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Diffstat (limited to 'src/aac-enc-dabplus.c')
| -rw-r--r-- | src/aac-enc-dabplus.c | 425 |
1 files changed, 425 insertions, 0 deletions
diff --git a/src/aac-enc-dabplus.c b/src/aac-enc-dabplus.c new file mode 100644 index 0000000..6a0b882 --- /dev/null +++ b/src/aac-enc-dabplus.c @@ -0,0 +1,425 @@ +/* ------------------------------------------------------------------ + * 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 <stdio.h> +#include <stdint.h> +#include <string.h> +#include <unistd.h> +#include <stdlib.h> +#include <getopt.h> +#include <assert.h> +#include "libAACenc/include/aacenc_lib.h" +#include "wavreader.h" + +#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" + +); + +} + + +#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; +} + |