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-rw-r--r--src/AlsaInput.h18
-rw-r--r--src/FileInput.cpp108
-rw-r--r--src/FileInput.h61
-rw-r--r--src/dabplus-enc-file-zmq.c620
-rw-r--r--src/dabplus-enc.cpp (renamed from src/dabplus-enc-alsa-zmq.cpp)197
5 files changed, 316 insertions, 688 deletions
diff --git a/src/AlsaInput.h b/src/AlsaInput.h
index 0d454a3..bd3a800 100644
--- a/src/AlsaInput.h
+++ b/src/AlsaInput.h
@@ -27,18 +27,19 @@
#include "SampleQueue.h"
-using namespace std;
-
// 16 bits per sample is fine for now
#define BYTES_PER_SAMPLE 2
// How many samples we insert into the queue each call
#define NUM_SAMPLES_PER_CALL 10 // 10 samples @ 32kHz = 3.125ms
+/* Common functionality for the direct alsa input and the
+ * threaded alsa input
+ */
class AlsaInput
{
public:
- AlsaInput(const string& alsa_dev,
+ AlsaInput(const std::string& alsa_dev,
unsigned int channels,
unsigned int rate) :
m_alsa_dev(alsa_dev),
@@ -57,12 +58,10 @@ class AlsaInput
/* Prepare the audio input */
int prepare();
- virtual void start() = 0;
-
protected:
ssize_t m_read(uint8_t* buf, snd_pcm_uframes_t length);
- string m_alsa_dev;
+ std::string m_alsa_dev;
unsigned int m_channels;
unsigned int m_rate;
@@ -75,13 +74,11 @@ class AlsaInput
class AlsaInputDirect : public AlsaInput
{
public:
- AlsaInputDirect(const string& alsa_dev,
+ AlsaInputDirect(const std::string& alsa_dev,
unsigned int channels,
unsigned int rate) :
AlsaInput(alsa_dev, channels, rate) { }
- virtual void start() { };
-
/* Read length Bytes from from the alsa device.
* length must be a multiple of channels * bytes_per_sample.
*
@@ -97,7 +94,7 @@ class AlsaInputDirect : public AlsaInput
class AlsaInputThreaded : public AlsaInput
{
public:
- AlsaInputThreaded(const string& alsa_dev,
+ AlsaInputThreaded(const std::string& alsa_dev,
unsigned int channels,
unsigned int rate,
SampleQueue<uint8_t>& queue) :
@@ -115,6 +112,7 @@ class AlsaInputThreaded : public AlsaInput
}
}
+ /* Start the ALSA thread that fills the queue */
virtual void start();
bool fault_detected() { return m_fault; };
diff --git a/src/FileInput.cpp b/src/FileInput.cpp
new file mode 100644
index 0000000..c8023dd
--- /dev/null
+++ b/src/FileInput.cpp
@@ -0,0 +1,108 @@
+/* ------------------------------------------------------------------
+ * Copyright (C) 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 "FileInput.h"
+#include "wavreader.h"
+#include <cstring>
+#include <cstdio>
+
+#include <stdint.h>
+
+int FileInput::prepare(void)
+{
+ if (m_raw_input) {
+ if (m_filename && strcmp(m_filename, "-")) {
+ m_in_fh = fopen(m_filename, "rb");
+ if (!m_in_fh) {
+ fprintf(stderr, "Can't open input file!\n");
+ return 1;
+ }
+ }
+ else {
+ m_in_fh = stdin;
+ }
+ }
+ else {
+ int bits_per_sample = 0;
+ int channels = 0;
+ int wav_format = 0;
+ int sample_rate = 0;
+
+ m_wav = wav_read_open(m_filename);
+ if (!m_wav) {
+ fprintf(stderr, "Unable to open wav file %s\n", m_filename);
+ return 1;
+ }
+ if (!wav_get_header(m_wav, &wav_format, &channels, &sample_rate,
+ &bits_per_sample, NULL)) {
+ fprintf(stderr, "Bad wav file %s\n", m_filename);
+ return 1;
+ }
+ if (wav_format != 1) {
+ fprintf(stderr, "Unsupported WAV format %d\n", wav_format);
+ return 1;
+ }
+ if (bits_per_sample != 16) {
+ fprintf(stderr, "Unsupported WAV sample depth %d\n", bits_per_sample);
+ return 1;
+ }
+ if (channels != 2) {
+ fprintf(stderr, "Unsupported WAV channels %d\n", channels);
+ return 1;
+ }
+ if (m_sample_rate != sample_rate) {
+ fprintf(stderr,
+ "WAV sample rate %d doesn't correspond to desired sample rate %d\n",
+ sample_rate, m_sample_rate);
+ return 1;
+ }
+ }
+
+ return 0;
+}
+
+ssize_t FileInput::read(uint8_t* buf, size_t length)
+{
+ ssize_t pcmread;
+
+ if (m_raw_input) {
+ if (fread(buf, length, 1, m_in_fh) == 1) {
+ pcmread = length;
+ }
+ else {
+ fprintf(stderr, "Unable to read from input!\n");
+ return 0;
+ }
+ }
+ else {
+ pcmread = wav_read_data(m_wav, buf, length);
+ }
+
+ return pcmread;
+}
+
+FileInput::~FileInput()
+{
+ if (m_raw_input && m_in_fh) {
+ fclose(m_in_fh);
+ }
+ else if (m_wav) {
+ wav_read_close(m_wav);
+ }
+}
+
diff --git a/src/FileInput.h b/src/FileInput.h
new file mode 100644
index 0000000..9330577
--- /dev/null
+++ b/src/FileInput.h
@@ -0,0 +1,61 @@
+/* ------------------------------------------------------------------
+ * Copyright (C) 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.
+ * -------------------------------------------------------------------
+ */
+
+#ifndef _FILE_INPUT_H_
+#define _FILE_INPUT_H_
+
+#include <stdint.h>
+#include <stdio.h>
+
+class FileInput
+{
+ public:
+ FileInput(const char* filename,
+ bool raw_input,
+ int sample_rate) :
+ m_filename(filename),
+ m_raw_input(raw_input),
+ m_sample_rate(sample_rate) { }
+
+ ~FileInput();
+
+ /* Open the file and prepare the wav decoder.
+ *
+ * Returns nonzero on error
+ */
+ int prepare(void);
+
+ /* Read length bytes into buf.
+ *
+ * Returns the number of bytes read.
+ */
+ ssize_t read(uint8_t* buf, size_t length);
+
+ protected:
+ const char* m_filename;
+ bool m_raw_input;
+ int m_sample_rate;
+
+ /* handle to the wav reader */
+ void *m_wav;
+
+ FILE* m_in_fh;
+};
+
+#endif
+
diff --git a/src/dabplus-enc-file-zmq.c b/src/dabplus-enc-file-zmq.c
deleted file mode 100644
index 1bfae12..0000000
--- a/src/dabplus-enc-file-zmq.c
+++ /dev/null
@@ -1,620 +0,0 @@
-/* ------------------------------------------------------------------
- * Copyright (C) 2011 Martin Storsjo
- * Copyright (C) 2013,2014 Matthias P. Braendli
- * Copyright (C) 2014 CSP Innovazione nelle ICT s.c.a r.l.
- * http://rd.csp.it/
- *
- * http://opendigitalradio.org
- *
- * This program is free software: you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as
- * published by the Free Software Foundation, either version 3 of the
- * License, or (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program If not, see <http://www.gnu.org/licenses/>.
- * -------------------------------------------------------------------
- */
-
-#include <stdio.h>
-#include <stdint.h>
-#include <string.h>
-#include <unistd.h>
-#include <stdlib.h>
-#include <getopt.h>
-#include <zmq.h>
-#include <assert.h>
-#include "libAACenc/include/aacenc_lib.h"
-#include "wavreader.h"
-#include "encryption.h"
-#include "utils.h"
-
-#include <fec.h>
-#include <sys/types.h>
-#include <sys/stat.h>
-#include <fcntl.h>
-#include <errno.h>
-
-#include "contrib/lib_crc.h"
-
-void usage(const char* name) {
- fprintf(stderr,
- "dabplus-enc-file-zmq %s is a HE-AACv2 encoder for DAB+\n"
- "based on fdk-aac-dabplus that can read from a file\n"
- "or pipe source and encode to a ZeroMQ output for ODR-DabMux,\n"
- "a file or standard output.\n"
- "\n"
- "It includes PAD (DLS and MOT Slideshow) support by http://rd.csp.it\n"
- "to be used with mot-encoder\n"
- "\n"
- " http://opendigitalradio.org\n"
- "\nUsage:\n"
- "%s [OPTION...]\n",
-#if defined(GITVERSION)
- GITVERSION
-#else
- PACKAGE_VERSION
-#endif
- , name);
-
- fprintf(stderr,
- " -b, --bitrate={ 8, 16, ..., 192 } Output bitrate in kbps. Must be 8 multiple.\n"
- " -i, --input=FILENAME Input filename (default: stdin).\n"
- " -o, --output=URI Output zmq uri or filename.\n"
- " Use '-' for standard output,\n"
- " 'tcp://odr-dabmux-host:9000' for ZeroMQ.\n"
- " Protocols supported: tcp, pgm, epgm\n"
- " -a, --afterburner Turn on AAC encoder quality increaser.\n"
- " -p, --pad=BYTES Set PAD size in bytes.\n"
- " -P, --pad-fifo=FILENAME Set PAD data input fifo name (default: /tmp/pad.fifo).\n"
- " -f, --format={ wav, raw } Set input file format (default: wav).\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"
- " -k, --secret-key=FILE Set the secret key for encryption.\n"
- " -l, --level Show level indication.\n"
- //" -v, --verbose=LEVEL Set verbosity level.\n"
- "\n"
- "Only the tcp:// zeromq transport has been tested until now.\n"
-
- );
-
-}
-
-
-#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;
- const char *infile = NULL;
- const char *outuri = NULL;
- FILE *in_fh;
- FILE *out_fh;
- void *wav;
- int wav_format, bits_per_sample, sample_rate=48000, channels=2;
- uint8_t* input_buf;
- int16_t* convert_buf;
-
- /* Keep track of peaks */
- int peak_left = 0;
- int peak_right = 0;
-
- void *rs_handler = NULL;
- int aot = AOT_DABPLUS_AAC_LC;
- int afterburner = 0, raw_input=0;
- HANDLE_AACENCODER handle;
- CHANNEL_MODE mode;
- AACENC_InfoStruct info = { 0 };
-
- char* pad_fifo = "/tmp/pad.fifo";
- int pad_fd;
- unsigned char pad_buf[128];
- int padlen;
-
- void *zmq_context = zmq_ctx_new();
- void *zmq_sock = NULL;
-
- int show_level = 0;
-
- /* Data for ZMQ CURVE authentication */
- char* keyfile = NULL;
- char secretkey[CURVE_KEYLEN+1];
-
- const struct option longopts[] = {
- {"bitrate", required_argument, 0, 'b'},
- {"input", required_argument, 0, 'i'},
- {"output", required_argument, 0, 'o'},
- {"format", required_argument, 0, 'f'},
- {"rate", required_argument, 0, 'r'},
- {"channels", required_argument, 0, 'c'},
- {"pad", required_argument, 0, 'p'},
- {"pad-fifo", required_argument, 0, 'P'},
- {"secret-key", required_argument, 0, 'k'},
- {"afterburner", no_argument, 0, 'a'},
- {"help", no_argument, 0, 'h'},
- {"level", no_argument, 0, 'l'},
- {0,0,0,0},
- };
-
- if (argc == 1) {
- usage(argv[0]);
- return 0;
- }
-
- int index;
- while(ch != -1) {
- ch = getopt_long(argc, argv, "tlhab:c:i:k:o:r:f:p:P:", longopts, &index);
- switch (ch) {
- case 'f':
- if(strcmp(optarg, "raw")==0) {
- raw_input = 1;
- } else if(strcmp(optarg, "wav")!=0)
- usage(argv[0]);
- 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 'i':
- infile = optarg;
- break;
- case 'k':
- keyfile = optarg;
- break;
- case 'o':
- outuri = optarg;
- break;
- case 'p':
- padlen = atoi(optarg);
- break;
- case 'P':
- pad_fifo = optarg;
- break;
- case 'l':
- show_level = 1;
- 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;
- }
- if(padlen != 0) {
- int flags;
- if (mkfifo(pad_fifo, S_IWUSR | S_IRUSR | S_IRGRP | S_IROTH) != 0) {
- if (errno != EEXIST) {
- fprintf(stderr, "Can't create pad file: %d!\n", errno);
- return 1;
- }
- }
- pad_fd = open(pad_fifo, O_RDONLY | O_NONBLOCK);
- if (pad_fd == -1) {
- fprintf(stderr, "Can't open pad file!\n");
- return 1;
- }
- flags = fcntl(pad_fd, F_GETFL, 0);
- if (fcntl(pad_fd, F_SETFL, flags | O_NONBLOCK)) {
- fprintf(stderr, "Can't set non-blocking mode in pad file!\n");
- return 1;
- }
- }
-
- if(raw_input) {
- if(infile && strcmp(infile, "-")) {
- in_fh = fopen(infile, "rb");
- if(!in_fh) {
- fprintf(stderr, "Can't open input file!\n");
- return 1;
- }
- } else {
- in_fh = stdin;
- }
- } else {
- wav = wav_read_open(infile);
- if (!wav) {
- fprintf(stderr, "Unable to open wav file %s\n", infile);
- return 1;
- }
- if (!wav_get_header(wav, &wav_format, &channels, &sample_rate, &bits_per_sample, NULL)) {
- fprintf(stderr, "Bad wav file %s\n", infile);
- return 1;
- }
- if (wav_format != 1) {
- fprintf(stderr, "Unsupported WAV format %d\n", wav_format);
- return 1;
- }
- if (bits_per_sample != 16) {
- fprintf(stderr, "Unsupported WAV sample depth %d\n", bits_per_sample);
- return 1;
- }
- if (channels > 2) {
- fprintf(stderr, "Unsupported WAV channels %d\n", channels);
- return 1;
- }
- }
-
- if (outuri) {
- if (strcmp(outuri, "-") == 0) {
- out_fh = stdout;
- }
- else if (strncmp(outuri, "file://", 7) == 0) {
- out_fh = fopen(&outuri[7], "wb");
-
- if(!out_fh) {
- fprintf(stderr, "Can't open output file!\n");
- return 1;
- }
- }
- else if ((strncmp(outuri, "tcp://", 6) == 0) ||
- (strncmp(outuri, "pgm://", 6) == 0) ||
- (strncmp(outuri, "epgm://", 7) == 0)) {
- 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 (keyfile) {
- fprintf(stderr, "Enabling encryption\n");
-
- int rc = readkey(keyfile, secretkey);
- if (rc) {
- fprintf(stderr, "Error reading secret key\n");
- return 2;
- }
-
- const int yes = 1;
- rc = zmq_setsockopt(zmq_sock, ZMQ_CURVE_SERVER,
- &yes, sizeof(yes));
- if (rc) {
- fprintf(stderr, "Error: %s\n", zmq_strerror(errno));
- return 2;
- }
-
- rc = zmq_setsockopt(zmq_sock, ZMQ_CURVE_SECRETKEY,
- secretkey, CURVE_KEYLEN);
- if (rc) {
- fprintf(stderr, "Error: %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 {
- out_fh = fopen(outuri, "wb");
-
- if(!out_fh) {
- fprintf(stderr, "Can't open output file!\n");
- return 1;
- }
- }
- } else {
- fprintf(stderr, "Output URI not defined\n");
- return 1;
- }
-
-
- switch (channels) {
- case 1: mode = MODE_1; break;
- case 2: mode = MODE_2; break;
- default:
- fprintf(stderr, "Unsupported channels number %d\n", channels);
- return 1;
- }
-
-
- if (aacEncOpen(&handle, 0x01|0x02|0x04, channels) != AACENC_OK) {
- fprintf(stderr, "Unable to open encoder\n");
- return 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");
- return 1;
- }
- if (aacEncoder_SetParam(handle, AACENC_SAMPLERATE, sample_rate) != AACENC_OK) {
- fprintf(stderr, "Unable to set the samplerate\n");
- return 1;
- }
- if (aacEncoder_SetParam(handle, AACENC_CHANNELMODE, mode) != AACENC_OK) {
- fprintf(stderr, "Unable to set the channel mode\n");
- return 1;
- }
- if (aacEncoder_SetParam(handle, AACENC_CHANNELORDER, 1) != AACENC_OK) {
- fprintf(stderr, "Unable to set the wav channel order\n");
- return 1;
- }
- if (aacEncoder_SetParam(handle, AACENC_GRANULE_LENGTH, 960) != AACENC_OK) {
- fprintf(stderr, "Unable to set the granule length\n");
- return 1;
- }
- if (aacEncoder_SetParam(handle, AACENC_TRANSMUX, TT_DABPLUS) != AACENC_OK) {
- fprintf(stderr, "Unable to set the RAW transmux\n");
- return 1;
- }
-
- /*if (aacEncoder_SetParam(handle, AACENC_BITRATEMODE, 7 *AACENC_BR_MODE_SFR*) != AACENC_OK) {
- fprintf(stderr, "Unable to set the bitrate mode\n");
- return 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");
- return 1;
- }
- if (aacEncoder_SetParam(handle, AACENC_AFTERBURNER, afterburner) != AACENC_OK) {
- fprintf(stderr, "Unable to set the afterburner mode\n");
- return 1;
- }
- if (aacEncoder_SetParam(handle, AACENC_ANCILLARY_BITRATE, 0) != AACENC_OK) {
- fprintf(stderr, "Unable to set the ancillary bitrate\n");
- return 1;
- }
- if (aacEncEncode(handle, NULL, NULL, NULL, NULL) != AACENC_OK) {
- fprintf(stderr, "Unable to initialize the encoder\n");
- return 1;
- }
- if (aacEncInfo(handle, &info) != AACENC_OK) {
- fprintf(stderr, "Unable to get the encoder info\n");
- return 1;
- }
-
- fprintf(stderr, "DAB+ Encoding: framelen=%d\n", info.frameLength);
-
- int input_size = channels*2*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");
- return 0;
- }
-
- int outbuf_size = subchannel_index*120;
- uint8_t zmqframebuf[2048];
- struct zmq_frame_header_t *zmq_frame_header =
- (struct zmq_frame_header_t*)zmqframebuf;
-
- uint8_t outbuf[2048];
-
- 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, IN_ANCILLRY_DATA};
- int in_size[2], in_elem_size[2];
- int out_identifier = OUT_BITSTREAM_DATA;
- int out_size, out_elem_size;
- int pcmread=0, i, ret;
- int send_error;
- void *in_ptr[2], *out_ptr;
- AACENC_ERROR err;
-
- // Read data from the PAD fifo
- if (padlen != 0) {
- ret = read(pad_fd, pad_buf, padlen);
- }
- else {
- ret = 0;
- }
-
-
- if(ret < 0 && errno == EAGAIN) {
- // If this condition passes, there is no data to be read
- in_buf.numBufs = 1; // Samples;
- }
- else if(ret >= 0) {
- // Otherwise, you're good to go and buffer should contain "count" bytes.
- in_buf.numBufs = 2; // Samples + Data;
- if (ret > 0)
- fprintf(stderr, "p");
- }
- else {
- // Some other error occurred during read.
- fprintf(stderr, "Unable to read from PAD!\n");
- break;
- }
-
- if(raw_input) {
- if(fread(input_buf, input_size, 1, in_fh) == 1) {
- pcmread = input_size;
- } else {
- fprintf(stderr, "Unable to read from input!\n");
- break;
- }
- } else {
- pcmread = wav_read_data(wav, input_buf, input_size);
- }
-
- for (i = 0; i < pcmread/2; i++) {
- const uint8_t* in = &input_buf[2*i];
- convert_buf[i] = in[0] | (in[1] << 8);
- }
-
- for (i = 0; i < pcmread/2; i+=2) {
- peak_left = MAX(peak_left, convert_buf[i]);
- peak_right = MAX(peak_right, convert_buf[i+1]);
- }
-
- if (pcmread <= 0) {
- in_args.numInSamples = -1;
- } else {
- in_ptr[0] = convert_buf;
- in_ptr[1] = pad_buf;
- in_size[0] = pcmread;
- in_size[1] = padlen;
-
- in_elem_size[0] = 2;
- in_elem_size[1] = sizeof(UCHAR);
-
- in_args.numInSamples = pcmread/2;
- in_args.numAncBytes = padlen;
-
- //in_buf.numBufs = 2; // Samples + Data
-
- 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;
-
- 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");
- return 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);
- }
- }
-
- if (zmq_sock) {
- zmq_frame_header->version = 1;
- zmq_frame_header->encoder = ZMQ_ENCODER_FDK;
- zmq_frame_header->datasize = outbuf_size;
- zmq_frame_header->audiolevel_left = peak_left;
- zmq_frame_header->audiolevel_right = peak_right;
-
- memcpy(ZMQ_FRAME_DATA(zmq_frame_header),
- outbuf, outbuf_size);
-
- send_error = zmq_send(zmq_sock, zmqframebuf,
- ZMQ_FRAME_SIZE(zmq_frame_header), ZMQ_DONTWAIT);
- if (send_error < 0) {
- fprintf(stderr, "ZeroMQ send failed! %s\n", zmq_strerror(errno));
- send_error_count ++;
- }
- }
- else {
- fwrite(outbuf, 1, /*out_args.numOutBytes*/ outbuf_size, out_fh);
- }
-
- 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 (show_level && out_args.numOutBytes + row*10 == outbuf_size) {
- fprintf(stderr, "\rIn: [%6s|%-6s]",
- level(0, &peak_left),
- level(1, &peak_right));
- }
-
- frame++;
- }
- fprintf(stderr, "\n");
-
- free(input_buf);
- free(convert_buf);
- if(raw_input) {
- fclose(in_fh);
- } else {
- wav_read_close(wav);
- }
-
- if (zmq_sock) {
- zmq_close(zmq_sock);
- zmq_ctx_term(zmq_context);
- }
- else {
- fclose(out_fh);
- }
-
- free_rs_char(rs_handler);
-
- aacEncClose(&handle);
-
- return 0;
-}
-
diff --git a/src/dabplus-enc-alsa-zmq.cpp b/src/dabplus-enc.cpp
index 699a256..82780d5 100644
--- a/src/dabplus-enc-alsa-zmq.cpp
+++ b/src/dabplus-enc.cpp
@@ -18,15 +18,16 @@
*/
#include "AlsaInput.h"
+#include "FileInput.h"
#include "SampleQueue.h"
#include "zmq.hpp"
extern "C" {
#include "encryption.h"
#include "utils.h"
+#include "wavreader.h"
}
-
#include <string>
#include <getopt.h>
#include <cstdio>
@@ -47,8 +48,8 @@ using namespace std;
void usage(const char* name) {
fprintf(stderr,
- "dabplus-enc-alsa-zmq %s is a HE-AACv2 encoder for DAB+\n"
- "based on fdk-aac-dabplus that can read from a ALSA source\n"
+ "dabplus-enc %s is a HE-AACv2 encoder for DAB+\n"
+ "based on fdk-aac-dabplus that can read from a ALSA or file source\n"
"and encode to a ZeroMQ output for ODR-DabMux.\n"
"\n"
"The -D option enables experimental sound card clock drift compensation.\n"
@@ -69,7 +70,7 @@ void usage(const char* name) {
"\n"
" http://opendigitalradio.org\n"
"\nUsage:\n"
- "%s [OPTION...]\n",
+ "%s (-i file|-d alsa_device) [OPTION...]\n",
#if defined(GITVERSION)
GITVERSION
#else
@@ -77,21 +78,27 @@ void usage(const char* name) {
#endif
, name);
fprintf(stderr,
- " -b, --bitrate={ 8, 16, ..., 192 } Output bitrate in kbps. Must be 8 multiple.\n"
+ " For the alsa input:\n"
+ " -d, --device=alsa_device Set ALSA input device (default: \"default\").\n"
" -D, --drift-comp Enable ALSA sound card drift compensation.\n"
- //" -i, --input=FILENAME Input filename (default: stdin).\n"
+ " For the file input:\n"
+ " -i, --input=FILENAME Input filename (default: stdin).\n"
+ " -f, --format={ wav, raw } Set input file format (default: wav).\n"
+ " Encoder parameters:\n"
+ " -b, --bitrate={ 8, 16, ..., 192 } Output bitrate in kbps. Must be 8 multiple.\n"
" -o, --output=URI Output zmq uri. (e.g. 'tcp://*:9000')\n"
+ " -or- Output file uri. (e.g. 'file.dab')\n"
+ " -or- a single dash '-' to denote stdout\n"
" -a, --afterburner Turn on AAC encoder quality increaser.\n"
" -p, --pad=BYTES Set PAD size in bytes.\n"
" -P, --pad-fifo=FILENAME Set PAD data input fifo name (default: /tmp/pad.fifo).\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"
- " -k, --secret-key=FILE Set the secret key for encryption.\n"
- " -l, --level Show level indication.\n"
- //" -V, --version Print version and exit.\n"
+ " -c, --channels={ 1, 2 } Nb of input channels (default: 2).\n"
+ " -r, --rate={ 32000, 48000 } Input sample rate (default: 48000).\n"
+ " -k, --secret-key=FILE Enable ZMQ encryption with the given secret key.\n"
+ " -l, --level Show peak audio level indication.\n"
"\n"
- "Only the tcp:// zeromq transport has been tested until now.\n"
+ "Only the tcp:// zeromq transport has been tested until now,\n"
+ " but epgm:// and pgm:// are also accepted\n"
);
}
@@ -196,7 +203,17 @@ int main(int argc, char *argv[])
{
int subchannel_index = 8; //64kbps subchannel
int ch=0;
- const char *alsa_device = "default";
+
+ // For the ALSA input
+ const char *alsa_device = NULL;
+
+ // For the file input
+ const char *infile = NULL;
+ int raw_input = 0;
+
+ // For the file output
+ FILE *out_fh;
+
const char *outuri = NULL;
int sample_rate=48000, channels=2;
const int bytes_per_sample = 2;
@@ -209,13 +226,16 @@ int main(int argc, char *argv[])
int peak_left = 0;
int peak_right = 0;
-
+ /* For MOT Slideshow and DLS insertion */
const char* pad_fifo = "/tmp/pad.fifo";
int pad_fd;
unsigned char pad_buf[128];
int padlen;
+ /* Encoder status, see the above STATUS macros */
int status = 0;
+
+ /* Whether to show the 'sox'-like measurement */
int show_level = 0;
/* Data for ZMQ CURVE authentication */
@@ -224,15 +244,17 @@ int main(int argc, char *argv[])
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'},
+ {"device", required_argument, 0, 'd'},
+ {"format", required_argument, 0, 'f'},
+ {"input", required_argument, 0, 'i'},
+ {"output", required_argument, 0, 'o'},
{"pad", required_argument, 0, 'p'},
{"pad-fifo", required_argument, 0, 'P'},
+ {"rate", required_argument, 0, 'r'},
{"secret-key", required_argument, 0, 'k'},
- {"drift-comp", no_argument, 0, 'D'},
{"afterburner", no_argument, 0, 'a'},
+ {"drift-comp", no_argument, 0, 'D'},
{"help", no_argument, 0, 'h'},
{"level", no_argument, 0, 'l'},
{0,0,0,0},
@@ -245,11 +267,8 @@ int main(int argc, char *argv[])
int index;
while(ch != -1) {
- ch = getopt_long(argc, argv, "ahDlb:c:k:o:r:d:p:P:", longopts, &index);
+ ch = getopt_long(argc, argv, "ahDlb:c:f:i:k:o:r:d:p:P:", longopts, &index);
switch (ch) {
- case 'd':
- alsa_device = optarg;
- break;
case 'a':
afterburner = true;
break;
@@ -259,17 +278,29 @@ int main(int argc, char *argv[])
case 'c':
channels = atoi(optarg);
break;
- case 'r':
- sample_rate = atoi(optarg);
+ case 'd':
+ alsa_device = optarg;
break;
- case 'o':
- outuri = optarg;
+ case 'D':
+ drift_compensation = true;
+ break;
+ case 'f':
+ if(strcmp(optarg, "raw")==0) {
+ raw_input = 1;
+ } else if(strcmp(optarg, "wav")!=0)
+ usage(argv[0]);
+ break;
+ case 'i':
+ infile = optarg;
break;
case 'k':
keyfile = optarg;
break;
- case 'D':
- drift_compensation = true;
+ case 'l':
+ show_level = 1;
+ break;
+ case 'o':
+ outuri = optarg;
break;
case 'p':
padlen = atoi(optarg);
@@ -277,8 +308,8 @@ int main(int argc, char *argv[])
case 'P':
pad_fifo = optarg;
break;
- case 'l':
- show_level = 1;
+ case 'r':
+ sample_rate = atoi(optarg);
break;
case '?':
case 'h':
@@ -287,7 +318,12 @@ int main(int argc, char *argv[])
}
}
- if(subchannel_index < 1 || subchannel_index > 24) {
+ if (alsa_device && infile) {
+ fprintf(stderr, "You must define either alsa or file input, not both\n");
+ return 1;
+ }
+
+ if (subchannel_index < 1 || subchannel_index > 24) {
fprintf(stderr, "Bad subchannels number: %d, try other bitrate.\n",
subchannel_index);
return 1;
@@ -298,10 +334,52 @@ int main(int argc, char *argv[])
return 1;
}
+ /* We assume that we need to call the encoder
+ * enc_calls_per_output before it gives us one encoded audio
+ * frame. This information is used when the alsa drift compensation
+ * is active
+ */
const int enc_calls_per_output = sample_rate / 16000;
- if (!outuri) {
- fprintf(stderr, "ZeroMQ output URI not defined\n");
+ zmq::context_t zmq_ctx;
+ zmq::socket_t zmq_sock(zmq_ctx, ZMQ_PUB);
+
+ if (outuri) {
+ if (strcmp(outuri, "-") == 0) {
+ out_fh = stdout;
+ }
+ else if ((strncmp(outuri, "tcp://", 6) == 0) ||
+ (strncmp(outuri, "pgm://", 6) == 0) ||
+ (strncmp(outuri, "epgm://", 7) == 0)) {
+ if (keyfile) {
+ fprintf(stderr, "Enabling encryption\n");
+
+ int rc = readkey(keyfile, secretkey);
+ if (rc) {
+ fprintf(stderr, "Error reading secret key\n");
+ return 2;
+ }
+
+ const int yes = 1;
+ zmq_sock.setsockopt(ZMQ_CURVE_SERVER,
+ &yes, sizeof(yes));
+
+ zmq_sock.setsockopt(ZMQ_CURVE_SECRETKEY,
+ secretkey, CURVE_KEYLEN);
+ }
+ zmq_sock.connect(outuri);
+ }
+ else { // We assume it's a file name
+ out_fh = fopen(outuri, "wb");
+
+ if (!out_fh) {
+ fprintf(stderr, "Can't open output file!\n");
+ return 1;
+ }
+ }
+ }
+ else {
+ fprintf(stderr, "Output URI not defined\n");
return 1;
}
@@ -325,26 +403,6 @@ int main(int argc, char *argv[])
}
}
- zmq::context_t zmq_ctx;
- zmq::socket_t zmq_sock(zmq_ctx, ZMQ_PUB);
-
- if (keyfile) {
- fprintf(stderr, "Enabling encryption\n");
-
- int rc = readkey(keyfile, secretkey);
- if (rc) {
- fprintf(stderr, "Error reading secret key\n");
- return 2;
- }
-
- const int yes = 1;
- zmq_sock.setsockopt(ZMQ_CURVE_SERVER,
- &yes, sizeof(yes));
-
- zmq_sock.setsockopt(ZMQ_CURVE_SECRETKEY,
- secretkey, CURVE_KEYLEN);
- }
- zmq_sock.connect(outuri);
HANDLE_AACENCODER encoder;
@@ -378,11 +436,23 @@ int main(int argc, char *argv[])
return 1;
}
- // We'll use either of the two possible alsa inputs.
+ /* No input defined ? default to alsa "default" */
+ if (!alsa_device) {
+ alsa_device = "default";
+ }
+
+ // We'll use one of the tree possible inputs
AlsaInputThreaded alsa_in_threaded(alsa_device, channels, sample_rate, queue);
- AlsaInputDirect alsa_in_direct(alsa_device, channels, sample_rate);
+ AlsaInputDirect alsa_in_direct(alsa_device, channels, sample_rate);
+ FileInput file_in(infile, raw_input, sample_rate);
- if (drift_compensation) {
+ if (infile) {
+ if (file_in.prepare() != 0) {
+ fprintf(stderr, "File input preparation failed\n");
+ return 1;
+ }
+ }
+ else if (drift_compensation) {
if (alsa_in_threaded.prepare() != 0) {
fprintf(stderr, "Alsa preparation failed\n");
return 1;
@@ -485,9 +555,20 @@ int main(int argc, char *argv[])
// -------------- Read Data
memset(outbuf, 0x00, outbuf_size);
+ memset(input_buf, 0x00, input_size);
ssize_t read;
- if (drift_compensation) {
+ if (infile) {
+ read = file_in.read(input_buf, input_size);
+ if (read < 0) {
+ break;
+ }
+ else if (read != input_size) {
+ fprintf(stderr, "Short file read !\n");
+ break;
+ }
+ }
+ else if (drift_compensation) {
if (alsa_in_threaded.fault_detected()) {
fprintf(stderr, "Detected fault in alsa input!\n");
break;