Remove dabplus-enc-file

3 files changed, 7 insertions, 447 deletions

diff --git a/Makefile.am b/Makefile.am
index 1097113..a357778 100644
--- a/Makefile.am
+++ b/Makefile.am
@@ -43,11 +43,6 @@ libfdk_aac_la_LDFLAGS = -version-info @FDK_AAC_VERSION@ -no-undefined \
 #aac_enc_SOURCES = src/aac-enc.c \
 #				  src/wavreader.c
 
-dabplus_enc_file_LDADD     = libfdk-aac.la -lfec
-dabplus_enc_file_CFLAGS    = $(AM_CPPFLAGS) $(GITVERSION_FLAGS)
-dabplus_enc_file_SOURCES   = src/dabplus-enc-file.c \
-							 src/wavreader.c
-
 dabplus_enc_file_zmq_LDADD     = libfdk-aac.la -lfec -lzmq
 dabplus_enc_file_zmq_CFLAGS    = $(AM_CPPFLAGS) $(GITVERSION_FLAGS)
 dabplus_enc_file_zmq_SOURCES   = src/dabplus-enc-file-zmq.c \
@@ -72,8 +67,7 @@ mot_encoder_SOURCES  = src/mot-encoder.c \
 					   contrib/lib_crc.h \
 					   contrib/lib_crc.c
 
-bin_PROGRAMS =  dabplus-enc-file$(EXEEXT) \
-				dabplus-enc-file-zmq$(EXEEXT) \
+bin_PROGRAMS =  dabplus-enc-file-zmq$(EXEEXT) \
 				dabplus-enc-alsa-zmq$(EXEEXT) \
 				mot-encoder$(EXEEXT)
 
diff --git a/README.md b/README.md
index 794424a..9d0caff 100644
--- a/README.md
+++ b/README.md
@@ -5,11 +5,9 @@ This package contains several tools that use the standalone library
 of the Fraunhofer FDK AAC code from Android, patched for
 960-transform to do DAB+ broadcast encoding.
 
-The first tool, *dabplus-enc-file* can encode from a file or pipe
-source, and encode into a file or pipe. There is no PAD support.
-
 The *dabplus-enc-file-zmq* can encode from a file or pipe source,
-and encode to a ZeroMQ output compatible with ODR-DabMux.
+and encode to a ZeroMQ output compatible with ODR-DabMux, to a
+file or to stdout.
 
 The *dabplus-enc-alsa-zmq* can encode from an ALSA soundcard,
 and encode to a ZeroMQ output compatible with ODR-DabMux. It supports
@@ -100,6 +98,7 @@ Then, you can use any media player that has an alsa output to play whatever sour
 
 Important: you must specify the correct sample rate on both "sides" of the virtual sound card.
 
+
 Scenario 3
 ----------
 Live Stream encoding and preparing for DAB muxer, with ZMQ output, at 32kHz, using sox.
@@ -120,16 +119,16 @@ Live Stream encoding and preparing for DAB muxer, with FIFO to odr-dabmux, 48kHz
 arecord.
 
     arecord -t raw -f S16_LE -c 2 -r 48000 -D plughw:CARD=Loopback,DEV=0,SUBDEV=0 | \
-    dabplus-enc-file -a -b 24 -f raw -c 2 -r 48000 -i /dev/stdin -o /dev/stdout 2>/dev/null | \
+    dabplus-enc-file-zmq -a -b 24 -f raw -c 2 -r 48000 -i /dev/stdin -o - | \
     mbuffer -q -m 10k -P 100 -s 360 > station1.fifo
 
-Here we are also using the ALSA plughw feature.
+Here we are using the ALSA plughw feature.
 
 Scenario 5
 ----------
 Wave file encoding, for non-realtime processing
 
-    dabplus-enc-file -a -b 64 -i wave_file.wav -o station1.dabp
+    dabplus-enc-file-zmq -a -b 64 -i wave_file.wav -o station1.dabp
 
 
 Usage of MOT Slideshow
diff --git a/src/dabplus-enc-file.c b/src/dabplus-enc-file.c
deleted file mode 100644
index 64c089d..0000000
--- a/src/dabplus-enc-file.c
+++ /dev/null
@@ -1,433 +0,0 @@
-/* ------------------------------------------------------------------
- * 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,
-    "dabplus-enc-file %s is a HE-AACv2 encoder for DAB+\n"
-    "based on fdk-aac-dabplus that can read from a file\n"
-    "or pipe source and encode into a file or pipe.\n"
-    "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"
-    ,
-#if defined(GITVERSION)
-    GITVERSION
-#else
-    PACKAGE_VERSION
-#endif
-    , 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;
-}
-