C++ PositionalAudioStream::isStereo方法代码示例

void AudioMixerSlave::addStream(AudioMixerClientData& listenerNodeData, const QUuid& sourceNodeID,
        const AvatarAudioStream& listeningNodeStream, const PositionalAudioStream& streamToAdd,
        bool throttle) {
    ++stats.totalMixes;

    // to reduce artifacts we call the HRTF functor for every source, even if throttled or silent
    // this ensures the correct tail from last mixed block and the correct spatialization of next first block

    // check if this is a server echo of a source back to itself
    bool isEcho = (&streamToAdd == &listeningNodeStream);

    glm::vec3 relativePosition = streamToAdd.getPosition() - listeningNodeStream.getPosition();

    float distance = glm::max(glm::length(relativePosition), EPSILON);
    float gain = computeGain(listenerNodeData, listeningNodeStream, streamToAdd, relativePosition, isEcho);
    float azimuth = isEcho ? 0.0f : computeAzimuth(listeningNodeStream, listeningNodeStream, relativePosition);
    const int HRTF_DATASET_INDEX = 1;

    if (!streamToAdd.lastPopSucceeded()) {
        bool forceSilentBlock = true;

        if (!streamToAdd.getLastPopOutput().isNull()) {
            bool isInjector = dynamic_cast<const InjectedAudioStream*>(&streamToAdd);

            // in an injector, just go silent - the injector has likely ended
            // in other inputs (microphone, &c.), repeat with fade to avoid the harsh jump to silence
            if (!isInjector) {
                // calculate its fade factor, which depends on how many times it's already been repeated.
                float fadeFactor = calculateRepeatedFrameFadeFactor(streamToAdd.getConsecutiveNotMixedCount() - 1);
                if (fadeFactor > 0.0f) {
                    // apply the fadeFactor to the gain
                    gain *= fadeFactor;
                    forceSilentBlock = false;
                }
            }
        }

        if (forceSilentBlock) {
            // call renderSilent with a forced silent block to reduce artifacts
            // (this is not done for stereo streams since they do not go through the HRTF)
            if (!streamToAdd.isStereo() && !isEcho) {
                // get the existing listener-source HRTF object, or create a new one
                auto& hrtf = listenerNodeData.hrtfForStream(sourceNodeID, streamToAdd.getStreamIdentifier());

                static int16_t silentMonoBlock[AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL] = {};
                hrtf.renderSilent(silentMonoBlock, _mixSamples, HRTF_DATASET_INDEX, azimuth, distance, gain,
                                  AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);

                ++stats.hrtfSilentRenders;
            }

            return;
        }
    }

    // grab the stream from the ring buffer
    AudioRingBuffer::ConstIterator streamPopOutput = streamToAdd.getLastPopOutput();

    // stereo sources are not passed through HRTF
    if (streamToAdd.isStereo()) {
        for (int i = 0; i < AudioConstants::NETWORK_FRAME_SAMPLES_STEREO; ++i) {
            _mixSamples[i] += float(streamPopOutput[i] * gain / AudioConstants::MAX_SAMPLE_VALUE);
        }

        ++stats.manualStereoMixes;
        return;
    }

    // echo sources are not passed through HRTF
    if (isEcho) {
        for (int i = 0; i < AudioConstants::NETWORK_FRAME_SAMPLES_STEREO; i += 2) {
            auto monoSample = float(streamPopOutput[i / 2] * gain / AudioConstants::MAX_SAMPLE_VALUE);
            _mixSamples[i] += monoSample;
            _mixSamples[i + 1] += monoSample;
        }

        ++stats.manualEchoMixes;
        return;
    }

    // get the existing listener-source HRTF object, or create a new one
    auto& hrtf = listenerNodeData.hrtfForStream(sourceNodeID, streamToAdd.getStreamIdentifier());

    streamPopOutput.readSamples(_bufferSamples, AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);

    if (streamToAdd.getLastPopOutputLoudness() == 0.0f) {
        // call renderSilent to reduce artifacts
        hrtf.renderSilent(_bufferSamples, _mixSamples, HRTF_DATASET_INDEX, azimuth, distance, gain,
                          AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);

        ++stats.hrtfSilentRenders;
        return;
    }

    if (throttle) {
        // call renderSilent with actual frame data and a gain of 0.0f to reduce artifacts
        hrtf.renderSilent(_bufferSamples, _mixSamples, HRTF_DATASET_INDEX, azimuth, distance, 0.0f,
                          AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);

        ++stats.hrtfThrottleRenders;
//.........这里部分代码省略.........

void AudioMixer::addStreamToMixForListeningNodeWithStream(AudioMixerClientData& listenerNodeData,
                                                          const PositionalAudioStream& streamToAdd,
                                                          const QUuid& sourceNodeID,
                                                          const AvatarAudioStream& listeningNodeStream) {


    // to reduce artifacts we calculate the gain and azimuth for every source for this listener
    // even if we are not going to end up mixing in this source

    ++_totalMixes;

    // this ensures that the tail of any previously mixed audio or the first block of new audio sounds correct

    // check if this is a server echo of a source back to itself
    bool isEcho = (&streamToAdd == &listeningNodeStream);

    glm::vec3 relativePosition = streamToAdd.getPosition() - listeningNodeStream.getPosition();

    // figure out the distance between source and listener
    float distance = glm::max(glm::length(relativePosition), EPSILON);

    // figure out the gain for this source at the listener
    float gain = gainForSource(streamToAdd, listeningNodeStream, relativePosition, isEcho);

    // figure out the azimuth to this source at the listener
    float azimuth = isEcho ? 0.0f : azimuthForSource(streamToAdd, listeningNodeStream, relativePosition);

    float repeatedFrameFadeFactor = 1.0f;

    static const int HRTF_DATASET_INDEX = 1;

    if (!streamToAdd.lastPopSucceeded()) {
        bool forceSilentBlock = true;

        if (_streamSettings._repetitionWithFade && !streamToAdd.getLastPopOutput().isNull()) {

            // reptition with fade is enabled, and we do have a valid previous frame to repeat
            // so we mix the previously-mixed block

            // this is preferable to not mixing it at all to avoid the harsh jump to silence

            // we'll repeat the last block until it has a block to mix
            // and we'll gradually fade that repeated block into silence.

            // calculate its fade factor, which depends on how many times it's already been repeated.

            repeatedFrameFadeFactor = calculateRepeatedFrameFadeFactor(streamToAdd.getConsecutiveNotMixedCount() - 1);
            if (repeatedFrameFadeFactor > 0.0f) {
                // apply the repeatedFrameFadeFactor to the gain
                gain *= repeatedFrameFadeFactor;

                forceSilentBlock = false;
            }
        }

        if (forceSilentBlock) {
            // we're deciding not to repeat either since we've already done it enough times or repetition with fade is disabled
            // in this case we will call renderSilent with a forced silent block
            // this ensures the correct tail from the previously mixed block and the correct spatialization of first block
            // of any upcoming audio

            if (!streamToAdd.isStereo() && !isEcho) {
                // get the existing listener-source HRTF object, or create a new one
                auto& hrtf = listenerNodeData.hrtfForStream(sourceNodeID, streamToAdd.getStreamIdentifier());

                // this is not done for stereo streams since they do not go through the HRTF
                static int16_t silentMonoBlock[AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL] = {};
                hrtf.renderSilent(silentMonoBlock, _mixedSamples, HRTF_DATASET_INDEX, azimuth, distance, gain,
                                  AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);

                ++_hrtfSilentRenders;;
            }

            return;
        }
    }

    // grab the stream from the ring buffer
    AudioRingBuffer::ConstIterator streamPopOutput = streamToAdd.getLastPopOutput();

    if (streamToAdd.isStereo() || isEcho) {
        // this is a stereo source or server echo so we do not pass it through the HRTF
        // simply apply our calculated gain to each sample
        if (streamToAdd.isStereo()) {
            for (int i = 0; i < AudioConstants::NETWORK_FRAME_SAMPLES_STEREO; ++i) {
                _mixedSamples[i] += float(streamPopOutput[i] * gain / AudioConstants::MAX_SAMPLE_VALUE);
            }

            ++_manualStereoMixes;
        } else {
            for (int i = 0; i < AudioConstants::NETWORK_FRAME_SAMPLES_STEREO; i += 2) {
                auto monoSample = float(streamPopOutput[i / 2] * gain / AudioConstants::MAX_SAMPLE_VALUE);
                _mixedSamples[i] += monoSample;
                _mixedSamples[i + 1] += monoSample;
            }

            ++_manualEchoMixes;
        }

        return;
//.........这里部分代码省略.........