gstreamer/subprojects/gst-plugins-bad/ext/isac/gstisacenc.c

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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"));
GST_ELEMENT_REGISTER_DEFINE (isacenc, "isacenc", GST_RANK_PRIMARY,
GST_TYPE_ISACENC);
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;
}