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authorMatthias P. Braendli <matthias.braendli@mpb.li>2016-02-17 20:16:12 +0100
committerMatthias P. Braendli <matthias.braendli@mpb.li>2016-02-17 20:16:12 +0100
commitce5773ca73462a9319c16d0afc692eb38e35af8d (patch)
tree5725cb73db488bcc82ef287b388f169df02224a8 /src
parent5671aa4f0f5536de2fc58b6f89f01947f4f8f1c1 (diff)
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Many libtoolame-dab fixes, at least filesize is right now
Diffstat (limited to 'src')
-rw-r--r--src/dabplus-enc.cpp159
1 files changed, 108 insertions, 51 deletions
diff --git a/src/dabplus-enc.cpp b/src/dabplus-enc.cpp
index 8abc82a..be3e593 100644
--- a/src/dabplus-enc.cpp
+++ b/src/dabplus-enc.cpp
@@ -117,11 +117,17 @@ void usage(const char* name) {
" Drift compensation\n"
" -D, --drift-comp Enable ALSA/VLC sound card drift compensation.\n"
" Encoder parameters:\n"
- " -a, --dab Encode in DAB and not in DAB+.\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"
@@ -248,7 +254,7 @@ int prepare_aac_encoder(
int main(int argc, char *argv[])
{
- int bitrate = 64; //64kbps subchannel
+ int bitrate = 0; // 0 is default
int ch=0;
encoder_selection_t selected_encoder = encoder_selection_t::fdk_dabplus;
@@ -282,6 +288,9 @@ 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;
@@ -325,6 +334,8 @@ int main(int argc, char *argv[])
{"write-icy-text", required_argument, 0, 'w'},
{"aaclc", no_argument, 0, 0 },
{"dab", no_argument, 0, 'a'},
+ {"dabmode", no_argument, 0, 4 },
+ {"dabpsy", no_argument, 0, 5 },
{"drift-comp", no_argument, 0, 'D'},
{"fifo-silence", no_argument, 0, 3 },
{"help", no_argument, 0, 'h'},
@@ -368,7 +379,11 @@ 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':
selected_encoder = encoder_selection_t::toolame_dab;
@@ -475,6 +490,10 @@ int main(int argc, char *argv[])
}
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) {
@@ -489,6 +508,10 @@ int main(int argc, char *argv[])
}
}
else if (selected_encoder == encoder_selection_t::toolame_dab) {
+ if (bitrate == 0) {
+ bitrate = 192;
+ }
+
if ( ! (sample_rate == 24000 || sample_rate == 48000)) {
fprintf(stderr, "Invalid sample rate. Possible values are: 24000, 48000.\n");
return 1;
@@ -603,7 +626,28 @@ int main(int argc, char *argv[])
int err = toolame_init();
if (err == 0) {
- toolame_set_bitrate(bitrate);
+ err = toolame_set_samplerate(sample_rate);
+ }
+
+ if (err == 0) {
+ err = toolame_set_bitrate(bitrate);
+ }
+
+ 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) {
@@ -613,7 +657,7 @@ int main(int argc, char *argv[])
// TODO int toolame_set_pad(int pad_len);
- input_buf.resize(2 * 1152);
+ input_buf.resize(2 * 1152 * BYTES_PER_SAMPLE);
}
/* We assume that we need to call the encoder
@@ -711,22 +755,23 @@ int main(int argc, char *argv[])
outbuf_size = bitrate/8*120;
outbuf.resize(24*120);
zmqframebuf.resize(ZMQ_HEADER_SIZE + 24*120);
+
+ 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 = 3 * bitrate;
+ outbuf_size = 4092;
outbuf.resize(outbuf_size);
zmqframebuf.resize(ZMQ_HEADER_SIZE + outbuf_size);
+ fprintf(stderr, "Setting outbuf size to %zu\n", outbuf.size());
+
}
zmq_frame_header_t *zmq_frame_header = (zmq_frame_header_t*)&zmqframebuf[0];
-
unsigned char pad_buf[padlen + 1];
- if(outbuf_size % 5 != 0) {
- fprintf(stderr, "(outbuf_size mod 5) = %d\n", outbuf_size % 5);
- }
-
fprintf(stderr, "Starting encoding\n");
int retval = 0;
@@ -735,15 +780,9 @@ 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 };
- 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) {
@@ -805,22 +844,21 @@ int main(int argc, char *argv[])
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[0], input_buf.size());
- if (read < 0) {
+ read_bytes = file_in.read(&input_buf[0], input_buf.size());
+ if (read_bytes < 0) {
break;
}
- else if (read != input_buf.size()) {
+ else if (read_bytes != input_buf.size()) {
if (inFifoSilence && file_in.eof()) {
memset(&input_buf[0], 0, input_buf.size());
- read = 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;
}
}
}
@@ -838,9 +876,9 @@ int main(int argc, char *argv[])
}
size_t overruns;
- read = queue.pop(input_buf, input_buf.size(), &overruns); // returns bytes
+ read_bytes = queue.pop(input_buf, input_buf.size(), &overruns); // returns bytes
- if (read != input_buf.size()) {
+ if (read_bytes != input_buf.size()) {
status |= STATUS_UNDERRUN;
}
@@ -851,12 +889,12 @@ int main(int argc, char *argv[])
else {
vlc_in = &vlc_in_direct;
- read = vlc_in_direct.read(input_buf, input_buf.size());
- if (read < 0) {
+ read_bytes = vlc_in_direct.read(input_buf, input_buf.size());
+ if (read_bytes < 0) {
fprintf(stderr, "Detected fault in VLC input!\n");
break;
}
- else if (read != input_buf.size()) {
+ else if (read_bytes != input_buf.size()) {
fprintf(stderr, "Short VLC read !\n");
break;
}
@@ -875,9 +913,9 @@ int main(int argc, char *argv[])
}
size_t overruns;
- read = queue.pop(&input_buf[0], input_buf.size(), &overruns); // returns bytes
+ read_bytes = queue.pop(&input_buf[0], input_buf.size(), &overruns); // returns bytes
- if (read != input_buf.size()) {
+ if (read_bytes != input_buf.size()) {
status |= STATUS_UNDERRUN;
}
@@ -886,16 +924,16 @@ int main(int argc, char *argv[])
}
}
else {
- read = alsa_in_direct.read(&input_buf[0], input_buf.size());
- if (read < 0) {
+ read_bytes = alsa_in_direct.read(&input_buf[0], input_buf.size());
+ if (read_bytes < 0) {
break;
}
- else if (read != input_buf.size()) {
+ else if (read_bytes != input_buf.size()) {
fprintf(stderr, "Short alsa read !\n");
}
}
- 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);
@@ -922,17 +960,24 @@ int main(int argc, char *argv[])
}
int numOutBytes = 0;
- if (selected_encoder == encoder_selection_t::fdk_dabplus) {
+ 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;
+ in_size[0] = read_bytes;
in_size[1] = calculated_padlen;
in_elem_size[0] = BYTES_PER_SAMPLE;
in_elem_size[1] = sizeof(uint8_t);
@@ -975,20 +1020,27 @@ int main(int argc, char *argv[])
short input_buffers[2][1152];
if (channels == 1) {
- memcpy(input_buffers[0], &input_buf[0], 1152);
+ memcpy(input_buffers[0], &input_buf[0], 1152 * BYTES_PER_SAMPLE);
}
else if (channels == 2) {
- for (int ch = 0; ch < 2; ch++) {
- for (int i = 0; i < 1152; i++) {
- input_buffers[ch][i] = input_buf[2*i + ch];
- }
+ 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");
}
- toolame_encode_frame(input_buffers, xpad_data, &outbuf[0]);
+ if (read_bytes) {
+ numOutBytes = toolame_encode_frame(input_buffers, xpad_data, &outbuf[0], outbuf.size());
+ }
+ else {
+ numOutBytes = toolame_finish(&outbuf[0], outbuf.size());
+ }
}
/* Check if the encoder has generated output data */
@@ -1020,11 +1072,13 @@ int main(int argc, char *argv[])
assert(subchannel_index * col + row < outbuf_size);
}
}
+
+ numOutBytes = outbuf_size;
}
if (numOutBytes != 0) {
if (out_fh) {
- fwrite(&outbuf[0], 1, outbuf_size, out_fh);
+ fwrite(&outbuf[0], 1, numOutBytes, out_fh);
}
else {
// ------------ ZeroMQ transmit
@@ -1036,14 +1090,14 @@ int main(int argc, char *argv[])
else if (selected_encoder == encoder_selection_t::toolame_dab) {
zmq_frame_header->encoder = ZMQ_ENCODER_TOOLAME;
}
- zmq_frame_header->datasize = outbuf_size;
+ 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) <= zmqframebuf.size());
memcpy(ZMQ_FRAME_DATA(zmq_frame_header),
- &outbuf[0], outbuf_size);
+ &outbuf[0], numOutBytes);
zmq_sock.send(&zmqframebuf[0], ZMQ_FRAME_SIZE(zmq_frame_header),
ZMQ_DONTWAIT);
@@ -1099,7 +1153,8 @@ int main(int argc, char *argv[])
}
fflush(stdout);
- }
+ } while (read_bytes > 0);
+
fprintf(stderr, "\n");
if (out_fh) {
@@ -1109,7 +1164,9 @@ 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;
}