diff options
| -rw-r--r-- | Makefile.am | 8 | ||||
| -rw-r--r-- | README.md | 13 | ||||
| -rw-r--r-- | src/dabplus-enc-file.c | 433 |
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) @@ -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; -} - |