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-rw-r--r--alsa-dabplus-zmq.c850
1 files changed, 425 insertions, 425 deletions
diff --git a/alsa-dabplus-zmq.c b/alsa-dabplus-zmq.c
index a51c722..b1b301d 100644
--- a/alsa-dabplus-zmq.c
+++ b/alsa-dabplus-zmq.c
@@ -6,7 +6,7 @@
* 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
+ * 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,
@@ -35,33 +35,33 @@
#include <fec.h>
static struct {
- snd_pcm_format_t format;
- unsigned int channels;
- unsigned int rate;
+ 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"
+ 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"
@@ -73,123 +73,123 @@ static snd_pcm_t *alsa_handle = NULL;
static void prg_exit(int code)
{
- if (alsa_handle) {
- snd_pcm_close(alsa_handle);
- }
- exit(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");
+ 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;
+ 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);
+ 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);
}
@@ -199,307 +199,307 @@ static void signal_handler(int sig)
#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 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;
+ 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]);
+ 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);
+ 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);
}
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