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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 #include <system/audio.h>
26 
27 namespace android {
28 // ----------------------------------------------------------------------------
29 
30 class ANDROID_API AudioResampler {
31 public:
32     // Determines quality of SRC.
33     //  LOW_QUALITY: linear interpolator (1st order)
34     //  MED_QUALITY: cubic interpolator (3rd order)
35     //  HIGH_QUALITY: fixed multi-tap FIR (e.g. 48KHz->44.1KHz)
36     // NOTE: high quality SRC will only be supported for
37     // certain fixed rate conversions. Sample rate cannot be
38     // changed dynamically.
39     enum src_quality {
40         DEFAULT_QUALITY=0,
41         LOW_QUALITY=1,
42         MED_QUALITY=2,
43         HIGH_QUALITY=3,
44         VERY_HIGH_QUALITY=4,
45         DYN_LOW_QUALITY=5,
46         DYN_MED_QUALITY=6,
47         DYN_HIGH_QUALITY=7,
48     };
49 
50     static const float UNITY_GAIN_FLOAT = 1.0f;
51 
52     static AudioResampler* create(audio_format_t format, int inChannelCount,
53             int32_t sampleRate, src_quality quality=DEFAULT_QUALITY);
54 
55     virtual ~AudioResampler();
56 
57     virtual void init() = 0;
58     virtual void setSampleRate(int32_t inSampleRate);
59     virtual void setVolume(float left, float right);
60     virtual void setLocalTimeFreq(uint64_t freq);
61 
62     // set the PTS of the next buffer output by the resampler
63     virtual void setPTS(int64_t pts);
64 
65     // Resample int16_t samples from provider and accumulate into 'out'.
66     // A mono provider delivers a sequence of samples.
67     // A stereo provider delivers a sequence of interleaved pairs of samples.
68     // Multi-channel providers are not supported.
69     // In either case, 'out' holds interleaved pairs of fixed-point Q4.27.
70     // That is, for a mono provider, there is an implicit up-channeling.
71     // Since this method accumulates, the caller is responsible for clearing 'out' initially.
72     // FIXME assumes provider is always successful; it should return the actual frame count.
73     virtual void resample(int32_t* out, size_t outFrameCount,
74             AudioBufferProvider* provider) = 0;
75 
76     virtual void reset();
getUnreleasedFrames()77     virtual size_t getUnreleasedFrames() const { return mInputIndex; }
78 
79     // called from destructor, so must not be virtual
getQuality()80     src_quality getQuality() const { return mQuality; }
81 
82 protected:
83     // number of bits for phase fraction - 30 bits allows nearly 2x downsampling
84     static const int kNumPhaseBits = 30;
85 
86     // phase mask for fraction
87     static const uint32_t kPhaseMask = (1LU<<kNumPhaseBits)-1;
88 
89     // multiplier to calculate fixed point phase increment
90     static const double kPhaseMultiplier;
91 
92     AudioResampler(int inChannelCount, int32_t sampleRate, src_quality quality);
93 
94     // prevent copying
95     AudioResampler(const AudioResampler&);
96     AudioResampler& operator=(const AudioResampler&);
97 
98     int64_t calculateOutputPTS(int outputFrameIndex);
99 
100     const int32_t mChannelCount;
101     const int32_t mSampleRate;
102     int32_t mInSampleRate;
103     AudioBufferProvider::Buffer mBuffer;
104     union {
105         int16_t mVolume[2];
106         uint32_t mVolumeRL;
107     };
108     int16_t mTargetVolume[2];
109     size_t mInputIndex;
110     int32_t mPhaseIncrement;
111     uint32_t mPhaseFraction;
112     uint64_t mLocalTimeFreq;
113     int64_t mPTS;
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     //
getInFrameCountRequired(size_t outFrameCount)142     inline size_t getInFrameCountRequired(size_t outFrameCount) {
143         return (static_cast<uint64_t>(outFrameCount)*mInSampleRate
144                 + (mSampleRate - 1))/mSampleRate;
145     }
146 
clampFloatVol(float volume)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 
156 private:
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 }
172 ; // namespace android
173 
174 #endif // ANDROID_AUDIO_RESAMPLER_H
175