| /* |
| * Copyright 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. |
| */ |
| |
| #ifndef OBOE_LATENCY_TUNER_ |
| #define OBOE_LATENCY_TUNER_ |
| |
| #include <atomic> |
| #include <cstdint> |
| #include "oboe/Definitions.h" |
| #include "oboe/AudioStream.h" |
| |
| namespace oboe { |
| |
| /** |
| * LatencyTuner can be used to dynamically tune the latency of an output stream. |
| * It adjusts the stream's bufferSize by monitoring the number of underruns. |
| * |
| * This only affects the latency associated with the first level of buffering that is closest |
| * to the application. It does not affect low latency in the HAL, or touch latency in the UI. |
| * |
| * Call tune() right before returning from your data callback function if using callbacks. |
| * Call tune() right before calling write() if using blocking writes. |
| * |
| * If you want to see the ongoing results of this tuning process then call |
| * stream->getBufferSize() periodically. |
| * |
| */ |
| class LatencyTuner { |
| public: |
| |
| /** |
| * Construct a new LatencyTuner object which will act on the given audio stream |
| * |
| * @param stream the stream who's latency will be tuned |
| */ |
| explicit LatencyTuner(AudioStream &stream); |
| |
| /** |
| * Construct a new LatencyTuner object which will act on the given audio stream. |
| * |
| * @param stream the stream who's latency will be tuned |
| * @param the maximum buffer size which the tune() operation will set the buffer size to |
| */ |
| explicit LatencyTuner(AudioStream &stream, int32_t maximumBufferSize); |
| |
| /** |
| * Adjust the bufferSizeInFrames to optimize latency. |
| * It will start with a low latency and then raise it if an underrun occurs. |
| * |
| * Latency tuning is only supported for AAudio. |
| * |
| * @return OK or negative error, ErrorUnimplemented for OpenSL ES |
| */ |
| Result tune(); |
| |
| /** |
| * This may be called from another thread. Then tune() will call reset(), |
| * which will lower the latency to the minimum and then allow it to rise back up |
| * if there are glitches. |
| * |
| * This is typically called in response to a user decision to minimize latency. In other words, |
| * call this from a button handler. |
| */ |
| void requestReset(); |
| |
| /** |
| * @return true if the audio stream's buffer size is at the maximum value. If no maximum value |
| * was specified when constructing the LatencyTuner then the value of |
| * stream->getBufferCapacityInFrames is used |
| */ |
| bool isAtMaximumBufferSize(); |
| |
| /** |
| * Set the minimum bufferSize in frames that is used when the tuner is reset. |
| * You may wish to call requestReset() after calling this. |
| * @param bufferSize |
| */ |
| void setMinimumBufferSize(int32_t bufferSize) { |
| mMinimumBufferSize = bufferSize; |
| } |
| |
| int32_t getMinimumBufferSize() const { |
| return mMinimumBufferSize; |
| } |
| |
| /** |
| * Set the amount the bufferSize will be incremented while tuning. |
| * By default, this will be one burst. |
| * |
| * Note that AAudio will quantize the buffer size to a multiple of the burstSize. |
| * So the final buffer sizes may not be a multiple of this increment. |
| * |
| * @param sizeIncrement |
| */ |
| void setBufferSizeIncrement(int32_t sizeIncrement) { |
| mBufferSizeIncrement = sizeIncrement; |
| } |
| |
| int32_t getBufferSizeIncrement() const { |
| return mBufferSizeIncrement; |
| } |
| |
| private: |
| |
| /** |
| * Drop the latency down to the minimum and then let it rise back up. |
| * This is useful if a glitch caused the latency to increase and it hasn't gone back down. |
| * |
| * This should only be called in the same thread as tune(). |
| */ |
| void reset(); |
| |
| enum class State { |
| Idle, |
| Active, |
| AtMax, |
| Unsupported |
| } ; |
| |
| // arbitrary number of calls to wait before bumping up the latency |
| static constexpr int32_t kIdleCount = 8; |
| static constexpr int32_t kDefaultNumBursts = 2; |
| |
| AudioStream &mStream; |
| State mState = State::Idle; |
| int32_t mMaxBufferSize = 0; |
| int32_t mPreviousXRuns = 0; |
| int32_t mIdleCountDown = 0; |
| int32_t mMinimumBufferSize; |
| int32_t mBufferSizeIncrement; |
| std::atomic<int32_t> mLatencyTriggerRequests{0}; // TODO user atomic requester from AAudio |
| std::atomic<int32_t> mLatencyTriggerResponses{0}; |
| }; |
| |
| } // namespace oboe |
| |
| #endif // OBOE_LATENCY_TUNER_ |