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-rw-r--r--aac-enc-dabplus.c698
1 files changed, 349 insertions, 349 deletions
diff --git a/aac-enc-dabplus.c b/aac-enc-dabplus.c
index 7516a15..cd86053 100644
--- a/aac-enc-dabplus.c
+++ b/aac-enc-dabplus.c
@@ -5,7 +5,7 @@
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
- * http://www.apache.org/licenses/LICENSE-2.0
+ * http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
@@ -29,26 +29,26 @@
#include <fec.h>
void usage(const char* name) {
- fprintf(stderr, "%s [OPTION...]\n", name);
- fprintf(stderr,
-" -b, --bitrate={ 8, 16, ..., 192 } Output bitrate in kbps. Must be 8 multiple.\n"
-//" -d, --data=FILENAME Set data filename.\n"
-//" -g, --fs-bug Turn on FS bug mitigation.\n"
-" -i, --input=FILENAME Input filename (default: stdin).\n"
-" -o, --output=FILENAME Output filename (default: stdout).\n"
-" -a, --afterburner Turn on AAC encoder quality increaser.\n"
-//" -m, --message Turn on AAC frame messages.\n"
-//" -p, --pad=BYTES Set PAD size in bytes.\n"
-" -f, --format={ wav, raw } Set input file format (default: wav).\n"
-" -c, --channels={ 1, 2 } Nb of input channels for raw input (default: 2).\n"
-" -r, --rate={ 32000, 48000 } Sample rate for raw input (default: 48000).\n"
-//" -t, --type=TYPE Set data type (dls|pad|packet|dg).\n"
-//" -v, --verbose=LEVEL Set verbosity level.\n"
-//" -V, --version Print version and exit.\n"
-//" --mi=[ 0, ... ] Set AAC frame messages interval in milliseconds.\n"
-//" --ma=[ 0, ... ] Set AAC frame messages attack time in milliseconds.\n"
-//" -t, --adts Set ADTS output format (for debugging).\n"
-//" -l, --lp Set frame size to 1024 instead of 960.\n"
+ fprintf(stderr, "%s [OPTION...]\n", name);
+ fprintf(stderr,
+" -b, --bitrate={ 8, 16, ..., 192 } Output bitrate in kbps. Must be 8 multiple.\n"
+//" -d, --data=FILENAME Set data filename.\n"
+//" -g, --fs-bug Turn on FS bug mitigation.\n"
+" -i, --input=FILENAME Input filename (default: stdin).\n"
+" -o, --output=FILENAME Output filename (default: stdout).\n"
+" -a, --afterburner Turn on AAC encoder quality increaser.\n"
+//" -m, --message Turn on AAC frame messages.\n"
+//" -p, --pad=BYTES Set PAD size in bytes.\n"
+" -f, --format={ wav, raw } Set input file format (default: wav).\n"
+" -c, --channels={ 1, 2 } Nb of input channels for raw input (default: 2).\n"
+" -r, --rate={ 32000, 48000 } Sample rate for raw input (default: 48000).\n"
+//" -t, --type=TYPE Set data type (dls|pad|packet|dg).\n"
+//" -v, --verbose=LEVEL Set verbosity level.\n"
+//" -V, --version Print version and exit.\n"
+//" --mi=[ 0, ... ] Set AAC frame messages interval in milliseconds.\n"
+//" --ma=[ 0, ... ] Set AAC frame messages attack time in milliseconds.\n"
+//" -t, --adts Set ADTS output format (for debugging).\n"
+//" -l, --lp Set frame size to 1024 instead of 960.\n"
);
@@ -71,8 +71,8 @@ int FindSRIndex(int sr)
{
int i;
for (i = 0; i < 16; i++) {
- if (sr == mpeg4audio_sample_rates[i])
- return i;
+ if (sr == mpeg4audio_sample_rates[i])
+ return i;
}
return 16 - 1;
}
@@ -84,341 +84,341 @@ void adts_hdr_up(char *buff, int size)
/* frame length, 13 bits */
buff[3] &= 0xFC;
- buff[3] |= ((len >> 11) & 0x03); /* 2b: aac_frame_length */
- buff[4] = len >> 3; /* 8b: aac_frame_length */
- buff[5] = (len << 5) & 0xE0; /* 3b: aac_frame_length */
+ buff[3] |= ((len >> 11) & 0x03); /* 2b: aac_frame_length */
+ buff[4] = len >> 3; /* 8b: aac_frame_length */
+ buff[5] = (len << 5) & 0xE0; /* 3b: aac_frame_length */
/* buffer fullness, 11 bits */
- buff[5] |= ((buffer_fullness >> 6) & 0x1F); /* 5b: adts_buffer_fullness */
- buff[6] = (buffer_fullness << 2) & 0xFC; /* 6b: adts_buffer_fullness */
- /* 2b: num_raw_data_blocks */
+ buff[5] |= ((buffer_fullness >> 6) & 0x1F); /* 5b: adts_buffer_fullness */
+ buff[6] = (buffer_fullness << 2) & 0xFC; /* 6b: adts_buffer_fullness */
+ /* 2b: num_raw_data_blocks */
}
int main(int argc, char *argv[]) {
- int subchannel_index = 8; //64kbps subchannel
- int ch=0;
- const char *infile, *outfile;
- FILE *in_fh, *out_fh;
- void *wav;
- int wav_format, bits_per_sample, sample_rate=48000, channels=2;
- uint8_t* input_buf;
- int16_t* convert_buf;
- void *rs_handler = NULL;
- int aot = AOT_DABPLUS_AAC_LC;
- int afterburner = 0, raw_input=0;
- HANDLE_AACENCODER handle;
- CHANNEL_MODE mode;
- AACENC_InfoStruct info = { 0 };
-
- const struct option longopts[] = {
- {"bitrate", required_argument, 0, 'b'},
- {"input", required_argument, 0, 'i'},
- {"output", required_argument, 0, 'o'},
- {"format", required_argument, 0, 'f'},
- {"rate", required_argument, 0, 'r'},
- {"channels", required_argument, 0, 'c'},
- //{"lp", no_argument, 0, 'l'},
- //{"adts", no_argument, 0, 't'},
- {"afterburner", no_argument, 0, 'a'},
- {"help", no_argument, 0, 'h'},
- {0,0,0,0},
- };
-
- int index;
- while(ch != -1) {
- ch = getopt_long(argc, argv, "tlhab:c:i:o:r:f:", longopts, &index);
- switch (ch) {
- case 'f':
- if(strcmp(optarg, "raw")==0) {
- raw_input = 1;
- } else if(strcmp(optarg, "wav")!=0)
- usage(argv[0]);
- break;
- case 'a':
- afterburner = 1;
- break;
- case 'b':
- subchannel_index = atoi(optarg) / 8;
- break;
- case 'c':
- channels = atoi(optarg);
- break;
- case 'r':
- sample_rate = atoi(optarg);
- break;
- case 'i':
- infile = optarg;
- break;
- case 'o':
- outfile = optarg;
- break;
- case '?':
- case 'h':
- usage(argv[0]);
- return 1;
- }
- }
-
- if(subchannel_index < 1 || subchannel_index > 24) {
- fprintf(stderr, "Bad subchannels number: %d, try other bitrate.\n", subchannel_index);
- return 1;
- }
-
- if(raw_input) {
- if(infile && strcmp(infile, "-")) {
- in_fh = fopen(infile, "rb");
- if(!in_fh) {
- fprintf(stderr, "Can't open input file!\n");
- return 1;
- }
- } else {
- in_fh = stdin;
- }
- } else {
- wav = wav_read_open(infile);
- if (!wav) {
- fprintf(stderr, "Unable to open wav file %s\n", infile);
- return 1;
- }
- if (!wav_get_header(wav, &wav_format, &channels, &sample_rate, &bits_per_sample, NULL)) {
- fprintf(stderr, "Bad wav file %s\n", infile);
- return 1;
- }
- if (wav_format != 1) {
- fprintf(stderr, "Unsupported WAV format %d\n", wav_format);
- return 1;
- }
- if (bits_per_sample != 16) {
- fprintf(stderr, "Unsupported WAV sample depth %d\n", bits_per_sample);
- return 1;
- }
- if (channels > 2) {
- fprintf(stderr, "Unsupported WAV channels %d\n", channels);
- return 1;
- }
- }
-
- if(outfile && strcmp(outfile, "-")) {
- out_fh = fopen(outfile, "wb");
- if(!out_fh) {
- fprintf(stderr, "Can't open output file!\n");
- return 1;
- }
- } else {
- out_fh = stdout;
- }
-
-
- switch (channels) {
- case 1: mode = MODE_1; break;
- case 2: mode = MODE_2; break;
- default:
- fprintf(stderr, "Unsupported channels number %d\n", channels);
- return 1;
- }
-
-
- if (aacEncOpen(&handle, 0x01|0x02|0x04, channels) != AACENC_OK) {
- fprintf(stderr, "Unable to open encoder\n");
- return 1;
- }
-
-
- if(channels == 2 && subchannel_index <= 6)
- aot = AOT_DABPLUS_PS;
- else if((channels == 1 && subchannel_index <= 8) || subchannel_index <= 10)
- aot = AOT_DABPLUS_SBR;
-
- fprintf(stderr, "Using %d subchannels. AAC type: %s%s%s. channels=%d, sample_rate=%d\n",
- subchannel_index,
- aot == AOT_DABPLUS_PS ? "HE-AAC v2" : "",
- aot == AOT_DABPLUS_SBR ? "HE-AAC" : "",
- aot == AOT_DABPLUS_AAC_LC ? "AAC-LC" : "",
- channels, sample_rate);
-
- if (aacEncoder_SetParam(handle, AACENC_AOT, aot) != AACENC_OK) {
- fprintf(stderr, "Unable to set the AOT\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 (aacEncoder_SetParam(handle, AACENC_GRANULE_LENGTH, 960) != AACENC_OK) {
- fprintf(stderr, "Unable to set the AOT\n");
- return 1;
- }
- if (aacEncoder_SetParam(handle, AACENC_TRANSMUX, TT_DABPLUS) != AACENC_OK) {
- fprintf(stderr, "Unable to set the RAW transmux\n");
- return 1;
- }
-
- /*if (aacEncoder_SetParam(handle, AACENC_BITRATEMODE, 7 *AACENC_BR_MODE_SFR*) != AACENC_OK) {
- fprintf(stderr, "Unable to set the bitrate mode\n");
- return 1;
- }*/
-
-
- fprintf(stderr, "AAC bitrate set to: %d\n", subchannel_index*8000);
- if (aacEncoder_SetParam(handle, AACENC_BITRATE, subchannel_index*8000) != AACENC_OK) {
- fprintf(stderr, "Unable to set the bitrate\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;
- }
-
- fprintf(stderr, "DAB+ Encoding: framelen=%d\n", info.frameLength);
-
- int input_size = channels*2*info.frameLength;
- input_buf = (uint8_t*) malloc(input_size);
- convert_buf = (int16_t*) malloc(input_size);
-
- /* symsize=8, gfpoly=0x11d, fcr=0, prim=1, nroots=10, pad=135 */
- rs_handler = init_rs_char(8, 0x11d, 0, 1, 10, 135);
- if (rs_handler == NULL) {
- perror("init_rs_char failed");
- return 0;
- }
+ int subchannel_index = 8; //64kbps subchannel
+ int ch=0;
+ const char *infile, *outfile;
+ FILE *in_fh, *out_fh;
+ void *wav;
+ int wav_format, bits_per_sample, sample_rate=48000, channels=2;
+ uint8_t* input_buf;
+ int16_t* convert_buf;
+ void *rs_handler = NULL;
+ int aot = AOT_DABPLUS_AAC_LC;
+ int afterburner = 0, raw_input=0;
+ HANDLE_AACENCODER handle;
+ CHANNEL_MODE mode;
+ AACENC_InfoStruct info = { 0 };
+
+ const struct option longopts[] = {
+ {"bitrate", required_argument, 0, 'b'},
+ {"input", required_argument, 0, 'i'},
+ {"output", required_argument, 0, 'o'},
+ {"format", required_argument, 0, 'f'},
+ {"rate", required_argument, 0, 'r'},
+ {"channels", required_argument, 0, 'c'},
+ //{"lp", no_argument, 0, 'l'},
+ //{"adts", no_argument, 0, 't'},
+ {"afterburner", no_argument, 0, 'a'},
+ {"help", no_argument, 0, 'h'},
+ {0,0,0,0},
+ };
+
+ int index;
+ while(ch != -1) {
+ ch = getopt_long(argc, argv, "tlhab:c:i:o:r:f:", longopts, &index);
+ switch (ch) {
+ case 'f':
+ if(strcmp(optarg, "raw")==0) {
+ raw_input = 1;
+ } else if(strcmp(optarg, "wav")!=0)
+ usage(argv[0]);
+ break;
+ case 'a':
+ afterburner = 1;
+ break;
+ case 'b':
+ subchannel_index = atoi(optarg) / 8;
+ break;
+ case 'c':
+ channels = atoi(optarg);
+ break;
+ case 'r':
+ sample_rate = atoi(optarg);
+ break;
+ case 'i':
+ infile = optarg;
+ break;
+ case 'o':
+ outfile = optarg;
+ break;
+ case '?':
+ case 'h':
+ usage(argv[0]);
+ return 1;
+ }
+ }
+
+ if(subchannel_index < 1 || subchannel_index > 24) {
+ fprintf(stderr, "Bad subchannels number: %d, try other bitrate.\n", subchannel_index);
+ return 1;
+ }
+
+ if(raw_input) {
+ if(infile && strcmp(infile, "-")) {
+ in_fh = fopen(infile, "rb");
+ if(!in_fh) {
+ fprintf(stderr, "Can't open input file!\n");
+ return 1;
+ }
+ } else {
+ in_fh = stdin;
+ }
+ } else {
+ wav = wav_read_open(infile);
+ if (!wav) {
+ fprintf(stderr, "Unable to open wav file %s\n", infile);
+ return 1;
+ }
+ if (!wav_get_header(wav, &wav_format, &channels, &sample_rate, &bits_per_sample, NULL)) {
+ fprintf(stderr, "Bad wav file %s\n", infile);
+ return 1;
+ }
+ if (wav_format != 1) {
+ fprintf(stderr, "Unsupported WAV format %d\n", wav_format);
+ return 1;
+ }
+ if (bits_per_sample != 16) {
+ fprintf(stderr, "Unsupported WAV sample depth %d\n", bits_per_sample);
+ return 1;
+ }
+ if (channels > 2) {
+ fprintf(stderr, "Unsupported WAV channels %d\n", channels);
+ return 1;
+ }
+ }
+
+ if(outfile && strcmp(outfile, "-")) {
+ out_fh = fopen(outfile, "wb");
+ if(!out_fh) {
+ fprintf(stderr, "Can't open output file!\n");
+ return 1;
+ }
+ } else {
+ out_fh = stdout;
+ }
+
+
+ switch (channels) {
+ case 1: mode = MODE_1; break;
+ case 2: mode = MODE_2; break;
+ default:
+ fprintf(stderr, "Unsupported channels number %d\n", channels);
+ return 1;
+ }
+
+
+ if (aacEncOpen(&handle, 0x01|0x02|0x04, channels) != AACENC_OK) {
+ fprintf(stderr, "Unable to open encoder\n");
+ return 1;
+ }
+
+
+ if(channels == 2 && subchannel_index <= 6)
+ aot = AOT_DABPLUS_PS;
+ else if((channels == 1 && subchannel_index <= 8) || subchannel_index <= 10)
+ aot = AOT_DABPLUS_SBR;
+
+ fprintf(stderr, "Using %d subchannels. AAC type: %s%s%s. channels=%d, sample_rate=%d\n",
+ subchannel_index,
+ aot == AOT_DABPLUS_PS ? "HE-AAC v2" : "",
+ aot == AOT_DABPLUS_SBR ? "HE-AAC" : "",
+ aot == AOT_DABPLUS_AAC_LC ? "AAC-LC" : "",
+ channels, sample_rate);
+
+ if (aacEncoder_SetParam(handle, AACENC_AOT, aot) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the AOT\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 (aacEncoder_SetParam(handle, AACENC_GRANULE_LENGTH, 960) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the AOT\n");
+ return 1;
+ }
+ if (aacEncoder_SetParam(handle, AACENC_TRANSMUX, TT_DABPLUS) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the RAW transmux\n");
+ return 1;
+ }
+
+ /*if (aacEncoder_SetParam(handle, AACENC_BITRATEMODE, 7 *AACENC_BR_MODE_SFR*) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the bitrate mode\n");
+ return 1;
+ }*/
+
+
+ fprintf(stderr, "AAC bitrate set to: %d\n", subchannel_index*8000);
+ if (aacEncoder_SetParam(handle, AACENC_BITRATE, subchannel_index*8000) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the bitrate\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;
+ }
+
+ fprintf(stderr, "DAB+ Encoding: framelen=%d\n", info.frameLength);
+
+ int input_size = channels*2*info.frameLength;
+ input_buf = (uint8_t*) malloc(input_size);
+ convert_buf = (int16_t*) malloc(input_size);
+
+ /* symsize=8, gfpoly=0x11d, fcr=0, prim=1, nroots=10, pad=135 */
+ rs_handler = init_rs_char(8, 0x11d, 0, 1, 10, 135);
+ if (rs_handler == NULL) {
+ perror("init_rs_char failed");
+ return 0;
+ }
int loops = 0;
int outbuf_size = subchannel_index*120;
- uint8_t outbuf[20480];
-
- if(outbuf_size % 5 != 0) {
- fprintf(stderr, "(outbuf_size mod 5) = %d\n", outbuf_size % 5);
- }
-
- fprintf(stderr, "outbuf_size: %d\n", outbuf_size);
- //outbuf_size += (4 * subchannel_index * (8*8)/8) - outbuf_size/5;
- fprintf(stderr, "outbuf_size: %d\n", outbuf_size);
-
- int frame=0;
- while (1) {
- memset(outbuf, 0x00, outbuf_size);
-
- 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=0, i;
- void *in_ptr, *out_ptr;
- AACENC_ERROR err;
-
- if(raw_input) {
- if(fread(input_buf, input_size, 1, in_fh) == 1) {
- read = input_size;
- } else {
- fprintf(stderr, "Unable to read from input!\n");
- break;
- }
- } else {
- 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;
+ uint8_t outbuf[20480];
+
+ if(outbuf_size % 5 != 0) {
+ fprintf(stderr, "(outbuf_size mod 5) = %d\n", outbuf_size % 5);
+ }
+
+ fprintf(stderr, "outbuf_size: %d\n", outbuf_size);
+ //outbuf_size += (4 * subchannel_index * (8*8)/8) - outbuf_size/5;
+ fprintf(stderr, "outbuf_size: %d\n", outbuf_size);
+
+ int frame=0;
+ while (1) {
+ memset(outbuf, 0x00, outbuf_size);
+
+ 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=0, i;
+ void *in_ptr, *out_ptr;
+ AACENC_ERROR err;
+
+ if(raw_input) {
+ if(fread(input_buf, input_size, 1, in_fh) == 1) {
+ read = input_size;
+ } else {
+ fprintf(stderr, "Unable to read from input!\n");
+ break;
+ }
+ } else {
+ 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;
#if 0
- unsigned char au_start[6];
- unsigned char* sfbuf = outbuf;
- au_start[0] = 6;
- au_start[1] = (*(sfbuf + 3) << 4) + ((*(sfbuf + 4)) >> 4);
- au_start[2] = ((*(sfbuf + 4) & 0x0f) << 8) + *(sfbuf + 5);
- fprintf (stderr, "au_start[0] = %d\n", au_start[0]);
- fprintf (stderr, "au_start[1] = %d\n", au_start[1]);
- fprintf (stderr, "au_start[2] = %d\n", au_start[2]);
+ unsigned char au_start[6];
+ unsigned char* sfbuf = outbuf;
+ au_start[0] = 6;
+ au_start[1] = (*(sfbuf + 3) << 4) + ((*(sfbuf + 4)) >> 4);
+ au_start[2] = ((*(sfbuf + 4) & 0x0f) << 8) + *(sfbuf + 5);
+ fprintf (stderr, "au_start[0] = %d\n", au_start[0]);
+ fprintf (stderr, "au_start[1] = %d\n", au_start[1]);
+ fprintf (stderr, "au_start[2] = %d\n", au_start[2]);
#endif
- int row, col;
- char buf_to_rs_enc[110];
- char rs_enc[10];
- for(row=0; row < subchannel_index; row++) {
- for(col=0;col < 110; col++) {
- buf_to_rs_enc[col] = outbuf[subchannel_index * col + row];
- }
-
- encode_rs_char(rs_handler, buf_to_rs_enc, rs_enc);
-
- for(col=110; col<120; col++) {
- outbuf[subchannel_index * col + row] = rs_enc[col-110];
- assert(subchannel_index * col + row < outbuf_size);
- }
- }
-
- fwrite(outbuf, 1, /*out_args.numOutBytes*/ outbuf_size, out_fh);
- //fprintf(stderr, "Written %d/%d bytes!\n", out_args.numOutBytes + row*10, outbuf_size);
- if(out_args.numOutBytes + row*10 == outbuf_size)
- fprintf(stderr, ".");
-
-// if(frame > 10)
-// break;
- frame++;
- }
- free(input_buf);
- free(convert_buf);
- if(raw_input) {
- fclose(in_fh);
- } else {
- wav_read_close(wav);
- }
- fclose(out_fh);
- free_rs_char(rs_handler);
-
- aacEncClose(&handle);
-
- return 0;
+ int row, col;
+ char buf_to_rs_enc[110];
+ char rs_enc[10];
+ for(row=0; row < subchannel_index; row++) {
+ for(col=0;col < 110; col++) {
+ buf_to_rs_enc[col] = outbuf[subchannel_index * col + row];
+ }
+
+ encode_rs_char(rs_handler, buf_to_rs_enc, rs_enc);
+
+ for(col=110; col<120; col++) {
+ outbuf[subchannel_index * col + row] = rs_enc[col-110];
+ assert(subchannel_index * col + row < outbuf_size);
+ }
+ }
+
+ fwrite(outbuf, 1, /*out_args.numOutBytes*/ outbuf_size, out_fh);
+ //fprintf(stderr, "Written %d/%d bytes!\n", out_args.numOutBytes + row*10, outbuf_size);
+ if(out_args.numOutBytes + row*10 == outbuf_size)
+ fprintf(stderr, ".");
+
+// if(frame > 10)
+// break;
+ frame++;
+ }
+ free(input_buf);
+ free(convert_buf);
+ if(raw_input) {
+ fclose(in_fh);
+ } else {
+ wav_read_close(wav);
+ }
+ fclose(out_fh);
+ free_rs_char(rs_handler);
+
+ aacEncClose(&handle);
+
+ return 0;
}
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-11 23:33:04 +0000