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 
29 namespace android {
30 // ----------------------------------------------------------------------------
31 
32 class ANDROID_API AudioResampler {
33 public:
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     virtual void setLocalTimeFreq(uint64_t freq);
63 
64     // set the PTS of the next buffer output by the resampler
65     virtual void setPTS(int64_t pts);
66 
67     // Resample int16_t samples from provider and accumulate into 'out'.
68     // A mono provider delivers a sequence of samples.
69     // A stereo provider delivers a sequence of interleaved pairs of samples.
70     //
71     // In either case, 'out' holds interleaved pairs of fixed-point Q4.27.
72     // That is, for a mono provider, there is an implicit up-channeling.
73     // Since this method accumulates, the caller is responsible for clearing 'out' initially.
74     //
75     // For a float resampler, 'out' holds interleaved pairs of float samples.
76     //
77     // Multichannel interleaved frames for n > 2 is supported for quality DYN_LOW_QUALITY,
78     // DYN_MED_QUALITY, and DYN_HIGH_QUALITY.
79     //
80     // Returns the number of frames resampled into the out buffer.
81     virtual size_t resample(int32_t* out, size_t outFrameCount,
82             AudioBufferProvider* provider) = 0;
83 
84     virtual void reset();
getUnreleasedFrames()85     virtual size_t getUnreleasedFrames() const { return mInputIndex; }
86 
87     // called from destructor, so must not be virtual
getQuality()88     src_quality getQuality() const { return mQuality; }
89 
90 protected:
91     // number of bits for phase fraction - 30 bits allows nearly 2x downsampling
92     static const int kNumPhaseBits = 30;
93 
94     // phase mask for fraction
95     static const uint32_t kPhaseMask = (1LU<<kNumPhaseBits)-1;
96 
97     // multiplier to calculate fixed point phase increment
98     static const double kPhaseMultiplier;
99 
100     AudioResampler(int inChannelCount, int32_t sampleRate, src_quality quality);
101 
102     // prevent copying
103     AudioResampler(const AudioResampler&);
104     AudioResampler& operator=(const AudioResampler&);
105 
106     int64_t calculateOutputPTS(int outputFrameIndex);
107 
108     const int32_t mChannelCount;
109     const int32_t mSampleRate;
110     int32_t mInSampleRate;
111     AudioBufferProvider::Buffer mBuffer;
112     union {
113         int16_t mVolume[2];
114         uint32_t mVolumeRL;
115     };
116     int16_t mTargetVolume[2];
117     size_t mInputIndex;
118     int32_t mPhaseIncrement;
119     uint32_t mPhaseFraction;
120     uint64_t mLocalTimeFreq;
121     int64_t mPTS;
122 
123     // returns the inFrameCount required to generate outFrameCount frames.
124     //
125     // Placed here to be a consistent for all resamplers.
126     //
127     // Right now, we use the upper bound without regards to the current state of the
128     // input buffer using integer arithmetic, as follows:
129     //
130     // (static_cast<uint64_t>(outFrameCount)*mInSampleRate + (mSampleRate - 1))/mSampleRate;
131     //
132     // The double precision equivalent (float may not be precise enough):
133     // ceil(static_cast<double>(outFrameCount) * mInSampleRate / mSampleRate);
134     //
135     // this relies on the fact that the mPhaseIncrement is rounded down from
136     // #phases * mInSampleRate/mSampleRate and the fact that Sum(Floor(x)) <= Floor(Sum(x)).
137     // http://www.proofwiki.org/wiki/Sum_of_Floors_Not_Greater_Than_Floor_of_Sums
138     //
139     // (so long as double precision is computed accurately enough to be considered
140     // greater than or equal to the Floor(x) value in int32_t arithmetic; thus this
141     // will not necessarily hold for floats).
142     //
143     // TODO:
144     // Greater accuracy and a tight bound is obtained by:
145     // 1) subtract and adjust for the current state of the AudioBufferProvider buffer.
146     // 2) using the exact integer formula where (ignoring 64b casting)
147     //  inFrameCount = (mPhaseIncrement * (outFrameCount - 1) + mPhaseFraction) / phaseWrapLimit;
148     //  phaseWrapLimit is the wraparound (1 << kNumPhaseBits), if not specified explicitly.
149     //
getInFrameCountRequired(size_t outFrameCount)150     inline size_t getInFrameCountRequired(size_t outFrameCount) {
151         return (static_cast<uint64_t>(outFrameCount)*mInSampleRate
152                 + (mSampleRate - 1))/mSampleRate;
153     }
154 
clampFloatVol(float volume)155     inline float clampFloatVol(float volume) {
156         if (volume > UNITY_GAIN_FLOAT) {
157             return UNITY_GAIN_FLOAT;
158         } else if (volume >= 0.) {
159             return volume;
160         }
161         return 0.;  // NaN or negative volume maps to 0.
162     }
163 
164 private:
165     const src_quality mQuality;
166 
167     // Return 'true' if the quality level is supported without explicit request
168     static bool qualityIsSupported(src_quality quality);
169 
170     // For pthread_once()
171     static void init_routine();
172 
173     // Return the estimated CPU load for specific resampler in MHz.
174     // The absolute number is irrelevant, it's the relative values that matter.
175     static uint32_t qualityMHz(src_quality quality);
176 };
177 
178 // ----------------------------------------------------------------------------
179 } // namespace android
180 
181 #endif // ANDROID_AUDIO_RESAMPLER_H
182