1/*
2 * Copyright (C) 2007 The Android Open Source Project
3 *
4 * Licensed under the Apache License, Version 2.0 (the "License");
5 * you may not use this file except in compliance with the License.
6 * You may obtain a copy of the License at
7 *
8 *      http://www.apache.org/licenses/LICENSE-2.0
9 *
10 * Unless required by applicable law or agreed to in writing, software
11 * distributed under the License is distributed on an "AS IS" BASIS,
12 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
13 * See the License for the specific language governing permissions and
14 * limitations under the License.
15 */
16
17#ifndef ANDROID_AUDIO_RESAMPLER_H
18#define ANDROID_AUDIO_RESAMPLER_H
19
20#include <stdint.h>
21#include <sys/types.h>
22
23#include <cutils/compiler.h>
24#include <utils/Compat.h>
25
26#include <media/AudioBufferProvider.h>
27#include <system/audio.h>
28
29namespace android {
30// ----------------------------------------------------------------------------
31
32class ANDROID_API AudioResampler {
33public:
34    // Determines quality of SRC.
35    //  LOW_QUALITY: linear interpolator (1st order)
36    //  MED_QUALITY: cubic interpolator (3rd order)
37    //  HIGH_QUALITY: fixed multi-tap FIR (e.g. 48KHz->44.1KHz)
38    // NOTE: high quality SRC will only be supported for
39    // certain fixed rate conversions. Sample rate cannot be
40    // changed dynamically.
41    enum src_quality {
42        DEFAULT_QUALITY=0,
43        LOW_QUALITY=1,
44        MED_QUALITY=2,
45        HIGH_QUALITY=3,
46        VERY_HIGH_QUALITY=4,
47        DYN_LOW_QUALITY=5,
48        DYN_MED_QUALITY=6,
49        DYN_HIGH_QUALITY=7,
50    };
51
52    static const CONSTEXPR float UNITY_GAIN_FLOAT = 1.0f;
53
54    static AudioResampler* create(audio_format_t format, int inChannelCount,
55            int32_t sampleRate, src_quality quality=DEFAULT_QUALITY);
56
57    virtual ~AudioResampler();
58
59    virtual void init() = 0;
60    virtual void setSampleRate(int32_t inSampleRate);
61    virtual void setVolume(float left, float right);
62
63    // Resample int16_t samples from provider and accumulate into 'out'.
64    // A mono provider delivers a sequence of samples.
65    // A stereo provider delivers a sequence of interleaved pairs of samples.
66    //
67    // In either case, 'out' holds interleaved pairs of fixed-point Q4.27.
68    // That is, for a mono provider, there is an implicit up-channeling.
69    // Since this method accumulates, the caller is responsible for clearing 'out' initially.
70    //
71    // For a float resampler, 'out' holds interleaved pairs of float samples.
72    //
73    // Multichannel interleaved frames for n > 2 is supported for quality DYN_LOW_QUALITY,
74    // DYN_MED_QUALITY, and DYN_HIGH_QUALITY.
75    //
76    // Returns the number of frames resampled into the out buffer.
77    virtual size_t resample(int32_t* out, size_t outFrameCount,
78            AudioBufferProvider* provider) = 0;
79
80    virtual void reset();
81    virtual size_t getUnreleasedFrames() const { return mInputIndex; }
82
83    // called from destructor, so must not be virtual
84    src_quality getQuality() const { return mQuality; }
85
86protected:
87    // number of bits for phase fraction - 30 bits allows nearly 2x downsampling
88    static const int kNumPhaseBits = 30;
89
90    // phase mask for fraction
91    static const uint32_t kPhaseMask = (1LU<<kNumPhaseBits)-1;
92
93    // multiplier to calculate fixed point phase increment
94    static const double kPhaseMultiplier;
95
96    AudioResampler(int inChannelCount, int32_t sampleRate, src_quality quality);
97
98    // prevent copying
99    AudioResampler(const AudioResampler&);
100    AudioResampler& operator=(const AudioResampler&);
101
102    const int32_t mChannelCount;
103    const int32_t mSampleRate;
104    int32_t mInSampleRate;
105    AudioBufferProvider::Buffer mBuffer;
106    union {
107        int16_t mVolume[2];
108        uint32_t mVolumeRL;
109    };
110    int16_t mTargetVolume[2];
111    size_t mInputIndex;
112    int32_t mPhaseIncrement;
113    uint32_t mPhaseFraction;
114
115    // returns the inFrameCount required to generate outFrameCount frames.
116    //
117    // Placed here to be a consistent for all resamplers.
118    //
119    // Right now, we use the upper bound without regards to the current state of the
120    // input buffer using integer arithmetic, as follows:
121    //
122    // (static_cast<uint64_t>(outFrameCount)*mInSampleRate + (mSampleRate - 1))/mSampleRate;
123    //
124    // The double precision equivalent (float may not be precise enough):
125    // ceil(static_cast<double>(outFrameCount) * mInSampleRate / mSampleRate);
126    //
127    // this relies on the fact that the mPhaseIncrement is rounded down from
128    // #phases * mInSampleRate/mSampleRate and the fact that Sum(Floor(x)) <= Floor(Sum(x)).
129    // http://www.proofwiki.org/wiki/Sum_of_Floors_Not_Greater_Than_Floor_of_Sums
130    //
131    // (so long as double precision is computed accurately enough to be considered
132    // greater than or equal to the Floor(x) value in int32_t arithmetic; thus this
133    // will not necessarily hold for floats).
134    //
135    // TODO:
136    // Greater accuracy and a tight bound is obtained by:
137    // 1) subtract and adjust for the current state of the AudioBufferProvider buffer.
138    // 2) using the exact integer formula where (ignoring 64b casting)
139    //  inFrameCount = (mPhaseIncrement * (outFrameCount - 1) + mPhaseFraction) / phaseWrapLimit;
140    //  phaseWrapLimit is the wraparound (1 << kNumPhaseBits), if not specified explicitly.
141    //
142    inline size_t getInFrameCountRequired(size_t outFrameCount) {
143        return (static_cast<uint64_t>(outFrameCount)*mInSampleRate
144                + (mSampleRate - 1))/mSampleRate;
145    }
146
147    inline float clampFloatVol(float volume) {
148        if (volume > UNITY_GAIN_FLOAT) {
149            return UNITY_GAIN_FLOAT;
150        } else if (volume >= 0.) {
151            return volume;
152        }
153        return 0.;  // NaN or negative volume maps to 0.
154    }
155
156private:
157    const src_quality mQuality;
158
159    // Return 'true' if the quality level is supported without explicit request
160    static bool qualityIsSupported(src_quality quality);
161
162    // For pthread_once()
163    static void init_routine();
164
165    // Return the estimated CPU load for specific resampler in MHz.
166    // The absolute number is irrelevant, it's the relative values that matter.
167    static uint32_t qualityMHz(src_quality quality);
168};
169
170// ----------------------------------------------------------------------------
171} // namespace android
172
173#endif // ANDROID_AUDIO_RESAMPLER_H
174