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-rw-r--r--src/dabplus-enc-file.c426
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diff --git a/src/dabplus-enc-file.c b/src/dabplus-enc-file.c
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+/* ------------------------------------------------------------------
+ * 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 <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 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;
+}
+