnbenc_get_property;
base_class->start = GST_DEBUG_FUNCPTR (gst_amrnbenc_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_amrnbenc_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_amrnbenc_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_amrnbenc_handle_frame);
g_object_class_install_property (object_class, PROP_BANDMODE,
g_param_spec_enum ("band-mode", "Band Mode",
"Encoding Band Mode (Kbps)", GST_AMRNBENC_BANDMODE_TYPE,
BANDMODE_DEFAULT,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
gst_element_class_add_static_pad_template (element_class, &sink_template);
gst_element_class_add_static_pad_template (element_class, &src_template);
gst_element_class_set_static_metadata (element_class, "AMR-NB audio encoder",
"Codec/Encoder/Audio",
"Adaptive Multi-Rate Narrow-Band audio encoder",
"Wim Taymans <wim.taymans@gmail.com>");
GST_DEBUG_CATEGORY_INIT (gst_amrnbenc_debug, "amrnbenc", 0,
"AMR-NB audio encoder");
}
static void
gst_amrnbenc_init (GstAmrnbEnc * amrnbenc)
{
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (amrnbenc));
}
static gboolean
gst_amrnbenc_start (GstAudioEncoder * enc)
{
GstAmrnbEnc *amrnbenc = GST_AMRNBENC (enc);
GST_DEBUG_OBJECT (amrnbenc, "start");
if (!(amrnbenc->handle = Encoder_Interface_init (0)))
return FALSE;
return TRUE;
}
static gboolean
gst_amrnbenc_stop (GstAudioEncoder * enc)
{
GstAmrnbEnc *amrnbenc = GST_AMRNBENC (enc);
GST_DEBUG_OBJECT (amrnbenc, "stop");
Encoder_Interface_exit (amrnbenc->handle);
return TRUE;
}
static gboolean
gst_amrnbenc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
{
GstAmrnbEnc *amrnbenc;
GstCaps *copy;
amrnbenc = GST_AMRNBENC (enc);
/* parameters already parsed for us */
amrnbenc->rate = GST_AUDIO_INFO_RATE (info);
amrnbenc->channels = GST_AUDIO_INFO_CHANNELS (info);
/* we do not really accept other input, but anyway ... */
/* this is not wrong but will sound bad */
if (amrnbenc->channels != 1) {
g_warning ("amrnbdec is only optimized for mono channels");
}
if (amrnbenc->rate != 8000) {
g_warning ("amrnbdec is only optimized for 8000 Hz samplerate");
}
/* create reverse caps */
copy = gst_caps_new_simple ("audio/AMR",
"channels", G_TYPE_INT, amrnbenc->channels,
"rate", G_TYPE_INT, amrnbenc->rate, NULL);
gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (amrnbenc), copy);
gst_caps_unref (copy);
/* report needs to base class: hand one frame at a time */
gst_audio_encoder_set_frame_samples_min (enc, 160);
gst_audio_encoder_set_frame_samples_max (enc, 160);
gst_audio_encoder_set_frame_max (enc, 1);
return TRUE;
}
static GstFlowReturn
gst_amrnbenc_handle_frame (GstAudioEncoder * enc, GstBuffer * buffer)
{
GstAmrnbEnc *amrnbenc;
GstFlowReturn ret;
GstBuffer *out;
GstMapInfo in_map, out_map;
gsize out_size;
amrnbenc = GST_AMRNBENC (enc);
g_return_val_if_fail (amrnbenc->handle, GST_FLOW_FLUSHING);
/* we don't deal with squeezing remnants, so simply discard those */
if (G_UNLIKELY (buffer == NULL)) {
GST_DEBUG_OBJECT (amrnbenc, "no data");
return GST_FLOW_OK;
}
gst_buffer_map (buffer, &in_map, GST_MAP_READ);
if (G_UNLIKELY (in_map.size < 320)) {
gst_buffer_unmap (buffer, &in_map);
GST_DEBUG_OBJECT (amrnbenc, "discarding trailing data of %" G_GSIZE_FORMAT
" bytes", in_map.size);
return gst_audio_encoder_finish_frame (enc, NULL, -1);
}
/* get output, max size is 32 */
out = gst_buffer_new_and_alloc (32);
/* AMR encoder actually writes into the source data buffers it gets */
/* should be able to handle that with what we are given */
gst_buffer_map (out, &out_map, GST_MAP_WRITE);
/* encode */
out_size =
Encoder_Interface_Encode (amrnbenc->handle, amrnbenc->bandmode,
(short *) in_map.data, out_map.data, 0);
gst_buffer_unmap (out, &out_map);
gst_buffer_resize (out, 0, out_size);
gst_buffer_unmap (buffer, &in_map);
GST_LOG_OBJECT (amrnbenc, "output data size %" G_GSIZE_FORMAT, out_size);
if (out_size) {
ret = gst_audio_encoder_finish_frame (enc, out, 160);
} else {
/* should not happen (without dtx or so at least) */
GST_WARNING_OBJECT (amrnbenc, "no encoded data; discarding input");
gst_buffer_unref (out);
ret = gst_audio_encoder_finish_frame (enc, NULL, -1);
}
return ret;
}