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182 lines
7.3 KiB
182 lines
7.3 KiB
/*
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* Copyright (C) 2007 The Android Open Source Project
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*
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* Licensed under the Apache License, Version 2.0 (the "License");
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* you may not use this file except in compliance with the License.
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* You may obtain a copy of the License at
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*
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* http://www.apache.org/licenses/LICENSE-2.0
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*
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* Unless required by applicable law or agreed to in writing, software
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* distributed under the License is distributed on an "AS IS" BASIS,
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* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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* See the License for the specific language governing permissions and
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* limitations under the License.
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*/
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#pragma once
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#include <cstdint>
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#if CC_PLATFORM == CC_PLATFORM_ANDROID
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#include <android/log.h>
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#include <sys/system_properties.h>
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#include <sys/types.h>
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#endif
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#include "audio/android/AudioBufferProvider.h"
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//#include <cutils/compiler.h>
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//#include <utils/Compat.h>
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//#include <media/AudioBufferProvider.h>
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//#include <system/audio.h>
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#include <cassert>
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#include "audio/android/audio.h"
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namespace cc {
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class AudioResampler {
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public:
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// Determines quality of SRC.
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// LOW_QUALITY: linear interpolator (1st order)
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// MED_QUALITY: cubic interpolator (3rd order)
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// HIGH_QUALITY: fixed multi-tap FIR (e.g. 48KHz->44.1KHz)
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// NOTE: high quality SRC will only be supported for
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// certain fixed rate conversions. Sample rate cannot be
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// changed dynamically.
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enum src_quality { // NOLINT(readability-identifier-naming)
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DEFAULT_QUALITY = 0,
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LOW_QUALITY = 1,
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MED_QUALITY = 2,
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HIGH_QUALITY = 3,
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VERY_HIGH_QUALITY = 4,
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};
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static const CONSTEXPR float UNITY_GAIN_FLOAT = 1.0F;
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static AudioResampler *create(audio_format_t format, int inChannelCount,
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int32_t sampleRate, src_quality quality = DEFAULT_QUALITY);
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virtual ~AudioResampler();
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virtual void init() = 0;
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virtual void setSampleRate(int32_t inSampleRate);
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virtual void setVolume(float left, float right);
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virtual void setLocalTimeFreq(uint64_t freq);
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// set the PTS of the next buffer output by the resampler
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virtual void setPTS(int64_t pts);
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// Resample int16_t samples from provider and accumulate into 'out'.
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// A mono provider delivers a sequence of samples.
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// A stereo provider delivers a sequence of interleaved pairs of samples.
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//
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// In either case, 'out' holds interleaved pairs of fixed-point Q4.27.
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// That is, for a mono provider, there is an implicit up-channeling.
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// Since this method accumulates, the caller is responsible for clearing 'out' initially.
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//
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// For a float resampler, 'out' holds interleaved pairs of float samples.
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//
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// Multichannel interleaved frames for n > 2 is supported for quality DYN_LOW_QUALITY,
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// DYN_MED_QUALITY, and DYN_HIGH_QUALITY.
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//
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// Returns the number of frames resampled into the out buffer.
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virtual size_t resample(int32_t *out, size_t outFrameCount,
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AudioBufferProvider *provider) = 0;
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virtual void reset();
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virtual size_t getUnreleasedFrames() const { return mInputIndex; }
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// called from destructor, so must not be virtual
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src_quality getQuality() const { return mQuality; }
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protected:
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// number of bits for phase fraction - 30 bits allows nearly 2x downsampling
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static const int kNumPhaseBits = 30; // NOLINT(readability-identifier-naming)
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// phase mask for fraction
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static const uint32_t kPhaseMask = (1LU << kNumPhaseBits) - 1; // NOLINT(readability-identifier-naming)
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// multiplier to calculate fixed point phase increment
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static const double kPhaseMultiplier; // NOLINT(readability-identifier-naming)
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AudioResampler(int inChannelCount, int32_t sampleRate, src_quality quality);
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// prevent copying
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AudioResampler(const AudioResampler &);
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AudioResampler &operator=(const AudioResampler &);
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int64_t calculateOutputPTS(int outputFrameIndex);
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const int32_t mChannelCount;// NOLINT(readability-identifier-naming)
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const int32_t mSampleRate;// NOLINT(readability-identifier-naming)
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int32_t mInSampleRate;// NOLINT(readability-identifier-naming)
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AudioBufferProvider::Buffer mBuffer;// NOLINT(readability-identifier-naming)
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union {
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int16_t mVolume[2];// NOLINT(readability-identifier-naming)
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uint32_t mVolumeRL;// NOLINT(readability-identifier-naming)
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};
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int16_t mTargetVolume[2];// NOLINT(readability-identifier-naming)
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size_t mInputIndex;// NOLINT(readability-identifier-naming)
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int32_t mPhaseIncrement;// NOLINT(readability-identifier-naming)
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uint32_t mPhaseFraction;// NOLINT(readability-identifier-naming)
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uint64_t mLocalTimeFreq;// NOLINT(readability-identifier-naming)
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int64_t mPTS;// NOLINT(readability-identifier-naming)
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// returns the inFrameCount required to generate outFrameCount frames.
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//
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// Placed here to be a consistent for all resamplers.
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//
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// Right now, we use the upper bound without regards to the current state of the
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// input buffer using integer arithmetic, as follows:
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//
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// (static_cast<uint64_t>(outFrameCount)*mInSampleRate + (mSampleRate - 1))/mSampleRate;
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//
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// The double precision equivalent (float may not be precise enough):
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// ceil(static_cast<double>(outFrameCount) * mInSampleRate / mSampleRate);
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//
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// this relies on the fact that the mPhaseIncrement is rounded down from
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// #phases * mInSampleRate/mSampleRate and the fact that Sum(Floor(x)) <= Floor(Sum(x)).
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// http://www.proofwiki.org/wiki/Sum_of_Floors_Not_Greater_Than_Floor_of_Sums
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//
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// (so long as double precision is computed accurately enough to be considered
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// greater than or equal to the Floor(x) value in int32_t arithmetic; thus this
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// will not necessarily hold for floats).
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//
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// REFINE:
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// Greater accuracy and a tight bound is obtained by:
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// 1) subtract and adjust for the current state of the AudioBufferProvider buffer.
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// 2) using the exact integer formula where (ignoring 64b casting)
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// inFrameCount = (mPhaseIncrement * (outFrameCount - 1) + mPhaseFraction) / phaseWrapLimit;
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// phaseWrapLimit is the wraparound (1 << kNumPhaseBits), if not specified explicitly.
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//
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inline size_t getInFrameCountRequired(size_t outFrameCount) const {
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return (static_cast<size_t>(outFrameCount) * mInSampleRate + (mSampleRate - 1)) / mSampleRate;
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}
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inline float clampFloatVol(float volume) {//NOLINT(readability-identifier-naming, readability-convert-member-functions-to-static)
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float ret = 0.0F;
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if (volume > UNITY_GAIN_FLOAT) {
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ret = UNITY_GAIN_FLOAT;
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} else if (volume >= 0.) {
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ret = volume;
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}
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return ret; // NaN or negative volume maps to 0.
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}
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private:
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const src_quality mQuality;// NOLINT(readability-identifier-naming)
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// Return 'true' if the quality level is supported without explicit request
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static bool qualityIsSupported(src_quality quality);
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// For pthread_once()
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static void init_routine(); // NOLINT(readability-identifier-naming)
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// Return the estimated CPU load for specific resampler in MHz.
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// The absolute number is irrelevant, it's the relative values that matter.
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static uint32_t qualityMHz(src_quality quality);
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};
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// ----------------------------------------------------------------------------
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} // namespace cc
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