isac: add iSAC plugin

Wrapper on the iSAC reference encoder and decoder from webrtc,
see https://en.wikipedia.org/wiki/Internet_Speech_Audio_Codec

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1124>
This commit is contained in:
Guillaume Desmottes 2020-03-19 15:07:47 +01:00
parent d1945de102
commit bfb9071081
10 changed files with 1001 additions and 0 deletions

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/* iSAC plugin
*
* 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.
*/
/**
* plugin-isac:
*
* Since: 1.20
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <string.h>
#include <gst/gst.h>
#include "gstisacenc.h"
#include "gstisacdec.h"
static gboolean
plugin_init (GstPlugin * plugin)
{
if (!gst_element_register (plugin, "isacenc", GST_RANK_PRIMARY,
GST_TYPE_ISACENC))
return FALSE;
if (!gst_element_register (plugin, "isacdec", GST_RANK_PRIMARY,
GST_TYPE_ISACDEC))
return FALSE;
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
isac,
"iSAC plugin", plugin_init, VERSION, "LGPL", PACKAGE_NAME,
GST_PACKAGE_ORIGIN)

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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"));
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;
}

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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.
*/
#ifndef __GST_ISAC_DEC_H__
#define __GST_ISAC_DEC_H__
#include <gst/audio/audio.h>
G_BEGIN_DECLS
#define GST_TYPE_ISACDEC gst_isacdec_get_type ()
G_DECLARE_FINAL_TYPE(GstIsacDec, gst_isacdec, GST, ISACDEC, GstAudioDecoder)
G_END_DECLS
#endif /* __GST_ISAC_DEC_H__ */

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

34
ext/isac/gstisacenc.h Normal file
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@ -0,0 +1,34 @@
/* iSAC encoder
*
* 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.
*/
#ifndef __GST_ISAC_ENC_H__
#define __GST_ISAC_ENC_H__
#include <gst/audio/audio.h>
G_BEGIN_DECLS
#define GST_TYPE_ISACENC gst_isacenc_get_type ()
G_DECLARE_FINAL_TYPE(GstIsacEnc, gst_isacenc, GST, ISACENC, GstAudioEncoder)
G_END_DECLS
#endif /* __GST_ISAC_ENC_H__ */

85
ext/isac/gstisacutils.c Normal file
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@ -0,0 +1,85 @@
/* iSAC plugin utils
*
* 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.
*/
#include "gstisacutils.h"
#include <modules/audio_coding/codecs/isac/main/source/settings.h>
const gchar *
isac_error_code_to_str (gint code)
{
switch (code) {
case ISAC_MEMORY_ALLOCATION_FAILED:
return "allocation failed";
case ISAC_MODE_MISMATCH:
return "mode mismatch";
case ISAC_DISALLOWED_BOTTLENECK:
return "disallowed bottleneck";
case ISAC_DISALLOWED_FRAME_LENGTH:
return "disallowed frame length";
case ISAC_UNSUPPORTED_SAMPLING_FREQUENCY:
return "unsupported sampling frequency";
case ISAC_RANGE_ERROR_BW_ESTIMATOR:
return "range error bandwitch estimator";
case ISAC_ENCODER_NOT_INITIATED:
return "encoder not initiated";
case ISAC_DISALLOWED_CODING_MODE:
return "disallowed coding mode";
case ISAC_DISALLOWED_FRAME_MODE_ENCODER:
return "disallowed frame mode encoder";
case ISAC_DISALLOWED_BITSTREAM_LENGTH:
return "disallowed bitstream length";
case ISAC_PAYLOAD_LARGER_THAN_LIMIT:
return "payload larger than limit";
case ISAC_DISALLOWED_ENCODER_BANDWIDTH:
return "disallowed encoder bandwith";
case ISAC_DECODER_NOT_INITIATED:
return "decoder not initiated";
case ISAC_EMPTY_PACKET:
return "empty packet";
case ISAC_DISALLOWED_FRAME_MODE_DECODER:
return "disallowed frame mode decoder";
case ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH:
return "range error decode frame length";
case ISAC_RANGE_ERROR_DECODE_BANDWIDTH:
return "range error decode bandwith";
case ISAC_RANGE_ERROR_DECODE_PITCH_GAIN:
return "range error decode pitch gain";
case ISAC_RANGE_ERROR_DECODE_PITCH_LAG:
return "range error decode pitch lag";
case ISAC_RANGE_ERROR_DECODE_LPC:
return "range error decode lpc";
case ISAC_RANGE_ERROR_DECODE_SPECTRUM:
return "range error decode spectrum";
case ISAC_LENGTH_MISMATCH:
return "length mismatch";
case ISAC_RANGE_ERROR_DECODE_BANDWITH:
return "range error decode bandwith";
case ISAC_DISALLOWED_BANDWIDTH_MODE_DECODER:
return "disallowed bandwitch mode decoder";
case ISAC_DISALLOWED_LPC_MODEL:
return "disallowed lpc model";
case ISAC_INCOMPATIBLE_FORMATS:
return "incompatible formats";
}
return "<unknown>";
}

40
ext/isac/gstisacutils.h Normal file
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@ -0,0 +1,40 @@
/* iSAC plugin utils
*
* 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.
*/
#ifndef __GST_ISAC_UTILS_H__
#define __GST_ISAC_UTILS_H__
#include <glib.h>
G_BEGIN_DECLS
const gchar * isac_error_code_to_str (gint code);
#define CHECK_ISAC_RET(ret, function) \
if (ret == -1) {\
gint16 code = WebRtcIsac_GetErrorCode (self->isac);\
GST_WARNING_OBJECT (self, "WebRtcIsac_"#function " call failed: %s (%d)", isac_error_code_to_str (code), code);\
return FALSE;\
}
G_END_DECLS
#endif /* __GST_ISAC_UTILS_H__ */

20
ext/isac/meson.build Normal file
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@ -0,0 +1,20 @@
webrtc_audio_coding_dep = dependency('webrtc-audio-coding-1', required: get_option('isac'))
if webrtc_audio_coding_dep.found()
isac_sources = [
'gstisac.c',
'gstisacenc.c',
'gstisacdec.c',
'gstisacutils.c',
]
gstisac = library('gstisac', isac_sources,
c_args : gst_plugins_bad_args,
include_directories : [configinc],
dependencies : [gstaudio_dep, webrtc_audio_coding_dep],
install : true,
install_dir : plugins_install_dir,
)
pkgconfig.generate(gstisac, install_dir : plugins_pkgconfig_install_dir)
plugins += [gstisac]
endif

View file

@ -21,6 +21,7 @@ subdir('gme')
subdir('gsm') subdir('gsm')
subdir('hls') subdir('hls')
subdir('iqa') subdir('iqa')
subdir('isac')
subdir('kate') subdir('kate')
subdir('ladspa') subdir('ladspa')
subdir('libde265') subdir('libde265')

View file

@ -169,6 +169,7 @@ option('zxing', type : 'feature', value : 'auto', description : 'Barcode image s
option('wpe', type : 'feature', value : 'auto', description : 'WPE Web browser plugin') option('wpe', type : 'feature', value : 'auto', description : 'WPE Web browser plugin')
option('magicleap', type : 'feature', value : 'auto', description : 'Magic Leap platform support') option('magicleap', type : 'feature', value : 'auto', description : 'Magic Leap platform support')
option('v4l2codecs', type : 'feature', value : 'auto', description : 'Video4Linux Stateless CODECs support') option('v4l2codecs', type : 'feature', value : 'auto', description : 'Video4Linux Stateless CODECs support')
option('isac', type : 'feature', value : 'auto', description : 'iSAC plugin')
# HLS plugin options # HLS plugin options
option('hls', type : 'feature', value : 'auto', description : 'HTTP Live Streaming plugin') option('hls', type : 'feature', value : 'auto', description : 'HTTP Live Streaming plugin')