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Diffstat (limited to 'alsa-dabplus-zmq.c')
| -rw-r--r-- | alsa-dabplus-zmq.c | 505 |
1 files changed, 505 insertions, 0 deletions
diff --git a/alsa-dabplus-zmq.c b/alsa-dabplus-zmq.c new file mode 100644 index 0000000..b1b301d --- /dev/null +++ b/alsa-dabplus-zmq.c @@ -0,0 +1,505 @@ +/* ------------------------------------------------------------------ + * 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 <stdio.h> +#include <stdint.h> +#include <string.h> +#include <alloca.h> +#include <math.h> +#include <unistd.h> +#include <stdlib.h> +#include <getopt.h> +#include <zmq.h> +#include <assert.h> +#include "libAACenc/include/aacenc_lib.h" +#include <error.h> +#include <signal.h> +#include <alsa/asoundlib.h> + +#include <fec.h> + +static struct { + snd_pcm_format_t format; + unsigned int channels; + unsigned int rate; +} hwparams; + + +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" + +); + +} + +static snd_pcm_t *alsa_handle = NULL; + +static void prg_exit(int code) +{ + if (alsa_handle) { + snd_pcm_close(alsa_handle); + } + exit(code); +} + +static void alsa_prepare(const char* alsa_dev, unsigned int rate, unsigned int channels) +{ + int err; + snd_pcm_hw_params_t *hw_params; + + fprintf(stderr, "Initialising ALSA...\n"); + + const int open_mode = 0; //|= SND_PCM_NONBLOCK; + + if ((err = snd_pcm_open(&alsa_handle, alsa_dev, SND_PCM_STREAM_CAPTURE, open_mode)) < 0) { + fprintf (stderr, "cannot open audio device %s (%s)\n", + alsa_dev, snd_strerror(err)); + prg_exit(1); + } + + const int nonblock = 0; //TODO remove dead code + if (nonblock) { + err = snd_pcm_nonblock(alsa_handle, 1); + if (err < 0) { + fprintf(stderr, "nonblock setting error: %s", snd_strerror(err)); + prg_exit(1); + } + } + + if ((err = snd_pcm_hw_params_malloc(&hw_params)) < 0) { + fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n", + snd_strerror(err)); + prg_exit(1); + } + + if ((err = snd_pcm_hw_params_any(alsa_handle, hw_params)) < 0) { + fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n", + snd_strerror(err)); + prg_exit(1); + } + + if ((err = snd_pcm_hw_params_set_access(alsa_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { + fprintf (stderr, "cannot set access type (%s)\n", + snd_strerror(err)); + prg_exit(1); + } + + if ((err = snd_pcm_hw_params_set_format(alsa_handle, hw_params, SND_PCM_FORMAT_S16_LE)) < 0) { + fprintf (stderr, "cannot set sample format (%s)\n", + snd_strerror(err)); + prg_exit(1); + } + + if ((err = snd_pcm_hw_params_set_rate_near(alsa_handle, hw_params, &rate, 0)) < 0) { + fprintf (stderr, "cannot set sample rate (%s)\n", + snd_strerror(err)); + prg_exit(1); + } + + if ((err = snd_pcm_hw_params_set_channels(alsa_handle, hw_params, channels)) < 0) { + fprintf (stderr, "cannot set channel count (%s)\n", + snd_strerror(err)); + prg_exit(1); + } + + if ((err = snd_pcm_hw_params(alsa_handle, hw_params)) < 0) { + fprintf (stderr, "cannot set parameters (%s)\n", + snd_strerror(err)); + prg_exit(1); + } + + snd_pcm_hw_params_free (hw_params); + + if ((err = snd_pcm_prepare(alsa_handle)) < 0) { + fprintf (stderr, "cannot prepare audio interface for use (%s)\n", + snd_strerror(err)); + prg_exit(1); + } + + fprintf(stderr, "ALSA init done.\n"); +} + +static size_t alsa_read(uint8_t* buf, snd_pcm_uframes_t length) +{ + int i; + int err; + + err = snd_pcm_readi(alsa_handle, buf, length); + + if (err != length) { + if (err < 0) { + fprintf (stderr, "read from audio interface failed (%s)\n", + snd_strerror(err)); + } + else { + fprintf(stderr, "short alsa read: %d\n", err); + } + } + + return err; +} + +static void signal_handler(int sig) +{ + fprintf(stderr, "Caught signal %d\n", sig); + if (alsa_handle) { + snd_pcm_abort(alsa_handle); + alsa_handle = NULL; + } + + if (sig == SIGABRT) { + /* do not call snd_pcm_close() and abort immediately */ + alsa_handle = NULL; + exit(EXIT_FAILURE); + } + signal(sig, signal_handler); +} + + + +#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; + uint8_t* input_buf; + int16_t* convert_buf; + void *rs_handler = NULL; + int aot = AOT_DABPLUS_AAC_LC; + int afterburner = 0; + HANDLE_AACENCODER handle; + CHANNEL_MODE mode; + AACENC_InfoStruct info = { 0 }; + + void *zmq_context = zmq_ctx_new(); + void *zmq_sock = NULL; + + 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'}, + //{"lp", no_argument, 0, 'l'}, + {"afterburner", no_argument, 0, 'a'}, + {"help", no_argument, 0, 'h'}, + {0,0,0,0}, + }; + + int index; + while(ch != -1) { + ch = getopt_long(argc, argv, "lhab: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) { + zmq_sock = zmq_socket(zmq_context, ZMQ_PUB); + if (zmq_sock == NULL) { + fprintf(stderr, "Error occurred during zmq_socket: %s\n", zmq_strerror(errno)); + return 2; + } + if (zmq_connect(zmq_sock, outuri) != 0) { + fprintf(stderr, "Error occurred during zmq_connect: %s\n", zmq_strerror(errno)); + return 2; + } + } else { + fprintf(stderr, "Output URI not defined\n"); + return 1; + } + + alsa_prepare(alsa_device, sample_rate, channels); + + signal(SIGINT, signal_handler); + signal(SIGTERM, signal_handler); + signal(SIGABRT, signal_handler); + + switch (channels) { + case 1: mode = MODE_1; break; + case 2: mode = MODE_2; break; + default: + fprintf(stderr, "Unsupported channels number %d\n", channels); + prg_exit(1); + } + + + if (aacEncOpen(&handle, 0x01|0x02|0x04, channels) != AACENC_OK) { + fprintf(stderr, "Unable to open encoder\n"); + prg_exit(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"); + prg_exit(1); + } + if (aacEncoder_SetParam(handle, AACENC_SAMPLERATE, sample_rate) != AACENC_OK) { + fprintf(stderr, "Unable to set the AOT\n"); + prg_exit(1); + } + if (aacEncoder_SetParam(handle, AACENC_CHANNELMODE, mode) != AACENC_OK) { + fprintf(stderr, "Unable to set the channel mode\n"); + prg_exit(1); + } + if (aacEncoder_SetParam(handle, AACENC_CHANNELORDER, 1) != AACENC_OK) { + fprintf(stderr, "Unable to set the wav channel order\n"); + prg_exit(1); + } + if (aacEncoder_SetParam(handle, AACENC_GRANULE_LENGTH, 960) != AACENC_OK) { + fprintf(stderr, "Unable to set the AOT\n"); + prg_exit(1); + } + if (aacEncoder_SetParam(handle, AACENC_TRANSMUX, TT_DABPLUS) != AACENC_OK) { + fprintf(stderr, "Unable to set the RAW transmux\n"); + prg_exit(1); + } + + /*if (aacEncoder_SetParam(handle, AACENC_BITRATEMODE, 7 *AACENC_BR_MODE_SFR*) != AACENC_OK) { + fprintf(stderr, "Unable to set the bitrate mode\n"); + prg_exit(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"); + prg_exit(1); + } + if (aacEncoder_SetParam(handle, AACENC_AFTERBURNER, afterburner) != AACENC_OK) { + fprintf(stderr, "Unable to set the afterburner mode\n"); + prg_exit(1); + } + if (aacEncEncode(handle, NULL, NULL, NULL, NULL) != AACENC_OK) { + fprintf(stderr, "Unable to initialize the encoder\n"); + prg_exit(1); + } + if (aacEncInfo(handle, &info) != AACENC_OK) { + fprintf(stderr, "Unable to get the encoder info\n"); + prg_exit(1); + } + + fprintf(stderr, "DAB+ Encoding: framelen=%d\n", info.frameLength); + + int input_size = channels * bytes_per_sample * 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"); + prg_exit(1); + } + + 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; + int send_error_count = 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; + int send_error; + void *in_ptr, *out_ptr; + AACENC_ERROR err; + + read = alsa_read(input_buf, info.frameLength); + if (read != info.frameLength) { + fprintf(stderr, "Unable to read enough data from input!\n"); + break; + } + + 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"); + prg_exit(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; + 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); + } + } + + send_error = zmq_send(zmq_sock, outbuf, outbuf_size, ZMQ_DONTWAIT); + if (send_error < 0) { + fprintf(stderr, "ZeroMQ send failed! %s\n", zmq_strerror(errno)); + send_error_count ++; + } + + if (send_error_count > 10) + { + fprintf(stderr, "ZeroMQ send failed ten times, aborting!\n"); + break; + } + //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++; + } + + zmq_close(zmq_sock); + free_rs_char(rs_handler); + + aacEncClose(&handle); + + zmq_ctx_term(zmq_context); + prg_exit(0); +} + |