summaryrefslogtreecommitdiffstats
diff options
context:
space:
mode:
Diffstat
-rw-r--r--libtoolame-dab.sym2
-rw-r--r--libtoolame-dab/bitstream.c79
-rw-r--r--libtoolame-dab/common.h7
-rw-r--r--libtoolame-dab/toolame.c35
-rw-r--r--libtoolame-dab/toolame.h17
-rw-r--r--src/dabplus-enc.cpp159
6 files changed, 175 insertions, 124 deletions
diff --git a/libtoolame-dab.sym b/libtoolame-dab.sym
index 2e8a1f1..13d6fa0 100644
--- a/libtoolame-dab.sym
+++ b/libtoolame-dab.sym
@@ -1,8 +1,10 @@
toolame_init
+toolame_finish
toolame_enable_downmix_stereo
toolame_enable_byteswap
toolame_set_channel_mode
toolame_set_psy_model
toolame_set_bitrate
+toolame_set_samplerate
toolame_set_pad
toolame_encode_frame
diff --git a/libtoolame-dab/bitstream.c b/libtoolame-dab/bitstream.c
index 410ef5b..d426ffc 100644
--- a/libtoolame-dab/bitstream.c
+++ b/libtoolame-dab/bitstream.c
@@ -42,89 +42,35 @@
/* in conformity of the norme ETS 300 401 http://www.etsi.org */
/* see toollame.c */
int minimum = MINIMUM;
-int refill_buffer (Bit_stream_struc * bs)
-{
- register int i = bs->buf_size - 2 - bs->buf_byte_idx;
- register unsigned long n = 1;
- register int index = 0;
- char val[2];
-
- while ((i >= 0) && (!bs->eob)) {
-
- if (bs->format == BINARY)
- n = fread (&bs->buf[i--], sizeof (unsigned char), 1, bs->pt);
-
- else {
- while ((index < 2) && n) {
- n = fread (&val[index], sizeof (char), 1, bs->pt);
- switch (val[index]) {
- case 0x30:
- case 0x31:
- case 0x32:
- case 0x33:
- case 0x34:
- case 0x35:
- case 0x36:
- case 0x37:
- case 0x38:
- case 0x39:
- case 0x41:
- case 0x42:
- case 0x43:
- case 0x44:
- case 0x45:
- case 0x46:
- index++;
- break;
- default:
- break;
- }
- }
-
- if (val[0] <= 0x39)
- bs->buf[i] = (val[0] - 0x30) << 4;
- else
- bs->buf[i] = (val[0] - 0x37) << 4;
- if (val[1] <= 0x39)
- bs->buf[i--] |= (val[1] - 0x30);
- else
- bs->buf[i--] |= (val[1] - 0x37);
- index = 0;
- }
-
- if (!n) {
- bs->eob = i + 1;
- }
-
- }
- return 0;
-}
/* empty the buffer to the output device when the buffer becomes full */
void empty_buffer (Bit_stream_struc * bs, int minimum)
{
- int i;
-
- if (bs->pt) {
- for (i = bs->buf_size - 1; i >= minimum; i--)
- fwrite (&bs->buf[i], sizeof (unsigned char), 1, bs->pt);
+ int j = 0;
+ for (int i = bs->buf_size - 1; i >= minimum; i--) {
+ if (j >= bs->output_buffer_size) {
+ fprintf(stderr, "Warning: libtoolame output buffer too small (%d vs %d)!\n",
+ bs->output_buffer_size, bs->buf_size - minimum);
+ break;
+ }
- fflush (bs->pt); /* NEW SS to assist in debugging */
+ bs->output_buffer[j] = bs->buf[i];
+ j++;
}
+ bs->output_buffer_written = j;
- for (i = minimum - 1; i >= 0; i--)
+ for (int i = minimum - 1; i >= 0; i--) {
bs->buf[bs->buf_size - minimum + i] = bs->buf[i];
+ }
bs->buf_byte_idx = bs->buf_size - 1 - minimum;
bs->buf_bit_idx = 8;
-
}
/* open the device to write the bit stream into it */
void open_bit_stream_w (Bit_stream_struc * bs, int size)
{
- bs->pt = NULL; // we're not using file output
alloc_buffer (bs, size);
bs->buf_byte_idx = size - 1;
bs->buf_bit_idx = 8;
@@ -139,7 +85,6 @@ void close_bit_stream_w (Bit_stream_struc * bs)
{
putbits (bs, 0, 7);
empty_buffer (bs, bs->buf_byte_idx + 1);
- if (bs->pt) fclose(bs->pt);
desalloc_buffer (bs);
}
diff --git a/libtoolame-dab/common.h b/libtoolame-dab/common.h
index 8551483..0a19af6 100644
--- a/libtoolame-dab/common.h
+++ b/libtoolame-dab/common.h
@@ -142,9 +142,10 @@ frame_info;
typedef struct bit_stream_struc
{
- FILE *pt; /* pointer to bit stream device */
- void *zmq_sock; /* zmq socket */
- int zmq_framesize; /* zmq frame size */
+ unsigned char *output_buffer; /* output buffer */
+ int output_buffer_size;
+ int output_buffer_written;
+
unsigned char *buf; /* bit stream buffer */
int buf_size; /* size of buffer (in number of bytes) */
long totbit; /* bit counter of bit stream */
diff --git a/libtoolame-dab/toolame.c b/libtoolame-dab/toolame.c
index a626552..bea186c 100644
--- a/libtoolame-dab/toolame.c
+++ b/libtoolame-dab/toolame.c
@@ -154,6 +154,19 @@ int toolame_init(void)
return 0;
}
+int toolame_finish(
+ unsigned char *output_buffer,
+ size_t output_buffer_size)
+{
+ bs.output_buffer = output_buffer;
+ bs.output_buffer_size = output_buffer_size;
+ bs.output_buffer_written = 0;
+
+ close_bit_stream_w(&bs);
+
+ return bs.output_buffer_written;
+}
+
int toolame_enable_downmix_stereo(void)
{
glopts.downmix = TRUE;
@@ -233,6 +246,17 @@ int toolame_set_bitrate(int brate)
return err;
}
+int toolame_set_samplerate(long sample_rate)
+{
+ int s_freq = SmpFrqIndex(sample_rate, &header.version);
+ if (s_freq < 0) {
+ return s_freq;
+ }
+
+ header.sampling_frequency = s_freq;
+ return 0;
+}
+
int toolame_set_pad(int pad_len)
{
header.dab_length = pad_len;
@@ -247,13 +271,20 @@ int toolame_set_pad(int pad_len)
int toolame_encode_frame(
short buffer[2][1152],
unsigned char *xpad_data,
- unsigned char *output_buffer)
+ unsigned char *output_buffer,
+ size_t output_buffer_size)
{
extern int minimum;
const int nch = frame.nch;
const int error_protection = header.error_protection;
+ bs.output_buffer = output_buffer;
+ bs.output_buffer_size = output_buffer_size;
+ bs.output_buffer_written = 0;
+
+#ifdef REFERENCECODE
short *win_buf[2] = {&buffer[0][0], &buffer[1][0]};
+#endif
int adb = available_bits (&header, &glopts);
int lg_frame = adb / 8;
@@ -503,6 +534,8 @@ int toolame_encode_frame(
else {
putbits (&bs, 0, 16); // FPAD is all-zero
}
+
+ return bs.output_buffer_written;
}
void smr_dump(double smr[2][SBLIMIT], int nch) {
diff --git a/libtoolame-dab/toolame.h b/libtoolame-dab/toolame.h
index d7f8198..1c29e24 100644
--- a/libtoolame-dab/toolame.h
+++ b/libtoolame-dab/toolame.h
@@ -7,6 +7,13 @@
/* Initialise toolame encoding library. */
int toolame_init(void);
+/* Finish encoding the pending samples.
+ * Returns number of bytes written to output_buffer
+ */
+int toolame_finish(
+ unsigned char *output_buffer,
+ size_t output_buffer_size);
+
int toolame_enable_downmix_stereo(void);
int toolame_enable_byteswap(void);
@@ -20,13 +27,19 @@ int toolame_set_psy_model(int new_model);
int toolame_set_bitrate(int brate);
+/* Set sample rate in Hz */
+int toolame_set_samplerate(long sample_rate);
+
/* Enable PAD insertion from the specified file with length */
int toolame_set_pad(int pad_len);
+/* Encodes one frame. Returns number of bytes written to output_buffer
+ */
int toolame_encode_frame(
short buffer[2][1152],
- unsigned char* xpad_data,
- unsigned char *output_buffer);
+ unsigned char *xpad_data,
+ unsigned char *output_buffer,
+ size_t output_buffer_size);
#endif // __TOOLAME_H_
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;
}
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-25 05:06:17 +0000