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 #include <cutils/log.h>
23
24 #include "AudioResampler.h"
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, mPTS);
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 sample;
79 int32_t x;
80
81 // calculate output sample
82 x = phaseFraction >> kPreInterpShift;
83 out[outputIndex++] += vl * interp(&left, x);
84 out[outputIndex++] += vr * interp(&right, x);
85 // out[outputIndex++] += vr * in[inputIndex*2];
86
87 // increment phase
88 phaseFraction += phaseIncrement;
89 uint32_t indexIncrement = (phaseFraction >> kNumPhaseBits);
90 phaseFraction &= kPhaseMask;
91
92 // time to fetch another sample
93 while (indexIncrement--) {
94
95 inputIndex++;
96 if (inputIndex == mBuffer.frameCount) {
97 inputIndex = 0;
98 provider->releaseBuffer(&mBuffer);
99 mBuffer.frameCount = inFrameCount;
100 provider->getNextBuffer(&mBuffer,
101 calculateOutputPTS(outputIndex / 2));
102 if (mBuffer.raw == NULL) {
103 goto save_state; // ugly, but efficient
104 }
105 in = mBuffer.i16;
106 // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
107 }
108
109 // advance sample state
110 advance(&left, in[inputIndex*2]);
111 advance(&right, in[inputIndex*2+1]);
112 }
113 }
114
115 save_state:
116 // ALOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction);
117 mInputIndex = inputIndex;
118 mPhaseFraction = phaseFraction;
119 return outputIndex / 2 /* channels for stereo */;
120 }
121
resampleMono16(int32_t * out,size_t outFrameCount,AudioBufferProvider * provider)122 size_t AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount,
123 AudioBufferProvider* provider) {
124
125 int32_t vl = mVolume[0];
126 int32_t vr = mVolume[1];
127
128 size_t inputIndex = mInputIndex;
129 uint32_t phaseFraction = mPhaseFraction;
130 uint32_t phaseIncrement = mPhaseIncrement;
131 size_t outputIndex = 0;
132 size_t outputSampleCount = outFrameCount * 2;
133 size_t inFrameCount = getInFrameCountRequired(outFrameCount);
134
135 // fetch first buffer
136 if (mBuffer.frameCount == 0) {
137 mBuffer.frameCount = inFrameCount;
138 provider->getNextBuffer(&mBuffer, mPTS);
139 if (mBuffer.raw == NULL) {
140 return 0;
141 }
142 // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
143 }
144 int16_t *in = mBuffer.i16;
145
146 while (outputIndex < outputSampleCount) {
147 int32_t sample;
148 int32_t x;
149
150 // calculate output sample
151 x = phaseFraction >> kPreInterpShift;
152 sample = interp(&left, x);
153 out[outputIndex++] += vl * sample;
154 out[outputIndex++] += vr * sample;
155
156 // increment phase
157 phaseFraction += phaseIncrement;
158 uint32_t indexIncrement = (phaseFraction >> kNumPhaseBits);
159 phaseFraction &= kPhaseMask;
160
161 // time to fetch another sample
162 while (indexIncrement--) {
163
164 inputIndex++;
165 if (inputIndex == mBuffer.frameCount) {
166 inputIndex = 0;
167 provider->releaseBuffer(&mBuffer);
168 mBuffer.frameCount = inFrameCount;
169 provider->getNextBuffer(&mBuffer,
170 calculateOutputPTS(outputIndex / 2));
171 if (mBuffer.raw == NULL) {
172 goto save_state; // ugly, but efficient
173 }
174 // ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
175 in = mBuffer.i16;
176 }
177
178 // advance sample state
179 advance(&left, in[inputIndex]);
180 }
181 }
182
183 save_state:
184 // ALOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction);
185 mInputIndex = inputIndex;
186 mPhaseFraction = phaseFraction;
187 return outputIndex;
188 }
189
190 // ----------------------------------------------------------------------------
191 } // namespace android
192