mirror of
https://github.com/AlexandreRouma/SDRPlusPlus.git
synced 2025-06-26 12:27:51 +02:00
Changed project structure
This commit is contained in:
AlexandreRouma
37
sink_modules/new_portaudio_sink/CMakeLists.txt
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37
sink_modules/new_portaudio_sink/CMakeLists.txt
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cmake_minimum_required(VERSION 3.13)
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project(new_portaudio_sink)
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if (MSVC)
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add_compile_options(/O2 /Ob2 /std:c++17 /EHsc)
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elseif (CMAKE_CXX_COMPILER_ID MATCHES "Clang")
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add_compile_options(-O3 -std=c++17 -Wno-unused-command-line-argument -undefined dynamic_lookup)
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else ()
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add_compile_options(-O3 -std=c++17 -fpermissive)
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endif (MSVC)
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file(GLOB SRC "src/*.cpp")
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include_directories("src/")
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add_library(new_portaudio_sink SHARED ${SRC})
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target_link_libraries(new_portaudio_sink PRIVATE sdrpp_core)
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set_target_properties(new_portaudio_sink PROPERTIES PREFIX "")
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if (MSVC)
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find_package(portaudio CONFIG REQUIRED)
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target_link_libraries(new_portaudio_sink PUBLIC portaudio)
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else (MSVC)
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find_package(PkgConfig)
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pkg_check_modules(PORTAUDIO REQUIRED portaudio-2.0)
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target_include_directories(new_portaudio_sink PUBLIC ${PORTAUDIO_INCLUDE_DIRS})
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target_link_directories(new_portaudio_sink PUBLIC ${PORTAUDIO_LIBRARY_DIRS})
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target_link_libraries(new_portaudio_sink PUBLIC ${PORTAUDIO_LIBRARIES})
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endif (MSVC)
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# Install directives
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install(TARGETS new_portaudio_sink DESTINATION lib/sdrpp/plugins)
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445
sink_modules/new_portaudio_sink/src/main.cpp
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445
sink_modules/new_portaudio_sink/src/main.cpp
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#include <imgui.h>
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#include <module.h>
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#include <gui/gui.h>
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#include <signal_path/signal_path.h>
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#include <signal_path/sink.h>
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#include <portaudio.h>
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#include <dsp/audio.h>
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#include <dsp/processing.h>
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#include <spdlog/spdlog.h>
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#include <config.h>
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#include <algorithm>
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#include <options.h>
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#define CONCAT(a, b) ((std::string(a) + b).c_str())
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#define BLOCK_SIZE_DIVIDER 60
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#define AUDIO_LATENCY 1.0 / 60.0
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SDRPP_MOD_INFO {
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/* Name: */ "new_portaudio_sink",
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/* Description: */ "Audio sink module for SDR++",
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/* Author: */ "Ryzerth;Maxime Biette",
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/* Version: */ 0, 1, 0,
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/* Max instances */ 1
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};
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ConfigManager config;
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class AudioSink : SinkManager::Sink {
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public:
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struct AudioDevice_t {
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const PaDeviceInfo* deviceInfo;
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const PaHostApiInfo* hostApiInfo;
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PaDeviceIndex id;
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int defaultSrId;
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PaStreamParameters outputParams;
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std::vector<double> sampleRates;
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std::string sampleRatesTxt;
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};
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AudioSink(SinkManager::Stream* stream, std::string streamName) {
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_stream = stream;
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_streamName = streamName;
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// Create config if it doesn't exist
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config.acquire();
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if (!config.conf.contains(_streamName)) {
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config.conf[_streamName]["device"] = "";
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config.conf[_streamName]["devices"] = json::object();
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}
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std::string selected = config.conf[_streamName]["device"];
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config.release(true);
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// Register the play state handler
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playStateHandler.handler = playStateChangeHandler;
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playStateHandler.ctx = this;
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gui::mainWindow.onPlayStateChange.bindHandler(&playStateHandler);
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// Initialize DSP blocks
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packer.init(_stream->sinkOut, 1024);
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s2m.init(&packer.out);
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// Refresh devices and select the one from the config
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refreshDevices();
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selectDevByName(selected);
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}
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~AudioSink() {
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stop();
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gui::mainWindow.onPlayStateChange.unbindHandler(&playStateHandler);
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}
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void start() {
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if (running || selectedDevName.empty()) { return; }
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// Get device and samplerate
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AudioDevice_t& dev = devices[deviceNames[devId]];
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double sampleRate = dev.sampleRates[srId];
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int blockSize = sampleRate / BLOCK_SIZE_DIVIDER;
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// Set the SDR++ stream sample rate
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_stream->setSampleRate(sampleRate);
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// Update the block size on the packer
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packer.setSampleCount(blockSize);
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// Clear read stop signals
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packer.out.clearReadStop();
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s2m.out.clearReadStop();
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// Open the stream
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PaError err;
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if (dev.deviceInfo->maxOutputChannels == 1) {
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packer.start();
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s2m.start();
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stereo = false;
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err = Pa_OpenStream(&devStream, NULL, &dev.outputParams, sampleRate, blockSize, paNoFlag, _mono_cb, this);
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}
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else {
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packer.start();
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stereo = true;
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err = Pa_OpenStream(&devStream, NULL, &dev.outputParams, sampleRate, blockSize, paNoFlag, _stereo_cb, this);
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}
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// In case of error, abort
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if (err) {
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spdlog::error("PortAudio error {0}: {1}", err, Pa_GetErrorText(err));
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return;
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}
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spdlog::info("Starting PortAudio stream at {0} S/s", sampleRate);
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// Start stream
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Pa_StartStream(devStream);
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running = true;
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}
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void stop() {
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if (!running || selectedDevName.empty()) { return; }
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// Send stop signal to the streams
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packer.out.stopReader();
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s2m.out.stopReader();
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// Stop DSP
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packer.stop();
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s2m.stop();
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// Stop stream
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Pa_AbortStream(devStream);
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// Close the stream
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Pa_CloseStream(devStream);
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running = false;
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}
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void menuHandler() {
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float menuWidth = ImGui::GetContentRegionAvailWidth();
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// Select device
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ImGui::SetNextItemWidth(menuWidth);
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if (ImGui::Combo("##audio_sink_dev_sel", &devId, deviceNamesTxt.c_str())) {
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selectDevByName(deviceNames[devId]);
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stop();
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start();
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if (selectedDevName != "") {
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config.acquire();
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config.conf[_streamName]["device"] = selectedDevName;
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config.release(true);
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}
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}
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// Select sample rate
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ImGui::SetNextItemWidth(menuWidth);
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if (ImGui::Combo("##audio_sink_sr_sel", &srId, selectedDev.sampleRatesTxt.c_str())) {
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stop();
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start();
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if (selectedDevName != "") {
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config.acquire();
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config.conf[_streamName]["devices"][selectedDevName] = selectedDev.sampleRates[srId];
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config.release(true);
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}
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}
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}
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int devId = 0;
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int srId = 0;
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bool stereo = false;
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private:
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static void playStateChangeHandler(bool newState, void* ctx) {
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AudioSink* _this = (AudioSink*)ctx;
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// Wake up reader to send nulls instead of data in preparation for shutoff
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if (newState) {
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if (_this->stereo) {
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_this->packer.out.stopReader();
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}
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else {
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_this->s2m.out.stopReader();
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}
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}
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else {
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if (_this->stereo) {
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_this->packer.out.clearReadStop();
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}
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else {
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_this->s2m.out.clearReadStop();
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}
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}
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}
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void refreshDevices() {
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// Clear current list
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devices.clear();
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deviceNames.clear();
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deviceNamesTxt.clear();
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// Get number of devices
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int devCount = Pa_GetDeviceCount();
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PaStreamParameters outputParams;
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char buffer[128];
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for (int i = 0; i < devCount; i++) {
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AudioDevice_t dev;
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// Get device info
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dev.deviceInfo = Pa_GetDeviceInfo(i);
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dev.hostApiInfo = Pa_GetHostApiInfo(dev.deviceInfo->hostApi);
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dev.id = i;
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// Check if device is usable
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if (dev.deviceInfo->maxOutputChannels == 0) { continue; }
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#ifdef _WIN32
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// On Windows, use only WASAPI
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if (dev.hostApiInfo->type == paMME || dev.hostApiInfo->type == paWDMKS) { continue; }
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#endif
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// Zero out output params
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dev.outputParams.device = i;
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dev.outputParams.sampleFormat = paFloat32;
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dev.outputParams.suggestedLatency = std::min<PaTime>(AUDIO_LATENCY, dev.deviceInfo->defaultLowOutputLatency);
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dev.outputParams.channelCount = std::min<int>(dev.deviceInfo->maxOutputChannels, 2);
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dev.outputParams.hostApiSpecificStreamInfo = NULL;
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// List available sample rates
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for (int sr = 12000; sr < 200000; sr += 12000) {
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if (Pa_IsFormatSupported(NULL, &dev.outputParams, sr) != paFormatIsSupported) { continue; }
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dev.sampleRates.push_back(sr);
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}
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for (int sr = 11025; sr < 192000; sr += 11025) {
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if (Pa_IsFormatSupported(NULL, &dev.outputParams, sr) != paFormatIsSupported) { continue; }
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dev.sampleRates.push_back(sr);
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}
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// If no sample rates are supported, cancel adding device
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if (dev.sampleRates.empty()) {
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continue;
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}
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// Sort sample rate list
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std::sort(dev.sampleRates.begin(), dev.sampleRates.end(), [](double a, double b) { return (a < b); });
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// Generate text list for UI
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int srId = 0;
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int _48kId = -1;
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for (auto sr : dev.sampleRates) {
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sprintf(buffer, "%d", (int)sr);
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dev.sampleRatesTxt += buffer;
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dev.sampleRatesTxt += '\0';
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// Save ID of the default sample rate and 48KHz
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if (sr == dev.deviceInfo->defaultSampleRate) { dev.defaultSrId = srId; }
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if (sr == 48000.0) { _48kId = srId; }
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srId++;
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}
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// If a 48KHz option was found, use it instead of the default
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if (_48kId >= 0) { dev.defaultSrId = _48kId; }
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std::string apiName = dev.hostApiInfo->name;
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#ifdef _WIN32
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// Shorten the names on windows
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if (apiName.rfind("Windows ", 0) == 0) {
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apiName = apiName.substr(8);
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}
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#endif
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// Create device name and save to list
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sprintf(buffer, "[%s] %s", apiName.c_str(), dev.deviceInfo->name);
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devices[buffer] = dev;
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deviceNames.push_back(buffer);
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deviceNamesTxt += buffer;
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deviceNamesTxt += '\0';
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}
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}
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void selectDefault() {
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if (devices.empty()) {
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selectedDevName = "";
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return;
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}
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// Search for the default device
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PaDeviceIndex defId = Pa_GetDefaultOutputDevice();
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for (auto const& [name, dev] : devices) {
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if (dev.id != defId) { continue; }
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selectDevByName(name);
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return;
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}
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// If default not found, select first
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selectDevByName(deviceNames[0]);
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}
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void selectDevByName(std::string name) {
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auto devIt = std::find(deviceNames.begin(), deviceNames.end(), name);
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if (devIt == deviceNames.end()) {
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selectDefault();
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return;
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}
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// Load the device name, device descriptor and device ID
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selectedDevName = name;
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selectedDev = devices[name];
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devId = std::distance(deviceNames.begin(), devIt);
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// Load config
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config.acquire();
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if (!config.conf[_streamName]["devices"].contains(name)) {
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config.conf[_streamName]["devices"][name] = selectedDev.sampleRates[selectedDev.defaultSrId];
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}
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config.release(true);
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// Find the sample rate ID, if not use default
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bool found = false;
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double selectedSr = config.conf[_streamName]["devices"][name];
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for (int i = 0; i < selectedDev.sampleRates.size(); i++) {
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if (selectedDev.sampleRates[i] != selectedSr) { continue; }
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srId = i;
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found = true;
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break;
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}
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if (!found) {
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srId = selectedDev.defaultSrId;
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}
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}
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static int _mono_cb(const void *input, void *output, unsigned long frameCount,
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const PaStreamCallbackTimeInfo* timeInfo, PaStreamCallbackFlags statusFlags, void *userData) {
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AudioSink* _this = (AudioSink*)userData;
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// For OSX, mute audio when not playing
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if (!gui::mainWindow.isPlaying()) {
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memset(output, 0, frameCount*sizeof(float));
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_this->s2m.out.flush();
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return 0;
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}
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// Write to buffer
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_this->s2m.out.read();
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memcpy(output, _this->s2m.out.readBuf, frameCount * sizeof(float));
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_this->s2m.out.flush();
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return 0;
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}
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static int _stereo_cb(const void *input, void *output, unsigned long frameCount,
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const PaStreamCallbackTimeInfo* timeInfo, PaStreamCallbackFlags statusFlags, void *userData) {
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AudioSink* _this = (AudioSink*)userData;
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// For OSX, mute audio when not playing
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if (!gui::mainWindow.isPlaying()) {
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memset(output, 0, frameCount*sizeof(dsp::stereo_t));
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_this->packer.out.flush();
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return 0;
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}
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// Write to buffer
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_this->packer.out.read();
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memcpy(output, _this->packer.out.readBuf, frameCount * sizeof(dsp::stereo_t));
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_this->packer.out.flush();
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return 0;
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}
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std::string _streamName;
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bool running = false;
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std::map<std::string, AudioDevice_t> devices;
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std::vector<std::string> deviceNames;
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std::string deviceNamesTxt;
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AudioDevice_t selectedDev;
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std::string selectedDevName;
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SinkManager::Stream* _stream;
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dsp::Packer<dsp::stereo_t> packer;
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dsp::StereoToMono s2m;
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PaStream *devStream;
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EventHandler<bool> playStateHandler;
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};
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class AudioSinkModule : public ModuleManager::Instance {
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public:
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AudioSinkModule(std::string name) {
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this->name = name;
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provider.create = create_sink;
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provider.ctx = this;
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Pa_Initialize();
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sigpath::sinkManager.registerSinkProvider("New Audio", provider);
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}
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~AudioSinkModule() {
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sigpath::sinkManager.unregisterSinkProvider("New Audio");
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Pa_Terminate();
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}
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void postInit() {}
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void enable() {
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enabled = true;
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}
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void disable() {
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enabled = false;
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}
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bool isEnabled() {
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return enabled;
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}
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private:
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static SinkManager::Sink* create_sink(SinkManager::Stream* stream, std::string streamName, void* ctx) {
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return (SinkManager::Sink*)(new AudioSink(stream, streamName));
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}
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std::string name;
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bool enabled = true;
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SinkManager::SinkProvider provider;
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};
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MOD_EXPORT void _INIT_() {
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config.setPath(options::opts.root + "/new_audio_sink_config.json");
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config.load(json::object());
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config.enableAutoSave();
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}
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MOD_EXPORT void* _CREATE_INSTANCE_(std::string name) {
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AudioSinkModule* instance = new AudioSinkModule(name);
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return instance;
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}
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MOD_EXPORT void _DELETE_INSTANCE_(void* instance) {
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delete (AudioSinkModule*)instance;
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}
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MOD_EXPORT void _END_() {
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config.disableAutoSave();
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config.save();
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}
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