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-rw-r--r--src/dabplus-enc.cpp423
1 files changed, 300 insertions, 123 deletions
diff --git a/src/dabplus-enc.cpp b/src/dabplus-enc.cpp
index 0f34d65..5c2f1cb 100644
--- a/src/dabplus-enc.cpp
+++ b/src/dabplus-enc.cpp
@@ -24,6 +24,7 @@
#include "VLCInput.h"
#include "SampleQueue.h"
#include "zmq.hpp"
+#include "common.h"
extern "C" {
#include "encryption.h"
@@ -41,21 +42,33 @@ extern "C" {
#include <sys/types.h>
#include <sys/stat.h>
#include <sys/ioctl.h>
+#include <fcntl.h>
#include "libAACenc/include/aacenc_lib.h"
extern "C" {
#include <fec.h>
+#include "libtoolame-dab/toolame.h"
}
+
+
+// Enumerate which encoder we can use
+enum class encoder_selection_t {
+ fdk_dabplus,
+ toolame_dab
+};
+
using namespace std;
void usage(const char* name) {
fprintf(stderr,
- "dabplus-enc %s is a HE-AACv2 encoder for DAB+\n"
- "based on fdk-aac-dabplus that can read from"
- "JACK, ALSA or a file source\n"
- "and encode to a ZeroMQ output for ODR-DabMux.\n"
+ "dabplus-enc %s is an audio encoder for both DAB and DAB+.\n"
+ "The DAB+ HE-AACv2 encoder is based on a Thirt-Party Modified\n"
+ "Version of the Fraunhofer FDK AAC Codec Library for Android,\n"
+ "and the DAB encoder is using the tooLAME MPEG\n"
+ "encoder sources. The encoder can read from JACK, ALSA or\n"
+ "a file source and encode to a ZeroMQ output for ODR-DabMux.\n"
"(Experimental!)It can also use libvlc as an input.\n"
"\n"
"The -D option enables experimental sound card clock drift compensation.\n"
@@ -74,7 +87,7 @@ void usage(const char* name) {
"This encoder includes PAD (DLS and MOT Slideshow) support by\n"
"http://rd.csp.it to be used with mot-encoder\n"
"\nUsage:\n"
- "%s (-i file|-d alsa_device) [OPTION...]\n",
+ "%s [INPUT SELECTION] [OPTION...]\n",
#if defined(GITVERSION)
GITVERSION
#else
@@ -83,7 +96,11 @@ void usage(const char* name) {
, name);
fprintf(stderr,
" For the alsa input:\n"
+#if HAVE_ALSA
" -d, --device=alsa_device Set ALSA input device (default: \"default\").\n"
+#else
+ " The Alsa input was disabled at compile time\n"
+#endif
" For the file input:\n"
" -i, --input=FILENAME Input filename (default: stdin).\n"
" -f, --format={ wav, raw } Set input file format (default: wav).\n"
@@ -113,9 +130,16 @@ void usage(const char* name) {
" -D, --drift-comp Enable ALSA/VLC sound card drift compensation.\n"
" Encoder parameters:\n"
" -b, --bitrate={ 8, 16, ..., 192 } Output bitrate in kbps. Must be a multiple of 8.\n"
- " -A, --no-afterburner Disable AAC encoder quality increaser.\n"
" -c, --channels={ 1, 2 } Nb of input channels (default: 2).\n"
" -r, --rate={ 32000, 48000 } Input sample rate (default: 48000).\n"
+ " DAB specific options\n"
+ " -a, --dab Encode in DAB and not in DAB+.\n"
+ " --dabmode=MODE Channel mode: s/d/j/m\n"
+ " (default: j if stereo, m if mono).\n"
+ " --dabpsy=PSY Psychoacoustic model 0/1/2/3\n"
+ " (default: 1).\n"
+ " DAB+ specific options\n"
+ " -A, --no-afterburner Disable AAC encoder quality increaser.\n"
" --aaclc Force the usage of AAC-LC (no SBR, no PS)\n"
" --sbr Force the usage of SBR\n"
" --ps Force the usage of PS\n"
@@ -242,9 +266,11 @@ int prepare_aac_encoder(
int main(int argc, char *argv[])
{
- int subchannel_index = 8; //64kbps subchannel
+ int bitrate = 0; // 0 is default
int ch=0;
+ encoder_selection_t selected_encoder = encoder_selection_t::fdk_dabplus;
+
// For the ALSA input
const char *alsa_device = NULL;
@@ -275,6 +301,9 @@ int main(int argc, char *argv[])
AACENC_InfoStruct info = { 0 };
int aot = AOT_NONE;
+ char dab_channel_mode = '\0';
+ int dab_psy_model = 1;
+
/* Keep track of peaks */
int peak_left = 0;
int peak_right = 0;
@@ -302,6 +331,8 @@ int main(int argc, char *argv[])
const struct option longopts[] = {
{"bitrate", required_argument, 0, 'b'},
{"channels", required_argument, 0, 'c'},
+ {"dabmode", required_argument, 0, 4 },
+ {"dabpsy", required_argument, 0, 5 },
{"device", required_argument, 0, 'd'},
{"format", required_argument, 0, 'f'},
{"input", required_argument, 0, 'i'},
@@ -318,7 +349,7 @@ int main(int argc, char *argv[])
{"vlc-opt", required_argument, 0, 'L'},
{"write-icy-text", required_argument, 0, 'w'},
{"aaclc", no_argument, 0, 0 },
- {"afterburner", no_argument, 0, 'a'},
+ {"dab", no_argument, 0, 'a'},
{"drift-comp", no_argument, 0, 'D'},
{"fifo-silence", no_argument, 0, 3 },
{"help", no_argument, 0, 'h'},
@@ -362,16 +393,20 @@ int main(int argc, char *argv[])
aot = AOT_DABPLUS_PS;
break;
case 3: // FIFO SILENCE
- inFifoSilence = true;
+ case 4: // DAB channel mode
+ dab_channel_mode = optarg[0];
+ break;
+ case 5: // DAB psy model
+ dab_psy_model = atoi(optarg);
break;
case 'a':
- fprintf(stderr, "Warning, -a option does not exist anymore!\n");
+ selected_encoder = encoder_selection_t::toolame_dab;
break;
case 'A':
afterburner = false;
break;
case 'b':
- subchannel_index = atoi(optarg) / 8;
+ bitrate = atoi(optarg);
break;
case 'c':
channels = atoi(optarg);
@@ -471,15 +506,33 @@ int main(int argc, char *argv[])
return 1;
}
- if (subchannel_index < 1 || subchannel_index > 24) {
- fprintf(stderr, "Bad subchannel index: %d, must be between 1 and 24. Try other bitrate.\n",
- subchannel_index);
- return 1;
+ if (selected_encoder == encoder_selection_t::fdk_dabplus) {
+ if (bitrate == 0) {
+ bitrate = 64;
+ }
+
+ int subchannel_index = bitrate / 8;
+
+ if (subchannel_index < 1 || subchannel_index > 24) {
+ fprintf(stderr, "Bad subchannel index: %d, must be between 1 and 24. Try other bitrate.\n",
+ subchannel_index);
+ return 1;
+ }
+
+ if ( ! (sample_rate == 32000 || sample_rate == 48000)) {
+ fprintf(stderr, "Invalid sample rate. Possible values are: 32000, 48000.\n");
+ return 1;
+ }
}
+ else if (selected_encoder == encoder_selection_t::toolame_dab) {
+ if (bitrate == 0) {
+ bitrate = 192;
+ }
- if ( ! (sample_rate == 32000 || sample_rate == 48000)) {
- fprintf(stderr, "Invalid sample rate. Possible values are: 32000, 48000.\n");
- return 1;
+ if ( ! (sample_rate == 24000 || sample_rate == 48000)) {
+ fprintf(stderr, "Invalid sample rate. Possible values are: 24000, 48000.\n");
+ return 1;
+ }
}
if (padlen < 0) {
@@ -561,12 +614,71 @@ int main(int argc, char *argv[])
}
+ std::vector<uint8_t> input_buf;
+
HANDLE_AACENCODER encoder;
- if (prepare_aac_encoder(&encoder, subchannel_index, channels,
- sample_rate, afterburner, &aot) != 0) {
- fprintf(stderr, "Encoder preparation failed\n");
- return 1;
+ if (selected_encoder == encoder_selection_t::fdk_dabplus) {
+ int subchannel_index = bitrate / 8;
+ if (prepare_aac_encoder(&encoder, subchannel_index, channels,
+ sample_rate, afterburner, &aot) != 0) {
+ fprintf(stderr, "Encoder preparation failed\n");
+ return 1;
+ }
+
+ 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);
+
+ input_buf.resize(input_size);
+ }
+ else if (selected_encoder == encoder_selection_t::toolame_dab) {
+ int err = toolame_init();
+
+ if (err == 0) {
+ err = toolame_set_samplerate(sample_rate);
+ }
+
+ if (err == 0) {
+ err = toolame_set_bitrate(bitrate);
+ }
+
+ if (err == 0) {
+ err = toolame_set_psy_model(dab_psy_model);
+ }
+
+ if (dab_channel_mode == '\0') {
+ if (channels == 2) {
+ dab_channel_mode = 'j'; // Default to joint-stereo
+ }
+ else if (channels == 1) {
+ dab_channel_mode = 'm'; // Default to mono
+ }
+ else {
+ fprintf(stderr, "Unsupported channels number %d\n", channels);
+ return 1;
+ }
+ }
+
+ if (err == 0) {
+ err = toolame_set_channel_mode(dab_channel_mode);
+ }
+
+ if (err) {
+ fprintf(stderr, "libtoolame-dab init failed: %d\n", err);
+ return err;
+ }
+
+ // TODO int toolame_set_pad(int pad_len);
+
+ input_buf.resize(2 * 1152 * BYTES_PER_SAMPLE);
}
/* We assume that we need to call the encoder
@@ -574,24 +686,13 @@ int main(int argc, char *argv[])
* frame. This information is used when the alsa drift compensation
* is active
*/
- const int enc_calls_per_output =
- (aot == AOT_DABPLUS_AAC_LC) ? sample_rate / 8000 : sample_rate / 16000;
-
+ int enc_calls_per_output = 1; // Valid for libtoolame-dab
- if (aacEncInfo(encoder, &info) != AACENC_OK) {
- fprintf(stderr, "Unable to get the encoder info\n");
- return 1;
+ if (selected_encoder == encoder_selection_t::fdk_dabplus) {
+ enc_calls_per_output = (aot == AOT_DABPLUS_AAC_LC) ? sample_rate / 8000 : sample_rate / 16000;
}
- // 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 = 8*input_size + NUM_SAMPLES_PER_CALL;
+ int max_size = 8*input_buf.size() + NUM_SAMPLES_PER_CALL;
SampleQueue<uint8_t> queue(BYTES_PER_SAMPLE, channels, max_size);
@@ -608,8 +709,10 @@ int main(int argc, char *argv[])
}
// We'll use one of the tree possible inputs
+#if HAVE_ALSA
AlsaInputThreaded alsa_in_threaded(alsa_device, channels, sample_rate, queue);
AlsaInputDirect alsa_in_direct(alsa_device, channels, sample_rate);
+#endif
FileInput file_in(infile, raw_input, sample_rate);
#if HAVE_JACK
JackInput jack_in(jack_name, channels, sample_rate, queue);
@@ -652,6 +755,7 @@ int main(int argc, char *argv[])
}
}
#endif
+#if HAVE_ALSA
else if (drift_compensation) {
if (alsa_in_threaded.prepare() != 0) {
fprintf(stderr, "Alsa with drift compensation: preparation failed\n");
@@ -667,19 +771,37 @@ int main(int argc, char *argv[])
return 1;
}
}
+#else
+ else {
+ fprintf(stderr, "No input defined\n");
+ return 1;
+ }
+#endif
- int outbuf_size = subchannel_index*120;
- uint8_t zmqframebuf[ZMQ_HEADER_SIZE + 24*120];
- zmq_frame_header_t *zmq_frame_header = (zmq_frame_header_t*)zmqframebuf;
-
- uint8_t outbuf[24*120];
+ int outbuf_size;
+ std::vector<uint8_t> zmqframebuf;
+ std::vector<uint8_t> outbuf;
+ if (selected_encoder == encoder_selection_t::fdk_dabplus) {
+ outbuf_size = bitrate/8*120;
+ outbuf.resize(24*120);
+ zmqframebuf.resize(ZMQ_HEADER_SIZE + 24*120);
- unsigned char pad_buf[padlen + 1];
+ if(outbuf_size % 5 != 0) {
+ fprintf(stderr, "Warning: (outbuf_size mod 5) = %d\n", outbuf_size % 5);
+ }
+ }
+ else if (selected_encoder == encoder_selection_t::toolame_dab) {
+ outbuf_size = 4092;
+ outbuf.resize(outbuf_size);
+ zmqframebuf.resize(ZMQ_HEADER_SIZE + outbuf_size);
+ fprintf(stderr, "Setting outbuf size to %zu\n", outbuf.size());
- if(outbuf_size % 5 != 0) {
- fprintf(stderr, "(outbuf_size mod 5) = %d\n", outbuf_size % 5);
}
+ zmq_frame_header_t *zmq_frame_header = (zmq_frame_header_t*)&zmqframebuf[0];
+
+ unsigned char pad_buf[padlen + 1];
+
fprintf(stderr, "Starting encoding\n");
int retval = 0;
@@ -688,17 +810,9 @@ int main(int argc, char *argv[])
clock_gettime(CLOCK_MONOTONIC, &tp_next);
int calls = 0; // for checking
- while (1) {
- int in_identifier[] = {IN_AUDIO_DATA, IN_ANCILLRY_DATA};
- int out_identifier = OUT_BITSTREAM_DATA;
-
+ ssize_t read_bytes = 0;
+ do {
AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 };
- AACENC_InArgs in_args = { 0 };
- AACENC_OutArgs out_args = { 0 };
- void *in_ptr[2], *out_ptr;
- int in_size[2], in_elem_size[2];
- int out_size, out_elem_size;
-
// -------------- wait the right amount of time
if (drift_compensation || jack_name) {
@@ -757,25 +871,24 @@ int main(int argc, char *argv[])
}
// -------------- Read Data
- memset(outbuf, 0x00, outbuf_size);
- memset(input_buf, 0x00, input_size);
+ memset(&outbuf[0], 0x00, outbuf_size);
+ memset(&input_buf[0], 0x00, input_buf.size());
- ssize_t read;
if (infile) {
- read = file_in.read(input_buf, input_size);
- if (read < 0) {
+ read_bytes = file_in.read(&input_buf[0], input_buf.size());
+ if (read_bytes < 0) {
break;
}
- else if (read != input_size) {
+ else if (read_bytes != input_buf.size()) {
if (inFifoSilence && file_in.eof()) {
- memset(input_buf, 0, input_size);
- read = input_size;
- usleep((long)input_size * 1000000 /
+ memset(&input_buf[0], 0, input_buf.size());
+ read_bytes = input_buf.size();
+ usleep((long)input_buf.size() * 1000000 /
(BYTES_PER_SAMPLE * channels * sample_rate));
}
else {
fprintf(stderr, "Short file read !\n");
- break;
+ read_bytes = 0;
}
}
}
@@ -793,9 +906,9 @@ int main(int argc, char *argv[])
}
size_t overruns;
- read = queue.pop(input_buf, input_size, &overruns); // returns bytes
+ read_bytes = queue.pop(&input_buf[0], input_buf.size(), &overruns); // returns bytes
- if (read != input_size) {
+ if (read_bytes != input_buf.size()) {
status |= STATUS_UNDERRUN;
}
@@ -806,12 +919,12 @@ int main(int argc, char *argv[])
else {
vlc_in = &vlc_in_direct;
- read = vlc_in_direct.read(input_buf, input_size);
- if (read < 0) {
+ read_bytes = vlc_in_direct.read(&input_buf[0], input_buf.size());
+ if (read_bytes < 0) {
fprintf(stderr, "Detected fault in VLC input!\n");
break;
}
- else if (read != input_size) {
+ else if (read_bytes != input_buf.size()) {
fprintf(stderr, "Short VLC read !\n");
break;
}
@@ -823,16 +936,18 @@ int main(int argc, char *argv[])
}
#endif
else if (drift_compensation || jack_name) {
+#if HAVE_ALSA
if (drift_compensation && alsa_in_threaded.fault_detected()) {
fprintf(stderr, "Detected fault in alsa input!\n");
retval = 5;
break;
}
+#endif
size_t overruns;
- read = queue.pop(input_buf, input_size, &overruns); // returns bytes
+ read_bytes = queue.pop(&input_buf[0], input_buf.size(), &overruns); // returns bytes
- if (read != input_size) {
+ if (read_bytes != input_buf.size()) {
status |= STATUS_UNDERRUN;
}
@@ -841,16 +956,18 @@ int main(int argc, char *argv[])
}
}
else {
- read = alsa_in_direct.read(input_buf, input_size);
- if (read < 0) {
+#if HAVE_ALSA
+ read_bytes = alsa_in_direct.read(&input_buf[0], input_buf.size());
+ if (read_bytes < 0) {
break;
}
- else if (read != input_size) {
+ else if (read_bytes != input_buf.size()) {
fprintf(stderr, "Short alsa read !\n");
}
+#endif
}
- for (int i = 0; i < read; i+=4) {
+ for (int i = 0; i < read_bytes; i+=4) {
int16_t l = input_buf[i] | (input_buf[i+1] << 8);
int16_t r = input_buf[i+2] | (input_buf[i+3] << 8);
peak_left = MAX(peak_left, l);
@@ -876,50 +993,94 @@ int main(int argc, char *argv[])
measured_silence_ms = 0;
}
- // -------------- AAC Encoding
-
- int calculated_padlen = ret > 0 ? pad_buf[padlen] : 0;
-
-
- in_ptr[0] = input_buf;
- in_ptr[1] = pad_buf + (padlen - calculated_padlen); // offset due to unused PAD bytes
- in_size[0] = read;
- in_size[1] = calculated_padlen;
- in_elem_size[0] = BYTES_PER_SAMPLE;
- in_elem_size[1] = sizeof(uint8_t);
- in_args.numInSamples = input_size/BYTES_PER_SAMPLE;
- in_args.numAncBytes = calculated_padlen;
-
- in_buf.bufs = (void**)&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) {
- fprintf(stderr, "encoder error: EOF reached\n");
+ int numOutBytes = 0;
+ if (read_bytes and
+ selected_encoder == encoder_selection_t::fdk_dabplus) {
+ AACENC_InArgs in_args = { 0 };
+ AACENC_OutArgs out_args = { 0 };
+ // -------------- AAC Encoding
+ //
+ int in_identifier[] = {IN_AUDIO_DATA, IN_ANCILLRY_DATA};
+ int out_identifier = OUT_BITSTREAM_DATA;
+ const int calculated_padlen = ret > 0 ? pad_buf[padlen] : 0;
+ const int subchannel_index = bitrate / 8;
+
+ void *in_ptr[2], *out_ptr;
+ int in_size[2], in_elem_size[2];
+ int out_size, out_elem_size;
+
+ in_ptr[0] = &input_buf[0];
+ in_ptr[1] = pad_buf + (padlen - calculated_padlen); // offset due to unused PAD bytes
+ in_size[0] = read_bytes;
+ in_size[1] = calculated_padlen;
+ in_elem_size[0] = BYTES_PER_SAMPLE;
+ in_elem_size[1] = sizeof(uint8_t);
+ in_args.numInSamples = input_buf.size()/BYTES_PER_SAMPLE;
+ in_args.numAncBytes = calculated_padlen;
+
+ in_buf.bufs = (void**)&in_ptr;
+ in_buf.bufferIdentifiers = in_identifier;
+ in_buf.bufSizes = in_size;
+ in_buf.bufElSizes = in_elem_size;
+
+ out_ptr = &outbuf[0];
+ out_size = outbuf.size();
+ 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) {
+ fprintf(stderr, "encoder error: EOF reached\n");
+ break;
+ }
+ fprintf(stderr, "Encoding failed (%d)\n", err);
+ retval = 3;
break;
}
- fprintf(stderr, "Encoding failed (%d)\n", err);
- retval = 3;
- break;
+ calls++;
+
+ numOutBytes = out_args.numOutBytes;
+ }
+ else if (selected_encoder == encoder_selection_t::toolame_dab) {
+ const int calculated_padlen = ret > 0 ? pad_buf[padlen] : 0;
+ uint8_t *xpad_data = pad_buf + (padlen - calculated_padlen); // offset due to unused PAD bytes
+
+ short input_buffers[2][1152];
+
+ if (channels == 1) {
+ memcpy(input_buffers[0], &input_buf[0], 1152 * BYTES_PER_SAMPLE);
+ }
+ else if (channels == 2) {
+ for (int i = 0; i < 1152; i++) {
+ int16_t l = input_buf[4*i] | (input_buf[4*i+1] << 8);
+ int16_t r = input_buf[4*i+2] | (input_buf[4*i+3] << 8);
+
+ input_buffers[0][i] = l;
+ input_buffers[1][i] = r;
+ }
+ }
+ else {
+ fprintf(stderr, "INTERNAL ERROR! invalid number of channels\n");
+ }
+
+ if (read_bytes) {
+ numOutBytes = toolame_encode_frame(input_buffers, xpad_data, &outbuf[0], outbuf.size());
+ }
+ else {
+ numOutBytes = toolame_finish(&outbuf[0], outbuf.size());
+ }
}
- calls++;
/* Check if the encoder has generated output data */
- if (out_args.numOutBytes != 0)
- {
+ if (numOutBytes != 0 and
+ selected_encoder == encoder_selection_t::fdk_dabplus) {
+
// Our timing code depends on this
if (calls != enc_calls_per_output) {
fprintf(stderr, "INTERNAL ERROR! calls=%d"
@@ -932,6 +1093,7 @@ int main(int argc, char *argv[])
int row, col;
unsigned char buf_to_rs_enc[110];
unsigned char rs_enc[10];
+ const int subchannel_index = bitrate / 8;
for(row=0; row < subchannel_index; row++) {
for(col=0;col < 110; col++) {
buf_to_rs_enc[col] = outbuf[subchannel_index * col + row];
@@ -945,24 +1107,33 @@ int main(int argc, char *argv[])
}
}
+ numOutBytes = outbuf_size;
+ }
+
+ if (numOutBytes != 0) {
if (out_fh) {
- fwrite(outbuf, 1, outbuf_size, out_fh);
+ fwrite(&outbuf[0], 1, numOutBytes, out_fh);
}
else {
// ------------ ZeroMQ transmit
try {
zmq_frame_header->version = 1;
- zmq_frame_header->encoder = ZMQ_ENCODER_FDK;
- zmq_frame_header->datasize = outbuf_size;
+ if (selected_encoder == encoder_selection_t::fdk_dabplus) {
+ zmq_frame_header->encoder = ZMQ_ENCODER_FDK;
+ }
+ else if (selected_encoder == encoder_selection_t::toolame_dab) {
+ zmq_frame_header->encoder = ZMQ_ENCODER_TOOLAME;
+ }
+ zmq_frame_header->datasize = numOutBytes;
zmq_frame_header->audiolevel_left = peak_left;
zmq_frame_header->audiolevel_right = peak_right;
- assert(ZMQ_FRAME_SIZE(zmq_frame_header) <= NUMOF(zmqframebuf));
+ assert(ZMQ_FRAME_SIZE(zmq_frame_header) <= zmqframebuf.size());
memcpy(ZMQ_FRAME_DATA(zmq_frame_header),
- outbuf, outbuf_size);
+ &outbuf[0], numOutBytes);
- zmq_sock.send(zmqframebuf, ZMQ_FRAME_SIZE(zmq_frame_header),
+ zmq_sock.send(&zmqframebuf[0], ZMQ_FRAME_SIZE(zmq_frame_header),
ZMQ_DONTWAIT);
}
catch (zmq::error_t& e) {
@@ -977,7 +1148,10 @@ int main(int argc, char *argv[])
break;
}
}
+ }
+ if (numOutBytes != 0)
+ {
if (show_level) {
if (channels == 1) {
fprintf(stderr, "\rIn: [%-6s] %1s %1s %1s",
@@ -1013,7 +1187,8 @@ int main(int argc, char *argv[])
}
fflush(stdout);
- }
+ } while (read_bytes > 0);
+
fprintf(stderr, "\n");
if (out_fh) {
@@ -1023,7 +1198,9 @@ int main(int argc, char *argv[])
zmq_sock.close();
free_rs_char(rs_handler);
- aacEncClose(&encoder);
+ if (selected_encoder == encoder_selection_t::fdk_dabplus) {
+ aacEncClose(&encoder);
+ }
return retval;
}