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f70a623418
Conflicts: docs/libs/Makefile.am ext/kate/gstkatetiger.c ext/opus/gstopusdec.c ext/xvid/gstxvidenc.c gst-libs/gst/basecamerabinsrc/Makefile.am gst-libs/gst/basecamerabinsrc/gstbasecamerasrc.c gst-libs/gst/basecamerabinsrc/gstbasecamerasrc.h gst-libs/gst/video/gstbasevideocodec.c gst-libs/gst/video/gstbasevideocodec.h gst-libs/gst/video/gstbasevideodecoder.c gst-libs/gst/video/gstbasevideoencoder.c gst/asfmux/gstasfmux.c gst/audiovisualizers/gstwavescope.c gst/camerabin2/gstcamerabin2.c gst/debugutils/gstcompare.c gst/frei0r/gstfrei0rmixer.c gst/mpegpsmux/mpegpsmux.c gst/mpegtsmux/mpegtsmux.c gst/mxf/mxfmux.c gst/videomeasure/gstvideomeasure_ssim.c gst/videoparsers/gsth264parse.c gst/videoparsers/gstmpeg4videoparse.c
310 lines
8.6 KiB
C
310 lines
8.6 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-voamrwbenc
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* @see_also: #GstAmrWbDec, #GstAmrWbParse
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*
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* 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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*
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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 ! voamrwbenc ! 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 "gstvoamrwbenc.h"
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#define MR660 0
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#define MR885 1
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#define MR1265 2
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#define MR1425 2
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#define MR1585 3
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#define MR1825 4
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#define MR1985 5
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#define MR2305 6
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#define MR2385 7
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#define MRDTX 8
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#define L_FRAME16k 320 /* Frame size at 16kHz */
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static GType
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gst_voamrwbenc_bandmode_get_type (void)
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{
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static GType gst_voamrwbenc_bandmode_type = 0;
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static GEnumValue gst_voamrwbenc_bandmode[] = {
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{MR660, "MR660", "MR660"},
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{MR885, "MR885", "MR885"},
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{MR1265, "MR1265", "MR1265"},
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{MR1425, "MR1425", "MR1425"},
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{MR1585, "MR1585", "MR1585"},
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{MR1825, "MR1825", "MR1825"},
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{MR1985, "MR1985", "MR1985"},
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{MR2305, "MR2305", "MR2305"},
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{MR2385, "MR2385", "MR2385"},
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{MRDTX, "MRDTX", "MRDTX"},
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{0, NULL, NULL},
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};
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if (!gst_voamrwbenc_bandmode_type) {
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gst_voamrwbenc_bandmode_type =
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g_enum_register_static ("GstVoAmrWbEncBandMode",
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gst_voamrwbenc_bandmode);
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}
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return gst_voamrwbenc_bandmode_type;
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}
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#define GST_VOAMRWBENC_BANDMODE_TYPE (gst_voamrwbenc_bandmode_get_type())
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#define BANDMODE_DEFAULT MR660
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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) 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_voamrwbenc_debug);
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#define GST_CAT_DEFAULT gst_voamrwbenc_debug
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static gboolean gst_voamrwbenc_start (GstAudioEncoder * enc);
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static gboolean gst_voamrwbenc_stop (GstAudioEncoder * enc);
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static gboolean gst_voamrwbenc_set_format (GstAudioEncoder * enc,
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GstAudioInfo * info);
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static GstFlowReturn gst_voamrwbenc_handle_frame (GstAudioEncoder * enc,
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GstBuffer * in_buf);
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GST_BOILERPLATE (GstVoAmrWbEnc, gst_voamrwbenc, GstAudioEncoder,
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GST_TYPE_AUDIO_ENCODER);
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static void
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gst_voamrwbenc_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstVoAmrWbEnc *self = GST_VOAMRWBENC (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_voamrwbenc_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstVoAmrWbEnc *self = GST_VOAMRWBENC (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_voamrwbenc_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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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_simple (element_class, "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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}
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static void
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gst_voamrwbenc_class_init (GstVoAmrWbEncClass * klass)
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{
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GObjectClass *object_class = G_OBJECT_CLASS (klass);
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GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass);
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object_class->set_property = gst_voamrwbenc_set_property;
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object_class->get_property = gst_voamrwbenc_get_property;
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base_class->start = GST_DEBUG_FUNCPTR (gst_voamrwbenc_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_voamrwbenc_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_voamrwbenc_set_format);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_voamrwbenc_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_VOAMRWBENC_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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}
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static void
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gst_voamrwbenc_init (GstVoAmrWbEnc * amrwbenc, GstVoAmrWbEncClass * klass)
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{
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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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}
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static gboolean
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gst_voamrwbenc_start (GstAudioEncoder * enc)
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{
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GstVoAmrWbEnc *voamrwbenc = GST_VOAMRWBENC (enc);
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GST_DEBUG_OBJECT (enc, "start");
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if (!(voamrwbenc->handle = E_IF_init ()))
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return FALSE;
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voamrwbenc->rate = 0;
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voamrwbenc->channels = 0;
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return TRUE;
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}
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static gboolean
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gst_voamrwbenc_stop (GstAudioEncoder * enc)
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{
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GstVoAmrWbEnc *voamrwbenc = GST_VOAMRWBENC (enc);
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GST_DEBUG_OBJECT (enc, "stop");
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if (voamrwbenc->handle) {
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E_IF_exit (voamrwbenc->handle);
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voamrwbenc->handle = NULL;
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}
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return TRUE;
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}
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static gboolean
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gst_voamrwbenc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
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{
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GstVoAmrWbEnc *amrwbenc;
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GstCaps *copy;
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amrwbenc = GST_VOAMRWBENC (benc);
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/* get channel count */
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amrwbenc->channels = GST_AUDIO_INFO_CHANNELS (info);
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amrwbenc->rate = GST_AUDIO_INFO_RATE (info);
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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 (GST_AUDIO_ENCODER_SRC_PAD (amrwbenc), copy);
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gst_caps_unref (copy);
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/* report needs to base class: one frame at a time */
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gst_audio_encoder_set_frame_samples_min (benc, L_FRAME16k);
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gst_audio_encoder_set_frame_samples_max (benc, L_FRAME16k);
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gst_audio_encoder_set_frame_max (benc, 1);
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return TRUE;
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}
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static GstFlowReturn
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gst_voamrwbenc_handle_frame (GstAudioEncoder * benc, GstBuffer * buffer)
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{
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GstVoAmrWbEnc *amrwbenc;
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GstFlowReturn ret = GST_FLOW_OK;
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const int buffer_size = sizeof (short) * L_FRAME16k;
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GstBuffer *out;
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gint outsize;
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amrwbenc = GST_VOAMRWBENC (benc);
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g_return_val_if_fail (amrwbenc->handle, GST_FLOW_NOT_NEGOTIATED);
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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 (amrwbenc, "no data");
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goto done;
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}
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if (G_UNLIKELY (GST_BUFFER_SIZE (buffer) < buffer_size)) {
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GST_DEBUG_OBJECT (amrwbenc, "discarding trailing data %d",
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buffer ? GST_BUFFER_SIZE (buffer) : 0);
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ret = gst_audio_encoder_finish_frame (benc, NULL, -1);
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goto done;
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}
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out = gst_buffer_new_and_alloc (buffer_size);
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/* encode */
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outsize = E_IF_encode (amrwbenc->handle, amrwbenc->bandmode,
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(const short *) GST_BUFFER_DATA (buffer),
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(unsigned char *) GST_BUFFER_DATA (out), 0);
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GST_LOG_OBJECT (amrwbenc, "encoded to %d bytes", outsize);
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GST_BUFFER_SIZE (out) = outsize;
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ret = gst_audio_encoder_finish_frame (benc, out, L_FRAME16k);
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done:
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return ret;
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}
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