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gstreamer/ext/fdkaac/gstfdkaacenc.c

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/*
* Copyright (C) 2016 Sebastian Dröge <sebastian@centricular.com>
*
* 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 St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstfdkaacenc.h"
#include <gst/pbutils/pbutils.h>
#include <string.h>
/* TODO:
* - Add support for other AOT / profiles
* - Expose more properties, e.g. afterburner and vbr
* - Signal encoder delay
* - LOAS / LATM support
*/
enum
{
PROP_0,
PROP_BITRATE
};
#define DEFAULT_BITRATE (0)
#define SAMPLE_RATES " 8000, " \
"11025, " \
"12000, " \
"16000, " \
"22050, " \
"24000, " \
"32000, " \
"44100, " \
"48000, " \
"64000, " \
"88200, " \
"96000"
static const struct
{
gint channels;
CHANNEL_MODE mode;
GstAudioChannelPosition positions[8];
} channel_layouts[] = {
{
1, MODE_1, {
GST_AUDIO_CHANNEL_POSITION_MONO}}, {
2, MODE_2, {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}}, {
3, MODE_1_2, {
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}}, {
3, MODE_2_1, {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_LFE1}}, {
4, MODE_1_2_1, {
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_CENTER}}, {
5, MODE_1_2_2, {
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}}, {
6, MODE_1_2_2_1, {
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
GST_AUDIO_CHANNEL_POSITION_LFE1}}, {
8, MODE_7_1_REAR_SURROUND, {
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_LFE1}}, {
8, MODE_7_1_FRONT_CENTER, {
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
GST_AUDIO_CHANNEL_POSITION_LFE1}}
};
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) ", "
"layout = (string) interleaved, "
"rate = (int) { " SAMPLE_RATES " }, "
"channels = (int) {1, 2, 3, 4, 5, 6, 8}")
);
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/mpeg, "
"mpegversion = (int) {2, 4}, "
"rate = (int) { " SAMPLE_RATES " }, "
"channels = (int) {1, 2, 3, 4, 5, 6, 8}, "
"stream-format = (string) { adts, adif, raw }, "
"base-profile = (string) lc")
);
GST_DEBUG_CATEGORY_STATIC (gst_fdkaacenc_debug);
#define GST_CAT_DEFAULT gst_fdkaacenc_debug
static void gst_fdkaacenc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_fdkaacenc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_fdkaacenc_start (GstAudioEncoder * enc);
static gboolean gst_fdkaacenc_stop (GstAudioEncoder * enc);
static gboolean gst_fdkaacenc_set_format (GstAudioEncoder * enc,
GstAudioInfo * info);
static GstFlowReturn gst_fdkaacenc_handle_frame (GstAudioEncoder * enc,
GstBuffer * in_buf);
static GstCaps *gst_fdkaacenc_get_caps (GstAudioEncoder * enc,
GstCaps * filter);
G_DEFINE_TYPE (GstFdkAacEnc, gst_fdkaacenc, GST_TYPE_AUDIO_ENCODER);
static void
gst_fdkaacenc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstFdkAacEnc *self = GST_FDKAACENC (object);
switch (prop_id) {
case PROP_BITRATE:
self->bitrate = g_value_get_int (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
return;
}
static void
gst_fdkaacenc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstFdkAacEnc *self = GST_FDKAACENC (object);
switch (prop_id) {
case PROP_BITRATE:
g_value_set_int (value, self->bitrate);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
return;
}
static gboolean
gst_fdkaacenc_start (GstAudioEncoder * enc)
{
GstFdkAacEnc *self = GST_FDKAACENC (enc);
GST_DEBUG_OBJECT (self, "start");
return TRUE;
}
static gboolean
gst_fdkaacenc_stop (GstAudioEncoder * enc)
{
GstFdkAacEnc *self = GST_FDKAACENC (enc);
GST_DEBUG_OBJECT (self, "stop");
if (self->enc)
aacEncClose (&self->enc);
return TRUE;
}
static GstCaps *
gst_fdkaacenc_get_caps (GstAudioEncoder * enc, GstCaps * filter)
{
GstCaps *res, *caps;
gint i;
caps = gst_caps_new_empty ();
for (i = 0; i < G_N_ELEMENTS (channel_layouts); i++) {
guint64 channel_mask;
GstCaps *tmp =
gst_caps_make_writable (gst_pad_get_pad_template_caps
(GST_AUDIO_ENCODER_SINK_PAD (enc)));
if (channel_layouts[i].channels == 1) {
gst_caps_set_simple (tmp, "channels", G_TYPE_INT,
channel_layouts[i].channels, NULL);
} else {
gst_audio_channel_positions_to_mask (channel_layouts[i].positions,
channel_layouts[i].channels, FALSE, &channel_mask);
gst_caps_set_simple (tmp, "channels", G_TYPE_INT,
channel_layouts[i].channels, "channel-mask", GST_TYPE_BITMASK,
channel_mask, NULL);
}
gst_caps_append (caps, tmp);
}
res = gst_audio_encoder_proxy_getcaps (enc, caps, filter);
gst_caps_unref (caps);
return res;
}
static gboolean
gst_fdkaacenc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
{
GstFdkAacEnc *self = GST_FDKAACENC (enc);
gboolean ret = FALSE;
GstCaps *allowed_caps;
GstCaps *src_caps;
AACENC_ERROR err;
gint transmux = 0, aot = AOT_AAC_LC;
gint mpegversion = 4;
CHANNEL_MODE channel_mode;
AACENC_InfoStruct enc_info = { 0 };
gint bitrate;
if (self->enc) {
/* drain */
gst_fdkaacenc_handle_frame (enc, NULL);
aacEncClose (&self->enc);
}
allowed_caps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (self));
GST_DEBUG_OBJECT (self, "allowed caps: %" GST_PTR_FORMAT, allowed_caps);
if (allowed_caps && gst_caps_get_size (allowed_caps) > 0) {
GstStructure *s = gst_caps_get_structure (allowed_caps, 0);
const gchar *str = NULL;
if ((str = gst_structure_get_string (s, "stream-format"))) {
if (strcmp (str, "adts") == 0) {
GST_DEBUG_OBJECT (self, "use ADTS format for output");
transmux = 2;
} else if (strcmp (str, "adif") == 0) {
GST_DEBUG_OBJECT (self, "use ADIF format for output");
transmux = 1;
} else if (strcmp (str, "raw") == 0) {
GST_DEBUG_OBJECT (self, "use RAW format for output");
transmux = 0;
}
}
gst_structure_get_int (s, "mpegversion", &mpegversion);
}
if (allowed_caps)
gst_caps_unref (allowed_caps);
if ((err =
aacEncOpen (&self->enc, 0,
GST_AUDIO_INFO_CHANNELS (info))) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to open encoder: %d\n", err);
return FALSE;
}
if (mpegversion == 4) {
aot = AOT_AAC_LC;
} else {
aot = AOT_MP2_AAC_LC;
}
if ((err = aacEncoder_SetParam (self->enc, AACENC_AOT, aot)) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to set AOT %d: %d\n", aot, err);
return FALSE;
}
if ((err = aacEncoder_SetParam (self->enc, AACENC_SAMPLERATE,
GST_AUDIO_INFO_RATE (info))) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to set sample rate %d: %d\n",
GST_AUDIO_INFO_RATE (info), err);
return FALSE;
}
if (GST_AUDIO_INFO_CHANNELS (info) == 1) {
channel_mode = MODE_1;
self->need_reorder = FALSE;
self->aac_positions = NULL;
} else {
guint64 in_channel_mask, out_channel_mask;
gint i;
for (i = 0; i < G_N_ELEMENTS (channel_layouts); i++) {
if (channel_layouts[i].channels != GST_AUDIO_INFO_CHANNELS (info))
continue;
gst_audio_channel_positions_to_mask (&GST_AUDIO_INFO_POSITION (info, 0),
GST_AUDIO_INFO_CHANNELS (info), FALSE, &in_channel_mask);
gst_audio_channel_positions_to_mask (channel_layouts[i].positions,
channel_layouts[i].channels, FALSE, &out_channel_mask);
if (in_channel_mask == out_channel_mask) {
channel_mode = channel_layouts[i].mode;
self->need_reorder =
memcmp (channel_layouts[i].positions,
&GST_AUDIO_INFO_POSITION (info, 0),
GST_AUDIO_INFO_CHANNELS (info) *
sizeof (GstAudioChannelPosition)) != 0;
self->aac_positions = channel_layouts[i].positions;
break;
}
}
if (i == G_N_ELEMENTS (channel_layouts)) {
GST_ERROR_OBJECT (self, "Couldn't find a valid channel layout");
return FALSE;
}
}
if ((err = aacEncoder_SetParam (self->enc, AACENC_CHANNELMODE,
channel_mode)) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to set channel mode %d: %d", channel_mode,
err);
return FALSE;
}
/* MPEG channel order */
if ((err = aacEncoder_SetParam (self->enc, AACENC_CHANNELORDER,
0)) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to set channel order %d: %d", channel_mode,
err);
return FALSE;
}
bitrate = self->bitrate;
/* See
* http://wiki.hydrogenaud.io/index.php?title=Fraunhofer_FDK_AAC#Recommended_Sampling_Rate_and_Bitrate_Combinations
*/
if (bitrate == 0) {
if (GST_AUDIO_INFO_CHANNELS (info) == 1) {
if (GST_AUDIO_INFO_RATE (info) < 16000) {
bitrate = 8000;
} else if (GST_AUDIO_INFO_RATE (info) == 16000) {
bitrate = 16000;
} else if (GST_AUDIO_INFO_RATE (info) < 32000) {
bitrate = 24000;
} else if (GST_AUDIO_INFO_RATE (info) == 32000) {
bitrate = 32000;
} else if (GST_AUDIO_INFO_RATE (info) <= 44100) {
bitrate = 56000;
} else {
bitrate = 160000;
}
} else if (GST_AUDIO_INFO_CHANNELS (info) == 2) {
if (GST_AUDIO_INFO_RATE (info) < 16000) {
bitrate = 16000;
} else if (GST_AUDIO_INFO_RATE (info) == 16000) {
bitrate = 24000;
} else if (GST_AUDIO_INFO_RATE (info) < 22050) {
bitrate = 32000;
} else if (GST_AUDIO_INFO_RATE (info) < 32000) {
bitrate = 40000;
} else if (GST_AUDIO_INFO_RATE (info) == 32000) {
bitrate = 96000;
} else if (GST_AUDIO_INFO_RATE (info) <= 44100) {
bitrate = 112000;
} else {
bitrate = 320000;
}
} else {
/* 5, 5.1 */
if (GST_AUDIO_INFO_RATE (info) < 32000) {
bitrate = 160000;
} else if (GST_AUDIO_INFO_RATE (info) <= 44100) {
bitrate = 240000;
} else {
bitrate = 320000;
}
}
}
if ((err = aacEncoder_SetParam (self->enc, AACENC_BITRATE,
bitrate)) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to set bitrate %d: %d", bitrate, err);
return FALSE;
}
if ((err = aacEncEncode (self->enc, NULL, NULL, NULL, NULL)) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to initialize encoder: %d", err);
return FALSE;
}
if ((err = aacEncInfo (self->enc, &enc_info)) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to get encoder info: %d", err);
return FALSE;
}
gst_audio_encoder_set_frame_max (enc, 1);
gst_audio_encoder_set_frame_samples_min (enc, enc_info.frameLength);
gst_audio_encoder_set_frame_samples_max (enc, enc_info.frameLength);
gst_audio_encoder_set_hard_min (enc, FALSE);
self->outbuf_size = enc_info.maxOutBufBytes;
self->samples_per_frame = enc_info.frameLength;
src_caps = gst_caps_new_simple ("audio/mpeg",
"mpegversion", G_TYPE_INT, mpegversion,
"channels", G_TYPE_INT, GST_AUDIO_INFO_CHANNELS (info),
"rate", G_TYPE_INT, GST_AUDIO_INFO_RATE (info), NULL);
/* raw */
if (transmux == 0) {
GstBuffer *codec_data =
gst_buffer_new_wrapped (g_memdup (enc_info.confBuf, enc_info.confSize),
enc_info.confSize);
gst_caps_set_simple (src_caps, "codec_data", GST_TYPE_BUFFER, codec_data,
"stream-format", G_TYPE_STRING, "raw", NULL);
gst_buffer_unref (codec_data);
} else if (transmux == 1) {
gst_caps_set_simple (src_caps, "stream-format", G_TYPE_STRING, "adif",
NULL);
} else if (transmux == 2) {
gst_caps_set_simple (src_caps, "stream-format", G_TYPE_STRING, "adts",
NULL);
} else {
g_assert_not_reached ();
}
gst_codec_utils_aac_caps_set_level_and_profile (src_caps, enc_info.confBuf,
enc_info.confSize);
ret = gst_audio_encoder_set_output_format (enc, src_caps);
gst_caps_unref (src_caps);
return ret;
}
static GstFlowReturn
gst_fdkaacenc_handle_frame (GstAudioEncoder * enc, GstBuffer * inbuf)
{
GstFdkAacEnc *self = GST_FDKAACENC (enc);
GstFlowReturn ret = GST_FLOW_OK;
GstAudioInfo *info;
GstMapInfo imap, omap;
GstBuffer *outbuf;
AACENC_BufDesc in_desc = { 0 };
AACENC_BufDesc out_desc = { 0 };
AACENC_InArgs in_args = { 0 };
AACENC_OutArgs out_args = { 0 };
gint in_id = IN_AUDIO_DATA, out_id = OUT_BITSTREAM_DATA;
gint in_sizes, out_sizes;
gint in_el_sizes, out_el_sizes;
AACENC_ERROR err;
info = gst_audio_encoder_get_audio_info (enc);
if (!inbuf) {
in_args.numInSamples = -1;
} else {
if (self->need_reorder) {
inbuf = gst_buffer_copy (inbuf);
gst_buffer_map (inbuf, &imap, GST_MAP_READWRITE);
gst_audio_reorder_channels (imap.data, imap.size,
GST_AUDIO_INFO_FORMAT (info), GST_AUDIO_INFO_CHANNELS (info),
&GST_AUDIO_INFO_POSITION (info, 0), self->aac_positions);
} else {
gst_buffer_map (inbuf, &imap, GST_MAP_READ);
}
in_args.numInSamples = imap.size / GST_AUDIO_INFO_BPS (info);
in_sizes = imap.size;
in_el_sizes = 2;
in_desc.bufferIdentifiers = &in_id;
in_desc.numBufs = 1;
in_desc.bufs = (void *) &imap.data;
in_desc.bufSizes = &in_sizes;
in_desc.bufElSizes = &in_el_sizes;
}
outbuf = gst_audio_encoder_allocate_output_buffer (enc, self->outbuf_size);
if (!outbuf) {
ret = GST_FLOW_ERROR;
goto out;
}
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
out_sizes = omap.size;
out_el_sizes = 1;
out_desc.bufferIdentifiers = &out_id;
out_desc.numBufs = 1;
out_desc.bufs = (void *) &omap.data;
out_desc.bufSizes = &out_sizes;
out_desc.bufElSizes = &out_el_sizes;
if ((err = aacEncEncode (self->enc, &in_desc, &out_desc, &in_args,
&out_args)) != AACENC_OK) {
if (!inbuf && err == AACENC_ENCODE_EOF)
goto out;
GST_ERROR_OBJECT (self, "Failed to encode data: %d", err);
ret = GST_FLOW_ERROR;
goto out;
}
if (inbuf) {
gst_buffer_unmap (inbuf, &imap);
if (self->need_reorder)
gst_buffer_unref (inbuf);
inbuf = NULL;
}
if (!out_args.numOutBytes)
goto out;
gst_buffer_unmap (outbuf, &omap);
gst_buffer_set_size (outbuf, out_args.numOutBytes);
ret = gst_audio_encoder_finish_frame (enc, outbuf, self->samples_per_frame);
outbuf = NULL;
out:
if (outbuf) {
gst_buffer_unmap (outbuf, &omap);
gst_buffer_unref (outbuf);
}
if (inbuf) {
gst_buffer_unmap (inbuf, &imap);
if (self->need_reorder)
gst_buffer_unref (inbuf);
}
return ret;
}
static void
gst_fdkaacenc_init (GstFdkAacEnc * self)
{
self->bitrate = DEFAULT_BITRATE;
self->enc = NULL;
gst_audio_encoder_set_drainable (GST_AUDIO_ENCODER (self), TRUE);
}
static void
gst_fdkaacenc_class_init (GstFdkAacEncClass * klass)
{
GObjectClass *object_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass);
object_class->set_property = GST_DEBUG_FUNCPTR (gst_fdkaacenc_set_property);
object_class->get_property = GST_DEBUG_FUNCPTR (gst_fdkaacenc_get_property);
base_class->start = GST_DEBUG_FUNCPTR (gst_fdkaacenc_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_fdkaacenc_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_fdkaacenc_set_format);
base_class->getcaps = GST_DEBUG_FUNCPTR (gst_fdkaacenc_get_caps);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_fdkaacenc_handle_frame);
g_object_class_install_property (object_class, PROP_BITRATE,
g_param_spec_int ("bitrate",
"Bitrate",
"Target Audio Bitrate (0 = fixed value based on "
" sample rate and channel count)",
0, G_MAXINT, DEFAULT_BITRATE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_add_static_pad_template (element_class, &sink_template);
gst_element_class_add_static_pad_template (element_class, &src_template);
gst_element_class_set_static_metadata (element_class, "FDK AAC audio encoder",
"Codec/Encoder/Audio", "FDK AAC audio encoder",
"Sebastian Dröge <sebastian@centricular.com>");
GST_DEBUG_CATEGORY_INIT (gst_fdkaacenc_debug, "fdkaacenc", 0,
"fdkaac encoder");
}
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