tidy dabplus-enc-file

1 files changed, 426 insertions, 0 deletions

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 <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;
+}
+