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Diffstat (limited to 'src/AlsaDabplus.cpp')
| -rw-r--r-- | src/AlsaDabplus.cpp | 252 |
1 files changed, 252 insertions, 0 deletions
diff --git a/src/AlsaDabplus.cpp b/src/AlsaDabplus.cpp new file mode 100644 index 0000000..e94f00a --- /dev/null +++ b/src/AlsaDabplus.cpp @@ -0,0 +1,252 @@ +/* ------------------------------------------------------------------ + * 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; + } + + |