kopia lustrzana https://github.com/f4exb/sdrangel
645 wiersze
17 KiB
C++
645 wiersze
17 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 2019 Edouard Griffiths, F4EXB //
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// Copyright (C) 2021 Jon Beniston, M7RCE //
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// //
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// This program is free software; you can redistribute it and/or modify //
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// it under the terms of the GNU General Public License as published by //
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// the Free Software Foundation as version 3 of the License, or //
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// (at your option) any later version. //
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// //
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// This program is distributed in the hope that it will be useful, //
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// but WITHOUT ANY WARRANTY; without even the implied warranty of //
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// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the //
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// GNU General Public License V3 for more details. //
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// //
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// You should have received a copy of the GNU General Public License //
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// along with this program. If not, see <http://www.gnu.org/licenses/>. //
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///////////////////////////////////////////////////////////////////////////////////
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#include <QDebug>
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#include <complex.h>
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#include "dsp/dspengine.h"
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#include "dsp/datafifo.h"
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#include "util/db.h"
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#include "pipes/pipeendpoint.h"
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#include "maincore.h"
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#include "dabdemod.h"
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#include "dabdemodsink.h"
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// Callbacks from DAB library
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void syncHandler(bool value, void *ctx)
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{
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(void)value;
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(void)ctx;
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}
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void systemDataHandler(bool sync, int16_t snr, int32_t freqOffset, void *ctx)
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{
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DABDemodSink *sink = (DABDemodSink *)ctx;
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sink->systemData(sync, snr, freqOffset);
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}
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void ensembleNameHandler(std::string name, int32_t id, void *ctx)
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{
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DABDemodSink *sink = (DABDemodSink *)ctx;
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sink->ensembleName(QString::fromStdString(name), id);
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}
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void programNameHandler(std::string name, int32_t id, void *ctx)
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{
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DABDemodSink *sink = (DABDemodSink *)ctx;
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sink->programName(QString::fromStdString(name), id);
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}
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void fibQualityHandler(int16_t percent, void *ctx)
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{
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DABDemodSink *sink = (DABDemodSink *)ctx;
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sink->fibQuality(percent);
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}
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void audioHandler(int16_t *buffer, int size, int samplerate, bool stereo, void *ctx)
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{
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DABDemodSink *sink = (DABDemodSink *)ctx;
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sink->audio(buffer, size, samplerate, stereo);
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}
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void dataHandler(std::string data, void *ctx)
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{
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DABDemodSink *sink = (DABDemodSink *)ctx;
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sink->data(QString::fromStdString(data));
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}
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void byteHandler(uint8_t *data, int16_t a, uint8_t b, void *ctx)
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{
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(void)data;
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(void)a;
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(void)b;
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(void)ctx;
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}
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// Note: North America has different table
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static const char *dabProgramType[] =
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{
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"No programme type",
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"News",
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"Current Affairs",
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"Information",
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"Sport",
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"Education",
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"Drama",
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"Culture",
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"Science",
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"Varied",
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"Pop Music",
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"Rock Music",
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"Easy Listening Music",
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"Light Classical",
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"Serious Classical",
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"Other Music",
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"Weather/meteorology",
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"Finance/Business",
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"Children's programmes",
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"Social Affairs",
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"Religion",
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"Phone In",
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"Travel",
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"Leisure",
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"Jazz Music",
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"Country Music",
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"National Music",
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"Oldies Music",
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"Folk Music",
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"Documentary",
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"Not used",
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"Not used",
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};
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static const char *dabLanguageCode[] =
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{
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"Unknown",
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"Albanian",
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"Breton",
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"Catalan",
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"Croatian",
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"Welsh",
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"Czech",
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"Danish",
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"German",
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"English",
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"Spanish",
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"Esperanto",
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"Estonian",
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"Basque",
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"Faroese",
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"French",
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"Frisian",
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"Irish",
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"Gaelic",
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"Galician",
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"Icelandic",
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"Italian",
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"Sami",
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"Latin",
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"Latvian",
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"Luxembourgian",
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"Lithuanian",
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"Hungarian",
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"Maltese",
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"Dutch",
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"Norwegian",
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"Occitan",
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"Polish",
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"Portuguese",
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"Romanian",
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"Romansh",
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"Serbian",
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"Slovak",
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"Slovene",
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"Finnish",
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"Swedish",
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"Turkish",
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"Flemish",
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"Walloon",
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"Reserved",
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"Reserved",
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"Reserved",
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"Reserved",
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"Reserved",
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"Reserved",
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"Reserved",
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"Reserved",
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"Reserved",
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"Reserved",
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"Reserved",
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"Reserved",
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"Reserved",
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"Reserved",
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"Reserved",
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"Reserved",
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"Reserved",
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"Reserved",
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"Reserved",
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"Reserved",
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"Background sound",
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"Reserved",
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"Reserved",
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"Reserved",
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"Reserved",
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"Zulu",
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"Vietnamese",
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"Uzbek",
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"Urdu",
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"Ukranian",
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"Thai",
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"Telugu",
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"Tatar",
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"Tamil",
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"Tadzhik",
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"Swahili",
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"Sranan Tongo",
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"Somali",
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"Sinhalese",
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"Shona",
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"Serbo-Croat",
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"Rusyn",
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"Russian",
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"Quechua",
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"Pushtu",
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"Punjabi",
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"Persian",
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"Papiamento",
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"Oriya",
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"Nepali",
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"Ndebele",
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"Marathi",
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"Moldavian",
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"Malaysian",
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"Malagasay",
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"Macedonian",
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"Laotian",
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"Korean",
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"Khmer",
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"Kazakh",
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"Kannada",
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"Japanese",
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"Indonesian",
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"Hindi",
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"Hebrew",
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"Hausa",
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"Gurani",
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"Gujurati",
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"Greek",
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"Georgian",
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"Fulani",
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"Dari",
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"Chuvash",
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"Chinese",
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"Burmese",
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"Bulgarian",
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"Bengali",
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"Belorussian",
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"Bambora",
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"Azerbaijani",
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"Assamese",
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"Armenian",
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"Arabic",
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"Amharic",
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};
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void programDataHandler(audiodata *data, void *ctx)
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{
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QString audio;
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if (data->ASCTy == 0)
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audio = "DAB";
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else if (data->ASCTy == 63)
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audio = "DAB+";
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else
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audio = "Unknown";
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QString language = "";
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if ((data->language < 0x80) && (data->language >= 0))
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language = dabLanguageCode[data->language & 0x7f];
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DABDemodSink *sink = (DABDemodSink *)ctx;
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sink->programData(data->bitRate, audio, language, dabProgramType[data->programType & 0x1f]);
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}
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void programQualityHandler(int16_t frames, int16_t rs, int16_t aac, void *ctx)
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{
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DABDemodSink *sink = (DABDemodSink *)ctx;
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sink->programQuality(frames, rs, aac);
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}
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void motDataHandler(std::string data, int a, void *ctx)
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{
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(void)data;
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(void)ctx;
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}
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void DABDemodSink::systemData(bool sync, int16_t snr, int32_t freqOffset)
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{
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if (getMessageQueueToChannel())
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{
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DABDemod::MsgDABSystemData *msg = DABDemod::MsgDABSystemData::create(sync, snr, freqOffset);
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getMessageQueueToChannel()->push(msg);
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}
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}
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void DABDemodSink::ensembleName(const QString& name, int id)
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{
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if (getMessageQueueToChannel())
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{
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DABDemod::MsgDABEnsembleName *msg = DABDemod::MsgDABEnsembleName::create(name, id);
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getMessageQueueToChannel()->push(msg);
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}
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}
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void DABDemodSink::programName(const QString& name, int id)
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{
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if (getMessageQueueToChannel())
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{
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DABDemod::MsgDABProgramName *msg = DABDemod::MsgDABProgramName::create(name, id);
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getMessageQueueToChannel()->push(msg);
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}
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}
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void DABDemodSink::programData(int bitrate, const QString& audio, const QString& language, const QString& programType)
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{
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if (getMessageQueueToChannel())
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{
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DABDemod::MsgDABProgramData *msg = DABDemod::MsgDABProgramData::create(bitrate, audio, language, programType);
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getMessageQueueToChannel()->push(msg);
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}
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}
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void DABDemodSink::fibQuality(int16_t percent)
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{
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if (getMessageQueueToChannel())
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{
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DABDemod::MsgDABFIBQuality *msg = DABDemod::MsgDABFIBQuality::create(percent);
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getMessageQueueToChannel()->push(msg);
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}
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}
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void DABDemodSink::programQuality(int16_t frames, int16_t rs, int16_t aac)
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{
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if (getMessageQueueToChannel())
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{
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DABDemod::MsgDABProgramQuality *msg = DABDemod::MsgDABProgramQuality::create(frames, rs, aac);
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getMessageQueueToChannel()->push(msg);
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}
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}
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void DABDemodSink::data(const QString& data)
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{
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if (getMessageQueueToChannel())
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{
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DABDemod::MsgDABData *msg = DABDemod::MsgDABData::create(data);
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getMessageQueueToChannel()->push(msg);
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}
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}
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static int16_t scale(int16_t sample, float factor)
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{
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int32_t prod = (int32_t)(((int32_t)sample) * factor);
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prod = std::min(prod, 32767);
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prod = std::max(prod, -32768);
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return (int16_t)prod;
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}
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void DABDemodSink::audio(int16_t *buffer, int size, int samplerate, bool stereo)
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{
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(void)stereo;
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(void)samplerate;
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if (samplerate != m_dabAudioSampleRate)
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{
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applyDABAudioSampleRate(samplerate);
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if (getMessageQueueToChannel())
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{
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DABDemod::MsgDABSampleRate *msg = DABDemod::MsgDABSampleRate::create(samplerate);
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getMessageQueueToChannel()->push(msg);
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}
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}
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// buffer is always 2 channels
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for (int i = 0; i < size; i+=2)
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{
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Complex ci, ca;
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if (!m_settings.m_audioMute)
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{
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ci.real(buffer[i]);
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ci.imag(buffer[i+1]);
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}
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else
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{
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ci.real(0.0f);
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ci.imag(0.0f);
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}
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if (m_audioInterpolatorDistance < 1.0f) // interpolate
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{
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while (!m_audioInterpolator.interpolate(&m_audioInterpolatorDistanceRemain, ci, &ca))
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{
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processOneAudioSample(ca);
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m_audioInterpolatorDistanceRemain += m_audioInterpolatorDistance;
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}
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}
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else // decimate
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{
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if (m_audioInterpolator.decimate(&m_audioInterpolatorDistanceRemain, ci, &ca))
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{
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processOneAudioSample(ca);
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m_audioInterpolatorDistanceRemain += m_audioInterpolatorDistance;
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}
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}
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}
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}
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void DABDemodSink::reset()
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{
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dabReset(m_dab);
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}
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void DABDemodSink::resetService()
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{
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dabReset_msc(m_dab);
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}
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void DABDemodSink::processOneAudioSample(Complex &ci)
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{
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float factor = m_settings.m_volume / 5.0f; // Should this be 5 or 10? 5 allows some positive gain
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qint16 l = scale(ci.real(), factor);
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qint16 r = scale(ci.real(), factor);
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m_audioBuffer[m_audioBufferFill].l = l;
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m_audioBuffer[m_audioBufferFill].r = r;
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++m_audioBufferFill;
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if (m_audioBufferFill >= m_audioBuffer.size())
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{
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uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill);
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if (res != m_audioBufferFill)
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{
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qDebug("DABDemodSink::audio: %u/%u audio samples written", res, m_audioBufferFill);
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m_audioFifo.clear();
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}
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m_audioBufferFill = 0;
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}
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m_demodBuffer[m_demodBufferFill++] = l; // FIXME: What about right channel?
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if (m_demodBufferFill >= m_demodBuffer.size())
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{
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QList<DataFifo*> *dataFifos = MainCore::instance()->getDataPipes().getFifos(m_channel, "demod");
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if (dataFifos)
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{
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QList<DataFifo*>::iterator it = dataFifos->begin();
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for (; it != dataFifos->end(); ++it) {
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(*it)->write((quint8*) &m_demodBuffer[0], m_demodBuffer.size() * sizeof(qint16));
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}
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}
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m_demodBufferFill = 0;
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}
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}
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DABDemodSink::DABDemodSink(DABDemod *packetDemod) :
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m_dabDemod(packetDemod),
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m_audioSampleRate(48000),
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m_dabAudioSampleRate(10000), // Unused value to begin with
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m_channelSampleRate(DABDEMOD_CHANNEL_SAMPLE_RATE),
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m_channelFrequencyOffset(0),
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m_magsqSum(0.0f),
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m_magsqPeak(0.0f),
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m_magsqCount(0),
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m_messageQueueToChannel(nullptr),
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m_audioFifo(48000)
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{
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m_audioBuffer.resize(1<<14);
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m_audioBufferFill = 0;
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m_magsq = 0.0;
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m_demodBuffer.resize(1<<12);
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m_demodBufferFill = 0;
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applySettings(m_settings, true);
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applyChannelSettings(m_channelSampleRate, m_channelFrequencyOffset, true);
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int mode = 1; // Latest DAB spec only has mode 1
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m_dab = dabInit(&m_device,
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mode,
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syncHandler,
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systemDataHandler,
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ensembleNameHandler,
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programNameHandler,
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fibQualityHandler,
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audioHandler,
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dataHandler,
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byteHandler,
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programDataHandler,
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programQualityHandler,
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motDataHandler,
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nullptr,
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nullptr,
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this);
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dabStartProcessing(m_dab);
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}
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DABDemodSink::~DABDemodSink()
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{
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dabExit(m_dab);
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}
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void DABDemodSink::feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end)
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{
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Complex ci;
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for (SampleVector::const_iterator it = begin; it != end; ++it)
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{
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Complex c(it->real(), it->imag());
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c *= m_nco.nextIQ();
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if (m_interpolatorDistance == 1.0f)
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{
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processOneSample(c);
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}
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else if (m_interpolatorDistance < 1.0f) // interpolate
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{
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while (!m_interpolator.interpolate(&m_interpolatorDistanceRemain, c, &ci))
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{
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processOneSample(ci);
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m_interpolatorDistanceRemain += m_interpolatorDistance;
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}
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}
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else // decimate
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{
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if (m_interpolator.decimate(&m_interpolatorDistanceRemain, c, &ci))
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{
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processOneSample(ci);
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m_interpolatorDistanceRemain += m_interpolatorDistance;
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}
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}
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}
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}
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void DABDemodSink::processOneSample(Complex &ci)
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{
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// Calculate average and peak levels for level meter
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double magsqRaw = ci.real()*ci.real() + ci.imag()*ci.imag();
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Real magsq = (Real)(magsqRaw / (SDR_RX_SCALED*SDR_RX_SCALED));
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m_movingAverage(magsq);
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m_magsq = m_movingAverage.asDouble();
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m_magsqSum += magsq;
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if (magsq > m_magsqPeak)
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{
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m_magsqPeak = magsq;
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}
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m_magsqCount++;
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// Send sample to DAB library
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std::complex<float> c;
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c.real(ci.real()/SDR_RX_SCALED);
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c.imag(ci.imag()/SDR_RX_SCALED);
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m_device.putSample(c);
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}
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void DABDemodSink::applyChannelSettings(int channelSampleRate, int channelFrequencyOffset, bool force)
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{
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qDebug() << "DABDemodSink::applyChannelSettings:"
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<< " channelSampleRate: " << channelSampleRate
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<< " channelFrequencyOffset: " << channelFrequencyOffset;
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if ((m_channelFrequencyOffset != channelFrequencyOffset) ||
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(m_channelSampleRate != channelSampleRate) || force)
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{
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m_nco.setFreq(-channelFrequencyOffset, channelSampleRate);
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}
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if ((m_channelSampleRate != channelSampleRate) || force)
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{
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m_interpolator.create(16, channelSampleRate, m_settings.m_rfBandwidth / 2.2);
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m_interpolatorDistance = (Real) channelSampleRate / (Real) DABDEMOD_CHANNEL_SAMPLE_RATE;
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m_interpolatorDistanceRemain = m_interpolatorDistance;
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}
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m_channelSampleRate = channelSampleRate;
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m_channelFrequencyOffset = channelFrequencyOffset;
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}
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void DABDemodSink::applySettings(const DABDemodSettings& settings, bool force)
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{
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qDebug() << "DABDemodSink::applySettings:"
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<< " force: " << force;
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|
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if ((settings.m_rfBandwidth != m_settings.m_rfBandwidth) || force)
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{
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m_interpolator.create(16, m_channelSampleRate, settings.m_rfBandwidth / 2.2);
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m_interpolatorDistance = (Real) m_channelSampleRate / (Real) DABDEMOD_CHANNEL_SAMPLE_RATE;
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m_interpolatorDistanceRemain = m_interpolatorDistance;
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}
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|
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if ((settings.m_program != m_settings.m_program) || force)
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{
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if (!settings.m_program.isEmpty())
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{
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QByteArray ba = settings.m_program.toLatin1();
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const char *program = ba.data();
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if (!is_audioService (m_dab, program))
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qWarning() << settings.m_program << " is not an audio service";
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else
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{
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dataforAudioService(m_dab, program, &m_ad, 0);
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if (!m_ad.defined)
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qWarning() << settings.m_program << " audio data is not defined";
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else
|
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{
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dabReset_msc(m_dab);
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set_audioChannel(m_dab, &m_ad);
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}
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|
}
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|
}
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|
}
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|
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m_settings = settings;
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}
|
|
|
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// Called when audio device sample rate changes
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|
void DABDemodSink::applyAudioSampleRate(int sampleRate)
|
|
{
|
|
if (sampleRate < 0)
|
|
{
|
|
qWarning("DABDemodSink::applyAudioSampleRate: invalid sample rate: %d", sampleRate);
|
|
return;
|
|
}
|
|
|
|
qDebug("DABDemodSink::applyAudioSampleRate: m_audioSampleRate: %d m_dabAudioSampleRate: %d", sampleRate, m_dabAudioSampleRate);
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|
|
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m_audioInterpolator.create(16, m_dabAudioSampleRate, m_dabAudioSampleRate/2.2f);
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m_audioInterpolatorDistanceRemain = 0;
|
|
m_audioInterpolatorDistance = (Real) m_dabAudioSampleRate / (Real) sampleRate;
|
|
|
|
m_audioFifo.setSize(sampleRate);
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|
|
|
m_audioSampleRate = sampleRate;
|
|
}
|
|
|
|
// Called when DAB audio sample rate changes
|
|
void DABDemodSink::applyDABAudioSampleRate(int sampleRate)
|
|
{
|
|
qDebug("DABDemodSink::applyDABAudioSampleRate: m_audioSampleRate: %d new m_dabAudioSampleRate: %d", m_audioSampleRate, sampleRate);
|
|
|
|
m_audioInterpolator.create(16, sampleRate, sampleRate/2.2f);
|
|
m_audioInterpolatorDistanceRemain = 0;
|
|
m_audioInterpolatorDistance = (Real) sampleRate / (Real) m_audioSampleRate;
|
|
|
|
m_dabAudioSampleRate = sampleRate;
|
|
}
|