mirror of
https://github.com/cjcliffe/CubicSDR.git
synced 2024-11-26 13:48:38 -05:00
Cleanup: audio file, audio threads
This commit is contained in:
parent
19b1ecd7c2
commit
0b97bd360a
@ -5,13 +5,9 @@
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#include "CubicSDR.h"
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#include <sstream>
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AudioFile::AudioFile() {
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AudioFile::AudioFile() = default;
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}
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AudioFile::~AudioFile() {
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}
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AudioFile::~AudioFile() = default;
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void AudioFile::setOutputFileName(std::string filename) {
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filenameBase = filename;
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@ -22,7 +22,8 @@ namespace little_endian_io
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template <typename Word>
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std::istream& read_word(std::istream& ins, Word& value, unsigned size = sizeof(Word)) {
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for (unsigned n = 0, value = 0; n < size; ++n) {
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value = 0;
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for (unsigned n = 0; n < size; ++n) {
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value |= ins.get() << (8 * n);
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}
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return ins;
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@ -50,11 +51,10 @@ namespace big_endian_io
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using namespace little_endian_io;
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AudioFileWAV::AudioFileWAV() : AudioFile() {
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AudioFileWAV::AudioFileWAV() : AudioFile(), dataChunkPos(0) {
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}
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AudioFileWAV::~AudioFileWAV() {
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}
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AudioFileWAV::~AudioFileWAV() = default;
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std::string AudioFileWAV::getExtension()
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@ -119,7 +119,7 @@ bool AudioFileWAV::closeFile()
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return true;
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}
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void AudioFileWAV::writeHeaderToFileStream(AudioThreadInputPtr input) {
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void AudioFileWAV::writeHeaderToFileStream(const AudioThreadInputPtr& input) {
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// Based on simple wav file output code from
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// http://www.cplusplus.com/forum/beginner/166954/
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@ -140,7 +140,7 @@ void AudioFileWAV::writeHeaderToFileStream(AudioThreadInputPtr input) {
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outputFileStream << "data----"; // (chunk size to be filled in later)
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}
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void AudioFileWAV::writePayloadToFileStream(AudioThreadInputPtr input, size_t startInputPosition, size_t endInputPosition) {
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void AudioFileWAV::writePayloadToFileStream(const AudioThreadInputPtr& input, size_t startInputPosition, size_t endInputPosition) {
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// Prevent clipping
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float intScale = (input->peak < 1.0) ? 32767.0f : (32767.0f / input->peak);
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@ -164,7 +164,7 @@ void AudioFileWAV::writePayloadToFileStream(AudioThreadInputPtr input, size_t st
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}
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}
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size_t AudioFileWAV::getMaxWritableNumberOfSamples(AudioThreadInputPtr input) {
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size_t AudioFileWAV::getMaxWritableNumberOfSamples(const AudioThreadInputPtr& input) const {
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long long remainingBytesInFile = (long long)(MAX_WAV_FILE_SIZE) - currentFileSize;
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@ -11,19 +11,19 @@ class AudioFileWAV : public AudioFile {
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public:
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AudioFileWAV();
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~AudioFileWAV();
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~AudioFileWAV() override;
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//override to manage name change with multi-part WAV.
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virtual void setOutputFileName(std::string filename);
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void setOutputFileName(std::string filename) override;
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//override of the base method to generate multi-part
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//WAV to overcome the WAV format size limit.
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virtual std::string getOutputFileName();
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std::string getOutputFileName() override;
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virtual std::string getExtension();
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std::string getExtension() override;
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virtual bool writeToFile(AudioThreadInputPtr input);
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virtual bool closeFile();
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bool writeToFile(AudioThreadInputPtr input) override;
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bool closeFile() override;
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protected:
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std::ofstream outputFileStream;
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@ -33,10 +33,10 @@ protected:
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private:
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size_t getMaxWritableNumberOfSamples(AudioThreadInputPtr input);
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size_t getMaxWritableNumberOfSamples(const AudioThreadInputPtr& input) const;
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void writeHeaderToFileStream(AudioThreadInputPtr input);
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void writeHeaderToFileStream(const AudioThreadInputPtr& input);
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//write [startInputPosition; endInputPosition[ samples from input into the file.
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void writePayloadToFileStream(AudioThreadInputPtr input, size_t startInputPosition, size_t endInputPosition);
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void writePayloadToFileStream(const AudioThreadInputPtr& input, size_t startInputPosition, size_t endInputPosition);
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};
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@ -11,7 +11,7 @@ class AudioSinkFileThread : public AudioSinkThread {
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public:
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AudioSinkFileThread();
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~AudioSinkFileThread();
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~AudioSinkFileThread() override;
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enum SquelchOption {
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SQUELCH_RECORD_SILENCE = 0, // default value, record as a user would hear it.
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@ -20,8 +20,8 @@ public:
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SQUELCH_RECORD_MAX
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};
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virtual void sink(AudioThreadInputPtr input);
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virtual void inputChanged(AudioThreadInput oldProps, AudioThreadInputPtr newProps);
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void sink(AudioThreadInputPtr input) override;
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void inputChanged(AudioThreadInput oldProps, AudioThreadInputPtr newProps) override;
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void setAudioFileHandler(AudioFile *output);
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@ -11,9 +11,7 @@ AudioSinkThread::AudioSinkThread() {
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setInputQueue("input", inputQueuePtr);
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}
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AudioSinkThread::~AudioSinkThread() {
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}
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AudioSinkThread::~AudioSinkThread() = default;
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void AudioSinkThread::run() {
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#ifdef __APPLE__
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@ -10,10 +10,10 @@ class AudioSinkThread : public IOThread {
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public:
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AudioSinkThread();
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virtual ~AudioSinkThread();
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~AudioSinkThread() override;
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virtual void run();
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virtual void terminate();
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void run() override;
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void terminate() override;
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virtual void sink(AudioThreadInputPtr input) = 0;
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virtual void inputChanged(AudioThreadInput oldProps, AudioThreadInputPtr newProps) = 0;
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@ -2,13 +2,10 @@
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// SPDX-License-Identifier: GPL-2.0+
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#include "AudioThread.h"
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#include "CubicSDRDefs.h"
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#include <vector>
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#include <algorithm>
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#include "CubicSDR.h"
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#include "DemodulatorThread.h"
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#include "DemodulatorInstance.h"
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#include <memory.h>
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#include <mutex>
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//50 ms
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@ -80,9 +77,9 @@ void AudioThread::deviceCleanup() {
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//NOT PROTECTED by m_device_mutex on purpose, to prevent deadlocks with i->second->controllerThread
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// it doesn't matter, it is only called when all "normal" audio threads are detached from the controller.
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//
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for (auto i = deviceController.begin(); i != deviceController.end(); i++) {
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for (auto & i : deviceController) {
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delete i->second;
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delete i.second;
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}
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deviceController.clear();
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@ -98,7 +95,7 @@ static int audioCallback(void *outputBuffer, void * /* inputBuffer */, unsigned
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::memset(out, 0, nBufferFrames * 2 * sizeof(float));
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//src in the controller thread:
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AudioThread *src = (AudioThread *)userData;
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auto *src = (AudioThread *)userData;
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//by construction, src is a controller thread, from deviceController:
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std::lock_guard<std::recursive_mutex> lock(src->getMutex());
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@ -157,7 +154,7 @@ static int audioCallback(void *outputBuffer, void * /* inputBuffer */, unsigned
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}
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if (srcmix->currentInput->channels == 0 || !srcmix->currentInput->data.size()) {
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if (srcmix->currentInput->channels == 0 || srcmix->currentInput->data.empty()) {
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if (!srcmix->inputQueue->empty()) {
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srcmix->audioQueuePtr = 0;
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if (srcmix->currentInput) {
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@ -195,7 +192,7 @@ static int audioCallback(void *outputBuffer, void * /* inputBuffer */, unsigned
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mixPeak = srcPeak;
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}
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}
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if (srcmix->currentInput && srcmix->currentInput->data.size()) {
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if (srcmix->currentInput && !srcmix->currentInput->data.empty()) {
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float v = srcmix->currentInput->data[srcmix->audioQueuePtr] * srcmix->gain;
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out[i * 2] += v;
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out[i * 2 + 1] += v;
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@ -204,7 +201,7 @@ static int audioCallback(void *outputBuffer, void * /* inputBuffer */, unsigned
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}
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}
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else {
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for (int i = 0, iMax = srcmix->currentInput->channels * nBufferFrames; i < iMax; i++) {
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for (unsigned int i = 0, iMax = srcmix->currentInput->channels * nBufferFrames; i < iMax; i++) {
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if (srcmix->audioQueuePtr >= srcmix->currentInput->data.size()) {
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srcmix->audioQueuePtr = 0;
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@ -222,7 +219,7 @@ static int audioCallback(void *outputBuffer, void * /* inputBuffer */, unsigned
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mixPeak = srcPeak;
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}
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}
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if (srcmix->currentInput && srcmix->currentInput->data.size()) {
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if (srcmix->currentInput && !srcmix->currentInput->data.empty()) {
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out[i] = out[i] + srcmix->currentInput->data[srcmix->audioQueuePtr] * srcmix->gain;
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}
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@ -248,9 +245,9 @@ static int audioCallback(void *outputBuffer, void * /* inputBuffer */, unsigned
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void AudioThread::enumerateDevices(std::vector<RtAudio::DeviceInfo> &devs) {
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RtAudio endac;
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int numDevices = endac.getDeviceCount();
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unsigned int numDevices = endac.getDeviceCount();
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for (int i = 0; i < numDevices; i++) {
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for (unsigned int i = 0; i < numDevices; i++) {
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RtAudio::DeviceInfo info = endac.getDeviceInfo(i);
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devs.push_back(info);
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@ -322,7 +319,7 @@ void AudioThread::setDeviceSampleRate(int deviceId, int sampleRate) {
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}
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}
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void AudioThread::setSampleRate(int sampleRate) {
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void AudioThread::setSampleRate(int sampleRate_in) {
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bool thisIsAController = false;
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@ -332,7 +329,7 @@ void AudioThread::setSampleRate(int sampleRate) {
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if (deviceController[outputDevice.load()] == this) {
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thisIsAController = true;
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deviceSampleRate[outputDevice.load()] = sampleRate;
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deviceSampleRate[outputDevice.load()] = sampleRate_in;
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}
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}
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@ -344,31 +341,30 @@ void AudioThread::setSampleRate(int sampleRate) {
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dac.closeStream();
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//Set bounded sample rate:
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for (size_t j = 0; j < boundThreads.size(); j++) {
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AudioThread *srcmix = boundThreads[j];
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srcmix->setSampleRate(sampleRate);
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for (auto srcmix : boundThreads) {
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srcmix->setSampleRate(sampleRate_in);
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}
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//make a local copy, snapshot of the list of demodulators
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std::vector<DemodulatorInstancePtr> demodulators = wxGetApp().getDemodMgr().getDemodulators();
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for (auto demod : demodulators) {
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for (const auto& demod : demodulators) {
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if (demod->getOutputDevice() == outputDevice.load()) {
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demod->setAudioSampleRate(sampleRate);
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demod->setAudioSampleRate(sampleRate_in);
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}
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}
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dac.openStream(¶meters, NULL, RTAUDIO_FLOAT32, sampleRate, &nBufferFrames, &audioCallback, (void *)this, &opts);
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dac.openStream(¶meters, nullptr, RTAUDIO_FLOAT32, sampleRate_in, &nBufferFrames, &audioCallback, (void *)this, &opts);
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dac.startStream();
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}
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this->sampleRate = sampleRate;
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sampleRate = sampleRate_in;
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}
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int AudioThread::getSampleRate() {
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std::lock_guard<std::recursive_mutex> lock(m_mutex);
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return this->sampleRate;
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return sampleRate;
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}
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void AudioThread::setupDevice(int deviceId) {
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@ -411,7 +407,7 @@ void AudioThread::setupDevice(int deviceId) {
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if (deviceController.find(parameters.deviceId) == deviceController.end()) {
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//Create a new controller thread for parameters.deviceId:
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AudioThread* newController = new AudioThread();
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auto* newController = new AudioThread();
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newController->setInitOutputDevice(parameters.deviceId, sampleRate);
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newController->bindThread(this);
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@ -422,7 +418,7 @@ void AudioThread::setupDevice(int deviceId) {
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else if (deviceController[parameters.deviceId] == this) {
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//Attach callback
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dac.openStream(¶meters, NULL, RTAUDIO_FLOAT32, sampleRate, &nBufferFrames, &audioCallback, (void *)this, &opts);
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dac.openStream(¶meters, nullptr, RTAUDIO_FLOAT32, sampleRate, &nBufferFrames, &audioCallback, (void *)this, &opts);
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dac.startStream();
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}
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else {
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@ -455,21 +451,21 @@ int AudioThread::getOutputDevice() {
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return outputDevice;
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}
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void AudioThread::setInitOutputDevice(int deviceId, int sampleRate) {
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void AudioThread::setInitOutputDevice(int deviceId, int sampleRate_in) {
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//global lock
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std::lock_guard<std::recursive_mutex> lock(m_device_mutex);
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outputDevice = deviceId;
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if (sampleRate == -1) {
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if (sampleRate_in == -1) {
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if (deviceSampleRate.find(parameters.deviceId) != deviceSampleRate.end()) {
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sampleRate = deviceSampleRate[deviceId];
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sampleRate_in = deviceSampleRate[deviceId];
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}
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}
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else {
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deviceSampleRate[deviceId] = sampleRate;
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deviceSampleRate[deviceId] = sampleRate_in;
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}
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this->sampleRate = sampleRate;
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sampleRate = sampleRate_in;
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}
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void AudioThread::run() {
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@ -517,20 +513,20 @@ void AudioThread::run() {
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//Nullify currentInput...
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currentInput = nullptr;
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//Stop : Retreive the matching controling thread in a scope lock:
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AudioThread* controllerThread = nullptr;
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//Stop : Retrieve the matching controlling thread in a scope lock:
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AudioThread* controllerMatchingThread;
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{
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std::lock_guard<std::recursive_mutex> global_lock(m_device_mutex);
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controllerThread = deviceController[parameters.deviceId];
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controllerMatchingThread = deviceController[parameters.deviceId];
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}
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if (controllerThread != this) {
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if (controllerMatchingThread != this) {
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//'this' is not the controller, so remove it from the bounded list:
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//beware, we must take the controller mutex, because the audio callback may use the list of bounded
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//threads at that moment:
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std::lock_guard<std::recursive_mutex> lock(controllerThread->getMutex());
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std::lock_guard<std::recursive_mutex> lock(controllerMatchingThread->getMutex());
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controllerThread->removeThread(this);
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controllerMatchingThread->removeThread(this);
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}
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else {
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// 'this' is a controller thread:
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@ -15,13 +15,13 @@
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class AudioThreadInput {
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public:
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long long frequency;
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int inputRate;
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int sampleRate;
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int channels;
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float peak;
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int type;
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bool is_squelch_active;
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long long frequency{};
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int inputRate{};
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int sampleRate{};
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int channels{};
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float peak{};
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int type{};
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bool is_squelch_active{};
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std::vector<float> data;
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@ -31,7 +31,7 @@ public:
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}
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AudioThreadInput(AudioThreadInput *copyFrom) {
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explicit AudioThreadInput(AudioThreadInput *copyFrom) {
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copy(copyFrom);
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}
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@ -47,9 +47,7 @@ public:
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}
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virtual ~AudioThreadInput() {
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}
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virtual ~AudioThreadInput() = default;
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};
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typedef std::shared_ptr<AudioThreadInput> AudioThreadInputPtr;
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@ -83,17 +81,17 @@ class AudioThread : public IOThread {
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public:
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AudioThread();
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virtual ~AudioThread();
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~AudioThread() override;
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static void enumerateDevices(std::vector<RtAudio::DeviceInfo> &devs);
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void setInitOutputDevice(int deviceId, int sampleRate = -1);
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void setInitOutputDevice(int deviceId, int sampleRate_in = -1);
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int getOutputDevice();
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int getSampleRate();
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virtual void run();
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virtual void terminate();
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void run() override;
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void terminate() override;
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bool isActive();
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void setActive(bool state);
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@ -141,7 +139,7 @@ private:
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std::recursive_mutex m_mutex;
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void setupDevice(int deviceId);
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void setSampleRate(int sampleRate);
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void setSampleRate(int sampleRate_in);
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void bindThread(AudioThread *other);
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void removeThread(AudioThread *other);
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