freedv-gui/src/integrations/common/MinimalTxRxThread.cpp

427 lines
15 KiB
C++

//=========================================================================
// Name: MinimalTxRxThread.cpp
// Purpose: Implements the main processing thread for audio I/O.
//
// Authors: Mooneer Salem
// License:
//
// All rights reserved.
//
// Redistribution and use in source and binary forms, with or without
// modification, are permitted provided that the following conditions
// are met:
//
// - Redistributions of source code must retain the above copyright
// notice, this list of conditions and the following disclaimer.
//
// - Redistributions in binary form must reproduce the above copyright
// notice, this list of conditions and the following disclaimer in the
// documentation and/or other materials provided with the distribution.
//
// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
// ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
// OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
// EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
// PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
// PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
// LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
// NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
// SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
//
//=========================================================================
#include <chrono>
using namespace std::chrono_literals;
#include "MinimalTxRxThread.h"
#include "../pipeline/paCallbackData.h"
#include "../../pipeline/AgcStep.h"
#include "../../pipeline/SpeexStep.h"
#include "../../pipeline/ResampleStep.h"
#include "../../pipeline/LevelAdjustStep.h"
#include "../../pipeline/RADEReceiveStep.h"
#include "../../pipeline/BandwidthExpandStep.h"
#include "../../util/logging/ulog.h"
#include "../../os/os_interface.h"
#include "../../pipeline/pipeline_defines.h"
#include "codec2_alloc.h"
extern std::atomic<bool> g_tx;
extern bool endingTx;
bool g_eoo_enqueued;
static const float TxScaleFactor_ = 2.0;
int MinimalTxRxThread::getTxNNomModemSamples() const
{
const int NUM_SAMPLES_SILENCE = 60 * RADE_MODEM_SAMPLE_RATE / 1000;
return std::max(rade_n_tx_out(rade_), rade_n_tx_eoo_out(rade_) + NUM_SAMPLES_SILENCE);
}
int MinimalTxRxThread::getRxNumSpeechSamples() const
{
return rade_n_features_in_out(rade_) * LPCNET_FRAME_SIZE / NB_TOTAL_FEATURES;
}
// Experimental options for potential future release:
//
// * ENABLE_FASTER_PLOTS: This uses a faster resampling algorithm to reduce the CPU
// usage required to generate various plots in the user interface. (Tech note: When
// enabled, libsamplerate is directed to use SRC_LINEAR for the plot resampling.)
// * ENABLE_PROCESSING_STATS: This causes execution statistics to be collected for RX and TX
// processing and output in the log after the user pushes Stop. (Define in .h file.)
#define ENABLE_FASTER_PLOTS
void MinimalTxRxThread::initializePipeline_()
{
pipeline_ = std::make_unique<AudioPipeline>(inputSampleRate_, outputSampleRate_);
if (m_tx)
{
txStep_ = new RADETransmitStep(rade_, encState_);
auto agcStep = new AgcStep(txStep_->getInputSampleRate());
auto speexStep = new SpeexStep(txStep_->getInputSampleRate());
pipeline_->appendPipelineStep(speexStep);
pipeline_->appendPipelineStep(agcStep);
pipeline_->appendPipelineStep(txStep_);
auto levelAdjustStep = new LevelAdjustStep(outputSampleRate_, +[]() FREEDV_NONBLOCKING { return TxScaleFactor_; });
pipeline_->appendPipelineStep(levelAdjustStep);
}
else
{
auto radeRxStep = new RADEReceiveStep(rade_, farganState_, radeText_, +[](RADEReceiveStep* step) FREEDV_NONBLOCKING {
MinimalTxRxThread* thisObj = (MinimalTxRxThread*)step->getStateObj();
thisObj->snr_.store(step->getSnr(), std::memory_order_release);
thisObj->sync_.store(step->getSync(), std::memory_order_release);
});
radeRxStep->setStateObj(this);
pipeline_->appendPipelineStep(radeRxStep);
auto bwExpandStep = new BandwidthExpandStep();
pipeline_->appendPipelineStep(bwExpandStep);
// Clear anything in the FIFO before resuming decode.
clearFifos_();
}
}
void* MinimalTxRxThread::Entry()
{
// Get raw pointer so we don't need to constantly access the shared_ptr
// and thus constantly increment/decrement refcounts.
IRealtimeHelper* helper = helper_.get();
// Ensure that O(1) memory allocator is used for Codec2
// instead of standard malloc().
codec2_initialize_realtime(CODEC2_REAL_TIME_MEMORY_SIZE);
initializePipeline_();
// Request real-time scheduling from the operating system.
helper->setHelperRealTime();
#if defined(ENABLE_PROCESSING_STATS)
resetStats_();
#endif // defined(ENABLE_PROCESSING_STATS)
#if defined(__linux__)
const char* threadName = nullptr;
if (m_tx) threadName = "FDV txThread";
else threadName = "FDV rxThread";
pthread_setname_np(pthread_self(), threadName);
#endif // defined(__linux__)
// Make sure we don't start processing until
// the main thread is ready.
readySem_.signal();
startSem_.wait();
clearFifos_();
while (m_run)
{
if (!m_run) break;
//log_info("thread woken up: m_tx=%d", (int)m_tx);
helper->startRealTimeWork();
if (m_tx) txProcessing_(helper);
else rxProcessing_(helper);
// Determine whether we need to pause for a shorter amount
// of time to avoid dropouts.
auto outFifo = cbData_->outfifo1;
if (!m_tx)
{
outFifo = cbData_->outfifo2;
}
auto totalFifoCapacity = outFifo->capacity();
auto fifoUsed = outFifo->numUsed();
helper->stopRealTimeWork(fifoUsed < totalFifoCapacity / 2);
}
#if defined(ENABLE_PROCESSING_STATS)
reportStats_();
#endif // defined(ENABLE_PROCESSING_STATS)
// Force pipeline to delete itself when we're done with the thread.
pipeline_ = nullptr;
// Return to normal scheduling
helper->clearHelperRealTime();
codec2_disable_realtime();
return NULL;
}
#if defined(ENABLE_PROCESSING_STATS)
void MinimalTxRxThread::resetStats_()
{
numTimeSamples_ = 0;
minDuration_ = 1e9;
maxDuration_ = 0;
sumDuration_ = 0;
sumDoubleDuration_ = 0;
}
void MinimalTxRxThread::startTimer_()
{
timeStart_ = std::chrono::high_resolution_clock::now();
}
void MinimalTxRxThread::endTimer_()
{
auto e = std::chrono::high_resolution_clock::now();
auto d = std::chrono::duration_cast<std::chrono::nanoseconds>(e - timeStart_).count();
numTimeSamples_++;
if (d < minDuration_)
{
minDuration_ = d;
minTime_ = std::chrono::system_clock::to_time_t(std::chrono::system_clock::now());
}
if (d > maxDuration_)
{
maxDuration_ = d;
maxTime_ = std::chrono::system_clock::to_time_t(std::chrono::system_clock::now());
}
sumDuration_ += d; sumDoubleDuration_ += pow(d, 2);
}
void MinimalTxRxThread::reportStats_()
{
if (numTimeSamples_ > 0)
{
std::tm * minTm = std::localtime(&minTime_);
std::tm * maxTm = std::localtime(&maxTime_);
char bufMin[32];
char bufMax[32];
std::strftime(bufMin, 32, "%H:%M:%S", minTm);
std::strftime(bufMax, 32, "%H:%M:%S", maxTm);
log_info("m_tx = %d, min = %f ns [%s], max = %f ns [%s], mean = %f ns, stdev = %f ns (n = %d)", m_tx, minDuration_, bufMin, maxDuration_, bufMax, sumDuration_ / numTimeSamples_, sqrt((sumDoubleDuration_ - pow(sumDuration_, 2)/numTimeSamples_) / (numTimeSamples_ - 1)), numTimeSamples_);
}
}
#endif // defined(ENABLE_PROCESSING_STATS)
void MinimalTxRxThread::clearFifos_()
{
if (m_tx)
{
cbData_->infifo1->reset();
cbData_->outfifo1->reset();
}
else
{
cbData_->infifo2->reset();
cbData_->outfifo2->reset();
}
}
//---------------------------------------------------------------------------------------------
// Main real time processing for tx and rx of FreeDV signals, run in its own threads
//---------------------------------------------------------------------------------------------
void MinimalTxRxThread::txProcessing_(IRealtimeHelper* helper) noexcept
#if defined(__clang__)
#if defined(__has_feature) && __has_feature(realtime_sanitizer)
[[clang::nonblocking]]
#endif // defined(__has_feature) && __has_feature(realtime_sanitizer)
#endif // defined(__clang__)
{
// Buffers re-used by tx and rx processing. We take samples from
// the sound card, and resample them for the freedv modem input
// sample rate. Typically the sound card is running at 48 or 44.1
// kHz, and the modem at 8kHz
//
// TX side processing --------------------------------------------
//
if (g_tx.load(std::memory_order_acquire)) {
if (deferReset_)
{
// We just entered TX from RX.
// Reset pipeline and wipe anything in the FIFO.
deferReset_ = false;
pipeline_->reset();
clearFifos_();
}
// This while loop locks the modulator to the sample rate of
// the input sound card. We want to make sure that modulator samples
// are uninterrupted by differences in sample rate between
// this sound card and the output sound card.
// Run code inside this while loop as soon as we have enough
// room for one frame of modem samples. Aim is to keep
// outfifo1 nice and full so we don't have any gaps in tx
// signal.
unsigned int nsam_one_modem_frame = (getTxNNomModemSamples() * outputSampleRate_) / RADE_MODEM_SAMPLE_RATE;
int nsam_in_48 = (inputSampleRate_ * FRAME_DURATION_MS) / MS_TO_SEC;
assert(nsam_in_48 > 0);
int nout;
while(!helper->mustStopWork() && (unsigned)cbData_->outfifo1->numFree() >= nsam_one_modem_frame) {
// OK to generate a frame of modem output samples we need
// an input frame of speech samples from the microphone.
#if defined(ENABLE_PROCESSING_STATS)
startTimer_();
#endif // defined(ENABLE_PROCESSING_STATS)
// infifo2 is written to by another sound card so it may
// over or underflow, but we don't really care. It will
// just result in a short interruption in audio being fed
// to codec2_enc, possibly making a click every now and
// again in the decoded audio at the other end.
// There may be recorded audio left to encode while ending TX. To handle this,
// we keep reading from the FIFO until we have less than nsam_in_48 samples available.
auto inputPtr = inputSamples_.get();
int nread = cbData_->infifo1->read(inputPtr, nsam_in_48);
if (nread != 0)
{
inputPtr = inputSamplesZeros_.get();
}
if (nread != 0 && endingTx)
{
if (!hasEooBeenSent_)
{
// Special case for handling RADE EOT
txStep_->restartVocoder();
hasEooBeenSent_ = true;
}
auto outputSamples = pipeline_->execute(inputPtr, 0, &nout);
if (nout > 0 && outputSamples != nullptr)
{
if (cbData_->outfifo1->write(outputSamples, nout) != 0)
{
log_warn("Could not inject resampled EOO samples (space remaining in FIFO = %d)", cbData_->outfifo1->numFree());
}
}
else
{
if (!g_eoo_enqueued)
{
// Add 40ms of additional silence as Flex will otherwise cut off EOO.
cbData_->outfifo1->write(inputSamplesZeros_.get(), 40 * outputSampleRate_ / 1000);
}
g_eoo_enqueued = true;
}
break;
}
else
{
g_eoo_enqueued = false;
hasEooBeenSent_ = false;
}
auto outputSamples = pipeline_->execute(inputPtr, nsam_in_48, &nout);
if (outputSamples != nullptr)
{
cbData_->outfifo1->write(outputSamples, nout);
}
#if defined(ENABLE_PROCESSING_STATS)
endTimer_();
#endif // defined(ENABLE_PROCESSING_STATS)
}
}
else
{
// Defer reset until next time we go into TX.
deferReset_ = true;
}
}
void MinimalTxRxThread::rxProcessing_(IRealtimeHelper* helper) noexcept
#if defined(__clang__)
#if defined(__has_feature) && __has_feature(realtime_sanitizer)
[[clang::nonblocking]]
#endif // defined(__has_feature) && __has_feature(realtime_sanitizer)
#endif // defined(__clang__)
{
// Buffers re-used by tx and rx processing. We take samples from
// the sound card, and resample them for the freedv modem input
// sample rate. Typically the sound card is running at 48 or 44.1
// kHz, and the modem at 8kHz.
//
// RX side processing --------------------------------------------
//
// Attempt to read one processing frame (about 20ms) of receive samples, we
// keep this frame duration constant across modes and sound card sample rates
int nsam = (inputSampleRate_ * FRAME_DURATION_MS) / MS_TO_SEC;
assert(nsam > 0);
int nout;
bool tmpTx = g_tx.load(std::memory_order_acquire);
bool processInputFifo = !tmpTx;
if (!processInputFifo)
{
clearFifos_();
}
int nsam_one_speech_frame = (getRxNumSpeechSamples() * outputSampleRate_) / RADE_SPEECH_SAMPLE_RATE;
auto outFifo = cbData_->outfifo2;
// while we have enough input samples available and enough space in the output FIFO ...
while (!helper->mustStopWork() && processInputFifo && outFifo->numFree() >= nsam_one_speech_frame && cbData_->infifo2->read(inputSamples_.get(), nsam) == 0) {
#if defined(ENABLE_PROCESSING_STATS)
startTimer_();
#endif // defined(ENABLE_PROCESSING_STATS)
// send latest squelch level to FreeDV API, as it handles squelch internally
auto outputSamples = pipeline_->execute(inputSamples_.get(), nsam, &nout);
if (nout > 0 && outputSamples != nullptr)
{
outFifo->write(outputSamples, nout);
}
tmpTx = g_tx.load(std::memory_order_acquire);
processInputFifo = !tmpTx;
#if defined(ENABLE_PROCESSING_STATS)
endTimer_();
#endif // defined(ENABLE_PROCESSING_STATS)
}
}