/*****************************************************************************
* CloseDecoder: decoder destruction
*****************************************************************************/
static void CloseEncoder(vlc_object_t *p_this)
{
encoder_t *p_enc = (encoder_t *)p_this;
encoder_sys_t *p_sys = p_enc->p_sys;
aacEncClose(&p_sys->handle);
free(p_sys);
}
static int aac_encode_close(AVCodecContext *avctx)
{
AACContext *s = avctx->priv_data;
if (s->handle)
aacEncClose(&s->handle);
av_freep(&avctx->extradata);
ff_af_queue_close(&s->afq);
return 0;
}
static void libfdk_destroy(void *data)
{
libfdk_encoder_t *enc = data;
aacEncClose(&enc->fdkhandle);
bfree(enc->packet_buffer);
bfree(enc);
blog(LOG_INFO, "libfdk_aac encoder destroyed");
}
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 int aac_encode_close(AVCodecContext *avctx)
{
AACContext *s = avctx->priv_data;
if (s->handle)
aacEncClose(&s->handle);
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
av_freep(&avctx->extradata);
ff_af_queue_close(&s->afq);
return 0;
}
static void AacEncoderDestroy(AacEncoderContext * _pAacCtx)
{
if (_pAacCtx == NULL) {
return;
}
if (((FdkaacConfig *)_pAacCtx)->pConvertBuf != NULL) {
free(((FdkaacConfig *)_pAacCtx)->pConvertBuf);
}
if (aacEncClose(&((FdkaacConfig *)_pAacCtx)->pEncoder) != AACENC_OK) {
return;
}
return;
}
AACFileWriter::~AACFileWriter()
{
if(m_advance_samples > 0)
{
m_in_args.numInSamples = m_advance_samples * sizeof( int16_t ) / m_channel;
m_encinBuf.bufs = (void **)&m_pInputbuf;
m_in_bufsize = m_advance_samples * sizeof( int16_t );
auto ret = aacEncEncode( m_hAacEncoder, &m_encinBuf, &m_encoutBuf, &m_in_args, &m_out_args );
if(ret != AACENC_OK)
{
return;
}
if(m_out_args.numOutBytes > 0)
{
fwrite( m_outofbyte, 1, m_out_args.numOutBytes, m_aacfile );
}
}
m_in_bufsize = 0;
m_encinBuf.bufs = nullptr;
m_in_args.numInSamples = -1;
auto ret = aacEncEncode( m_hAacEncoder, &m_encinBuf, &m_encoutBuf, &m_in_args, &m_out_args );
if(ret != AACENC_OK)
{
return;
}
if(m_out_args.numOutBytes > 0)
{
fwrite( m_outofbyte, 1, m_out_args.numOutBytes, m_aacfile );
}
if(m_aacfile) fclose( m_aacfile );
m_aacfile = nullptr;
if(m_outofbyte) delete[] m_outofbyte;
if(m_pInputbuf) delete[] m_pInputbuf;
aacEncClose( &m_hAacEncoder );
}
SoftAACEncoder2::~SoftAACEncoder2() {
aacEncClose(&mAACEncoder);
onReset();
}
int main(int argc, char *argv[]) {
int bitrate = 64000;
int ch;
const char *infile, *outfile;
FILE *out;
void *wav;
int format, sample_rate, channels, bits_per_sample;
int input_size;
uint8_t* input_buf;
int16_t* convert_buf;
int aot = 2;
int afterburner = 1;
int eld_sbr = 0;
int vbr = 0;
HANDLE_AACENCODER handle;
CHANNEL_MODE mode;
AACENC_InfoStruct info = { 0 };
while ((ch = getopt(argc, argv, "r:t:a:s:v:")) != -1) {
switch (ch) {
case 'r':
bitrate = atoi(optarg);
break;
case 't':
aot = atoi(optarg);
break;
case 'a':
afterburner = atoi(optarg);
break;
case 's':
eld_sbr = atoi(optarg);
break;
case 'v':
vbr = atoi(optarg);
break;
case '?':
default:
usage(argv[0]);
return 1;
}
}
if (argc - optind < 2) {
usage(argv[0]);
return 1;
}
infile = argv[optind];
outfile = argv[optind + 1];
wav = wav_read_open(infile);
if (!wav) {
fprintf(stderr, "Unable to open wav file %s\n", infile);
return 1;
}
if (!wav_get_header(wav, &format, &channels, &sample_rate, &bits_per_sample, NULL)) {
fprintf(stderr, "Bad wav file %s\n", infile);
return 1;
}
if (format != 1) {
fprintf(stderr, "Unsupported WAV format %d\n", format);
return 1;
}
if (bits_per_sample != 16) {
fprintf(stderr, "Unsupported WAV sample depth %d\n", bits_per_sample);
return 1;
}
switch (channels) {
case 1: mode = MODE_1; break;
case 2: mode = MODE_2; break;
case 3: mode = MODE_1_2; break;
case 4: mode = MODE_1_2_1; break;
case 5: mode = MODE_1_2_2; break;
case 6: mode = MODE_1_2_2_1; break;
default:
fprintf(stderr, "Unsupported WAV channels %d\n", channels);
return 1;
}
if (aacEncOpen(&handle, 0, channels) != AACENC_OK) {
fprintf(stderr, "Unable to open encoder\n");
return 1;
}
if (aacEncoder_SetParam(handle, AACENC_AOT, aot) != AACENC_OK) {
fprintf(stderr, "Unable to set the AOT\n");
return 1;
}
if (aot == 39 && eld_sbr) {
if (aacEncoder_SetParam(handle, AACENC_SBR_MODE, 1) != AACENC_OK) {
fprintf(stderr, "Unable to set SBR mode for ELD\n");
return 1;
}
}
if (aacEncoder_SetParam(handle, AACENC_SAMPLERATE, sample_rate) != AACENC_OK) {
fprintf(stderr, "Unable to set the AOT\n");
return 1;
}
if (aacEncoder_SetParam(handle, AACENC_CHANNELMODE, mode) != AACENC_OK) {
fprintf(stderr, "Unable to set the channel mode\n");
return 1;
}
if (aacEncoder_SetParam(handle, AACENC_CHANNELORDER, 1) != AACENC_OK) {
fprintf(stderr, "Unable to set the wav channel order\n");
return 1;
}
if (vbr) {
if (aacEncoder_SetParam(handle, AACENC_BITRATEMODE, vbr) != AACENC_OK) {
fprintf(stderr, "Unable to set the VBR bitrate mode\n");
return 1;
}
} else {
if (aacEncoder_SetParam(handle, AACENC_BITRATE, bitrate) != AACENC_OK) {
fprintf(stderr, "Unable to set the bitrate\n");
return 1;
}
}
if (aacEncoder_SetParam(handle, AACENC_TRANSMUX, 2) != AACENC_OK) {
fprintf(stderr, "Unable to set the ADTS transmux\n");
return 1;
}
if (aacEncoder_SetParam(handle, AACENC_AFTERBURNER, afterburner) != AACENC_OK) {
fprintf(stderr, "Unable to set the afterburner mode\n");
return 1;
}
if (aacEncEncode(handle, NULL, NULL, NULL, NULL) != AACENC_OK) {
fprintf(stderr, "Unable to initialize the encoder\n");
return 1;
}
if (aacEncInfo(handle, &info) != AACENC_OK) {
fprintf(stderr, "Unable to get the encoder info\n");
return 1;
}
out = fopen(outfile, "wb");
if (!out) {
perror(outfile);
return 1;
}
input_size = channels*2*info.frameLength;
input_buf = (uint8_t*) malloc(input_size);
convert_buf = (int16_t*) malloc(input_size);
while (1) {
AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 };
AACENC_InArgs in_args = { 0 };
AACENC_OutArgs out_args = { 0 };
int in_identifier = IN_AUDIO_DATA;
int in_size, in_elem_size;
int out_identifier = OUT_BITSTREAM_DATA;
int out_size, out_elem_size;
int read, i;
void *in_ptr, *out_ptr;
uint8_t outbuf[20480];
AACENC_ERROR err;
read = wav_read_data(wav, input_buf, input_size);
for (i = 0; i < read/2; i++) {
const uint8_t* in = &input_buf[2*i];
convert_buf[i] = in[0] | (in[1] << 8);
}
if (read <= 0) {
in_args.numInSamples = -1;
} else {
in_ptr = convert_buf;
in_size = read;
in_elem_size = 2;
in_args.numInSamples = read/2;
in_buf.numBufs = 1;
in_buf.bufs = &in_ptr;
in_buf.bufferIdentifiers = &in_identifier;
in_buf.bufSizes = &in_size;
in_buf.bufElSizes = &in_elem_size;
}
out_ptr = outbuf;
out_size = sizeof(outbuf);
out_elem_size = 1;
out_buf.numBufs = 1;
out_buf.bufs = &out_ptr;
out_buf.bufferIdentifiers = &out_identifier;
out_buf.bufSizes = &out_size;
out_buf.bufElSizes = &out_elem_size;
if ((err = aacEncEncode(handle, &in_buf, &out_buf, &in_args, &out_args)) != AACENC_OK) {
if (err == AACENC_ENCODE_EOF)
break;
fprintf(stderr, "Encoding failed\n");
return 1;
}
if (out_args.numOutBytes == 0)
continue;
fwrite(outbuf, 1, out_args.numOutBytes, out);
}
free(input_buf);
free(convert_buf);
fclose(out);
wav_read_close(wav);
aacEncClose(&handle);
return 0;
}
SoftAACEncoder2::~SoftAACEncoder2() {
aacEncClose(&mAACEncoder);
delete[] mInputFrame;
mInputFrame = NULL;
}
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);
err = aacEncOpen (&self->enc, 0, GST_AUDIO_INFO_CHANNELS (info));
if (err != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to open encoder: %d\n", err);
return FALSE;
}
aot = AOT_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_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 ((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_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 void *libfdk_create(obs_data_t settings, obs_encoder_t encoder)
{
bool hasFdkHandle = false;
libfdk_encoder_t *enc = 0;
int bitrate = (int)obs_data_get_int(settings, "bitrate") * 1000;
int afterburner = obs_data_get_bool(settings, "afterburner") ? 1 : 0;
audio_t audio = obs_encoder_audio(encoder);
int mode = 0;
AACENC_ERROR err;
if (!bitrate) {
blog(LOG_ERROR, "Invalid bitrate");
return NULL;
}
enc = bzalloc(sizeof(libfdk_encoder_t));
enc->encoder = encoder;
enc->channels = (int)audio_output_get_channels(audio);
enc->sample_rate = audio_output_get_sample_rate(audio);
switch(enc->channels) {
case 1:
mode = MODE_1;
break;
case 2:
mode = MODE_2;
break;
case 3:
mode = MODE_1_2;
break;
case 4:
mode = MODE_1_2_1;
break;
case 5:
mode = MODE_1_2_2;
break;
case 6:
mode = MODE_1_2_2_1;
break;
default:
blog(LOG_ERROR, "Invalid channel count");
goto fail;
}
CHECK_LIBFDK(aacEncOpen(&enc->fdkhandle, 0, enc->channels));
hasFdkHandle = true;
CHECK_LIBFDK(aacEncoder_SetParam(enc->fdkhandle, AACENC_AOT,
2)); // MPEG-4 AAC-LC
CHECK_LIBFDK(aacEncoder_SetParam(enc->fdkhandle, AACENC_SAMPLERATE,
enc->sample_rate));
CHECK_LIBFDK(aacEncoder_SetParam(enc->fdkhandle, AACENC_CHANNELMODE, mode));
CHECK_LIBFDK(aacEncoder_SetParam(enc->fdkhandle, AACENC_CHANNELORDER, 1));
CHECK_LIBFDK(aacEncoder_SetParam(enc->fdkhandle, AACENC_BITRATEMODE, 0));
CHECK_LIBFDK(aacEncoder_SetParam(enc->fdkhandle, AACENC_BITRATE, bitrate));
CHECK_LIBFDK(aacEncoder_SetParam(enc->fdkhandle, AACENC_TRANSMUX, 0));
CHECK_LIBFDK(aacEncoder_SetParam(enc->fdkhandle, AACENC_AFTERBURNER,
afterburner));
CHECK_LIBFDK(aacEncEncode(enc->fdkhandle, NULL, NULL, NULL, NULL));
CHECK_LIBFDK(aacEncInfo(enc->fdkhandle, &enc->info));
enc->frame_size_bytes = enc->info.frameLength * 2 * enc->channels;
enc->packet_buffer_size = enc->channels * 768;
if(enc->packet_buffer_size < 8192)
enc->packet_buffer_size = 8192;
enc->packet_buffer = bmalloc(enc->packet_buffer_size);
blog(LOG_INFO, "libfdk_aac encoder created");
blog(LOG_INFO, "libfdk_aac bitrate: %d, channels: %d",
bitrate / 1000, enc->channels);
return enc;
fail:
if(hasFdkHandle)
aacEncClose(&enc->fdkhandle);
if(enc->packet_buffer)
bfree(enc->packet_buffer);
if(enc)
bfree(enc);
blog(LOG_WARNING, "libfdk_aac encoder creation failed");
return 0;
}
int aacenc_init(HANDLE_AACENCODER *encoder, const aacenc_param_t *params,
const pcm_sample_description_t *format,
AACENC_InfoStruct *info)
{
int channel_mode;
int aot;
LIB_INFO lib_info;
*encoder = 0;
aacenc_get_lib_info(&lib_info);
if ((channel_mode = aacenc_channel_mode(format)) == 0) {
fprintf(stderr, "ERROR: unsupported channel layout\n");
goto FAIL;
}
if (aacEncOpen(encoder, 0, 0) != AACENC_OK) {
fprintf(stderr, "ERROR: aacEncOpen() failed\n");
goto FAIL;
}
aot = (params->profile ? params->profile : AOT_AAC_LC);
if (aacEncoder_SetParam(*encoder, AACENC_AOT, aot) != AACENC_OK) {
fprintf(stderr, "ERROR: unsupported profile\n");
goto FAIL;
}
if (params->bitrate_mode == 0)
aacEncoder_SetParam(*encoder, AACENC_BITRATE, params->bitrate);
else if (aacEncoder_SetParam(*encoder, AACENC_BITRATEMODE,
params->bitrate_mode) != AACENC_OK) {
fprintf(stderr, "ERROR: unsupported bitrate mode\n");
goto FAIL;
}
if (aacEncoder_SetParam(*encoder, AACENC_SAMPLERATE,
format->sample_rate) != AACENC_OK) {
fprintf(stderr, "ERROR: unsupported sample rate\n");
goto FAIL;
}
if (aacEncoder_SetParam(*encoder, AACENC_CHANNELMODE,
channel_mode) != AACENC_OK) {
fprintf(stderr, "ERROR: unsupported channel mode\n");
goto FAIL;
}
aacEncoder_SetParam(*encoder, AACENC_BANDWIDTH, params->bandwidth);
aacEncoder_SetParam(*encoder, AACENC_CHANNELORDER, 1);
aacEncoder_SetParam(*encoder, AACENC_AFTERBURNER, !!params->afterburner);
aacEncoder_SetParam(*encoder, AACENC_SBR_MODE, params->lowdelay_sbr);
#if AACENCODER_LIB_VL0 > 3 || (AACENCODER_LIB_VL0==3 && AACENCODER_LIB_VL1>=4)
if (lib_info.version > 0x03040800)
aacEncoder_SetParam(*encoder, AACENC_SBR_RATIO, params->sbr_ratio);
#endif
if (aacEncoder_SetParam(*encoder, AACENC_TRANSMUX,
params->transport_format) != AACENC_OK) {
fprintf(stderr, "ERROR: unsupported transport format\n");
goto FAIL;
}
if (aacEncoder_SetParam(*encoder, AACENC_SIGNALING_MODE,
params->sbr_signaling) != AACENC_OK) {
fprintf(stderr, "ERROR: failed to set SBR signaling mode\n");
goto FAIL;
}
if (params->adts_crc_check)
aacEncoder_SetParam(*encoder, AACENC_PROTECTION, 1);
if (params->header_period)
aacEncoder_SetParam(*encoder, AACENC_HEADER_PERIOD,
params->header_period);
if (aacEncEncode(*encoder, 0, 0, 0, 0) != AACENC_OK) {
fprintf(stderr, "ERROR: encoder initialization failed\n");
goto FAIL;
}
if (aacEncInfo(*encoder, info) != AACENC_OK) {
fprintf(stderr, "ERROR: cannot retrieve encoder info\n");
goto FAIL;
}
return 0;
FAIL:
if (encoder)
aacEncClose(encoder);
return -1;
}
