| /* |
| * Copyright 2016 The Chromium Authors. All rights reserved. |
| * Copyright (C) 2020, Apple Inc. All rights reserved. |
| * |
| * Redistribution and use in source and binary forms, with or without |
| * modification, are permitted provided that the following conditions |
| * are met: |
| * 1. Redistributions of source code must retain the above copyright |
| * notice, this list of conditions and the following disclaimer. |
| * 2. 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 APPLE INC. AND ITS 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 APPLE INC. OR ITS 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 "config.h" |
| |
| #if ENABLE(WEB_AUDIO) |
| #include "IIRFilter.h" |
| |
| #include "AudioArray.h" |
| #include "AudioUtilities.h" |
| #include "VectorMath.h" |
| #include <algorithm> |
| #include <complex> |
| #include <wtf/MathExtras.h> |
| |
| namespace WebCore { |
| |
| // The length of the memory buffers for the IIR filter. This MUST be a power of |
| // two and must be greater than the possible length of the filter coefficients. |
| constexpr int bufferLength = 32; |
| static_assert(bufferLength >= IIRFilter::maxOrder + 1, "Internal IIR buffer length must be greater than maximum IIR Filter order."); |
| |
| static std::complex<double> evaluatePolynomial(const double* coefficients, std::complex<double> z, int order) |
| { |
| // Use Horner's method to evaluate the polynomial P(z) = sum(coef[k]*z^k, k, 0, order); |
| std::complex<double> result = 0; |
| for (int k = order; k >= 0; --k) |
| result = result * z + std::complex<double>(coefficients[k]); |
| return result; |
| } |
| |
| IIRFilter::IIRFilter(const Vector<double>& feedforward, const Vector<double>& feedback) |
| : m_feedforward(feedforward) |
| , m_feedback(feedback) |
| { |
| m_xBuffer.fill(0, bufferLength); |
| m_yBuffer.fill(0, bufferLength); |
| } |
| |
| void IIRFilter::reset() |
| { |
| m_xBuffer.fill(0); |
| m_yBuffer.fill(0); |
| m_bufferIndex = 0; |
| } |
| |
| void IIRFilter::process(const float* source, float* destination, size_t framesToProcess) |
| { |
| // Compute |
| // |
| // y[n] = sum(b[k] * x[n - k], k = 0, M) - sum(a[k] * y[n - k], k = 1, N) |
| // |
| // where b[k] are the feedforward coefficients and a[k] are the feedback |
| // coefficients of the filter. |
| // This is a Direct Form I implementation of an IIR Filter. Should we |
| // consider doing a different implementation such as Transposed Direct Form |
| // II? |
| const double* feedback = m_feedback.data(); |
| const double* feedforward = m_feedforward.data(); |
| |
| ASSERT(feedback); |
| ASSERT(feedforward); |
| |
| // Sanity check to see if the feedback coefficients have been scaled appropriately. It must be EXACTLY 1! |
| ASSERT(feedback[0] == 1); |
| |
| int feedbackLength = m_feedback.size(); |
| int feedforwardLength = m_feedforward.size(); |
| int minLength = std::min(feedbackLength, feedforwardLength); |
| |
| double* xBuffer = m_xBuffer.data(); |
| double* yBuffer = m_yBuffer.data(); |
| |
| for (size_t n = 0; n < framesToProcess; ++n) { |
| // To help minimize roundoff, we compute using double's, even though the |
| // filter coefficients only have single precision values. |
| double yn = feedforward[0] * source[n]; |
| |
| // Run both the feedforward and feedback terms together, when possible. |
| for (int k = 1; k < minLength; ++k) { |
| int m = (m_bufferIndex - k) & (bufferLength - 1); |
| yn += feedforward[k] * xBuffer[m]; |
| yn -= feedback[k] * yBuffer[m]; |
| } |
| |
| // Handle any remaining feedforward or feedback terms. |
| for (int k = minLength; k < feedforwardLength; ++k) |
| yn += feedforward[k] * xBuffer[(m_bufferIndex - k) & (bufferLength - 1)]; |
| for (int k = minLength; k < feedbackLength; ++k) |
| yn -= feedback[k] * yBuffer[(m_bufferIndex - k) & (bufferLength - 1)]; |
| |
| // Save the current input and output values in the memory buffers for the |
| // next output. |
| m_xBuffer[m_bufferIndex] = source[n]; |
| m_yBuffer[m_bufferIndex] = yn; |
| |
| m_bufferIndex = (m_bufferIndex + 1) & (bufferLength - 1); |
| |
| destination[n] = yn; |
| } |
| } |
| |
| void IIRFilter::getFrequencyResponse(unsigned length, const float* frequency, float* magResponse, float* phaseResponse) |
| { |
| // Evaluate the z-transform of the filter at the given normalized frequencies |
| // from 0 to 1. (One corresponds to the Nyquist frequency.) |
| // |
| // The z-tranform of the filter is |
| // |
| // H(z) = sum(b[k]*z^(-k), k, 0, M) / sum(a[k]*z^(-k), k, 0, N); |
| // |
| // The desired frequency response is H(exp(j*omega)), where omega is in [0, |
| // 1). |
| // |
| // Let P(x) = sum(c[k]*x^k, k, 0, P) be a polynomial of order P. Then each of |
| // the sums in H(z) is equivalent to evaluating a polynomial at the point |
| // 1/z. |
| |
| for (unsigned k = 0; k < length; ++k) { |
| if (frequency[k] < 0 || frequency[k] > 1) { |
| // Out-of-bounds frequencies should return NaN. |
| magResponse[k] = std::nanf(""); |
| phaseResponse[k] = std::nanf(""); |
| } else { |
| // zRecip = 1/z = exp(-j*frequency) |
| double omega = -piDouble * frequency[k]; |
| auto zRecip = std::complex<double>(cos(omega), sin(omega)); |
| |
| auto numerator = evaluatePolynomial(m_feedforward.data(), zRecip, m_feedforward.size() - 1); |
| auto denominator = evaluatePolynomial(m_feedback.data(), zRecip, m_feedback.size() - 1); |
| auto response = numerator / denominator; |
| magResponse[k] = static_cast<float>(abs(response)); |
| phaseResponse[k] = static_cast<float>(atan2(imag(response), real(response))); |
| } |
| } |
| } |
| |
| double IIRFilter::tailTime(double sampleRate, bool isFilterStable) |
| { |
| // The maximum tail time. This is somewhat arbitrary, but we're assuming that |
| // no one is going to expect the IIRFilter to produce an output after this |
| // much time after the inputs have stopped. |
| constexpr double maxTailTime = 10; |
| |
| // If the maximum amplitude of the impulse response is less than this, we |
| // assume that we've reached the tail of the response. Currently, this means |
| // that the impulse is less than 1 bit of a 16-bit PCM value. |
| constexpr float maxTailAmplitude = 1 / 32768.0; |
| |
| // If filter is not stable, just return max tail. Since the filter is not |
| // stable, the impulse response won't converge to zero, so we don't need to |
| // find the impulse response to find the actual tail time. |
| if (!isFilterStable || !sampleRate) |
| return maxTailTime; |
| |
| // How to compute the tail time? We're going to filter an impulse |
| // for |maxTailTime| seconds, in blocks of kRenderQuantumFrames at |
| // a time. The maximum magnitude of this block is saved. After all |
| // of the samples have been computed, find the last block with a |
| // maximum magnitude greater than |kMaxTaileAmplitude|. That block |
| // index + 1 will be the tail time. We don't need to be |
| // super-accurate in computing the tail time since we process on |
| // blocks, block accuracy is good enough, and the value just needs |
| // to be larger than the "real" tail time, so we don't prematurely |
| // zero out the output of the node. |
| |
| // Number of render quanta needed to reach the max tail time. |
| int numberOfBlocks = std::ceil(sampleRate * maxTailTime / AudioUtilities::renderQuantumSize); |
| RELEASE_ASSERT(numberOfBlocks); |
| |
| // Input and output buffers for filtering. |
| AudioFloatArray input(AudioUtilities::renderQuantumSize); |
| AudioFloatArray output(AudioUtilities::renderQuantumSize); |
| |
| // Array to hold the max magnitudes |
| AudioFloatArray magnitudes(numberOfBlocks); |
| |
| // Create the impulse input signal. |
| input[0] = 1; |
| |
| // Process the first block and get the max magnitude of the output. |
| process(input.data(), output.data(), AudioUtilities::renderQuantumSize); |
| magnitudes[0] = VectorMath::maximumMagnitude(output.data(), AudioUtilities::renderQuantumSize); |
| |
| // Process the rest of the signal, getting the max magnitude of the |
| // output for each block. |
| input[0] = 0; |
| |
| for (int k = 1; k < numberOfBlocks; ++k) { |
| process(input.data(), output.data(), AudioUtilities::renderQuantumSize); |
| magnitudes[k] = VectorMath::maximumMagnitude(output.data(), AudioUtilities::renderQuantumSize); |
| } |
| |
| // Done computing the impulse response; reset the state so the actual node |
| // starts in the expected initial state. |
| reset(); |
| |
| // Find the last block with amplitude greater than the threshold. |
| int index = numberOfBlocks - 1; |
| for (int k = index; k >= 0; --k) { |
| if (magnitudes[k] > maxTailAmplitude) { |
| index = k; |
| break; |
| } |
| } |
| |
| // The magnitude first become lower than the threshold at the next block. |
| // Compute the corresponding time value value; that's the tail time. |
| return (index + 1) * AudioUtilities::renderQuantumSize / sampleRate; |
| } |
| |
| } // namespace WebCore |
| |
| #endif // ENABLE(WEB_AUDIO) |