From 1a0fc3ef6348f0c85a2bdbc97ebf5422217dabdd Mon Sep 17 00:00:00 2001 From: "Matthias P. Braendli" Date: Sun, 9 Mar 2014 16:50:18 +0100 Subject: tidy dabplus-enc-file --- src/aac-enc-dabplus.c | 425 ------------------------------------------------ src/dabplus-enc-file.c | 426 +++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 426 insertions(+), 425 deletions(-) delete mode 100644 src/aac-enc-dabplus.c create mode 100644 src/dabplus-enc-file.c (limited to 'src') diff --git a/src/aac-enc-dabplus.c b/src/aac-enc-dabplus.c deleted file mode 100644 index 6a0b882..0000000 --- a/src/aac-enc-dabplus.c +++ /dev/null @@ -1,425 +0,0 @@ -/* ------------------------------------------------------------------ - * Copyright (C) 2011 Martin Storsjo - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * 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 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either - * express or implied. - * See the License for the specific language governing permissions - * and limitations under the License. - * ------------------------------------------------------------------- - */ - -#include -#include -#include -#include -#include -#include -#include -#include "libAACenc/include/aacenc_lib.h" -#include "wavreader.h" - -#include - -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" - -); - -} - - -#define no_argument 0 -#define required_argument 1 -#define optional_argument 2 - -#define ADTS_HEADER_SIZE 7 -#define ADTS_MPEG_ID 1 /* 0: MPEG-4, 1: MPEG-2 */ -#define ADTS_MPEG_PROFILE 1 -const int mpeg4audio_sample_rates[16] = { - 96000, 88200, 64000, 48000, 44100, 32000, - 24000, 22050, 16000, 12000, 11025, 8000, 7350 -}; - -int FindSRIndex(int sr) -{ - int i; - for (i = 0; i < 16; i++) { - if (sr == mpeg4audio_sample_rates[i]) - return i; - } - return 16 - 1; -} - -void adts_hdr_up(char *buff, int size) -{ - unsigned short len = size + ADTS_HEADER_SIZE; - unsigned short buffer_fullness = 0x07FF; - - /* 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 */ - /* 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 */ -} - -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 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); - // returns bytes read - } - - 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]); -#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; -} - diff --git a/src/dabplus-enc-file.c b/src/dabplus-enc-file.c new file mode 100644 index 0000000..c6aff07 --- /dev/null +++ b/src/dabplus-enc-file.c @@ -0,0 +1,426 @@ +/* ------------------------------------------------------------------ + * Copyright (C) 2011 Martin Storsjo + * Copyright (C) 2014 Matthias P. Braendli + * + * http://opendigitalradio.org + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * 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 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either + * express or implied. + * See the License for the specific language governing permissions + * and limitations under the License. + * ------------------------------------------------------------------- + */ + +#include +#include +#include +#include +#include +#include +#include +#include "libAACenc/include/aacenc_lib.h" +#include "wavreader.h" + +#include + +void usage(const char* name) { + fprintf(stderr, + "%s is a HE-AACv2 encoder for DAB+ based on fdk-aac-dabplus\n" + "that can encode from a file or pipe source, and encode\n" + "into a file or pipe. There is no PAD support.\n\n" + "Usage:\n" + "%s [OPTION...]\n\n" + " -b, --bitrate={ 8, 16, ..., 192 } Output bitrate in kbps. Must be 8 multiple.\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" + //" -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" + //" -v, --verbose=LEVEL Set verbosity level.\n" + , name, name); + +} + +#define no_argument 0 +#define required_argument 1 +#define optional_argument 2 + +#define ADTS_HEADER_SIZE 7 +#define ADTS_MPEG_ID 1 /* 0: MPEG-4, 1: MPEG-2 */ +#define ADTS_MPEG_PROFILE 1 +const int mpeg4audio_sample_rates[16] = { + 96000, 88200, 64000, 48000, 44100, 32000, + 24000, 22050, 16000, 12000, 11025, 8000, 7350 +}; + +int FindSRIndex(int sr) +{ + int i; + for (i = 0; i < 16; i++) { + if (sr == mpeg4audio_sample_rates[i]) + return i; + } + return 16 - 1; +} + +void adts_hdr_up(char *buff, int size) +{ + unsigned short len = size + ADTS_HEADER_SIZE; + unsigned short buffer_fullness = 0x07FF; + + /* 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 */ + /* 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 */ +} + +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}, + }; + + if (argc == 1) { + usage(argv[0]); + return 1; + } + + 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); + // returns bytes read + } + + 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]); +#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; +} + -- cgit v1.2.3