72dafb20e0
The calculation done in prepareTracks_l() for the minimum amount off frames needed to mix one output buffer had 2 issues: - the additional sample needed for interpolation was not included - the fact that the resampler does not acknowledge the frames consumed immediately after each mixing round but only once all frames requested have been used was not taken into account. Thus the number of frames available in track buffer could be considered sufficient although it was not and the resampler would abort producing a short silence perceived as a click. Issue 5727099. Change-Id: I7419847a7474c7d9f9170bedd0a636132262142c
97 lines
2.9 KiB
C++
97 lines
2.9 KiB
C++
/*
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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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#ifndef ANDROID_AUDIO_RESAMPLER_H
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#define ANDROID_AUDIO_RESAMPLER_H
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#include <stdint.h>
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#include <sys/types.h>
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#include "AudioBufferProvider.h"
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namespace android {
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// ----------------------------------------------------------------------------
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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 {
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DEFAULT=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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};
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static AudioResampler* create(int bitDepth, int inChannelCount,
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int32_t sampleRate, int quality=DEFAULT);
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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(int16_t left, int16_t right);
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virtual void 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() { return mInputIndex; }
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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;
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// phase mask for fraction
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static const uint32_t kPhaseMask = (1LU<<kNumPhaseBits)-1;
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// multiplier to calculate fixed point phase increment
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static const double kPhaseMultiplier = 1L << kNumPhaseBits;
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enum format {MONO_16_BIT, STEREO_16_BIT};
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AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate);
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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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int32_t mBitDepth;
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int32_t mChannelCount;
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int32_t mSampleRate;
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int32_t mInSampleRate;
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AudioBufferProvider::Buffer mBuffer;
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union {
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int16_t mVolume[2];
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uint32_t mVolumeRL;
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};
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int16_t mTargetVolume[2];
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format mFormat;
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size_t mInputIndex;
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int32_t mPhaseIncrement;
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uint32_t mPhaseFraction;
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};
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// ----------------------------------------------------------------------------
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}
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; // namespace android
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#endif // ANDROID_AUDIO_RESAMPLER_H
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