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31831eb47e
doesn't align on 20 millisecond frame size. The AMR-WB codec imposes a fixed 20 millisecond frame size. In its current form, the `voamrwbenc` plugin deals with this limitation by discarding any audio at the end of the stream that falls short of 20 milliseconds. This patch keeps the audio data, and appends silence to the end to preserve frame size alignment. The patch also adds tests to check for the updated behavior. I noticed that tests weren't being built, so I changed the build to allow for building the tests when the `tests` and `voamrwbenc` options are set. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3027>
320 lines
9 KiB
C
320 lines
9 KiB
C
/* GStreamer Adaptive Multi-Rate Wide-Band (AMR-WB) plugin
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* Copyright (C) 2006 Edgard Lima <edgard.lima@gmail.com>
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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-voamrwbenc
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* @title: 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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* [reference codec implementation](http://www.penguin.cz/~utx/amr).
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*
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* ## Example launch line
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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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*
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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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#include <string.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 const 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, "
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"format = (string) " GST_AUDIO_NE (S16) ", "
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"layout = (string) interleaved, "
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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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G_DEFINE_TYPE (GstVoAmrWbEnc, gst_voamrwbenc, GST_TYPE_AUDIO_ENCODER);
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GST_ELEMENT_REGISTER_DEFINE (voamrwbenc, "voamrwbenc",
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GST_RANK_SECONDARY, GST_TYPE_VOAMRWBENC);
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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_class_init (GstVoAmrWbEncClass * 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_voamrwbenc_set_property;
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object_class->get_property = gst_voamrwbenc_get_property;
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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-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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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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GST_DEBUG_CATEGORY_INIT (gst_voamrwbenc_debug, "voamrwbenc", 0,
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"voamrwb encoder");
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gst_type_mark_as_plugin_api (GST_VOAMRWBENC_BANDMODE_TYPE, 0);
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}
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static void
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gst_voamrwbenc_init (GstVoAmrWbEnc * amrwbenc)
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{
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GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (amrwbenc));
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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_audio_encoder_set_output_format (GST_AUDIO_ENCODER (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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GstMapInfo map, omap;
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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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gst_buffer_map (buffer, &map, GST_MAP_READ);
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out = gst_buffer_new_and_alloc (buffer_size);
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gst_buffer_map (out, &omap, GST_MAP_WRITE);
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/* encode */
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if (G_UNLIKELY (map.size < buffer_size)) {
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short input_buffer[L_FRAME16k] = { 0, };
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GST_DEBUG_OBJECT (amrwbenc, "add silence to packet of size %d",
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(gint) map.size);
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memcpy ((void *) input_buffer, map.data, map.size);
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outsize = E_IF_encode (amrwbenc->handle, amrwbenc->bandmode,
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(const short *) input_buffer, (unsigned char *) omap.data, 0);
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} else {
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outsize = E_IF_encode (amrwbenc->handle, amrwbenc->bandmode,
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(const short *) map.data, (unsigned char *) omap.data, 0);
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}
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GST_LOG_OBJECT (amrwbenc, "encoded to %d bytes", outsize);
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gst_buffer_unmap (out, &omap);
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gst_buffer_unmap (buffer, &map);
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gst_buffer_resize (out, 0, 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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