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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 
26 namespace android {
27 // ----------------------------------------------------------------------------
28 
29 class ANDROID_API AudioResampler {
30 public:
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();
getUnreleasedFrames()71     virtual size_t getUnreleasedFrames() const { return mInputIndex; }
72 
73     // called from destructor, so must not be virtual
getQuality()74     src_quality getQuality() const { return mQuality; }
75 
76 protected:
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 
110 private:
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