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437 lines
13 KiB
C
437 lines
13 KiB
C
/* iSAC encoder
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*
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* Copyright (C) 2020 Collabora Ltd.
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* Author: Guillaume Desmottes <guillaume.desmottes@collabora.com>, Collabora Ltd.
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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 Free
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* Software Foundation, Inc., 51 Franklin Street, 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-isacenc
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* @title: isacenc
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* @short_description: iSAC audio encoder
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*
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* Since: 1.20
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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 "gstisacenc.h"
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#include "gstisacutils.h"
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#include <modules/audio_coding/codecs/isac/main/include/isac.h>
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GST_DEBUG_CATEGORY_STATIC (isacenc_debug);
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#define GST_CAT_DEFAULT isacenc_debug
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/* Buffer size used in the simpleKenny.c test app from webrtc */
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#define OUTPUT_BUFFER_SIZE 1200
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#define GST_TYPE_ISACENC_OUTPUT_FRAME_LEN (gst_isacenc_output_frame_len_get_type ())
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static GType
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gst_isacenc_output_frame_len_get_type (void)
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{
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static GType qtype = 0;
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if (qtype == 0) {
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static const GEnumValue values[] = {
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{30, "30 ms", "30 ms"},
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{60, "60 ms", "60 ms, only usable in wideband mode (16 kHz)"},
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{0, NULL, NULL}
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};
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qtype = g_enum_register_static ("GstIsacEncOutputFrameLen", values);
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}
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return qtype;
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}
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enum
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{
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PROP_0,
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PROP_OUTPUT_FRAME_LEN,
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PROP_BITRATE,
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PROP_MAX_PAYLOAD_SIZE,
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PROP_MAX_RATE,
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};
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#define GST_ISACENC_OUTPUT_FRAME_LEN_DEFAULT (30)
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#define GST_ISACENC_BITRATE_DEFAULT (32000)
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#define GST_ISACENC_MAX_PAYLOAD_SIZE_DEFAULT (-1)
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#define GST_ISACENC_MAX_RATE_DEFAULT (-1)
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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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"rate = (int) { 16000, 32000 }, "
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"layout = (string) interleaved, " "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/isac, "
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"rate = (int) { 16000, 32000 }, " "channels = (int) 1")
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);
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typedef enum
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{
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ENCODER_MODE_WIDEBAND, /* 16 kHz */
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ENCODER_MODE_SUPER_WIDEBAND, /* 32 kHz */
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} EncoderMode;
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struct _GstIsacEnc
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{
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/*< private > */
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GstAudioEncoder parent;
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ISACStruct *isac;
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EncoderMode mode;
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gint samples_per_frame; /* number of samples in one input frame */
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gsize frame_size; /* size, in bytes, of one input frame */
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guint nb_processed_input_frames; /* number of input frames processed by the encoder since the last produced encoded data */
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/* properties */
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gint output_frame_len;
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gint bitrate;
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gint max_payload_size;
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gint max_rate;
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};
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#define gst_isacenc_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstIsacEnc, gst_isacenc,
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GST_TYPE_AUDIO_ENCODER,
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GST_DEBUG_CATEGORY_INIT (isacenc_debug, "isacenc", 0,
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"debug category for isacenc element"));
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GST_ELEMENT_REGISTER_DEFINE (isacenc, "isacenc", GST_RANK_PRIMARY,
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GST_TYPE_ISACENC);
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static gboolean
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gst_isacenc_start (GstAudioEncoder * enc)
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{
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GstIsacEnc *self = GST_ISACENC (enc);
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gint16 ret;
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g_assert (!self->isac);
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ret = WebRtcIsac_Create (&self->isac);
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CHECK_ISAC_RET (ret, Create);
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self->nb_processed_input_frames = 0;
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return TRUE;
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}
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static gboolean
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gst_isacenc_stop (GstAudioEncoder * enc)
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{
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GstIsacEnc *self = GST_ISACENC (enc);
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if (self->isac) {
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gint16 ret;
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ret = WebRtcIsac_Free (self->isac);
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CHECK_ISAC_RET (ret, Free);
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self->isac = 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_isacenc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
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{
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GstIsacEnc *self = GST_ISACENC (enc);
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GstCaps *input_caps, *output_caps;
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gint16 ret;
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gboolean result;
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switch (GST_AUDIO_INFO_RATE (info)) {
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case 16000:
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self->mode = ENCODER_MODE_WIDEBAND;
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break;
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case 32000:
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self->mode = ENCODER_MODE_SUPER_WIDEBAND;
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break;
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default:
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g_assert_not_reached ();
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return FALSE;
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}
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input_caps = gst_audio_info_to_caps (info);
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output_caps = gst_caps_new_simple ("audio/isac",
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"channels", G_TYPE_INT, GST_AUDIO_INFO_CHANNELS (info),
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"rate", G_TYPE_INT, GST_AUDIO_INFO_RATE (info), NULL);
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GST_DEBUG_OBJECT (self, "input caps: %" GST_PTR_FORMAT, input_caps);
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GST_DEBUG_OBJECT (self, "output caps: %" GST_PTR_FORMAT, output_caps);
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ret = WebRtcIsac_SetEncSampRate (self->isac, GST_AUDIO_INFO_RATE (info));
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CHECK_ISAC_RET (ret, SetEncSampleRate);
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/* TODO: add support for automatically adjusted bit rate and frame
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* length (codingMode = 0). */
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ret = WebRtcIsac_EncoderInit (self->isac, 1);
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CHECK_ISAC_RET (ret, EncoderInit);
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if (self->mode == ENCODER_MODE_SUPER_WIDEBAND && self->output_frame_len != 30) {
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GST_ERROR_OBJECT (self,
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"Only output-frame-len=30 is supported in super-wideband mode (32 kHz)");
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return FALSE;
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}
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if (self->mode == ENCODER_MODE_WIDEBAND && (self->bitrate < 10000
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|| self->bitrate > 32000)) {
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GST_ERROR_OBJECT (self,
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"bitrate range is 10000 to 32000 bps in wideband mode (16 kHz)");
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return FALSE;
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} else if (self->mode == ENCODER_MODE_SUPER_WIDEBAND && (self->bitrate < 10000
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|| self->bitrate > 56000)) {
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GST_ERROR_OBJECT (self,
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"bitrate range is 10000 to 56000 bps in super-wideband mode (32 kHz)");
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return FALSE;
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}
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ret = WebRtcIsac_Control (self->isac, self->bitrate, self->output_frame_len);
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CHECK_ISAC_RET (ret, Control);
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if (self->max_payload_size != GST_ISACENC_MAX_PAYLOAD_SIZE_DEFAULT) {
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GST_DEBUG_OBJECT (self, "set max payload size to %d bytes",
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self->max_payload_size);
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ret = WebRtcIsac_SetMaxPayloadSize (self->isac, self->max_payload_size);
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CHECK_ISAC_RET (ret, SetMaxPayloadSize);
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}
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if (self->max_rate != GST_ISACENC_MAX_RATE_DEFAULT) {
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GST_DEBUG_OBJECT (self, "set max rate to %d bits/sec", self->max_rate);
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ret = WebRtcIsac_SetMaxRate (self->isac, self->max_rate);
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CHECK_ISAC_RET (ret, SetMaxRate);
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}
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result = gst_audio_encoder_set_output_format (enc, output_caps);
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/* input size is 10ms */
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self->samples_per_frame = GST_AUDIO_INFO_RATE (info) / 100;
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self->frame_size = self->samples_per_frame * GST_AUDIO_INFO_BPS (info);
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GST_DEBUG_OBJECT (self, "input frame: %d samples, %" G_GSIZE_FORMAT " bytes",
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self->samples_per_frame, self->frame_size);
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gst_audio_encoder_set_frame_samples_min (enc, self->samples_per_frame);
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gst_audio_encoder_set_frame_samples_max (enc, self->samples_per_frame);
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gst_audio_encoder_set_hard_min (enc, TRUE);
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gst_caps_unref (input_caps);
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gst_caps_unref (output_caps);
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return result;
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}
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static GstFlowReturn
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gst_isacenc_handle_frame (GstAudioEncoder * enc, GstBuffer * input)
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{
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GstIsacEnc *self = GST_ISACENC (enc);
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GstMapInfo map_read;
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gint16 ret;
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GstFlowReturn flow_ret = GST_FLOW_ERROR;
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gsize offset = 0;
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/* Can't drain the encoder */
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if (!input)
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return GST_FLOW_OK;
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if (!gst_buffer_map (input, &map_read, GST_MAP_READ)) {
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GST_ELEMENT_ERROR (self, RESOURCE, READ, ("Failed to map input buffer"),
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(NULL));
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return GST_FLOW_ERROR;
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}
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GST_LOG_OBJECT (self, "Received %" G_GSIZE_FORMAT " bytes", map_read.size);
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while (offset + self->frame_size <= map_read.size) {
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GstBuffer *output;
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GstMapInfo map_write;
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output = gst_audio_encoder_allocate_output_buffer (enc, OUTPUT_BUFFER_SIZE);
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if (!gst_buffer_map (output, &map_write, GST_MAP_WRITE)) {
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GST_ELEMENT_ERROR (self, RESOURCE, WRITE, ("Failed to map output buffer"),
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(NULL));
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gst_buffer_unref (output);
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goto out;
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}
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ret =
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WebRtcIsac_Encode (self->isac,
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(const gint16 *) (map_read.data + offset), map_write.data);
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gst_buffer_unmap (output, &map_write);
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self->nb_processed_input_frames++;
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offset += self->frame_size;
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if (ret == 0) {
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/* buffering */
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gst_buffer_unref (output);
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continue;
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} else if (ret < 0) {
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/* error */
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gint16 code = WebRtcIsac_GetErrorCode (self->isac);
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GST_ELEMENT_ERROR (self, LIBRARY, ENCODE, ("Failed to encode frame"),
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("Failed to encode: %s (%d)", isac_error_code_to_str (code), code));
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gst_buffer_unref (output);
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goto out;
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} else {
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/* encoded */
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GST_LOG_OBJECT (self, "Encoded %d input frames to %d bytes",
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self->nb_processed_input_frames, ret);
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gst_buffer_set_size (output, ret);
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flow_ret =
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gst_audio_encoder_finish_frame (enc, output,
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self->nb_processed_input_frames * self->samples_per_frame);
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if (flow_ret != GST_FLOW_OK)
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goto out;
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self->nb_processed_input_frames = 0;
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}
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}
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flow_ret = GST_FLOW_OK;
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out:
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gst_buffer_unmap (input, &map_read);
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return flow_ret;
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}
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static void
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gst_isacenc_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstIsacEnc *self = GST_ISACENC (object);
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switch (prop_id) {
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case PROP_OUTPUT_FRAME_LEN:
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self->output_frame_len = g_value_get_enum (value);
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break;
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case PROP_BITRATE:
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self->bitrate = g_value_get_int (value);
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break;
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case PROP_MAX_PAYLOAD_SIZE:
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self->max_payload_size = g_value_get_int (value);
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break;
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case PROP_MAX_RATE:
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self->max_rate = g_value_get_int (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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}
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static void
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gst_isacenc_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstIsacEnc *self = GST_ISACENC (object);
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switch (prop_id) {
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case PROP_OUTPUT_FRAME_LEN:
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g_value_set_enum (value, self->output_frame_len);
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break;
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case PROP_BITRATE:
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g_value_set_int (value, self->bitrate);
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break;
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case PROP_MAX_PAYLOAD_SIZE:
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g_value_set_int (value, self->max_payload_size);
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break;
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case PROP_MAX_RATE:
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g_value_set_int (value, self->max_rate);
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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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}
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static void
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gst_isacenc_class_init (GstIsacEncClass * klass)
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{
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
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GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass);
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gobject_class->set_property = gst_isacenc_set_property;
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gobject_class->get_property = gst_isacenc_get_property;
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base_class->start = GST_DEBUG_FUNCPTR (gst_isacenc_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_isacenc_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_isacenc_set_format);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_isacenc_handle_frame);
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g_object_class_install_property (gobject_class, PROP_OUTPUT_FRAME_LEN,
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g_param_spec_enum ("output-frame-len", "Output Frame Length",
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"Length, in ms, of output frames",
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GST_TYPE_ISACENC_OUTPUT_FRAME_LEN,
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GST_ISACENC_OUTPUT_FRAME_LEN_DEFAULT,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
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GST_PARAM_MUTABLE_READY));
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g_object_class_install_property (gobject_class, PROP_BITRATE,
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g_param_spec_int ("bitrate", "Bitrate",
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"Average Bitrate (ABR) in bits/sec",
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10000, 56000,
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GST_ISACENC_BITRATE_DEFAULT,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
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GST_PARAM_MUTABLE_READY));
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g_object_class_install_property (gobject_class, PROP_MAX_PAYLOAD_SIZE,
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g_param_spec_int ("max-payload-size", "Max Payload Size",
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"Maximum payload size, in bytes. Range is 120 to 400 at 16 kHz "
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"and 120 to 600 at 32 kHz (-1 = encoder default)",
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-1, 600,
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GST_ISACENC_MAX_PAYLOAD_SIZE_DEFAULT,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
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GST_PARAM_MUTABLE_READY));
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g_object_class_install_property (gobject_class, PROP_MAX_RATE,
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g_param_spec_int ("max-rate", "Max Rate",
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"Maximum rate, in bits/sec, which the codec may not exceed for any "
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"signal packet. Range is 32000 to 53400 at 16 kHz "
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"and 32000 to 160000 at 32 kHz (-1 = encoder default)",
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-1, 160000,
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GST_ISACENC_MAX_PAYLOAD_SIZE_DEFAULT,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
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GST_PARAM_MUTABLE_READY));
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gst_element_class_set_static_metadata (gstelement_class, "iSAC encoder",
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"Codec/Encoder/Audio",
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"iSAC audio encoder",
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"Guillaume Desmottes <guillaume.desmottes@collabora.com>");
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gst_element_class_add_static_pad_template (gstelement_class, &sink_template);
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gst_element_class_add_static_pad_template (gstelement_class, &src_template);
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}
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static void
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gst_isacenc_init (GstIsacEnc * self)
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{
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self->output_frame_len = GST_ISACENC_OUTPUT_FRAME_LEN_DEFAULT;
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self->bitrate = GST_ISACENC_BITRATE_DEFAULT;
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self->max_payload_size = GST_ISACENC_MAX_PAYLOAD_SIZE_DEFAULT;
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self->max_rate = GST_ISACENC_MAX_RATE_DEFAULT;
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}
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