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>

ext/isac/gstisac.c

@@ -0,0 +1,56 @@
+/* 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)

ext/isac/gstisacdec.c

@@ -0,0 +1,295 @@
+/* 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;
+}

ext/isac/gstisacdec.h

@@ -0,0 +1,34 @@
+/* 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__ */

ext/isac/gstisacenc.c

@@ -0,0 +1,435 @@
+/* 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;
+}

ext/isac/gstisacenc.h

@@ -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__ */

ext/isac/gstisacutils.c

@@ -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>";
+}

ext/isac/gstisacutils.h

@@ -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__ */

ext/isac/meson.build

@@ -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

ext/meson.build

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

meson_options.txt

@@ -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('magicleap', type : 'feature', value : 'auto', description : 'Magic Leap platform support')
 option('v4l2codecs', type : 'feature', value : 'auto', description : 'Video4Linux Stateless CODECs support')
+option('isac', type : 'feature', value : 'auto', description : 'iSAC plugin')
 
 # HLS plugin options
 option('hls', type : 'feature', value : 'auto', description : 'HTTP Live Streaming plugin')