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gstreamer/subprojects/gst-plugins-bad/ext/fdkaac/gstfdkaacenc.c
Edward Hervey 8ca2a2a230 fdkaacenc: Properly terminate GEnumValue table 
It should be terminated with a NULL entry, otherwise we just stray into the
realms of cryptographic libraries^W^W random memory usage.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3319>
2022-11-04 17:59:21 +00:00

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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 "gstfdkaac.h"
#include "gstfdkaacenc.h"
#include <gst/pbutils/pbutils.h>
#include <string.h>
/* TODO:
* - Add support for other AOT / profiles
* - Signal encoder delay
* - LOAS / LATM support
*/
enum
{
PROP_0,
PROP_AFTERBURNER,
PROP_BITRATE,
PROP_PEAK_BITRATE,
PROP_RATE_CONTROL,
PROP_VBR_PRESET,
};
#define DEFAULT_BITRATE (0)
#define DEFAULT_PEAK_BITRATE (0)
#define DEFAULT_RATE_CONTROL (GST_FDK_AAC_RATE_CONTROL_CONSTANT_BITRATE)
#define DEFAULT_VBR_PRESET (GST_FDK_AAC_VBR_PRESET_MEDIUM)
#define SAMPLE_RATES " 8000, " \
"11025, " \
"12000, " \
"16000, " \
"22050, " \
"24000, " \
"32000, " \
"44100, " \
"48000, " \
"64000, " \
"88200, " \
"96000"
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) 4, "
"rate = (int) { " SAMPLE_RATES " }, "
"channels = (int) {1, 2, 3, 4, 5, 6, 8}, "
"stream-format = (string) { adts, adif, raw }, "
"profile = (string) { lc, he-aac-v1, he-aac-v2, ld }, "
"framed = (boolean) true")
);
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);
static void gst_fdkaacenc_flush (GstAudioEncoder * enc);
G_DEFINE_TYPE (GstFdkAacEnc, gst_fdkaacenc, GST_TYPE_AUDIO_ENCODER);
GST_ELEMENT_REGISTER_DEFINE (fdkaacenc, "fdkaacenc", GST_RANK_PRIMARY,
GST_TYPE_FDKAACENC);
#define GST_FDK_AAC_VBR_PRESET (gst_fdk_aac_vbr_preset_get_type ())
static GType
gst_fdk_aac_vbr_preset_get_type (void)
{
static GType fdk_aac_vbr_preset_type = 0;
static const GEnumValue vbr_preset_types[] = {
{GST_FDK_AAC_VBR_PRESET_VERY_LOW, "Very Low Variable Bitrate", "very-low"},
{GST_FDK_AAC_VBR_PRESET_LOW, "Low Variable Bitrate", "low"},
{GST_FDK_AAC_VBR_PRESET_MEDIUM, "Medium Variable Bitrate", "medium"},
{GST_FDK_AAC_VBR_PRESET_HIGH, "High Variable Bitrate", "high"},
{GST_FDK_AAC_VBR_PRESET_VERY_HIGH, "Very High Variable Bitrate",
"very-high"},
{0, NULL, NULL}
};
if (!fdk_aac_vbr_preset_type)
fdk_aac_vbr_preset_type =
g_enum_register_static ("GstFdkAacVbrPreset", vbr_preset_types);
return fdk_aac_vbr_preset_type;
}
#define GST_FDK_AAC_RATE_CONTROL (gst_fdk_aac_rate_control_get_type ())
static GType
gst_fdk_aac_rate_control_get_type (void)
{
static GType fdk_aac_rate_control_type = 0;
static const GEnumValue rate_control_types[] = {
{GST_FDK_AAC_RATE_CONTROL_CONSTANT_BITRATE, "Constant Bitrate", "cbr"},
{GST_FDK_AAC_RATE_CONTROL_VARIABLE_BITRATE, "Variable Bitrate", "vbr"},
{0, NULL, NULL}
};
if (!fdk_aac_rate_control_type)
fdk_aac_rate_control_type =
g_enum_register_static ("GstFdkAacRateControl", rate_control_types);
return fdk_aac_rate_control_type;
}
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;
case PROP_AFTERBURNER:
self->afterburner = g_value_get_boolean (value);
break;
case PROP_PEAK_BITRATE:
self->peak_bitrate = g_value_get_int (value);
break;
case PROP_RATE_CONTROL:
self->rate_control = g_value_get_enum (value);
break;
case PROP_VBR_PRESET:
self->vbr_preset = g_value_get_enum (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;
case PROP_AFTERBURNER:
g_value_set_boolean (value, self->afterburner);
break;
case PROP_PEAK_BITRATE:
g_value_set_int (value, self->peak_bitrate);
break;
case PROP_RATE_CONTROL:
g_value_set_enum (value, self->rate_control);
break;
case PROP_VBR_PRESET:
g_value_set_enum (value, self->vbr_preset);
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);
self->enc = NULL;
}
self->is_drained = TRUE;
return TRUE;
}
static GstCaps *
gst_fdkaacenc_get_caps (GstAudioEncoder * enc, GstCaps * filter)
{
const GstFdkAacChannelLayout *layout;
GstCaps *res, *caps, *allowed_caps;
gboolean allow_mono = TRUE;
allowed_caps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (enc));
GST_DEBUG_OBJECT (enc, "allowed caps %" GST_PTR_FORMAT, allowed_caps);
/* We need at least 2 channels if Parametric Stereo is in use. */
if (allowed_caps && gst_caps_get_size (allowed_caps) > 0) {
GstStructure *s = gst_caps_get_structure (allowed_caps, 0);
const gchar *profile = NULL;
if ((profile = gst_structure_get_string (s, "profile"))
&& strcmp (profile, "he-aac-v2") == 0) {
allow_mono = FALSE;
}
}
gst_clear_caps (&allowed_caps);
caps = gst_caps_new_empty ();
for (layout = channel_layouts; layout->channels; layout++) {
GstCaps *tmp;
gint channels = layout->channels;
if (channels == 1 && !allow_mono)
continue;
tmp = gst_caps_make_writable (gst_pad_get_pad_template_caps
(GST_AUDIO_ENCODER_SINK_PAD (enc)));
if (channels == 1) {
gst_caps_set_simple (tmp, "channels", G_TYPE_INT, channels, NULL);
} else {
guint64 channel_mask;
gst_audio_channel_positions_to_mask (layout->positions, channels, FALSE,
&channel_mask);
gst_caps_set_simple (tmp, "channels", G_TYPE_INT, 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;
gint mpegversion = 4;
gint aot = AOT_AAC_LC;
const gchar *profile_str = "lc";
CHANNEL_MODE channel_mode;
AACENC_InfoStruct enc_info = { 0 };
gint bitrate, signaling_mode;
guint bitrate_mode;
if (self->enc && !self->is_drained) {
/* drain */
gst_fdkaacenc_handle_frame (enc, NULL);
aacEncClose (&self->enc);
self->is_drained = TRUE;
}
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;
}
}
if ((str = gst_structure_get_string (s, "profile"))) {
if (strcmp (str, "lc") == 0) {
GST_DEBUG_OBJECT (self, "using AAC-LC profile for output");
aot = AOT_AAC_LC;
profile_str = "lc";
} else if (strcmp (str, "he-aac-v1") == 0) {
GST_DEBUG_OBJECT (self, "using SBR (HE-AACv1) profile for output");
aot = AOT_SBR;
profile_str = "he-aac-v1";
} else if (strcmp (str, "he-aac-v2") == 0) {
GST_DEBUG_OBJECT (self, "using PS (HE-AACv2) profile for output");
aot = AOT_PS;
profile_str = "he-aac-v2";
} else if (strcmp (str, "ld") == 0) {
GST_DEBUG_OBJECT (self, "using AAC-LD profile for output");
aot = AOT_ER_AAC_LD;
profile_str = "ld";
}
}
gst_structure_get_int (s, "mpegversion", &mpegversion);
}
if (allowed_caps)
gst_caps_unref (allowed_caps);
err = aacEncOpen (&self->enc, 0, GST_AUDIO_INFO_CHANNELS (info));
if (err != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to open encoder: %d", err);
return FALSE;
}
if ((err = aacEncoder_SetParam (self->enc, AACENC_AOT, aot)) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to set AOT %d: %d", aot, err);
return FALSE;
}
/* Use explicit hierarchical signaling (2) with raw output stream-format
* and implicit signaling (0) with ADTS/ADIF */
if (transmux == 0)
signaling_mode = 2;
else
signaling_mode = 0;
if ((err = aacEncoder_SetParam (self->enc, AACENC_SIGNALING_MODE,
signaling_mode)) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to set signaling mode %d: %d",
signaling_mode, 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",
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 {
gint in_channels = GST_AUDIO_INFO_CHANNELS (info);
const GstAudioChannelPosition *in_positions =
&GST_AUDIO_INFO_POSITION (info, 0);
guint64 in_channel_mask;
const GstFdkAacChannelLayout *layout;
gst_audio_channel_positions_to_mask (in_positions, in_channels, FALSE,
&in_channel_mask);
for (layout = channel_layouts; layout->channels; layout++) {
gint channels = layout->channels;
const GstAudioChannelPosition *positions = layout->positions;
guint64 channel_mask;
if (channels != in_channels)
continue;
gst_audio_channel_positions_to_mask (positions, channels, FALSE,
&channel_mask);
if (channel_mask != in_channel_mask)
continue;
channel_mode = layout->mode;
self->need_reorder = memcmp (positions, in_positions,
channels * sizeof *positions) != 0;
self->aac_positions = positions;
break;
}
if (!layout->channels) {
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_TRANSMUX,
transmux)) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to set transmux %d: %d", transmux, err);
return FALSE;
}
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 (self->rate_control == GST_FDK_AAC_RATE_CONTROL_CONSTANT_BITRATE) {
/*
* Note that the `bitrate` property is honoured only when using
* constant bit rate.
*/
bitrate_mode = 0; // Constant Bitrate
} else {
bitrate_mode = self->vbr_preset;
}
if ((err = aacEncoder_SetParam (self->enc, AACENC_BITRATEMODE,
bitrate_mode)) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to set bitrate mode %d: %d",
bitrate_mode, err);
return FALSE;
}
if (self->peak_bitrate) {
if ((err = aacEncoder_SetParam (self->enc, AACENC_PEAK_BITRATE,
self->peak_bitrate)) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to set peak bitrate %d: %d",
self->peak_bitrate, err);
return FALSE;
}
GST_INFO_OBJECT (self, "Setting peak bitrate to %d", self->peak_bitrate);
}
if (self->afterburner) {
if ((err =
aacEncoder_SetParam (self->enc, AACENC_AFTERBURNER,
1)) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Could not enable afterburner: %d", err);
return FALSE;
}
GST_INFO_OBJECT (self, "Afterburner enabled");
}
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),
"framed", G_TYPE_BOOLEAN, TRUE,
"rate", G_TYPE_INT, GST_AUDIO_INFO_RATE (info), NULL);
/* raw */
if (transmux == 0) {
GstBuffer *codec_data =
gst_buffer_new_memdup (enc_info.confBuf, 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);
/* The above only parses the "base" profile, which is always going to be LC.
* Set actual profile. */
gst_caps_set_simple (src_caps, "profile", G_TYPE_STRING, profile_str, NULL);
/* An AAC-LC-only decoder will not decode a stream that uses explicit
* hierarchical signaling */
if (signaling_mode == 2 && aot != AOT_AAC_LC) {
gst_structure_remove_field (gst_caps_get_structure (src_caps, 0),
"base-profile");
}
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) {
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 = GST_AUDIO_INFO_BPS (info);
in_desc.numBufs = 1;
} else {
in_args.numInSamples = -1;
in_sizes = 0;
in_el_sizes = 0;
in_desc.numBufs = 0;
}
/* We unset is_drained even if there's no inbuf. Basically this is a
* workaround for aacEncEncode always producing 1024 bytes even without any
* input, thus messing up with the base class counting */
self->is_drained = FALSE;
in_desc.bufferIdentifiers = &in_id;
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;
err = aacEncEncode (self->enc, &in_desc, &out_desc, &in_args, &out_args);
if (err == AACENC_ENCODE_EOF && !inbuf)
goto out;
else if (err != AACENC_OK) {
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_flush (GstAudioEncoder * enc)
{
GstFdkAacEnc *self = GST_FDKAACENC (enc);
GstAudioInfo *info = gst_audio_encoder_get_audio_info (enc);
aacEncClose (&self->enc);
self->enc = NULL;
self->is_drained = TRUE;
if (GST_AUDIO_INFO_IS_VALID (info))
gst_fdkaacenc_set_format (enc, info);
}
static void
gst_fdkaacenc_init (GstFdkAacEnc * self)
{
self->bitrate = DEFAULT_BITRATE;
self->enc = NULL;
self->is_drained = TRUE;
self->afterburner = FALSE;
self->peak_bitrate = DEFAULT_PEAK_BITRATE;
self->rate_control = DEFAULT_RATE_CONTROL;
self->vbr_preset = DEFAULT_VBR_PRESET;
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);
base_class->flush = GST_DEBUG_FUNCPTR (gst_fdkaacenc_flush);
g_object_class_install_property (object_class, PROP_BITRATE,
g_param_spec_int ("bitrate",
"Bitrate",
"Target Audio Bitrate. Only applicable if rate-control=cbr. "
"(0 = fixed value based on sample rate and channel count)",
0, G_MAXINT, DEFAULT_BITRATE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstFdkAacEnc:peak-bitrate:
*
* Peak Bitrate to adjust maximum bits per audio frame.
*
* Since: 1.22
*/
g_object_class_install_property (object_class, PROP_PEAK_BITRATE,
g_param_spec_int ("peak-bitrate",
"Peak Bitrate",
"Peak Bitrate to adjust maximum bits per audio frame. "
"Bitrate is in bits/second. Only applicable if rate-control=vbr. (0 = Not set)",
0, G_MAXINT, DEFAULT_PEAK_BITRATE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstFdkAacEnc:afterburner:
*
* Afterburner - Quality Parameter.
*
* Since: 1.22
*/
g_object_class_install_property (object_class, PROP_AFTERBURNER,
g_param_spec_boolean ("afterburner", "Afterburner - Quality Parameter",
"Additional quality control parameter. Can cause workload increase.",
FALSE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstFdkAacEnc:rate-control:
*
* Rate Control.
*
* Since: 1.22
*/
g_object_class_install_property (object_class, PROP_RATE_CONTROL,
g_param_spec_enum ("rate-control", "Rate Control",
"Whether Constant or Variable Bitrate should be used.",
GST_FDK_AAC_RATE_CONTROL, DEFAULT_RATE_CONTROL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstFdkAacEnc:vbr-preset:
*
* AAC Variable Bitrate configurations.
*
* Since: 1.22
*/
g_object_class_install_property (object_class, PROP_VBR_PRESET,
g_param_spec_enum ("vbr-preset", "Variable Bitrate Preset",
"AAC Variable Bitrate configurations. Requires rate-control as vbr.",
GST_FDK_AAC_VBR_PRESET, DEFAULT_VBR_PRESET,
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/Converter", "FDK AAC audio encoder",
"Sebastian Dröge <sebastian@centricular.com>");
GST_DEBUG_CATEGORY_INIT (gst_fdkaacenc_debug, "fdkaacenc", 0,
"fdkaac encoder");
gst_type_mark_as_plugin_api (GST_FDK_AAC_VBR_PRESET, 0);
gst_type_mark_as_plugin_api (GST_FDK_AAC_RATE_CONTROL, 0);
}
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