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/* ------------------------------------------------------------------
* Copyright (C) 2011 Martin Storsjo
* Copyright (C) 2013,2014 Matthias P. Braendli
*
* 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 "AlsaInput.h"
#include "SampleQueue.h"
#include "zmq.hpp"
#include <string>
#include <getopt.h>
#include <cstdio>
#include <stdint.h>
#include "libAACenc/include/aacenc_lib.h"
using namespace std;
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=URI Output zmq uri. (e.g. 'tcp://*:9000')\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"
" -d, --device=alsa_device Set ALSA input device (default: \"default\").\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"
//" -l, --lp Set frame size to 1024 instead of 960.\n"
"\n"
"Only the tcp:// zeromq transport has been tested until now.\n"
);
}
int prepare_aac_encoder(
HANDLE_AACENCODER *encoder,
int subchannel_index,
int channels,
int sample_rate,
int afterburner)
{
HANDLE_AACENCODER handle = *encoder;
int aot = AOT_DABPLUS_AAC_LC;
CHANNEL_MODE mode;
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 sample rate\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 granule length\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;
}
return 0;
}
#define no_argument 0
#define required_argument 1
#define optional_argument 2
int main(int argc, char *argv[]) {
int subchannel_index = 8; //64kbps subchannel
int ch=0;
int err;
const char *alsa_device = "default";
const char *outuri = NULL;
int sample_rate=48000, channels=2;
const int bytes_per_sample = 2;
void *rs_handler = NULL;
int afterburner = 0;
HANDLE_AACENCODER handle;
AACENC_InfoStruct info = { 0 };
const struct option longopts[] = {
{"bitrate", required_argument, 0, 'b'},
{"output", required_argument, 0, 'o'},
{"device", required_argument, 0, 'd'},
{"rate", required_argument, 0, 'r'},
{"channels", required_argument, 0, 'c'},
{"afterburner", no_argument, 0, 'a'},
{"help", no_argument, 0, 'h'},
{0,0,0,0},
};
int index;
while(ch != -1) {
ch = getopt_long(argc, argv, "hab:c:o:r:d:", longopts, &index);
switch (ch) {
case 'd':
alsa_device = optarg;
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 'o':
outuri = 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;
}
fprintf(stderr, "Setting up ZeroMQ socket\n");
if (!outuri) {
fprintf(stderr, "ZeroMQ output URI not defined\n");
return 1;
}
zmq::context_t zmq_ctx;
zmq::socket_t zmq_sock(zmq_ctx, ZMQ_PUB);
zmq_sock.connect(outuri);
HANDLE_AACENCODER encoder;
if (prepare_aac_encoder(&encoder, subchannel_index, channels,
sample_rate, afterburner) != 0) {
fprintf(stderr, "Encoder preparation failed\n");
return 2;
}
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);
// Each DAB+ frame will need input_size audio bytes
int input_size = channels * bytes_per_sample * info.frameLength;
uint8_t input_buf[input_size];
int max_size = input_size + NUM_SAMPLES_PER_CALL;
SampleQueue<uint8_t> queue(BYTES_PER_SAMPLE, channels, max_size);
AlsaInput alsa_in(alsa_device, channels, sample_rate, queue);
if (alsa_in.prepare() != 0) {
fprintf(stderr, "Alsa preparation failed\n");
return 1;
}
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