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Diffstat (limited to 'src/AlsaDabplus.cpp')
| -rw-r--r-- | src/AlsaDabplus.cpp | 408 |
1 files changed, 408 insertions, 0 deletions
diff --git a/src/AlsaDabplus.cpp b/src/AlsaDabplus.cpp new file mode 100644 index 0000000..50dcddc --- /dev/null +++ b/src/AlsaDabplus.cpp @@ -0,0 +1,408 @@ +/* ------------------------------------------------------------------ + * 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 <time.h> +#include <unistd.h> + +#include "libAACenc/include/aacenc_lib.h" + +extern "C" { +#include <fec.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; + } + + *encoder = handle; + + 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; + 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; + 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(encoder, &info) != AACENC_OK) { + fprintf(stderr, "Unable to get the encoder info\n"); + return 1; + } + + // Each DAB+ frame will need input_size audio bytes + const int input_size = channels * bytes_per_sample * info.frameLength; + fprintf(stderr, "DAB+ Encoding: framelen=%d (%dB)\n", + info.frameLength, + input_size); + + 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); + + /* 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 1; + } + + AlsaInput alsa_in(alsa_device, channels, sample_rate, queue); + + if (alsa_in.prepare() != 0) { + fprintf(stderr, "Alsa preparation failed\n"); + return 1; + } + + fprintf(stderr, "Start ALSA capture thread\n"); + alsa_in.start(); + + fprintf(stderr, "Starting queue preroll\n"); + // Preroll until queue full + while (queue.size() < input_size) { + usleep(1000); + } + + 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, "Starting encoding\n"); + + int send_error_count = 0; + struct timespec tp; + clock_gettime(CLOCK_MONOTONIC, &tp); + + int calls; + while (1) { + int in_identifier = IN_AUDIO_DATA; + int out_identifier = OUT_BITSTREAM_DATA; + + AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 }; + AACENC_InArgs in_args = { 0 }; + AACENC_OutArgs out_args = { 0 }; + void *in_ptr, *out_ptr; + int in_size, in_elem_size; + int out_size, out_elem_size; + + memset(outbuf, 0x00, outbuf_size); + + int read = queue.pop(input_buf, input_size); + + if (read != input_size) { + fprintf(stderr, "Short read\n"); + } + + // -------------- AAC Encoding + + in_ptr = input_buf; + in_size = read; + in_elem_size = BYTES_PER_SAMPLE; + in_args.numInSamples = input_size; + 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; + + AACENC_ERROR err; + if ((err = aacEncEncode(encoder, &in_buf, &out_buf, &in_args, &out_args)) + != AACENC_OK) { + if (err == AACENC_ENCODE_EOF) + break; + fprintf(stderr, "Encoding failed\n"); + break; + } + + calls++; + + /* Check if the encoder has generated output data */ + if (out_args.numOutBytes != 0) + { + fprintf(stderr, "data out after %d calls\n", calls); + calls = 0; + // ----------- RS encoding + int row, col; + unsigned char buf_to_rs_enc[110]; + unsigned 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); + } + } + + // ------------ ZeroMQ transmit + try { + zmq_sock.send(outbuf, outbuf_size, ZMQ_DONTWAIT); + } + catch (zmq::error_t& e) { + fprintf(stderr, "ZeroMQ send error !\n"); + send_error_count ++; + } + + if (send_error_count > 10) + { + fprintf(stderr, "ZeroMQ send failed ten times, aborting!\n"); + break; + } + + if (out_args.numOutBytes + row*10 == outbuf_size) + fprintf(stderr, "."); + } + + // -------------- wait the right amount of time + tp.tv_nsec += 60000000; + if (tp.tv_nsec > 1000000000L) { + tp.tv_nsec -= 1000000000L; + tp.tv_sec += 1; + } + + fprintf(stderr, "sleep %ld.%ld\n", tp.tv_sec, tp.tv_nsec); + struct timespec tp_now; + do { + usleep(1000); + clock_gettime(CLOCK_MONOTONIC, &tp_now); + } while (tp_now.tv_sec < tp.tv_sec || + ( tp_now.tv_sec == tp.tv_sec && + tp_now.tv_nsec < tp.tv_nsec) ); + + tp.tv_sec = tp_now.tv_sec; + tp.tv_nsec = tp_now.tv_nsec; + + } + + zmq_sock.close(); + free_rs_char(rs_handler); + + aacEncClose(&encoder); + +} + |