feat(audio): make echo cancellation work and improve some minor stuff

src/audio/audio.h

@@ -100,7 +100,7 @@ public:
     static constexpr uint32_t AUDIO_FRAME_DURATION = 20;
     static constexpr uint32_t AUDIO_FRAME_SAMPLE_COUNT =
         AUDIO_FRAME_DURATION * AUDIO_SAMPLE_RATE / 1000;
-    static constexpr uint32_t AUDIO_CHANNELS = 2;
+    static constexpr uint32_t AUDIO_CHANNELS = 1;
 
 signals:
     void frameAvailable(const int16_t* pcm, size_t sample_count, uint8_t channels,

src/audio/backend/openal2.cpp

@@ -316,7 +316,7 @@ bool OpenAL2::initOutput(const QString& deviceName)
     }
 
     // check for the needed extensions for echo cancelation
-    if (echoCancelSupported = (alcIsExtensionPresent(NULL, "ALC_SOFT_loopback") == AL_TRUE)) {
+    if (echoCancelSupported = (alcIsExtensionPresent(alOutDev, "ALC_SOFT_loopback") == AL_TRUE)) {
         qDebug() << "Device supports loopback";
     }
     if (alIsExtensionPresent("AL_SOFT_source_latency") == AL_TRUE) {
@@ -540,17 +540,10 @@ void OpenAL2::playMono16SoundCleanup()
 }
 
 /**
- * @brief Called on the captureTimer events to capture audio
+ * @brief Handle audio output
  */
-void OpenAL2::doAudio()
+void OpenAL2::doOutput()
 {
-    QMutexLocker lock(&audioLock);
-
-    static int outBufcnt = 0;
-    static double outLatency = 0;
-
-    // output section
-    if(echoCancelSupported && outputInitialized) {
     alcMakeContextCurrent(alOutContext);
     ALuint bufids[PROXY_BUFFER_COUNT];
     ALint processed = 0, queued = 0;
@@ -572,10 +565,11 @@ void OpenAL2::doAudio()
     LPALGETSOURCEDVSOFT alGetSourcedvSOFT =
             reinterpret_cast<LPALGETSOURCEDVSOFT> (alcGetProcAddress(alOutDev, "alGetSourcedvSOFT"));
     ALdouble latency[2] = {0};
+    if(alGetSourcedvSOFT) {
         alGetSourcedvSOFT(alProxySource, AL_SEC_OFFSET_LATENCY_SOFT, latency);
         checkAlError();
+    }
     //qDebug() << "Playback latency: " << latency[1] << "offset: " << latency[0];
-        outLatency = latency[1];
 
     ALshort outBuf[AUDIO_FRAME_SAMPLE_COUNT * AUDIO_CHANNELS] = {0};
     alcMakeContextCurrent(alProxyContext);
@@ -587,18 +581,18 @@ void OpenAL2::doAudio()
     alBufferData(bufids[0], (AUDIO_CHANNELS == 1) ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16, outBuf,
                  AUDIO_FRAME_SAMPLE_COUNT * 2 * AUDIO_CHANNELS, AUDIO_SAMPLE_RATE);
     alSourceQueueBuffers(alProxySource, 1, bufids);
-        ++outBufcnt;
 
     // initialize echo canceler if supported
     if(!filterer) {
         filterer = new_filter_audio(AUDIO_SAMPLE_RATE);
-            int16_t filterLatency = outLatency*1000 + AUDIO_FRAME_DURATION;
+        int16_t filterLatency = latency[1]*1000*2 + AUDIO_FRAME_DURATION;
         qDebug() << "Setting filter delay to: " << filterLatency << "ms";
         set_echo_delay_ms(filterer, filterLatency);
+        enable_disable_filters(filterer, 1, 1, 1, 0);
     }
 
     // do echo cancel
-        pass_audio_output(filterer, outBuf, AUDIO_FRAME_SAMPLE_COUNT);
+    int retVal = pass_audio_output(filterer, outBuf, AUDIO_FRAME_SAMPLE_COUNT);
 
     ALint state;
     alGetSourcei(alProxySource, AL_SOURCE_STATE, &state);
@@ -608,30 +602,27 @@ void OpenAL2::doAudio()
     }
 }
 
-    // input section
-
-    if (!alInDev || !inSubscriptions)
-        return;
-
+/**
+ * @brief handles recording of audio frames
+ */
+void OpenAL2::doInput()
+{
     ALint curSamples = 0;
     alcGetIntegerv(alInDev, ALC_CAPTURE_SAMPLES, sizeof(curSamples), &curSamples);
-    if (curSamples < AUDIO_FRAME_SAMPLE_COUNT)
+    if (curSamples < AUDIO_FRAME_SAMPLE_COUNT) {
         return;
-    // check for dropped buffers in echo cancel
-    if(outBufcnt > 1) {
-        qDebug() << "Echo cancel frame dropped";
     }
-    outBufcnt = 0;
 
     int16_t buf[AUDIO_FRAME_SAMPLE_COUNT * AUDIO_CHANNELS];
     alcCaptureSamples(alInDev, buf, AUDIO_FRAME_SAMPLE_COUNT);
 
+    int retVal = 0;
     if(echoCancelSupported && filterer) {
-        filter_audio(filterer, buf, AUDIO_FRAME_SAMPLE_COUNT);
+        retVal = filter_audio(filterer, buf, AUDIO_FRAME_SAMPLE_COUNT);
     }
 
-    for (quint32 i = 0; i < AUDIO_FRAME_SAMPLE_COUNT * AUDIO_CHANNELS; ++i) {
     // gain amplification with clipping to 16-bit boundaries
+    for (quint32 i = 0; i < AUDIO_FRAME_SAMPLE_COUNT * AUDIO_CHANNELS; ++i) {
         int ampPCM =
             qBound<int>(std::numeric_limits<int16_t>::min(), qRound(buf[i] * inputGainFactor()),
                         std::numeric_limits<int16_t>::max());
@@ -642,6 +633,27 @@ void OpenAL2::doAudio()
     emit Audio::frameAvailable(buf, AUDIO_FRAME_SAMPLE_COUNT, AUDIO_CHANNELS, AUDIO_SAMPLE_RATE);
 }
 
+/**
+ * @brief Called on the captureTimer events to capture audio
+ */
+void OpenAL2::doAudio()
+{
+    QMutexLocker lock(&audioLock);
+
+    // output section
+    if(echoCancelSupported && outputInitialized && !peerSources.isEmpty()) {
+        doOutput();
+    } else {
+        kill_filter_audio(filterer);
+        filterer = nullptr;
+    }
+
+    // input section
+    if (alInDev && inSubscriptions) {
+        doInput();
+    }
+}
+
 /**
  * @brief Returns true if the output device is open
  */

src/audio/backend/openal2.h

@@ -106,6 +106,8 @@ private:
     void playMono16SoundCleanup();
     void doAudio();
     qreal inputGainFactor() const;
+    void doInput();
+    void doOutput();
 
 private:
     QThread* audioThread;