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authorMatthias P. Braendli <matthias.braendli@mpb.li>2016-07-08 11:50:26 +0200
committerMatthias P. Braendli <matthias.braendli@mpb.li>2016-07-08 11:50:26 +0200
commit64b729f3284ead6a71fbeb7f8e0dd22187b21786 (patch)
tree60efb272d174f31bb93cfe561ac1c7ceeb3b412e /src
parent561b6adce76dd209ebc547fcd98f525501742d97 (diff)
downloadODR-AudioEnc-64b729f3284ead6a71fbeb7f8e0dd22187b21786.tar.gz
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Add doxygen configuration and target
Diffstat (limited to 'src')
-rw-r--r--src/AlsaInput.cpp2
-rw-r--r--src/AlsaInput.h15
-rw-r--r--src/FileInput.cpp2
-rw-r--r--src/FileInput.h20
-rw-r--r--src/JackInput.cpp7
-rw-r--r--src/JackInput.h18
-rw-r--r--src/SampleQueue.h53
-rw-r--r--src/VLCInput.cpp33
-rw-r--r--src/VLCInput.h55
-rw-r--r--src/charset.h16
-rw-r--r--src/dabplus-enc.cpp77
-rw-r--r--src/encryption.h1
-rw-r--r--src/mot-encoder.cpp65
-rw-r--r--src/utils.h5
14 files changed, 267 insertions, 102 deletions
diff --git a/src/AlsaInput.cpp b/src/AlsaInput.cpp
index e5fd420..293232f 100644
--- a/src/AlsaInput.cpp
+++ b/src/AlsaInput.cpp
@@ -35,7 +35,7 @@ int AlsaInput::prepare()
fprintf(stderr, "Initialising ALSA...\n");
- const int open_mode = 0; //|= SND_PCM_NONBLOCK;
+ const int open_mode = 0;
if ((err = snd_pcm_open(&m_alsa_handle, m_alsa_dev.c_str(),
SND_PCM_STREAM_CAPTURE, open_mode)) < 0) {
diff --git a/src/AlsaInput.h b/src/AlsaInput.h
index 34886df..a17134b 100644
--- a/src/AlsaInput.h
+++ b/src/AlsaInput.h
@@ -16,6 +16,10 @@
* and limitations under the License.
* -------------------------------------------------------------------
*/
+/*! \section ALSA Input
+ *
+ * This input uses libasound to get audio data.
+ */
#ifndef __ALSA_H_
#define __ALSA_H_
@@ -34,8 +38,9 @@
#include "SampleQueue.h"
#include "common.h"
-/* Common functionality for the direct alsa input and the
- * threaded alsa input
+/*! Common functionality for the direct alsa input and the
+ * threaded alsa input. The threaded one is used for
+ * drift compensation.
*/
class AlsaInput
{
@@ -80,10 +85,10 @@ class AlsaInputDirect : public AlsaInput
unsigned int rate) :
AlsaInput(alsa_dev, channels, rate) { }
- /* Read length Bytes from from the alsa device.
+ /*! Read length Bytes from from the alsa device.
* length must be a multiple of channels * bytes_per_sample.
*
- * Returns the number of bytes read.
+ * \return the number of bytes read.
*/
ssize_t read(uint8_t* buf, size_t length);
@@ -112,7 +117,7 @@ class AlsaInputThreaded : public AlsaInput
}
}
- /* Start the ALSA thread that fills the queue */
+ /*! 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
index eae484f..d6c33cf 100644
--- a/src/FileInput.cpp
+++ b/src/FileInput.cpp
@@ -1,5 +1,5 @@
/* ------------------------------------------------------------------
- * Copyright (C) 2014 Matthias P. Braendli
+ * Copyright (C) 2016 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.
diff --git a/src/FileInput.h b/src/FileInput.h
index 57cc5a0..937cbd4 100644
--- a/src/FileInput.h
+++ b/src/FileInput.h
@@ -1,5 +1,5 @@
/* ------------------------------------------------------------------
- * Copyright (C) 2014 Matthias P. Braendli
+ * Copyright (C) 2016 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.
@@ -15,6 +15,16 @@
* and limitations under the License.
* -------------------------------------------------------------------
*/
+/*! \section File Input
+ *
+ * This input reads a wav or raw file.
+ *
+ * The raw input needs to be signed 16-bit per sample data, with
+ * the number of channels corresponding to the command line.
+ *
+ * The wav input must also correspond to the parameters on the command
+ * line (number of channels, rate)
+ */
#ifndef _FILE_INPUT_H_
#define _FILE_INPUT_H_
@@ -35,15 +45,15 @@ class FileInput
~FileInput();
- /* Open the file and prepare the wav decoder.
+ /*! Open the file and prepare the wav decoder.
*
- * Returns nonzero on error
+ * \return nonzero on error
*/
int prepare(void);
- /* Read length bytes into buf.
+ /*! Read length bytes into buf.
*
- * Returns the number of bytes read.
+ * \return the number of bytes read.
*/
ssize_t read(uint8_t* buf, size_t length);
int eof();
diff --git a/src/JackInput.cpp b/src/JackInput.cpp
index 51de6e4..a94f2e7 100644
--- a/src/JackInput.cpp
+++ b/src/JackInput.cpp
@@ -1,6 +1,6 @@
/* ------------------------------------------------------------------
* Copyright (C) 2011 Martin Storsjo
- * Copyright (C) 2013,2014 Matthias P. Braendli
+ * Copyright (C) 2016 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.
@@ -104,7 +104,10 @@ int JackInput::prepare()
void JackInput::jack_process(jack_nframes_t nframes)
{
- // Convert samples to shorts
+ /*! JACK works with float samples, we need to convert
+ * them to shorts first. This is done using a saturated
+ * conversion to avoid glitches.
+ */
std::vector<int16_t> buffer(m_channels * nframes);
for (int chan = 0; chan < m_channels; chan++) {
diff --git a/src/JackInput.h b/src/JackInput.h
index 36cd34f..ba4834e 100644
--- a/src/JackInput.h
+++ b/src/JackInput.h
@@ -1,6 +1,6 @@
/* ------------------------------------------------------------------
* Copyright (C) 2011 Martin Storsjo
- * Copyright (C) 2013,2014 Matthias P. Braendli
+ * Copyright (C) 2016 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.
@@ -16,6 +16,11 @@
* and limitations under the License.
* -------------------------------------------------------------------
*/
+/*! \section JACK Input
+ *
+ * This input uses JACK to get audio data. This always uses drift
+ * compensation, because there is no blocking way to read from JACK.
+ */
#ifndef __JACK_INPUT_H
#define __JACK_INPUT_H
@@ -53,13 +58,16 @@ class JackInput
}
}
- /* Prepare the audio input */
+ /*! Prepare the audio input
+ *
+ * \return 0 on success
+ */
int prepare();
private:
JackInput(const JackInput& other);
- jack_client_t* m_client;
+ jack_client_t *m_client;
std::vector<jack_port_t*> m_input_ports;
@@ -82,13 +90,13 @@ class JackInput
SampleQueue<uint8_t>& m_queue;
// Static functions for JACK callbacks
- static int process_cb(jack_nframes_t nframes, void* arg)
+ static int process_cb(jack_nframes_t nframes, void *arg)
{
((JackInput*)arg)->jack_process(nframes);
return 0;
}
- static void shutdown_cb(void* arg)
+ static void shutdown_cb(void *arg)
{
((JackInput*)arg)->jack_shutdown();
}
diff --git a/src/SampleQueue.h b/src/SampleQueue.h
index 09b67c7..994672f 100644
--- a/src/SampleQueue.h
+++ b/src/SampleQueue.h
@@ -1,10 +1,28 @@
/*
- Copyright (C) 2013, 2014, 2015
- Matthias P. Braendli, matthias.braendli@mpb.li
+ * Copyright (C) 2016 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.
+ *
+ * Matthias P. Braendli, matthias.braendli@mpb.li
+ */
- An implementation for a threadsafe queue using the C++11 thread library
- for audio samples.
-*/
+/*!
+ * \section SampleQueue
+ *
+ * An implementation for a threadsafe queue using the C++11 thread library
+ * for audio samples.
+ */
#ifndef _SAMPLE_QUEUE_H_
#define _SAMPLE_QUEUE_H_
@@ -21,7 +39,7 @@
#include <cstdio>
#include <cmath>
-/* This queue is meant to be used by two threads. One producer
+/*! This queue is meant to be used by two threads. One producer
* that pushes elements into the queue, and one consumer that
* retrieves the elements.
*
@@ -55,7 +73,10 @@ public:
m_overruns(0) {}
- /* Push a bunch of samples into the buffer */
+ /*! Push a bunch of samples into the buffer
+ *
+ * \return size of the queue after the push
+ */
size_t push(const T *val, size_t len)
{
size_t new_size = 0;
@@ -97,11 +118,11 @@ public:
return m_queue.size();
}
- /* Wait until len elements in the queue are available,
+ /*! Wait until len elements in the queue are available,
* and then fill the buf. If the timeout_ms (expressed in milliseconds
* expires), fill the available number of elements.
- * Returns the number
- * of elemets written into buf
+ *
+ * \return the number of elemets written into buf
*/
size_t pop_wait(T* buf, size_t len, int timeout_ms)
{
@@ -158,8 +179,9 @@ public:
return num_to_copy;
}
- /* Get up to len elements, place them into the buf array
- * Returns the number of elements it was able to take
+ /*! Get up to len elements, place them into the buf array
+ *
+ * \return the number of elements it was able to take
* from the queue
*/
size_t pop(T* buf, size_t len)
@@ -168,10 +190,11 @@ public:
return pop(buf, len, ovr);
}
- /* Get up to len elements, place them into the buf array.
+ /*! Get up to len elements, place them into the buf array.
* Also update the overrun variable with the information
* of how many overruns we saw since the last pop.
- * Returns the number of elements it was able to take
+ *
+ * \return the number of elements it was able to take
* from the queue
*/
size_t pop(T* buf, size_t len, size_t* overruns)
@@ -243,7 +266,7 @@ private:
unsigned int m_bytes_per_sample;
size_t m_max_size;
- /* Counter to keep track of number of overruns between calls
+ /*! Counter to keep track of number of overruns between calls
* to pop()
*/
size_t m_overruns;
diff --git a/src/VLCInput.cpp b/src/VLCInput.cpp
index 33c4594..4831915 100644
--- a/src/VLCInput.cpp
+++ b/src/VLCInput.cpp
@@ -34,7 +34,11 @@ int check_vlc_uses_size_t();
using namespace std;
-// VLC Audio prerender callback
+/*! VLC callback functions have to be C functions.
+ * These wrappers call the VLCInput functions
+ */
+
+//! VLC Audio prerender callback
void prepareRender_size_t(
void* p_audio_data,
uint8_t** pp_pcm_buffer,
@@ -45,6 +49,7 @@ void prepareRender_size_t(
in->preRender_cb(pp_pcm_buffer, size);
}
+//! VLC Audio prepare render callback
void prepareRender(
void* p_audio_data,
uint8_t** pp_pcm_buffer,
@@ -56,7 +61,7 @@ void prepareRender(
}
-// Audio postrender callback
+//! Audio postrender callback for VLC versions that use size_t
void handleStream_size_t(
void* p_audio_data,
uint8_t* p_pcm_buffer,
@@ -79,7 +84,9 @@ void handleStream_size_t(
in->postRender_cb();
}
-// convert from unsigned int size to size_t size
+/*! Audio postrender callback for VLC versions that use unsigned int.
+ * Convert from unsigned int size to size_t size
+ */
void handleStream(
void* p_audio_data,
uint8_t* p_pcm_buffer,
@@ -101,7 +108,7 @@ void handleStream(
pts);
}
-// VLC Exit callback
+/*! VLC Exit callback */
void handleVLCExit(void* opaque)
{
((VLCInput*)opaque)->exit_cb();
@@ -109,7 +116,6 @@ void handleVLCExit(void* opaque)
int VLCInput::prepare()
{
- int err;
fprintf(stderr, "Initialising VLC...\n");
long long int handleStream_address;
@@ -330,10 +336,11 @@ ssize_t VLCInput::m_read(uint8_t* buf, size_t length)
}
/* Write the corresponding text to a file readable by mot-encoder, with optional
- * DL+ information. The text is passed as a copy because we actually use the m_nowplaying
- * variable which is also accessed in another thread, so better make a copy.
+ * DL+ information. The text is passed as a copy because we actually use the
+ * m_nowplaying variable which is also accessed in another thread, so better
+ * make a copy.
*
- * Returns false on failure
+ * \return false on failure
*/
bool write_icy_to_file(const std::string text, const std::string& filename, bool dl_plus)
{
@@ -379,6 +386,9 @@ void VLCInput::write_icy_text(const std::string& filename, bool dl_plus)
std::lock_guard<std::mutex> lock(m_nowplaying_mutex);
if (m_nowplaying_previous != m_nowplaying) {
+ /*! We write the ICY text in a separate task because
+ * we do not want to have a delay due to IO
+ */
icy_text_written = std::async(std::launch::async,
std::bind(write_icy_to_file, m_nowplaying, filename, dl_plus));
@@ -400,8 +410,9 @@ void VLCInput::start()
}
}
-// How many samples we insert into the queue each call
-// 10 samples @ 32kHz = 3.125ms
+/*! How many samples we insert into the queue each call
+ * 10 samples @ 32kHz = 3.125ms
+ */
#define NUM_BYTES_PER_CALL (10 * BYTES_PER_SAMPLE)
void VLCInput::process()
@@ -424,7 +435,7 @@ void VLCInput::process()
-/* VLC up to version 2.1.0 used a different callback function signature.
+/*! VLC up to version 2.1.0 used a different callback function signature.
* VLC 2.2.0 uses size_t
*
* \return 1 if the callback with size_t size should be used.
diff --git a/src/VLCInput.h b/src/VLCInput.h
index 1ea0bee..6f91e89 100644
--- a/src/VLCInput.h
+++ b/src/VLCInput.h
@@ -15,6 +15,11 @@
* and limitations under the License.
* -------------------------------------------------------------------
*/
+/*! \section VLC Input
+ *
+ * This input uses libvlc to get audio data. It is extremely useful, and allows
+ * the encoder to use all inputs VLC supports.
+ */
#ifndef __VLC_INPUT_H_
#define __VLC_INPUT_H_
@@ -41,8 +46,8 @@ extern "C" {
#include "utils.h"
}
-/* Common functionality for the direct libvlc input and the
- * threaded libvlc input
+/*! Common functionality for the direct libvlc input and the
+ * threaded libvlc input
*/
class VLCInput
{
@@ -80,28 +85,31 @@ class VLCInput
cleanup();
}
- /* Prepare the audio input */
+ /*! Prepare the audio input
+ *
+ * \return 0 on success
+ */
int prepare();
- /* Start the libVLC thread that fills the samplequeue */
+ /*! Start the libVLC thread that fills m_samplequeue */
void start();
- /* Write the last received ICY-Text to the
+ /*! Write the last received ICY-Text to the
* file.
*/
void write_icy_text(const std::string& filename, bool dl_plus);
- // Callbacks for VLC
+ //! Callbacks for VLC
- /* Notification of VLC exit */
+ /*! Notification of VLC exit */
void exit_cb(void);
- /* Prepare a buffer for VLC */
+ /*! Prepare a buffer for VLC */
void preRender_cb(
uint8_t** pp_pcm_buffer,
size_t size);
- /* Notification from VLC that the buffer is now filled
+ /*! Notification from VLC that the buffer is now filled
*/
void postRender_cb();
@@ -112,14 +120,23 @@ class VLCInput
bool fault_detected() { return m_fault; };
private:
+ /*! Stop the player and release resources
+ */
void cleanup(void);
- // Fill exactly length bytes into buf. Blocking.
+ /*! Fill exactly length bytes into buf. Blocking.
+ *
+ * \return number of bytes written into buf, or
+ * -1 in case of error
+ */
ssize_t m_read(uint8_t* buf, size_t length);
+ /*! Buffer used in the callback functions for VLC */
std::vector<uint8_t> m_current_buf;
std::mutex m_queue_mutex;
+
+ /*! Buffer containing all available samples from VLC */
std::deque<uint8_t> m_queue;
std::string m_uri;
@@ -127,16 +144,20 @@ class VLCInput
unsigned m_channels;
int m_rate;
- // Whether to enable network caching in VLC or not
+ //! Whether to enable network caching in VLC or not
std::string m_cache;
- // Given as-is to libvlc
+ //! Given as-is to libvlc, useful for additional arguments
std::vector<std::string> m_additional_opts;
- // value for the VLC compressor filter
+ /*! value for the VLC compressor filter --compressor-makeup
+ * setting. Many more compressor settings could be set.
+ */
std::string m_gain;
-
+ /*! VLC can give us the ICY-Text from an Icecast stream,
+ * which we optionnally write into a text file for mot-encoder
+ */
std::future<bool> icy_text_written;
std::mutex m_nowplaying_mutex;
std::string m_nowplaying;
@@ -148,12 +169,16 @@ class VLCInput
// For the thread
- /* The function runnin in the thread */
+ /* The thread running process takes samples from m_queue and writes
+ * them into m_samplequeue. This decouples m_queue from m_samplequeue
+ * which is directly used by dabplus-enc.cpp
+ */
void process();
std::atomic<bool> m_fault;
std::atomic<bool> m_running;
std::thread m_thread;
+
SampleQueue<uint8_t>& m_samplequeue;
};
diff --git a/src/charset.h b/src/charset.h
index 82d81cf..8d1e1a2 100644
--- a/src/charset.h
+++ b/src/charset.h
@@ -13,13 +13,13 @@
You should have received a copy of the GNU General Public License
along with this program. If not, see <http://www.gnu.org/licenses/>.
+*/
+/*!
+ \file charset.h
+ \brief Define the EBU charset according to ETSI TS 101 756v1.8.1 for DLS encoding
- charset.h
- Define the EBU charset according to ETSI TS 101 756v1.8.1 for DLS encoding
-
- Authors:
- Matthias P. Braendli <matthias@mpb.li>
- Lindsay Cornell
+ \author Matthias P. Braendli <matthias@mpb.li>
+ \author Lindsay Cornell
*/
#ifndef __CHARSET_H_
@@ -65,7 +65,7 @@ class CharsetConverter
{
public:
CharsetConverter() {
- /* Build the converstion table that contains the known code points,
+ /*! Build the converstion table that contains the known code points,
* at the indices corresponding to the EBU Latin table
*/
using namespace std;
@@ -77,7 +77,7 @@ class CharsetConverter
}
}
- /* Convert a UTF-8 encoded text line into an EBU Latin encoded byte stream
+ /*! Convert a UTF-8 encoded text line into an EBU Latin encoded byte stream
*/
std::string convert(std::string line_utf8) {
using namespace std;
diff --git a/src/dabplus-enc.cpp b/src/dabplus-enc.cpp
index 5b2b403..953073d 100644
--- a/src/dabplus-enc.cpp
+++ b/src/dabplus-enc.cpp
@@ -17,6 +17,33 @@
* -------------------------------------------------------------------
*/
+/*! \mainpage Introduction
+ * The ODR-mmbTools FDK-AAC-DABplus Audio encoder can encode audio for
+ * ODR-DabMux, both DAB and DAB+. The DAB encoder is based on toolame. The
+ * DAB+ encoder uses a modified library of the Fraunhofer FDK AAC code from
+ * Android, patched for 960-transform to do DAB+ broadcast encoding.
+ *
+ * This document describes some internals of the encoder, and is intended
+ * to help developers understand and improve the software package.
+ *
+ * User documentation is available in the README and in the ODR-mmbTools
+ * Guide, available on the www.opendigitalradio.org website.
+ *
+ * The readme for the whole package is \ref md_README
+ *
+ * Interesting starting points for the encoder
+ * - \ref dabplus-enc.cpp
+ * - \ref VLC Input
+ * - \ref Alsa Input
+ * - \ref JACK Input
+ * - \ref SampleQueue
+ * - \ref charset.h
+ * - \ref libtoolame API
+ *
+ * For the mot-encoder:
+ * - \ref mot-encoder.cpp
+ */
+
#include "config.h"
#include "AlsaInput.h"
#include "FileInput.h"
@@ -56,7 +83,7 @@ extern "C" {
-// Enumerate which encoder we can use
+//! Enumeration of encoders we can use
enum class encoder_selection_t {
fdk_dabplus,
toolame_dab
@@ -166,6 +193,10 @@ void usage(const char* name) {
}
+/*! Setup the FDK AAC encoder
+ *
+ * \return 0 on success
+ */
int prepare_aac_encoder(
HANDLE_AACENCODER *encoder,
int subchannel_index,
@@ -262,6 +293,9 @@ int prepare_aac_encoder(
chrono::steady_clock::time_point timepoint_last_compensation;
+/*! Wait the proper amount of time to throttle down to nominal encoding
+ * rate, if drift compensation is enabled.
+ */
void drift_compensation_delay(int sample_rate, int channels, size_t bytes)
{
const size_t bytes_per_second = sample_rate * BYTES_PER_SAMPLE * channels;
@@ -726,6 +760,9 @@ int main(int argc, char *argv[])
int max_size = 8*input_buf.size() + NUM_SAMPLES_PER_CALL;
+ /*! The SampleQueue \c queue is given to the inputs, so that they
+ * can fill it.
+ */
SampleQueue<uint8_t> queue(BYTES_PER_SAMPLE, channels, max_size);
/* symsize=8, gfpoly=0x11d, fcr=0, prim=1, nroots=10, pad=135 */
@@ -867,6 +904,18 @@ int main(int argc, char *argv[])
memset(&outbuf[0], 0x00, outbuf_size);
memset(&input_buf[0], 0x00, input_buf.size());
+ /*! \section Data input
+ * We read data input either in a blocking way (file input, VLC or ALSA
+ * without drift compensation) or in a non-blocking way (VLC or ALSA
+ * with drift compensation, JACK).
+ *
+ * The file input doesn't need the queue at all. But the other inputs
+ * do, and either use \c pop() or \c pop_wait() depending on if it's blocking or not
+ *
+ * In non-blocking, the \c queue makes the data available without delay, and the
+ * \c drift_compensation_delay() function handles rate throttling.
+ */
+
if (infile) {
read_bytes = file_in.read(&input_buf[0], input_buf.size());
if (read_bytes < 0) {
@@ -911,6 +960,10 @@ int main(int argc, char *argv[])
else {
const int timeout_ms = 1000;
read_bytes = input_buf.size();
+
+ /*! pop_wait() must return after a timeout, otherwise the silence detector cannot do
+ * its job.
+ */
size_t bytes_from_queue = queue.pop_wait(&input_buf[0], read_bytes, timeout_ms); // returns bytes
if (bytes_from_queue < read_bytes) {
@@ -960,6 +1013,12 @@ int main(int argc, char *argv[])
#endif
}
+ /*! Audio level measurement is always done assuming we have two
+ * channels, and is formally wrong in mono, but still gives
+ * numbers one can use.
+ *
+ * \todo fix level measurement in mono
+ */
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);
@@ -967,7 +1026,12 @@ int main(int argc, char *argv[])
peak_right = MAX(peak_right, r);
}
- /* Silence detection */
+ /*! Silence detection, looks at the audio level and is
+ * only useful if the connection dropped, or if no data is available. It is not
+ * useful if the source is nearly silent (some noise present), because the
+ * threshold is 0, and not configurable. The rationale is that we want to
+ * guard against connection issues, not source level issues
+ */
if (die_on_silence && MAX(peak_left, peak_right) == 0) {
const unsigned int frame_time_msec = 1000ul *
read_bytes / (BYTES_PER_SAMPLE * channels * sample_rate);
@@ -1050,6 +1114,9 @@ int main(int argc, char *argv[])
}
}
+ /*! toolame expects the audio to be in another shape as
+ * we have in input_buf, and we need to convert first
+ */
short input_buffers[2][1152];
if (channels == 1) {
@@ -1076,7 +1143,10 @@ int main(int argc, char *argv[])
}
}
- /* Check if the encoder has generated output data */
+ /*! Check if the encoder has generated output data.
+ * DAB+ requires RS encoding, which is not done in ODR-DabMux and not necessary
+ * for DAB.
+ */
if (numOutBytes != 0 and
selected_encoder == encoder_selection_t::fdk_dabplus) {
@@ -1088,7 +1158,6 @@ int main(int argc, char *argv[])
}
calls = 0;
- // ----------- RS encoding
int row, col;
unsigned char buf_to_rs_enc[110];
unsigned char rs_enc[10];
diff --git a/src/encryption.h b/src/encryption.h
index 936578b..bfe1fc3 100644
--- a/src/encryption.h
+++ b/src/encryption.h
@@ -15,6 +15,7 @@
* and limitations under the License.
* -------------------------------------------------------------------
*/
+/* \brief Helper functions for the ZMQ encryption */
#ifndef _ENCRYPTION_H_
#define _ENCRYPTION_H_
diff --git a/src/mot-encoder.cpp b/src/mot-encoder.cpp
index 8d8d9ce..00757f6 100644
--- a/src/mot-encoder.cpp
+++ b/src/mot-encoder.cpp
@@ -17,14 +17,14 @@
You should have received a copy of the GNU General Public License
along with this program. If not, see <http://www.gnu.org/licenses/>.
+*/
+/*!
+ \file mot-encoder.c
+ \brief Generete PAD data for MOT Slideshow and DLS
- mot-encoder.c
- Generete PAD data for MOT Slideshow and DLS
-
- Authors:
- Sergio Sagliocco <sergio.sagliocco@csp.it>
- Matthias P. Braendli <matthias@mpb.li>
- Stefan Pöschel <odr@basicmaster.de>
+ \author Sergio Sagliocco <sergio.sagliocco@csp.it>
+ \author Matthias P. Braendli <matthias@mpb.li>
+ \author Stefan Pöschel <odr@basicmaster.de>
*/
#include <cstdio>
@@ -82,12 +82,12 @@ extern "C" {
#define MINQUALITY 40
// Charsets from TS 101 756
-#define CHARSET_COMPLETE_EBU_LATIN 0 // Complete EBU Latin based repertoire
-#define CHARSET_EBU_LATIN_CY_GR 1 // EBU Latin based common core, Cyrillic, Greek
-#define CHARSET_EBU_LATIN_AR_HE_CY_GR 2 // EBU Latin based core, Arabic, Hebrew, Cyrillic and Greek
-#define CHARSET_ISO_LATIN_ALPHABET_2 3 // ISO Latin Alphabet No 2
-#define CHARSET_UCS2_BE 6 // ISO/IEC 10646 using UCS-2 transformation format, big endian byte order
-#define CHARSET_UTF8 15 // ISO Latin Alphabet No 2
+#define CHARSET_COMPLETE_EBU_LATIN 0 //!< Complete EBU Latin based repertoire
+#define CHARSET_EBU_LATIN_CY_GR 1 //!< EBU Latin based common core, Cyrillic, Greek
+#define CHARSET_EBU_LATIN_AR_HE_CY_GR 2 //!< EBU Latin based core, Arabic, Hebrew, Cyrillic and Greek
+#define CHARSET_ISO_LATIN_ALPHABET_2 3 //!< ISO Latin Alphabet No 2
+#define CHARSET_UCS2_BE 6 //!< ISO/IEC 10646 using UCS-2 transformation format, big endian byte order
+#define CHARSET_UTF8 15 //!< ISO Latin Alphabet No 2
typedef std::vector<uint8_t> uint8_vector_t;
@@ -121,7 +121,7 @@ struct MSCDG {
unsigned short int crc; // 16 bits
};
-/* Between collection of slides and transmission, the slide data is saved
+/*! Between collection of slides and transmission, the slide data is saved
* in this structure.
*/
struct slide_metadata_t {
@@ -138,7 +138,7 @@ struct slide_metadata_t {
}
};
-/* A simple fingerprint for each slide transmitted.
+/*! A simple fingerprint for each slide transmitted.
* Allows us to reuse the same fidx if the same slide
* is transmitted more than once.
*/
@@ -153,8 +153,9 @@ struct fingerprint_t {
// assigned fidx, -1 means invalid
int fidx;
- // The comparison is not done on fidx, only
- // on the file-specific data
+ /*! The comparison is not done on fidx, only
+ * on the file-specific data
+ */
bool operator==(const fingerprint_t& other) const {
return (((s_name == other.s_name &&
s_size == other.s_size) &&
@@ -183,6 +184,13 @@ struct fingerprint_t {
}
};
+/*! We keep track of transmitted files so that we can retransmit
+ * identical slides with the same index, in case the receivers cache
+ * them.
+ *
+ * \c MAXHISTORYLEN defines for how how many slides we want to keep this
+ * history.
+ */
class History {
public:
History(size_t hist_size) :
@@ -594,7 +602,7 @@ void PADPacketizer::AddCI(int apptype, int len_index) {
int PADPacketizer::OptimalSubFieldSizeIndex(size_t available_bytes) {
- /* Return the index of the optimal sub-field size by stepwise search (regards only Variable Size X-PAD):
+ /*! Return the index of the optimal sub-field size by stepwise search (regards only Variable Size X-PAD):
* - find the smallest sub-field able to hold (at least) all available bytes
* - find the biggest regarding sub-field we have space for (which definitely exists - otherwise previously the PAD would have been flushed)
* - if the wasted space is at least as big as the smallest possible sub-field, use a sub-field one size smaller
@@ -620,7 +628,7 @@ int PADPacketizer::WriteDGToSubField(DATA_GROUP* dg, size_t len) {
bool PADPacketizer::AppendDG(DATA_GROUP* dg) {
- /* use X-PAD w/o CIs instead of X-PAD w/ CIs, if we can save some bytes or at least do not waste additional bytes
+ /*! use X-PAD w/o CIs instead of X-PAD w/ CIs, if we can save some bytes or at least do not waste additional bytes
*
* Omit CI list in case:
* 1. no pending data sub-fields
@@ -1051,12 +1059,13 @@ void warnOnSmallerImage(size_t height, size_t width, std::string& fname) {
}
-// Scales the image down if needed,
-// so that it is 320x240 pixels.
-// Automatically reduces the quality to make sure the
-// blobsize is not too large.
-//
-// Returns: the blobsize
+/*! Scales the image down if needed,
+ * so that it is 320x240 pixels.
+ * Automatically reduces the quality to make sure the
+ * blobsize is not too large.
+ *
+ * \return the blobsize
+ */
#if HAVE_MAGICKWAND
size_t resizeImage(MagickWand* m_wand, unsigned char** blob, std::string& fname)
{
@@ -1127,17 +1136,17 @@ int encodeFile(int output_fd, std::string& fname, int fidx, bool raw_slides)
size_t orig_quality;
char* orig_format = NULL;
- /* We handle JPEG differently, because we want to avoid recompressing the
+ /*! We handle JPEG differently, because we want to avoid recompressing the
* image if it is suitable as is
*/
bool orig_is_jpeg = false;
- /* If the original is a PNG, we transmit it as is, if the resolution is correct
+ /*! If the original is a PNG, we transmit it as is, if the resolution is correct
* and the file is not too large. Otherwise it gets resized and sent as JPEG.
*/
bool orig_is_png = false;
- /* By default, we do resize the image to 320x240, with a quality such that
+ /*! By default, we do resize the image to 320x240, with a quality such that
* the blobsize is at most MAXSLIDESIZE.
*
* For JPEG input files that are already at the right resolution and at the
diff --git a/src/utils.h b/src/utils.h
index e411963..83b3e4d 100644
--- a/src/utils.h
+++ b/src/utils.h
@@ -12,12 +12,13 @@
#define linear_to_dB(x) (log10(x) * 20)
-/* Calculate the little string containing a bargraph
+/*! Calculate the little string containing a bargraph
* 'VU-meter' from the peak value measured
*/
const char* level(int channel, int peak);
-/* This defines the on-wire representation of a ZMQ message header.
+/*! This defines the on-wire representation of a ZMQ message header.
+ * It must be compatible with the definition in ODR-DabMux.
*
* The data follows right after this header */
struct zmq_frame_header_t