4bb21c496b
The problem is that when an AudioRecord using the resampler is restarted, the resampler state is not reset (as there is no reset function in the resampler). The consequence is that the first time the record thread loop runs, it calls the resampler which consumes the remaining data in the input buffer and when this buffer is released the input index is incremented over the limit. The fix consists in implementing a reset function in the resampler. A similar problem was also present for playback but unoticed because the track buffer is always drained by the mixer when a track stops. The only problem for playback was that the initial phase fraction was wrong when restarting a track after stop (it was correct after a pause). Change-Id: Ifc2585d685f4402d29f4afc63f6efd1d69265de3
96 lines
2.8 KiB
C++
96 lines
2.8 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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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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