예제 #1
0
    void DspLimiter::Process(DspChunk& chunk)
    {
        if (chunk.IsEmpty())
            return;

        if (!m_exclusive || (chunk.GetFormat() != DspFormat::Float &&
                             chunk.GetFormat() != DspFormat::Double))
        {
            m_active = false;
            return;
        }

        m_active = true;

        // Analyze samples
        float peak;
        if (chunk.GetFormat() == DspFormat::Double)
        {
            double largePeak = GetPeak((double*)chunk.GetData(), chunk.GetSampleCount());
            peak = std::nexttoward((float)largePeak, largePeak);
        }
        else
        {
            assert(chunk.GetFormat() == DspFormat::Float);
            peak = GetPeak((float*)chunk.GetData(), chunk.GetSampleCount());
        }

        // Configure limiter
        if (peak > 1.0f)
        {
            if (m_holdWindow <= 0)
            {
                NewTreshold(std::max(peak, 1.4f));
            }
            else if (peak > m_peak)
            {
                NewTreshold(peak);
            }

            m_holdWindow = (int64_t)m_rate * m_channels * 10; // 10 seconds
        }

        // Apply limiter
        if (m_holdWindow > 0)
        {
            if (chunk.GetFormat() == DspFormat::Double)
            {
                ApplyLimiter<double>((double*)chunk.GetData(), chunk.GetSampleCount(), m_threshold);
            }
            else
            {
                assert(chunk.GetFormat() == DspFormat::Float);
                ApplyLimiter((float*)chunk.GetData(), chunk.GetSampleCount(), m_threshold);
            }

            m_holdWindow -= chunk.GetSampleCount();
        }
    }
예제 #2
0
    void AudioRenderer::ApplyRateCorrection(DspChunk& chunk)
    {
        CAutoLock objectLock(this);
        assert(m_device);
        assert(!m_device->IsBitstream());
        assert(m_state == State_Running);

        if (chunk.IsEmpty())
            return;

        const REFERENCE_TIME latency = llMulDiv(chunk.GetFrameCount(), OneSecond, chunk.GetRate(), 0) +
                                       m_device->GetStreamLatency() + OneMillisecond * 10;

        const REFERENCE_TIME remaining = m_device->GetEnd() - m_device->GetPosition();

        REFERENCE_TIME deltaTime = 0;

        if (m_live)
        {
            // Rate matching.
            if (remaining > latency)
            {
                size_t dropFrames = (size_t)llMulDiv(m_device->GetWaveFormat()->nSamplesPerSec,
                                                     remaining - latency, OneSecond, 0);

                dropFrames = std::min(dropFrames, chunk.GetFrameCount());

                chunk.ShrinkHead(chunk.GetFrameCount() - dropFrames);

                DebugOut("AudioRenderer drop", dropFrames, "frames for rate matching");
            }
        }
        else
        {
            // Clock matching.
            assert(m_externalClock);

            REFERENCE_TIME graphTime, myTime, myStartTime;
            if (SUCCEEDED(m_myClock.GetAudioClockStartTime(&myStartTime)) &&
                SUCCEEDED(m_myClock.GetAudioClockTime(&myTime, nullptr)) &&
                SUCCEEDED(m_graphClock->GetTime(&graphTime)) &&
                myTime > myStartTime)
            {
                myTime -= m_device->GetSilence();

                if (myTime > graphTime)
                {
                    // Pad and adjust backwards.
                    REFERENCE_TIME padTime = myTime - graphTime;
                    assert(padTime >= 0);

                    size_t padFrames = (size_t)llMulDiv(m_device->GetWaveFormat()->nSamplesPerSec,
                                                        padTime, OneSecond, 0);

                    if (padFrames > m_device->GetWaveFormat()->nSamplesPerSec / 33) // ~30ms threshold
                    {
                        DspChunk tempChunk(chunk.GetFormat(), chunk.GetChannelCount(),
                                           chunk.GetFrameCount() + padFrames, chunk.GetRate());

                        size_t padBytes = tempChunk.GetFrameSize() * padFrames;
                        ZeroMemory(tempChunk.GetData(), padBytes);
                        memcpy(tempChunk.GetData() + padBytes, chunk.GetData(), chunk.GetSize());

                        chunk = std::move(tempChunk);

                        REFERENCE_TIME paddedTime = llMulDiv(padFrames, OneSecond,
                                                             m_device->GetWaveFormat()->nSamplesPerSec, 0);

                        m_myClock.OffsetSlavedClock(-paddedTime);
                        padTime -= paddedTime;
                        assert(padTime >= 0);

                        DebugOut("AudioRenderer pad", paddedTime / 10000., "ms for clock matching at",
                                 m_sampleCorrection.GetLastFrameEnd() / 10000., "frame position");
                    }

                    // Correct the rest with variable rate.
                    m_dspRealtimeRate.Adjust(padTime);
                    m_myClock.OffsetSlavedClock(-padTime);
                }
                else if (remaining > latency)
                {
                    // Crop and adjust forwards.
                    assert(myTime <= graphTime);
                    REFERENCE_TIME dropTime = std::min(graphTime - myTime, remaining - latency);
                    assert(dropTime >= 0);

                    size_t dropFrames = (size_t)llMulDiv(m_device->GetWaveFormat()->nSamplesPerSec,
                                                         dropTime, OneSecond, 0);

                    dropFrames = std::min(dropFrames, chunk.GetFrameCount());

                    if (dropFrames > m_device->GetWaveFormat()->nSamplesPerSec / 33) // ~30ms threshold
                    {
                        chunk.ShrinkHead(chunk.GetFrameCount() - dropFrames);

                        REFERENCE_TIME droppedTime = llMulDiv(dropFrames, OneSecond,
                                                              m_device->GetWaveFormat()->nSamplesPerSec, 0);

                        m_myClock.OffsetSlavedClock(droppedTime);
                        dropTime -= droppedTime;
                        assert(dropTime >= 0);

                        DebugOut("AudioRenderer drop", droppedTime / 10000., "ms for clock matching at",
                                 m_sampleCorrection.GetLastFrameEnd() / 10000., "frame position");
                    }

                    // Correct the rest with variable rate.
                    m_dspRealtimeRate.Adjust(-dropTime);
                    m_myClock.OffsetSlavedClock(dropTime);
                }
            }
        }
    }