1 /*
2 * Copyright (C) 2017 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 "AAudioMixer"
18 //#define LOG_NDEBUG 0
19 #include <utils/Log.h>
20
21 #define ATRACE_TAG ATRACE_TAG_AUDIO
22
23 #include <cstring>
24 #include <utils/Trace.h>
25
26 #include "AAudioMixer.h"
27
28 #ifndef AAUDIO_MIXER_ATRACE_ENABLED
29 #define AAUDIO_MIXER_ATRACE_ENABLED 1
30 #endif
31
32 using android::WrappingBuffer;
33 using android::FifoBuffer;
34 using android::fifo_frames_t;
35
~AAudioMixer()36 AAudioMixer::~AAudioMixer() {
37 delete[] mOutputBuffer;
38 }
39
allocate(int32_t samplesPerFrame,int32_t framesPerBurst)40 void AAudioMixer::allocate(int32_t samplesPerFrame, int32_t framesPerBurst) {
41 mSamplesPerFrame = samplesPerFrame;
42 mFramesPerBurst = framesPerBurst;
43 int32_t samplesPerBuffer = samplesPerFrame * framesPerBurst;
44 mOutputBuffer = new float[samplesPerBuffer];
45 mBufferSizeInBytes = samplesPerBuffer * sizeof(float);
46 }
47
clear()48 void AAudioMixer::clear() {
49 memset(mOutputBuffer, 0, mBufferSizeInBytes);
50 }
51
mix(int streamIndex,FifoBuffer * fifo,bool allowUnderflow)52 int32_t AAudioMixer::mix(int streamIndex, FifoBuffer *fifo, bool allowUnderflow) {
53 WrappingBuffer wrappingBuffer;
54 float *destination = mOutputBuffer;
55
56 #if AAUDIO_MIXER_ATRACE_ENABLED
57 ATRACE_BEGIN("aaMix");
58 #endif /* AAUDIO_MIXER_ATRACE_ENABLED */
59
60 // Gather the data from the client. May be in two parts.
61 fifo_frames_t fullFrames = fifo->getFullDataAvailable(&wrappingBuffer);
62 #if AAUDIO_MIXER_ATRACE_ENABLED
63 if (ATRACE_ENABLED()) {
64 char rdyText[] = "aaMixRdy#";
65 char letter = 'A' + (streamIndex % 26);
66 rdyText[sizeof(rdyText) - 2] = letter;
67 ATRACE_INT(rdyText, fullFrames);
68 }
69 #else /* MIXER_ATRACE_ENABLED */
70 (void) trackIndex;
71 #endif /* AAUDIO_MIXER_ATRACE_ENABLED */
72
73 // If allowUnderflow then always advance by one burst even if we do not have the data.
74 // Otherwise the stream timing will drift whenever there is an underflow.
75 // This actual underflow can then be detected by the client for XRun counting.
76 //
77 // Generally, allowUnderflow will be false when stopping a stream and we want to
78 // use up whatever data is in the queue.
79 fifo_frames_t framesDesired = mFramesPerBurst;
80 if (!allowUnderflow && fullFrames < framesDesired) {
81 framesDesired = fullFrames; // just use what is available then stop
82 }
83
84 // Mix data in one or two parts.
85 int partIndex = 0;
86 int32_t framesLeft = framesDesired;
87 while (framesLeft > 0 && partIndex < WrappingBuffer::SIZE) {
88 fifo_frames_t framesToMixFromPart = framesLeft;
89 fifo_frames_t framesAvailableFromPart = wrappingBuffer.numFrames[partIndex];
90 if (framesAvailableFromPart > 0) {
91 if (framesToMixFromPart > framesAvailableFromPart) {
92 framesToMixFromPart = framesAvailableFromPart;
93 }
94 mixPart(destination, (float *)wrappingBuffer.data[partIndex],
95 framesToMixFromPart);
96
97 destination += framesToMixFromPart * mSamplesPerFrame;
98 framesLeft -= framesToMixFromPart;
99 }
100 partIndex++;
101 }
102 fifo->advanceReadIndex(framesDesired);
103
104 #if AAUDIO_MIXER_ATRACE_ENABLED
105 ATRACE_END();
106 #endif /* AAUDIO_MIXER_ATRACE_ENABLED */
107
108 return (framesDesired - framesLeft); // framesRead
109 }
110
mixPart(float * destination,float * source,int32_t numFrames)111 void AAudioMixer::mixPart(float *destination, float *source, int32_t numFrames) {
112 int32_t numSamples = numFrames * mSamplesPerFrame;
113 // TODO maybe optimize using SIMD
114 for (int sampleIndex = 0; sampleIndex < numSamples; sampleIndex++) {
115 *destination++ += *source++;
116 }
117 }
118
getOutputBuffer()119 float *AAudioMixer::getOutputBuffer() {
120 return mOutputBuffer;
121 }
122