merge file and alsa encoders into dabplus-enc

There was a lot of redundant code between the two

7 files changed, 333 insertions, 714 deletions

diff --git a/.gitignore b/.gitignore
index 2014f3a..094130a 100644
--- a/.gitignore
+++ b/.gitignore
@@ -30,3 +30,4 @@ mot-encoder
 dabplus-enc-file
 dabplus-enc-file-zmq
 dabplus-enc-alsa-zmq
+dabplus-enc-zmq
diff --git a/Makefile.am b/Makefile.am
index 7801cc1..8662153 100644
--- a/Makefile.am
+++ b/Makefile.am
@@ -43,31 +43,22 @@ libfdk_aac_la_LDFLAGS = -version-info @FDK_AAC_VERSION@ -no-undefined \
 #aac_enc_SOURCES = src/aac-enc.c \
 #				  src/wavreader.c
 
-dabplus_enc_file_zmq_LDADD     = libfdk-aac.la -lfec -lzmq
-dabplus_enc_file_zmq_CFLAGS    = $(AM_CPPFLAGS) $(GITVERSION_FLAGS)
-dabplus_enc_file_zmq_SOURCES   = src/dabplus-enc-file-zmq.c \
-								 src/wavreader.c \
-								 src/encryption.c \
-								 src/encryption.h \
-								 src/utils.c \
-								 src/utils.h \
-								 contrib/lib_crc.h \
-								 contrib/lib_crc.c
-
-dabplus_enc_alsa_zmq_LDFLAGS     = -no-install
-dabplus_enc_alsa_zmq_LDADD       = libfdk-aac.la -lfec -lzmq -lasound \
+dabplus_enc_LDFLAGS     = -no-install
+dabplus_enc_LDADD       = libfdk-aac.la -lfec -lzmq -lasound \
 								   -lrt -lboost_thread
-dabplus_enc_alsa_zmq_CPPFLAGS    = $(AM_CPPFLAGS) $(GITVERSION_FLAGS)
-dabplus_enc_alsa_zmq_SOURCES     = src/dabplus-enc-alsa-zmq.cpp \
-								   src/AlsaInput.cpp \
-								   src/AlsaInput.h \
-								   src/SampleQueue.h \
-								   src/encryption.c \
-								   src/encryption.h \
-								   src/zmq.hpp \
-								   src/utils.c \
-								   src/utils.h
-
+dabplus_enc_CPPFLAGS    = $(AM_CPPFLAGS) $(GITVERSION_FLAGS)
+dabplus_enc_SOURCES     = src/dabplus-enc.cpp \
+						  src/FileInput.cpp \
+						  src/FileInput.h \
+						  src/AlsaInput.cpp \
+						  src/AlsaInput.h \
+						  src/SampleQueue.h \
+						  src/encryption.c \
+						  src/encryption.h \
+						  src/zmq.hpp \
+						  src/utils.c \
+						  src/utils.h \
+						  src/wavreader.c
 
 mot_encoder_CFLAGS   = $(GITVERSION_FLAGS) @MAGICKWAND_CFLAGS@ -Icontrib
 mot_encoder_LDADD    = @MAGICKWAND_LDADD@
@@ -75,8 +66,7 @@ mot_encoder_SOURCES  = src/mot-encoder.c \
 					   contrib/lib_crc.h \
 					   contrib/lib_crc.c
 
-bin_PROGRAMS =  dabplus-enc-file-zmq$(EXEEXT) \
-				dabplus-enc-alsa-zmq$(EXEEXT) \
+bin_PROGRAMS =  dabplus-enc$(EXEEXT) \
 				mot-encoder$(EXEEXT)
 
 noinst_HEADERS = src/wavreader.h
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;