AudioResampler.h revision 34af02647b387a252fb02bab8e2cb9f7bd9c8abb 
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#include <cutils/compiler.h>
23
24#include <media/AudioBufferProvider.h>
25
26namespace android {
27// ----------------------------------------------------------------------------
28
29class ANDROID_API AudioResampler {
30public:
31    // Determines quality of SRC.
32    //  LOW_QUALITY: linear interpolator (1st order)
33    //  MED_QUALITY: cubic interpolator (3rd order)
34    //  HIGH_QUALITY: fixed multi-tap FIR (e.g. 48KHz->44.1KHz)
35    // NOTE: high quality SRC will only be supported for
36    // certain fixed rate conversions. Sample rate cannot be
37    // changed dynamically.
38    enum src_quality {
39        DEFAULT_QUALITY=0,
40        LOW_QUALITY=1,
41        MED_QUALITY=2,
42        HIGH_QUALITY=3,
43        VERY_HIGH_QUALITY=4,
44    };
45
46    static AudioResampler* create(int bitDepth, int inChannelCount,
47            int32_t sampleRate, src_quality quality=DEFAULT_QUALITY);
48
49    virtual ~AudioResampler();
50
51    virtual void init() = 0;
52    virtual void setSampleRate(int32_t inSampleRate);
53    virtual void setVolume(int16_t left, int16_t right);
54    virtual void setLocalTimeFreq(uint64_t freq);
55
56    // set the PTS of the next buffer output by the resampler
57    virtual void setPTS(int64_t pts);
58
59    // Resample int16_t samples from provider and accumulate into 'out'.
60    // A mono provider delivers a sequence of samples.
61    // A stereo provider delivers a sequence of interleaved pairs of samples.
62    // Multi-channel providers are not supported.
63    // In either case, 'out' holds interleaved pairs of fixed-point signed Q19.12.
64    // That is, for a mono provider, there is an implicit up-channeling.
65    // Since this method accumulates, the caller is responsible for clearing 'out' initially.
66    // FIXME assumes provider is always successful; it should return the actual frame count.
67    virtual void resample(int32_t* out, size_t outFrameCount,
68            AudioBufferProvider* provider) = 0;
69
70    virtual void reset();
71    virtual size_t getUnreleasedFrames() const { return mInputIndex; }
72
73    // called from destructor, so must not be virtual
74    src_quality getQuality() const { return mQuality; }
75
76protected:
77    // number of bits for phase fraction - 30 bits allows nearly 2x downsampling
78    static const int kNumPhaseBits = 30;
79
80    // phase mask for fraction
81    static const uint32_t kPhaseMask = (1LU<<kNumPhaseBits)-1;
82
83    // multiplier to calculate fixed point phase increment
84    static const double kPhaseMultiplier = 1L << kNumPhaseBits;
85
86    AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate, src_quality quality);
87
88    // prevent copying
89    AudioResampler(const AudioResampler&);
90    AudioResampler& operator=(const AudioResampler&);
91
92    int64_t calculateOutputPTS(int outputFrameIndex);
93
94    const int32_t mBitDepth;
95    const int32_t mChannelCount;
96    const int32_t mSampleRate;
97    int32_t mInSampleRate;
98    AudioBufferProvider::Buffer mBuffer;
99    union {
100        int16_t mVolume[2];
101        uint32_t mVolumeRL;
102    };
103    int16_t mTargetVolume[2];
104    size_t mInputIndex;
105    int32_t mPhaseIncrement;
106    uint32_t mPhaseFraction;
107    uint64_t mLocalTimeFreq;
108    int64_t mPTS;
109
110private:
111    const src_quality mQuality;
112
113    // Return 'true' if the quality level is supported without explicit request
114    static bool qualityIsSupported(src_quality quality);
115
116    // For pthread_once()
117    static void init_routine();
118
119    // Return the estimated CPU load for specific resampler in MHz.
120    // The absolute number is irrelevant, it's the relative values that matter.
121    static uint32_t qualityMHz(src_quality quality);
122};
123
124// ----------------------------------------------------------------------------
125}
126; // namespace android
127
128#endif // ANDROID_AUDIO_RESAMPLER_H
129