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 #define LOG_TAG "AudioResamplerCubic"
18
19 #include <stdint.h>
20 #include <string.h>
21 #include <sys/types.h>
22
23 #include <log/log.h>
24
25 #include "AudioResamplerCubic.h"
26
27 namespace android {
28 // ----------------------------------------------------------------------------
29
init()30 void AudioResamplerCubic::init() {
31 memset(&left, 0, sizeof(state));
32 memset(&right, 0, sizeof(state));
33 }
34
resample(int32_t * out,size_t outFrameCount,AudioBufferProvider * provider)35 size_t AudioResamplerCubic::resample(int32_t* out, size_t outFrameCount,
36 AudioBufferProvider* provider) {
37
38 // should never happen, but we overflow if it does
39 // ALOG_ASSERT(outFrameCount < 32767);
40
41 // select the appropriate resampler
42 switch (mChannelCount) {
43 case 1:
44 return resampleMono16(out, outFrameCount, provider);
45 case 2:
46 return resampleStereo16(out, outFrameCount, provider);
47 default:
48 LOG_ALWAYS_FATAL("invalid channel count: %d", mChannelCount);
49 return 0;
50 }
51 }
52
resampleStereo16(int32_t * out,size_t outFrameCount,AudioBufferProvider * provider)53 size_t AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount,
54 AudioBufferProvider* provider) {
55
56 int32_t vl = mVolume[0];
57 int32_t vr = mVolume[1];
58
59 size_t inputIndex = mInputIndex;
60 uint32_t phaseFraction = mPhaseFraction;
61 uint32_t phaseIncrement = mPhaseIncrement;
62 size_t outputIndex = 0;
63 size_t outputSampleCount = outFrameCount * 2;
64 size_t inFrameCount = getInFrameCountRequired(outFrameCount);
65
66 // fetch first buffer
67 if (mBuffer.frameCount == 0) {
68 mBuffer.frameCount = inFrameCount;
69 provider->getNextBuffer(&mBuffer);
70 if (mBuffer.raw == NULL) {
71 return 0;
72 }
73 // ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
74 }
75 int16_t *in = mBuffer.i16;
76
77 while (outputIndex < outputSampleCount) {
78 int32_t x;
79
80 // calculate output sample
81 x = phaseFraction >> kPreInterpShift;
82 out[outputIndex++] += vl * interp(&left, x);
83 out[outputIndex++] += vr * interp(&right, x);
84 // out[outputIndex++] += vr * in[inputIndex*2];
85
86 // increment phase
87 phaseFraction += phaseIncrement;
88 uint32_t indexIncrement = (phaseFraction >> kNumPhaseBits);
89 phaseFraction &= kPhaseMask;
90
91 // time to fetch another sample
92 while (indexIncrement--) {
93
94 inputIndex++;
95 if (inputIndex == mBuffer.frameCount) {
96 inputIndex = 0;
97 provider->releaseBuffer(&mBuffer);
98 mBuffer.frameCount = inFrameCount;
99 provider->getNextBuffer(&mBuffer);
100 if (mBuffer.raw == NULL) {
101 goto save_state; // ugly, but efficient
102 }
103 in = mBuffer.i16;
104 // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
105 }
106
107 // advance sample state
108 advance(&left, in[inputIndex*2]);
109 advance(&right, in[inputIndex*2+1]);
110 }
111 }
112
113 save_state:
114 // ALOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction);
115 mInputIndex = inputIndex;
116 mPhaseFraction = phaseFraction;
117 return outputIndex / 2 /* channels for stereo */;
118 }
119
resampleMono16(int32_t * out,size_t outFrameCount,AudioBufferProvider * provider)120 size_t AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount,
121 AudioBufferProvider* provider) {
122
123 int32_t vl = mVolume[0];
124 int32_t vr = mVolume[1];
125
126 size_t inputIndex = mInputIndex;
127 uint32_t phaseFraction = mPhaseFraction;
128 uint32_t phaseIncrement = mPhaseIncrement;
129 size_t outputIndex = 0;
130 size_t outputSampleCount = outFrameCount * 2;
131 size_t inFrameCount = getInFrameCountRequired(outFrameCount);
132
133 // fetch first buffer
134 if (mBuffer.frameCount == 0) {
135 mBuffer.frameCount = inFrameCount;
136 provider->getNextBuffer(&mBuffer);
137 if (mBuffer.raw == NULL) {
138 return 0;
139 }
140 // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
141 }
142 int16_t *in = mBuffer.i16;
143
144 while (outputIndex < outputSampleCount) {
145 int32_t sample;
146 int32_t x;
147
148 // calculate output sample
149 x = phaseFraction >> kPreInterpShift;
150 sample = interp(&left, x);
151 out[outputIndex++] += vl * sample;
152 out[outputIndex++] += vr * sample;
153
154 // increment phase
155 phaseFraction += phaseIncrement;
156 uint32_t indexIncrement = (phaseFraction >> kNumPhaseBits);
157 phaseFraction &= kPhaseMask;
158
159 // time to fetch another sample
160 while (indexIncrement--) {
161
162 inputIndex++;
163 if (inputIndex == mBuffer.frameCount) {
164 inputIndex = 0;
165 provider->releaseBuffer(&mBuffer);
166 mBuffer.frameCount = inFrameCount;
167 provider->getNextBuffer(&mBuffer);
168 if (mBuffer.raw == NULL) {
169 goto save_state; // ugly, but efficient
170 }
171 // ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
172 in = mBuffer.i16;
173 }
174
175 // advance sample state
176 advance(&left, in[inputIndex]);
177 }
178 }
179
180 save_state:
181 // ALOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction);
182 mInputIndex = inputIndex;
183 mPhaseFraction = phaseFraction;
184 return outputIndex;
185 }
186
187 // ----------------------------------------------------------------------------
188 } // namespace android
189