gstreamer/ext/isac/gstisacdec.c

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/* iSAC decoder
*
* Copyright (C) 2020 Collabora Ltd.
* Author: Guillaume Desmottes <guillaume.desmottes@collabora.com>, Collabora Ltd.
*
* 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 Street, Fifth Floor,
* Boston, MA 02110-1301 USA.
*/
/**
* SECTION:element-isacdec
* @title: isacdec
* @short_description: iSAC audio decoder
*
* Since: 1.20
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstisacdec.h"
#include "gstisacutils.h"
#include <modules/audio_coding/codecs/isac/main/include/isac.h>
GST_DEBUG_CATEGORY_STATIC (isacdec_debug);
#define GST_CAT_DEFAULT isacdec_debug
#define SAMPLE_SIZE 2 /* 16-bits samples */
#define MAX_OUTPUT_SAMPLES 960 /* decoder produces max 960 samples */
#define MAX_OUTPUT_SIZE (SAMPLE_SIZE * MAX_OUTPUT_SAMPLES)
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/isac, "
"rate = (int) { 16000, 32000 }, " "channels = (int) 1")
);
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_NE (S16) ", "
"rate = (int) { 16000, 32000 }, "
"layout = (string) interleaved, " "channels = (int) 1")
);
struct _GstIsacDec
{
/*< private > */
GstAudioDecoder parent;
ISACStruct *isac;
/* properties */
};
#define gst_isacdec_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstIsacDec, gst_isacdec,
GST_TYPE_AUDIO_DECODER,
GST_DEBUG_CATEGORY_INIT (isacdec_debug, "isacdec", 0,
"debug category for isacdec element"));
GST_ELEMENT_REGISTER_DEFINE (isacdec, "isacdec", GST_RANK_PRIMARY,
GST_TYPE_ISACDEC);
static gboolean
gst_isacdec_start (GstAudioDecoder * dec)
{
GstIsacDec *self = GST_ISACDEC (dec);
gint16 ret;
g_assert (!self->isac);
ret = WebRtcIsac_Create (&self->isac);
CHECK_ISAC_RET (ret, Create);
return TRUE;
}
static gboolean
gst_isacdec_stop (GstAudioDecoder * dec)
{
GstIsacDec *self = GST_ISACDEC (dec);
if (self->isac) {
gint16 ret;
ret = WebRtcIsac_Free (self->isac);
CHECK_ISAC_RET (ret, Free);
self->isac = NULL;
}
return TRUE;
}
static gboolean
gst_isacdec_set_format (GstAudioDecoder * dec, GstCaps * input_caps)
{
GstIsacDec *self = GST_ISACDEC (dec);
GstAudioInfo output_format;
gint16 ret;
gboolean result;
GstStructure *s;
gint rate, channels;
GstCaps *output_caps;
GST_DEBUG_OBJECT (self, "input caps: %" GST_PTR_FORMAT, input_caps);
s = gst_caps_get_structure (input_caps, 0);
if (!s)
return FALSE;
if (!gst_structure_get_int (s, "rate", &rate)) {
GST_ERROR_OBJECT (self, "'rate' missing in input caps: %" GST_PTR_FORMAT,
input_caps);
return FALSE;
}
if (!gst_structure_get_int (s, "channels", &channels)) {
GST_ERROR_OBJECT (self,
"'channels' missing in input caps: %" GST_PTR_FORMAT, input_caps);
return FALSE;
}
gst_audio_info_set_format (&output_format, GST_AUDIO_FORMAT_S16LE, rate,
channels, NULL);
output_caps = gst_audio_info_to_caps (&output_format);
GST_DEBUG_OBJECT (self, "output caps: %" GST_PTR_FORMAT, output_caps);
gst_caps_unref (output_caps);
ret = WebRtcIsac_SetDecSampRate (self->isac, rate);
CHECK_ISAC_RET (ret, SetDecSampleRate);
WebRtcIsac_DecoderInit (self->isac);
result = gst_audio_decoder_set_output_format (dec, &output_format);
gst_audio_decoder_set_plc_aware (dec, TRUE);
return result;
}
static GstFlowReturn
gst_isacdec_plc (GstIsacDec * self, GstClockTime duration)
{
GstAudioDecoder *dec = GST_AUDIO_DECODER (self);
guint nb_plc_frames;
GstBuffer *output;
GstMapInfo map_write;
size_t ret;
/* Decoder produces 30 ms PLC frames */
nb_plc_frames = duration / (30 * GST_MSECOND);
GST_DEBUG_OBJECT (self,
"GAP of %" GST_TIME_FORMAT " detected, request PLC for %d frames",
GST_TIME_ARGS (duration), nb_plc_frames);
output =
gst_audio_decoder_allocate_output_buffer (dec,
nb_plc_frames * MAX_OUTPUT_SIZE);
if (!gst_buffer_map (output, &map_write, GST_MAP_WRITE)) {
GST_ERROR_OBJECT (self, "Failed to map output buffer");
gst_buffer_unref (output);
return GST_FLOW_ERROR;
}
ret =
WebRtcIsac_DecodePlc (self->isac, (gint16 *) map_write.data,
nb_plc_frames);
gst_buffer_unmap (output, &map_write);
if (ret < 0) {
/* error */
gint16 code = WebRtcIsac_GetErrorCode (self->isac);
GST_WARNING_OBJECT (self, "Failed to produce PLC: %s (%d)",
isac_error_code_to_str (code), code);
gst_buffer_unref (output);
return GST_FLOW_ERROR;
} else if (ret == 0) {
GST_DEBUG_OBJECT (self, "Decoder didn't produce any PLC frame");
gst_buffer_unref (output);
return GST_FLOW_OK;
}
gst_buffer_set_size (output, ret * SAMPLE_SIZE);
GST_LOG_OBJECT (self, "Produced %" G_GSIZE_FORMAT " PLC samples", ret);
return gst_audio_decoder_finish_frame (dec, output, 1);
}
static GstFlowReturn
gst_isacdec_handle_frame (GstAudioDecoder * dec, GstBuffer * input)
{
GstIsacDec *self = GST_ISACDEC (dec);
GstMapInfo map_read, map_write;
GstBuffer *output;
gint16 ret, speech_type[1];
gsize input_size;
/* Can't drain the decoder */
if (!input)
return GST_FLOW_OK;
if (!gst_buffer_get_size (input)) {
/* Base class detected a gap in the stream, try to do PLC */
return gst_isacdec_plc (self, GST_BUFFER_DURATION (input));
}
if (!gst_buffer_map (input, &map_read, GST_MAP_READ)) {
GST_ELEMENT_ERROR (self, RESOURCE, READ, ("Failed to map input buffer"),
(NULL));
return GST_FLOW_ERROR;
}
input_size = map_read.size;
output = gst_audio_decoder_allocate_output_buffer (dec, MAX_OUTPUT_SIZE);
if (!gst_buffer_map (output, &map_write, GST_MAP_WRITE)) {
GST_ELEMENT_ERROR (self, RESOURCE, WRITE, ("Failed to map output buffer"),
(NULL));
gst_buffer_unref (output);
gst_buffer_unmap (input, &map_read);
return GST_FLOW_ERROR;
}
ret = WebRtcIsac_Decode (self->isac, map_read.data, map_read.size,
(gint16 *) map_write.data, speech_type);
gst_buffer_unmap (input, &map_read);
gst_buffer_unmap (output, &map_write);
if (ret < 0) {
/* error */
gint16 code = WebRtcIsac_GetErrorCode (self->isac);
GST_WARNING_OBJECT (self, "Failed to decode: %s (%d)",
isac_error_code_to_str (code), code);
gst_buffer_unref (output);
/* Give a chance to decode next frames */
return GST_FLOW_OK;
} else if (ret == 0) {
GST_DEBUG_OBJECT (self, "Decoder didn't produce any frame");
gst_buffer_unref (output);
output = NULL;
} else {
gst_buffer_set_size (output, ret * SAMPLE_SIZE);
}
GST_LOG_OBJECT (self, "Decoded %d samples from %" G_GSIZE_FORMAT " bytes",
ret, input_size);
return gst_audio_decoder_finish_frame (dec, output, 1);
}
static void
gst_isacdec_class_init (GstIsacDecClass * klass)
{
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
base_class->start = GST_DEBUG_FUNCPTR (gst_isacdec_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_isacdec_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_isacdec_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_isacdec_handle_frame);
gst_element_class_set_static_metadata (gstelement_class, "iSAC decoder",
"Codec/Decoder/Audio",
"iSAC audio decoder",
"Guillaume Desmottes <guillaume.desmottes@collabora.com>");
gst_element_class_add_static_pad_template (gstelement_class, &sink_template);
gst_element_class_add_static_pad_template (gstelement_class, &src_template);
}
static void
gst_isacdec_init (GstIsacDec * self)
{
self->isac = NULL;
}