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Phil Burkc0c70e32017-02-09 13:18:38 -08001/*
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
Phil Burkfbf031e2017-10-12 15:58:31 -070017#define LOG_TAG "AAudioMixer"
Phil Burkc0c70e32017-02-09 13:18:38 -080018//#define LOG_NDEBUG 0
19#include <utils/Log.h>
20
Phil Burkfd34a932017-07-19 07:03:52 -070021#define ATRACE_TAG ATRACE_TAG_AUDIO
22
Phil Burkc0c70e32017-02-09 13:18:38 -080023#include <cstring>
Phil Burkfd34a932017-07-19 07:03:52 -070024#include <utils/Trace.h>
25
Phil Burkc0c70e32017-02-09 13:18:38 -080026#include "AAudioMixer.h"
27
Phil Burkfd34a932017-07-19 07:03:52 -070028#ifndef AAUDIO_MIXER_ATRACE_ENABLED
29#define AAUDIO_MIXER_ATRACE_ENABLED 1
30#endif
31
Phil Burkc0c70e32017-02-09 13:18:38 -080032using android::WrappingBuffer;
33using android::FifoBuffer;
34using android::fifo_frames_t;
35
Phil Burkc0c70e32017-02-09 13:18:38 -080036void AAudioMixer::allocate(int32_t samplesPerFrame, int32_t framesPerBurst) {
37 mSamplesPerFrame = samplesPerFrame;
38 mFramesPerBurst = framesPerBurst;
39 int32_t samplesPerBuffer = samplesPerFrame * framesPerBurst;
Phil Burkcdfd7932020-07-15 22:44:43 +000040 mOutputBuffer = std::make_unique<float[]>(samplesPerBuffer);
Phil Burkc0c70e32017-02-09 13:18:38 -080041 mBufferSizeInBytes = samplesPerBuffer * sizeof(float);
42}
43
44void AAudioMixer::clear() {
Phil Burkcdfd7932020-07-15 22:44:43 +000045 memset(mOutputBuffer.get(), 0, mBufferSizeInBytes);
Phil Burkc0c70e32017-02-09 13:18:38 -080046}
47
Phil Burk8f4fe502020-07-15 23:54:50 +000048int32_t AAudioMixer::mix(int streamIndex, std::shared_ptr<FifoBuffer> fifo, bool allowUnderflow) {
Phil Burkc0c70e32017-02-09 13:18:38 -080049 WrappingBuffer wrappingBuffer;
Phil Burkcdfd7932020-07-15 22:44:43 +000050 float *destination = mOutputBuffer.get();
Phil Burkc0c70e32017-02-09 13:18:38 -080051
Phil Burkfd34a932017-07-19 07:03:52 -070052#if AAUDIO_MIXER_ATRACE_ENABLED
53 ATRACE_BEGIN("aaMix");
54#endif /* AAUDIO_MIXER_ATRACE_ENABLED */
55
Phil Burkc0c70e32017-02-09 13:18:38 -080056 // Gather the data from the client. May be in two parts.
Phil Burkfd34a932017-07-19 07:03:52 -070057 fifo_frames_t fullFrames = fifo->getFullDataAvailable(&wrappingBuffer);
58#if AAUDIO_MIXER_ATRACE_ENABLED
59 if (ATRACE_ENABLED()) {
60 char rdyText[] = "aaMixRdy#";
Phil Burk83fb8442017-10-05 16:55:17 -070061 char letter = 'A' + (streamIndex % 26);
Phil Burkfd34a932017-07-19 07:03:52 -070062 rdyText[sizeof(rdyText) - 2] = letter;
63 ATRACE_INT(rdyText, fullFrames);
64 }
65#else /* MIXER_ATRACE_ENABLED */
66 (void) trackIndex;
Phil Burkfd34a932017-07-19 07:03:52 -070067#endif /* AAUDIO_MIXER_ATRACE_ENABLED */
Phil Burkc0c70e32017-02-09 13:18:38 -080068
Phil Burk83fb8442017-10-05 16:55:17 -070069 // If allowUnderflow then always advance by one burst even if we do not have the data.
70 // Otherwise the stream timing will drift whenever there is an underflow.
71 // This actual underflow can then be detected by the client for XRun counting.
72 //
73 // Generally, allowUnderflow will be false when stopping a stream and we want to
74 // use up whatever data is in the queue.
75 fifo_frames_t framesDesired = mFramesPerBurst;
76 if (!allowUnderflow && fullFrames < framesDesired) {
77 framesDesired = fullFrames; // just use what is available then stop
78 }
79
Phil Burkc0c70e32017-02-09 13:18:38 -080080 // Mix data in one or two parts.
81 int partIndex = 0;
Phil Burk83fb8442017-10-05 16:55:17 -070082 int32_t framesLeft = framesDesired;
Phil Burkc0c70e32017-02-09 13:18:38 -080083 while (framesLeft > 0 && partIndex < WrappingBuffer::SIZE) {
Phil Burk83fb8442017-10-05 16:55:17 -070084 fifo_frames_t framesToMixFromPart = framesLeft;
85 fifo_frames_t framesAvailableFromPart = wrappingBuffer.numFrames[partIndex];
86 if (framesAvailableFromPart > 0) {
87 if (framesToMixFromPart > framesAvailableFromPart) {
88 framesToMixFromPart = framesAvailableFromPart;
Phil Burkc0c70e32017-02-09 13:18:38 -080089 }
Phil Burk83fb8442017-10-05 16:55:17 -070090 mixPart(destination, (float *)wrappingBuffer.data[partIndex],
91 framesToMixFromPart);
Phil Burkc0c70e32017-02-09 13:18:38 -080092
Phil Burk83fb8442017-10-05 16:55:17 -070093 destination += framesToMixFromPart * mSamplesPerFrame;
94 framesLeft -= framesToMixFromPart;
Phil Burkc0c70e32017-02-09 13:18:38 -080095 }
96 partIndex++;
97 }
Phil Burk882c5202018-04-23 10:32:45 -070098 fifo->advanceReadIndex(framesDesired);
Phil Burkfaeb8b22017-07-25 15:15:07 -070099
Phil Burkfd34a932017-07-19 07:03:52 -0700100#if AAUDIO_MIXER_ATRACE_ENABLED
101 ATRACE_END();
102#endif /* AAUDIO_MIXER_ATRACE_ENABLED */
103
Phil Burk23296382017-11-20 15:45:11 -0800104 return (framesDesired - framesLeft); // framesRead
Phil Burkc0c70e32017-02-09 13:18:38 -0800105}
106
Phil Burk83fb8442017-10-05 16:55:17 -0700107void AAudioMixer::mixPart(float *destination, float *source, int32_t numFrames) {
Phil Burkc0c70e32017-02-09 13:18:38 -0800108 int32_t numSamples = numFrames * mSamplesPerFrame;
109 // TODO maybe optimize using SIMD
110 for (int sampleIndex = 0; sampleIndex < numSamples; sampleIndex++) {
Phil Burk83fb8442017-10-05 16:55:17 -0700111 *destination++ += *source++;
Phil Burkc0c70e32017-02-09 13:18:38 -0800112 }
113}
114
115float *AAudioMixer::getOutputBuffer() {
Phil Burkcdfd7932020-07-15 22:44:43 +0000116 return mOutputBuffer.get();
Phil Burkc0c70e32017-02-09 13:18:38 -0800117}