| /* |
| * Copyright (C) 2017 The Android Open Source Project |
| * |
| * Licensed under the Apache License, Version 2.0 (the "License"); |
| * you may not use this file except in compliance with the License. |
| * You may obtain a copy of the License at |
| * |
| * http://www.apache.org/licenses/LICENSE-2.0 |
| * |
| * Unless required by applicable law or agreed to in writing, software |
| * distributed under the License is distributed on an "AS IS" BASIS, |
| * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. |
| * See the License for the specific language governing permissions and |
| * limitations under the License. |
| */ |
| |
| #define LOG_TAG "AAudioMixer" |
| //#define LOG_NDEBUG 0 |
| #include <utils/Log.h> |
| |
| #define ATRACE_TAG ATRACE_TAG_AUDIO |
| |
| #include <cstring> |
| #include <utils/Trace.h> |
| |
| #include "AAudioMixer.h" |
| |
| #ifndef AAUDIO_MIXER_ATRACE_ENABLED |
| #define AAUDIO_MIXER_ATRACE_ENABLED 1 |
| #endif |
| |
| using android::WrappingBuffer; |
| using android::FifoBuffer; |
| using android::fifo_frames_t; |
| |
| void AAudioMixer::allocate(int32_t samplesPerFrame, int32_t framesPerBurst) { |
| mSamplesPerFrame = samplesPerFrame; |
| mFramesPerBurst = framesPerBurst; |
| int32_t samplesPerBuffer = samplesPerFrame * framesPerBurst; |
| mOutputBuffer = std::make_unique<float[]>(samplesPerBuffer); |
| mBufferSizeInBytes = samplesPerBuffer * sizeof(float); |
| } |
| |
| void AAudioMixer::clear() { |
| memset(mOutputBuffer.get(), 0, mBufferSizeInBytes); |
| } |
| |
| int32_t AAudioMixer::mix( |
| int streamIndex, const std::shared_ptr<FifoBuffer>& fifo, bool allowUnderflow) { |
| WrappingBuffer wrappingBuffer; |
| float *destination = mOutputBuffer.get(); |
| |
| #if AAUDIO_MIXER_ATRACE_ENABLED |
| ATRACE_BEGIN("aaMix"); |
| #endif /* AAUDIO_MIXER_ATRACE_ENABLED */ |
| |
| // Gather the data from the client. May be in two parts. |
| fifo_frames_t fullFrames = fifo->getFullDataAvailable(&wrappingBuffer); |
| #if AAUDIO_MIXER_ATRACE_ENABLED |
| if (ATRACE_ENABLED()) { |
| char rdyText[] = "aaMixRdy#"; |
| char letter = 'A' + (streamIndex % 26); |
| rdyText[sizeof(rdyText) - 2] = letter; |
| ATRACE_INT(rdyText, fullFrames); |
| } |
| #else /* MIXER_ATRACE_ENABLED */ |
| (void) trackIndex; |
| #endif /* AAUDIO_MIXER_ATRACE_ENABLED */ |
| |
| // If allowUnderflow then always advance by one burst even if we do not have the data. |
| // Otherwise the stream timing will drift whenever there is an underflow. |
| // This actual underflow can then be detected by the client for XRun counting. |
| // |
| // Generally, allowUnderflow will be false when stopping a stream and we want to |
| // use up whatever data is in the queue. |
| fifo_frames_t framesDesired = mFramesPerBurst; |
| if (!allowUnderflow && fullFrames < framesDesired) { |
| framesDesired = fullFrames; // just use what is available then stop |
| } |
| |
| // Mix data in one or two parts. |
| int partIndex = 0; |
| int32_t framesLeft = framesDesired; |
| while (framesLeft > 0 && partIndex < WrappingBuffer::SIZE) { |
| fifo_frames_t framesToMixFromPart = framesLeft; |
| fifo_frames_t framesAvailableFromPart = wrappingBuffer.numFrames[partIndex]; |
| if (framesAvailableFromPart > 0) { |
| if (framesToMixFromPart > framesAvailableFromPart) { |
| framesToMixFromPart = framesAvailableFromPart; |
| } |
| mixPart(destination, (float *)wrappingBuffer.data[partIndex], |
| framesToMixFromPart); |
| |
| destination += framesToMixFromPart * mSamplesPerFrame; |
| framesLeft -= framesToMixFromPart; |
| } |
| partIndex++; |
| } |
| fifo->advanceReadIndex(framesDesired); |
| |
| #if AAUDIO_MIXER_ATRACE_ENABLED |
| ATRACE_END(); |
| #endif /* AAUDIO_MIXER_ATRACE_ENABLED */ |
| |
| return (framesDesired - framesLeft); // framesRead |
| } |
| |
| void AAudioMixer::mixPart(float *destination, float *source, int32_t numFrames) { |
| int32_t numSamples = numFrames * mSamplesPerFrame; |
| // TODO maybe optimize using SIMD |
| for (int sampleIndex = 0; sampleIndex < numSamples; sampleIndex++) { |
| *destination++ += *source++; |
| } |
| } |
| |
| float *AAudioMixer::getOutputBuffer() { |
| return mOutputBuffer.get(); |
| } |