mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 20:21:24 +00:00
604 lines
17 KiB
C
604 lines
17 KiB
C
/* GStreamer AAC encoder plugin
|
|
* Copyright (C) 2011 Kan Hu <kan.hu@linaro.org>
|
|
*
|
|
* 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.
|
|
*/
|
|
|
|
/**
|
|
* SECTION:element-voaacenc
|
|
*
|
|
* AAC audio encoder based on vo-aacenc library
|
|
* <ulink url="http://sourceforge.net/projects/opencore-amr/files/vo-aacenc/">vo-aacenc library source file</ulink>.
|
|
*
|
|
* <refsect2>
|
|
* <title>Example launch line</title>
|
|
* |[
|
|
* gst-launch filesrc location=abc.wav ! wavparse ! audioresample ! audioconvert ! voaacenc ! filesink location=abc.aac
|
|
* ]|
|
|
* </refsect2>
|
|
*/
|
|
|
|
#ifdef HAVE_CONFIG_H
|
|
#include "config.h"
|
|
#endif
|
|
|
|
#include <string.h>
|
|
|
|
#include <gst/pbutils/codec-utils.h>
|
|
|
|
#include "gstvoaacenc.h"
|
|
|
|
#define VOAAC_ENC_DEFAULT_BITRATE (128000)
|
|
#define VOAAC_ENC_DEFAULT_OUTPUTFORMAT (0) /* RAW */
|
|
#define VOAAC_ENC_MPEGVERSION (4)
|
|
#define VOAAC_ENC_CODECDATA_LEN (2)
|
|
#define VOAAC_ENC_BITS_PER_SAMPLE (16)
|
|
|
|
enum
|
|
{
|
|
PROP_0,
|
|
PROP_BITRATE
|
|
};
|
|
|
|
#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]")
|
|
);
|
|
|
|
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], "
|
|
"stream-format = (string) { adts, raw }, " "base-profile = (string) lc")
|
|
);
|
|
|
|
GST_DEBUG_CATEGORY_STATIC (gst_voaacenc_debug);
|
|
#define GST_CAT_DEFAULT gst_voaacenc_debug
|
|
|
|
static gboolean voaacenc_core_init (GstVoAacEnc * voaacenc);
|
|
static gboolean voaacenc_core_set_parameter (GstVoAacEnc * voaacenc);
|
|
static void voaacenc_core_uninit (GstVoAacEnc * voaacenc);
|
|
|
|
static gboolean gst_voaacenc_start (GstAudioEncoder * enc);
|
|
static gboolean gst_voaacenc_stop (GstAudioEncoder * enc);
|
|
static gboolean gst_voaacenc_set_format (GstAudioEncoder * enc,
|
|
GstAudioInfo * info);
|
|
static GstFlowReturn gst_voaacenc_handle_frame (GstAudioEncoder * enc,
|
|
GstBuffer * in_buf);
|
|
static GstCaps *gst_voaacenc_getcaps (GstAudioEncoder * enc, GstCaps * filter);
|
|
|
|
G_DEFINE_TYPE (GstVoAacEnc, gst_voaacenc, GST_TYPE_AUDIO_ENCODER);
|
|
|
|
static void
|
|
gst_voaacenc_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstVoAacEnc *self = GST_VOAACENC (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_voaacenc_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstVoAacEnc *self = GST_VOAACENC (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 void
|
|
gst_voaacenc_class_init (GstVoAacEncClass * 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_voaacenc_set_property);
|
|
object_class->get_property = GST_DEBUG_FUNCPTR (gst_voaacenc_get_property);
|
|
|
|
base_class->start = GST_DEBUG_FUNCPTR (gst_voaacenc_start);
|
|
base_class->stop = GST_DEBUG_FUNCPTR (gst_voaacenc_stop);
|
|
base_class->set_format = GST_DEBUG_FUNCPTR (gst_voaacenc_set_format);
|
|
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_voaacenc_handle_frame);
|
|
base_class->getcaps = GST_DEBUG_FUNCPTR (gst_voaacenc_getcaps);
|
|
|
|
g_object_class_install_property (object_class, PROP_BITRATE,
|
|
g_param_spec_int ("bitrate",
|
|
"Bitrate",
|
|
"Target Audio Bitrate",
|
|
0, G_MAXINT, VOAAC_ENC_DEFAULT_BITRATE,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
gst_element_class_add_pad_template (element_class,
|
|
gst_static_pad_template_get (&sink_template));
|
|
gst_element_class_add_pad_template (element_class,
|
|
gst_static_pad_template_get (&src_template));
|
|
|
|
gst_element_class_set_static_metadata (element_class, "AAC audio encoder",
|
|
"Codec/Encoder/Audio", "AAC audio encoder", "Kan Hu <kan.hu@linaro.org>");
|
|
|
|
GST_DEBUG_CATEGORY_INIT (gst_voaacenc_debug, "voaacenc", 0, "voaac encoder");
|
|
}
|
|
|
|
static void
|
|
gst_voaacenc_init (GstVoAacEnc * voaacenc)
|
|
{
|
|
voaacenc->bitrate = VOAAC_ENC_DEFAULT_BITRATE;
|
|
voaacenc->output_format = VOAAC_ENC_DEFAULT_OUTPUTFORMAT;
|
|
|
|
/* init rest */
|
|
voaacenc->handle = NULL;
|
|
}
|
|
|
|
static gboolean
|
|
gst_voaacenc_start (GstAudioEncoder * enc)
|
|
{
|
|
GstVoAacEnc *voaacenc = GST_VOAACENC (enc);
|
|
|
|
GST_DEBUG_OBJECT (enc, "start");
|
|
|
|
if (voaacenc_core_init (voaacenc) == FALSE)
|
|
return FALSE;
|
|
|
|
voaacenc->rate = 0;
|
|
voaacenc->channels = 0;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_voaacenc_stop (GstAudioEncoder * enc)
|
|
{
|
|
GstVoAacEnc *voaacenc = GST_VOAACENC (enc);
|
|
|
|
GST_DEBUG_OBJECT (enc, "stop");
|
|
voaacenc_core_uninit (voaacenc);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
#define VOAAC_ENC_MAX_CHANNELS 6
|
|
/* describe the channels position */
|
|
static const GstAudioChannelPosition
|
|
aac_channel_positions[][VOAAC_ENC_MAX_CHANNELS] = {
|
|
{ /* 1 ch: Mono */
|
|
GST_AUDIO_CHANNEL_POSITION_MONO},
|
|
{ /* 2 ch: front left + front right (front stereo) */
|
|
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
|
|
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
|
|
{ /* 3 ch: front center + front stereo */
|
|
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
|
|
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
|
|
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
|
|
{ /* 4 ch: front center + front stereo + back center */
|
|
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 ch: front center + front stereo + back stereo */
|
|
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
|
|
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
|
|
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
|
|
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
|
|
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT},
|
|
{ /* 6ch: front center + front stereo + back stereo + LFE */
|
|
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
|
|
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
|
|
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
|
|
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
|
|
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
|
|
GST_AUDIO_CHANNEL_POSITION_LFE1}
|
|
};
|
|
|
|
static gpointer
|
|
gst_voaacenc_generate_sink_caps (gpointer data)
|
|
{
|
|
GstCaps *caps;
|
|
gint i, c;
|
|
static const int rates[] = {
|
|
8000, 11025, 12000, 16000, 22050, 24000,
|
|
32000, 44100, 48000, 64000, 88200, 96000
|
|
};
|
|
GValue rates_arr = { 0, };
|
|
GValue tmp = { 0, };
|
|
GstStructure *s, *t;
|
|
|
|
g_value_init (&rates_arr, GST_TYPE_LIST);
|
|
g_value_init (&tmp, G_TYPE_INT);
|
|
for (i = 0; i < G_N_ELEMENTS (rates); i++) {
|
|
g_value_set_int (&tmp, rates[i]);
|
|
gst_value_list_append_value (&rates_arr, &tmp);
|
|
}
|
|
g_value_unset (&tmp);
|
|
|
|
s = gst_structure_new ("audio/x-raw",
|
|
"format", G_TYPE_STRING, GST_AUDIO_NE (S16),
|
|
"layout", G_TYPE_STRING, "interleaved", NULL);
|
|
gst_structure_set_value (s, "rate", &rates_arr);
|
|
|
|
caps = gst_caps_new_empty ();
|
|
|
|
for (i = 1; i <= 2 /* VOAAC_ENC_MAX_CHANNELS */ ; i++) {
|
|
guint64 channel_mask = 0;
|
|
t = gst_structure_copy (s);
|
|
|
|
gst_structure_set (t, "channels", G_TYPE_INT, i, NULL);
|
|
if (i > 1) {
|
|
for (c = 0; c < i; c++)
|
|
channel_mask |=
|
|
G_GUINT64_CONSTANT (1) << aac_channel_positions[i - 1][c];
|
|
|
|
gst_structure_set (t, "channel-mask", GST_TYPE_BITMASK, channel_mask,
|
|
NULL);
|
|
}
|
|
gst_caps_append_structure (caps, t);
|
|
}
|
|
|
|
gst_structure_free (s);
|
|
g_value_unset (&rates_arr);
|
|
|
|
GST_DEBUG ("generated sink caps: %" GST_PTR_FORMAT, caps);
|
|
return caps;
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_voaacenc_get_sink_caps (void)
|
|
{
|
|
static GOnce g_once = G_ONCE_INIT;
|
|
GstCaps *caps;
|
|
|
|
g_once (&g_once, gst_voaacenc_generate_sink_caps, NULL);
|
|
caps = g_once.retval;
|
|
|
|
return caps;
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_voaacenc_getcaps (GstAudioEncoder * benc, GstCaps * filter)
|
|
{
|
|
return gst_audio_encoder_proxy_getcaps (benc, gst_voaacenc_get_sink_caps (),
|
|
filter);
|
|
}
|
|
|
|
/* check downstream caps to configure format */
|
|
static void
|
|
gst_voaacenc_negotiate (GstVoAacEnc * voaacenc)
|
|
{
|
|
GstCaps *caps;
|
|
|
|
caps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (voaacenc));
|
|
|
|
GST_DEBUG_OBJECT (voaacenc, "allowed caps: %" GST_PTR_FORMAT, caps);
|
|
|
|
if (caps && gst_caps_get_size (caps) > 0) {
|
|
GstStructure *s = gst_caps_get_structure (caps, 0);
|
|
const gchar *str = NULL;
|
|
|
|
if ((str = gst_structure_get_string (s, "stream-format"))) {
|
|
if (strcmp (str, "adts") == 0) {
|
|
GST_DEBUG_OBJECT (voaacenc, "use ADTS format for output");
|
|
voaacenc->output_format = 1;
|
|
} else if (strcmp (str, "raw") == 0) {
|
|
GST_DEBUG_OBJECT (voaacenc, "use RAW format for output");
|
|
voaacenc->output_format = 0;
|
|
} else {
|
|
GST_DEBUG_OBJECT (voaacenc, "unknown stream-format: %s", str);
|
|
voaacenc->output_format = VOAAC_ENC_DEFAULT_OUTPUTFORMAT;
|
|
}
|
|
}
|
|
}
|
|
|
|
if (caps)
|
|
gst_caps_unref (caps);
|
|
}
|
|
|
|
static gint
|
|
gst_voaacenc_get_rate_index (gint rate)
|
|
{
|
|
static const gint rate_table[] = {
|
|
96000, 88200, 64000, 48000, 44100, 32000,
|
|
24000, 22050, 16000, 12000, 11025, 8000
|
|
};
|
|
gint i;
|
|
for (i = 0; i < G_N_ELEMENTS (rate_table); ++i) {
|
|
if (rate == rate_table[i]) {
|
|
return i;
|
|
}
|
|
}
|
|
return -1;
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_voaacenc_create_source_pad_caps (GstVoAacEnc * voaacenc)
|
|
{
|
|
GstCaps *caps = NULL;
|
|
gint index;
|
|
GstBuffer *codec_data;
|
|
GstMapInfo map;
|
|
|
|
if ((index = gst_voaacenc_get_rate_index (voaacenc->rate)) >= 0) {
|
|
codec_data = gst_buffer_new_and_alloc (VOAAC_ENC_CODECDATA_LEN);
|
|
gst_buffer_map (codec_data, &map, GST_MAP_WRITE);
|
|
/* LC profile only */
|
|
map.data[0] = ((0x02 << 3) | (index >> 1));
|
|
map.data[1] = ((index & 0x01) << 7) | (voaacenc->channels << 3);
|
|
|
|
caps = gst_caps_new_simple ("audio/mpeg",
|
|
"mpegversion", G_TYPE_INT, VOAAC_ENC_MPEGVERSION,
|
|
"channels", G_TYPE_INT, voaacenc->channels,
|
|
"rate", G_TYPE_INT, voaacenc->rate,
|
|
"stream-format", G_TYPE_STRING,
|
|
(voaacenc->output_format ? "adts" : "raw")
|
|
, NULL);
|
|
|
|
gst_codec_utils_aac_caps_set_level_and_profile (caps, map.data,
|
|
VOAAC_ENC_CODECDATA_LEN);
|
|
gst_buffer_unmap (codec_data, &map);
|
|
|
|
if (!voaacenc->output_format) {
|
|
gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, codec_data,
|
|
NULL);
|
|
}
|
|
gst_buffer_unref (codec_data);
|
|
}
|
|
|
|
return caps;
|
|
}
|
|
|
|
static gboolean
|
|
gst_voaacenc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
|
|
{
|
|
gboolean ret = FALSE;
|
|
GstVoAacEnc *voaacenc;
|
|
GstCaps *src_caps;
|
|
|
|
voaacenc = GST_VOAACENC (benc);
|
|
|
|
/* get channel count */
|
|
voaacenc->channels = GST_AUDIO_INFO_CHANNELS (info);
|
|
voaacenc->rate = GST_AUDIO_INFO_RATE (info);
|
|
|
|
/* precalc buffer size as it's constant now */
|
|
voaacenc->inbuf_size = voaacenc->channels * 2 * 1024;
|
|
|
|
gst_voaacenc_negotiate (voaacenc);
|
|
|
|
/* create reverse caps */
|
|
src_caps = gst_voaacenc_create_source_pad_caps (voaacenc);
|
|
|
|
if (src_caps) {
|
|
gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (voaacenc),
|
|
src_caps);
|
|
gst_caps_unref (src_caps);
|
|
ret = voaacenc_core_set_parameter (voaacenc);
|
|
}
|
|
|
|
/* report needs to base class */
|
|
gst_audio_encoder_set_frame_samples_min (benc, 1024);
|
|
gst_audio_encoder_set_frame_samples_max (benc, 1024);
|
|
gst_audio_encoder_set_frame_max (benc, 1);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_voaacenc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
|
|
{
|
|
GstVoAacEnc *voaacenc;
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
GstBuffer *out;
|
|
VO_AUDIO_OUTPUTINFO output_info = { {0} };
|
|
VO_CODECBUFFER input = { 0 };
|
|
VO_CODECBUFFER output = { 0 };
|
|
GstMapInfo map, omap;
|
|
GstAudioInfo *info = gst_audio_encoder_get_audio_info (benc);
|
|
|
|
voaacenc = GST_VOAACENC (benc);
|
|
|
|
g_return_val_if_fail (voaacenc->handle, GST_FLOW_NOT_NEGOTIATED);
|
|
|
|
/* we don't deal with squeezing remnants, so simply discard those */
|
|
if (G_UNLIKELY (buf == NULL)) {
|
|
GST_DEBUG_OBJECT (benc, "no data");
|
|
goto exit;
|
|
}
|
|
|
|
if (memcmp (info->position, aac_channel_positions[info->channels - 1],
|
|
sizeof (GstAudioChannelPosition) * info->channels) != 0) {
|
|
buf = gst_buffer_make_writable (buf);
|
|
gst_audio_buffer_reorder_channels (buf, info->finfo->format,
|
|
info->channels, info->position,
|
|
aac_channel_positions[info->channels - 1]);
|
|
}
|
|
|
|
gst_buffer_map (buf, &map, GST_MAP_READ);
|
|
|
|
if (G_UNLIKELY (map.size < voaacenc->inbuf_size)) {
|
|
gst_buffer_unmap (buf, &map);
|
|
GST_DEBUG_OBJECT (voaacenc, "discarding trailing data %d", (gint) map.size);
|
|
ret = gst_audio_encoder_finish_frame (benc, NULL, -1);
|
|
goto exit;
|
|
}
|
|
|
|
/* max size */
|
|
out = gst_buffer_new_and_alloc (voaacenc->inbuf_size);
|
|
gst_buffer_map (out, &omap, GST_MAP_WRITE);
|
|
|
|
output.Buffer = omap.data;
|
|
output.Length = voaacenc->inbuf_size;
|
|
|
|
g_assert (map.size == voaacenc->inbuf_size);
|
|
input.Buffer = map.data;
|
|
input.Length = voaacenc->inbuf_size;
|
|
voaacenc->codec_api.SetInputData (voaacenc->handle, &input);
|
|
|
|
/* encode */
|
|
if (voaacenc->codec_api.GetOutputData (voaacenc->handle, &output,
|
|
&output_info) != VO_ERR_NONE) {
|
|
gst_buffer_unmap (buf, &map);
|
|
gst_buffer_unmap (out, &omap);
|
|
gst_buffer_unref (out);
|
|
goto encode_failed;
|
|
}
|
|
|
|
GST_LOG_OBJECT (voaacenc, "encoded to %lu bytes", output.Length);
|
|
gst_buffer_unmap (buf, &map);
|
|
gst_buffer_unmap (out, &omap);
|
|
gst_buffer_resize (out, 0, output.Length);
|
|
|
|
ret = gst_audio_encoder_finish_frame (benc, out, 1024);
|
|
|
|
exit:
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
encode_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (voaacenc, STREAM, ENCODE, (NULL), ("encode failed"));
|
|
ret = GST_FLOW_ERROR;
|
|
goto exit;
|
|
}
|
|
}
|
|
|
|
static VO_U32
|
|
voaacenc_core_mem_alloc (VO_S32 uID, VO_MEM_INFO * pMemInfo)
|
|
{
|
|
if (!pMemInfo)
|
|
return VO_ERR_INVALID_ARG;
|
|
|
|
pMemInfo->VBuffer = g_malloc (pMemInfo->Size);
|
|
return 0;
|
|
}
|
|
|
|
static VO_U32
|
|
voaacenc_core_mem_free (VO_S32 uID, VO_PTR pMem)
|
|
{
|
|
g_free (pMem);
|
|
return 0;
|
|
}
|
|
|
|
static VO_U32
|
|
voaacenc_core_mem_set (VO_S32 uID, VO_PTR pBuff, VO_U8 uValue, VO_U32 uSize)
|
|
{
|
|
memset (pBuff, uValue, uSize);
|
|
return 0;
|
|
}
|
|
|
|
static VO_U32
|
|
voaacenc_core_mem_copy (VO_S32 uID, VO_PTR pDest, VO_PTR pSource, VO_U32 uSize)
|
|
{
|
|
memcpy (pDest, pSource, uSize);
|
|
return 0;
|
|
}
|
|
|
|
static VO_U32
|
|
voaacenc_core_mem_check (VO_S32 uID, VO_PTR pBuffer, VO_U32 uSize)
|
|
{
|
|
return 0;
|
|
}
|
|
|
|
static gboolean
|
|
voaacenc_core_init (GstVoAacEnc * voaacenc)
|
|
{
|
|
VO_CODEC_INIT_USERDATA user_data = { 0 };
|
|
voGetAACEncAPI (&voaacenc->codec_api);
|
|
|
|
voaacenc->mem_operator.Alloc = voaacenc_core_mem_alloc;
|
|
voaacenc->mem_operator.Copy = voaacenc_core_mem_copy;
|
|
voaacenc->mem_operator.Free = voaacenc_core_mem_free;
|
|
voaacenc->mem_operator.Set = voaacenc_core_mem_set;
|
|
voaacenc->mem_operator.Check = voaacenc_core_mem_check;
|
|
user_data.memflag = VO_IMF_USERMEMOPERATOR;
|
|
user_data.memData = &voaacenc->mem_operator;
|
|
voaacenc->codec_api.Init (&voaacenc->handle, VO_AUDIO_CodingAAC, &user_data);
|
|
|
|
if (voaacenc->handle == NULL) {
|
|
return FALSE;
|
|
}
|
|
return TRUE;
|
|
|
|
}
|
|
|
|
static gboolean
|
|
voaacenc_core_set_parameter (GstVoAacEnc * voaacenc)
|
|
{
|
|
AACENC_PARAM params = { 0 };
|
|
guint32 ret;
|
|
|
|
params.sampleRate = voaacenc->rate;
|
|
params.bitRate = voaacenc->bitrate;
|
|
params.nChannels = voaacenc->channels;
|
|
if (voaacenc->output_format) {
|
|
params.adtsUsed = 1;
|
|
} else {
|
|
params.adtsUsed = 0;
|
|
}
|
|
|
|
ret =
|
|
voaacenc->codec_api.SetParam (voaacenc->handle, VO_PID_AAC_ENCPARAM,
|
|
¶ms);
|
|
if (ret != VO_ERR_NONE) {
|
|
GST_ERROR_OBJECT (voaacenc, "Failed to set encoder parameters");
|
|
return FALSE;
|
|
}
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
voaacenc_core_uninit (GstVoAacEnc * voaacenc)
|
|
{
|
|
if (voaacenc->handle) {
|
|
voaacenc->codec_api.Uninit (voaacenc->handle);
|
|
voaacenc->handle = NULL;
|
|
}
|
|
}
|