| /* |
| * Copyright 2016 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. |
| */ |
| |
| #include <cassert> |
| #include <stdint.h> |
| #include <stdlib.h> |
| |
| #include "aaudio/AAudioLoader.h" |
| #include "aaudio/AudioStreamAAudio.h" |
| #include "common/AudioClock.h" |
| #include "common/OboeDebug.h" |
| #include "oboe/Utilities.h" |
| #include "AAudioExtensions.h" |
| |
| #ifdef __ANDROID__ |
| #include <sys/system_properties.h> |
| #include <common/QuirksManager.h> |
| |
| #endif |
| |
| #ifndef OBOE_FIX_FORCE_STARTING_TO_STARTED |
| // Workaround state problems in AAudio |
| // TODO Which versions does this occur in? Verify fixed in Q. |
| #define OBOE_FIX_FORCE_STARTING_TO_STARTED 1 |
| #endif // OBOE_FIX_FORCE_STARTING_TO_STARTED |
| |
| using namespace oboe; |
| AAudioLoader *AudioStreamAAudio::mLibLoader = nullptr; |
| |
| // 'C' wrapper for the data callback method |
| static aaudio_data_callback_result_t oboe_aaudio_data_callback_proc( |
| AAudioStream *stream, |
| void *userData, |
| void *audioData, |
| int32_t numFrames) { |
| |
| AudioStreamAAudio *oboeStream = reinterpret_cast<AudioStreamAAudio*>(userData); |
| if (oboeStream != nullptr) { |
| return static_cast<aaudio_data_callback_result_t>( |
| oboeStream->callOnAudioReady(stream, audioData, numFrames)); |
| |
| } else { |
| return static_cast<aaudio_data_callback_result_t>(DataCallbackResult::Stop); |
| } |
| } |
| |
| // This runs in its own thread. |
| // Only one of these threads will be launched from internalErrorCallback(). |
| // It calls app error callbacks from a static function in case the stream gets deleted. |
| static void oboe_aaudio_error_thread_proc(AudioStreamAAudio *oboeStream, |
| Result error) { |
| LOGD("%s(,%d) - entering >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>", __func__, error); |
| AudioStreamErrorCallback *errorCallback = oboeStream->getErrorCallback(); |
| if (errorCallback == nullptr) return; // should be impossible |
| bool isErrorHandled = errorCallback->onError(oboeStream, error); |
| |
| if (!isErrorHandled) { |
| oboeStream->requestStop(); |
| errorCallback->onErrorBeforeClose(oboeStream, error); |
| oboeStream->close(); |
| // Warning, oboeStream may get deleted by this callback. |
| errorCallback->onErrorAfterClose(oboeStream, error); |
| } |
| LOGD("%s() - exiting <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<", __func__); |
| } |
| |
| // This runs in its own thread. |
| // Only one of these threads will be launched from internalErrorCallback(). |
| // Prevents deletion of the stream if the app is using AudioStreamBuilder::openSharedStream() |
| static void oboe_aaudio_error_thread_proc_shared(std::shared_ptr<AudioStream> sharedStream, |
| Result error) { |
| AudioStreamAAudio *oboeStream = reinterpret_cast<AudioStreamAAudio*>(sharedStream.get()); |
| oboe_aaudio_error_thread_proc(oboeStream, error); |
| } |
| |
| namespace oboe { |
| |
| /* |
| * Create a stream that uses Oboe Audio API. |
| */ |
| AudioStreamAAudio::AudioStreamAAudio(const AudioStreamBuilder &builder) |
| : AudioStream(builder) |
| , mAAudioStream(nullptr) { |
| mCallbackThreadEnabled.store(false); |
| mLibLoader = AAudioLoader::getInstance(); |
| } |
| |
| bool AudioStreamAAudio::isSupported() { |
| mLibLoader = AAudioLoader::getInstance(); |
| int openResult = mLibLoader->open(); |
| return openResult == 0; |
| } |
| |
| // Static method for the error callback. |
| // We use a method so we can access protected methods on the stream. |
| // Launch a thread to handle the error. |
| // That other thread can safely stop, close and delete the stream. |
| void AudioStreamAAudio::internalErrorCallback( |
| AAudioStream *stream, |
| void *userData, |
| aaudio_result_t error) { |
| oboe::Result oboeResult = static_cast<Result>(error); |
| AudioStreamAAudio *oboeStream = reinterpret_cast<AudioStreamAAudio*>(userData); |
| |
| // Coerce the error code if needed to workaround a regression in RQ1A that caused |
| // the wrong code to be passed when headsets plugged in. See b/173928197. |
| if (OboeGlobals::areWorkaroundsEnabled() |
| && getSdkVersion() == __ANDROID_API_R__ |
| && oboeResult == oboe::Result::ErrorTimeout) { |
| oboeResult = oboe::Result::ErrorDisconnected; |
| LOGD("%s() ErrorTimeout changed to ErrorDisconnected to fix b/173928197", __func__); |
| } |
| |
| oboeStream->mErrorCallbackResult = oboeResult; |
| |
| // Prevents deletion of the stream if the app is using AudioStreamBuilder::openStream(shared_ptr) |
| std::shared_ptr<AudioStream> sharedStream = oboeStream->lockWeakThis(); |
| |
| // These checks should be enough because we assume that the stream close() |
| // will join() any active callback threads and will not allow new callbacks. |
| if (oboeStream->wasErrorCallbackCalled()) { // block extra error callbacks |
| LOGE("%s() multiple error callbacks called!", __func__); |
| } else if (stream != oboeStream->getUnderlyingStream()) { |
| LOGW("%s() stream already closed or closing", __func__); // might happen if there are bugs |
| } else if (sharedStream) { |
| // Handle error on a separate thread using shared pointer. |
| std::thread t(oboe_aaudio_error_thread_proc_shared, sharedStream, oboeResult); |
| t.detach(); |
| } else { |
| // Handle error on a separate thread. |
| std::thread t(oboe_aaudio_error_thread_proc, oboeStream, oboeResult); |
| t.detach(); |
| } |
| } |
| |
| void AudioStreamAAudio::beginPerformanceHintInCallback() { |
| if (isPerformanceHintEnabled()) { |
| if (!mAdpfOpenAttempted) { |
| int64_t targetDurationNanos = (mFramesPerBurst * 1e9) / getSampleRate(); |
| // This has to be called from the callback thread so we get the right TID. |
| int adpfResult = mAdpfWrapper.open(gettid(), targetDurationNanos); |
| if (adpfResult < 0) { |
| LOGW("WARNING ADPF not supported, %d\n", adpfResult); |
| } else { |
| LOGD("ADPF is now active\n"); |
| } |
| mAdpfOpenAttempted = true; |
| } |
| mAdpfWrapper.onBeginCallback(); |
| } else if (!isPerformanceHintEnabled() && mAdpfOpenAttempted) { |
| LOGD("ADPF closed\n"); |
| mAdpfWrapper.close(); |
| mAdpfOpenAttempted = false; |
| } |
| } |
| |
| void AudioStreamAAudio::endPerformanceHintInCallback(int32_t numFrames) { |
| if (mAdpfWrapper.isOpen()) { |
| // Scale the measured duration based on numFrames so it is normalized to a full burst. |
| double durationScaler = static_cast<double>(mFramesPerBurst) / numFrames; |
| // Skip this callback if numFrames is very small. |
| // This can happen when buffers wrap around, particularly when doing sample rate conversion. |
| if (durationScaler < 2.0) { |
| mAdpfWrapper.onEndCallback(durationScaler); |
| } |
| } |
| } |
| |
| void AudioStreamAAudio::logUnsupportedAttributes() { |
| int sdkVersion = getSdkVersion(); |
| |
| // These attributes are not supported pre Android "P" |
| if (sdkVersion < __ANDROID_API_P__) { |
| if (mUsage != Usage::Media) { |
| LOGW("Usage [AudioStreamBuilder::setUsage()] " |
| "is not supported on AAudio streams running on pre-Android P versions."); |
| } |
| |
| if (mContentType != ContentType::Music) { |
| LOGW("ContentType [AudioStreamBuilder::setContentType()] " |
| "is not supported on AAudio streams running on pre-Android P versions."); |
| } |
| |
| if (mSessionId != SessionId::None) { |
| LOGW("SessionId [AudioStreamBuilder::setSessionId()] " |
| "is not supported on AAudio streams running on pre-Android P versions."); |
| } |
| } |
| } |
| |
| Result AudioStreamAAudio::open() { |
| Result result = Result::OK; |
| |
| if (mAAudioStream != nullptr) { |
| return Result::ErrorInvalidState; |
| } |
| |
| result = AudioStream::open(); |
| if (result != Result::OK) { |
| return result; |
| } |
| |
| AAudioStreamBuilder *aaudioBuilder; |
| result = static_cast<Result>(mLibLoader->createStreamBuilder(&aaudioBuilder)); |
| if (result != Result::OK) { |
| return result; |
| } |
| |
| // Do not set INPUT capacity below 4096 because that prevents us from getting a FAST track |
| // when using the Legacy data path. |
| // If the app requests > 4096 then we allow it but we are less likely to get LowLatency. |
| // See internal bug b/80308183 for more details. |
| // Fixed in Q but let's still clip the capacity because high input capacity |
| // does not increase latency. |
| int32_t capacity = mBufferCapacityInFrames; |
| constexpr int kCapacityRequiredForFastLegacyTrack = 4096; // matches value in AudioFinger |
| if (OboeGlobals::areWorkaroundsEnabled() |
| && mDirection == oboe::Direction::Input |
| && capacity != oboe::Unspecified |
| && capacity < kCapacityRequiredForFastLegacyTrack |
| && mPerformanceMode == oboe::PerformanceMode::LowLatency) { |
| capacity = kCapacityRequiredForFastLegacyTrack; |
| LOGD("AudioStreamAAudio.open() capacity changed from %d to %d for lower latency", |
| static_cast<int>(mBufferCapacityInFrames), capacity); |
| } |
| mLibLoader->builder_setBufferCapacityInFrames(aaudioBuilder, capacity); |
| |
| if (mLibLoader->builder_setSessionId != nullptr) { |
| mLibLoader->builder_setSessionId(aaudioBuilder, |
| static_cast<aaudio_session_id_t>(mSessionId)); |
| // Output effects do not support PerformanceMode::LowLatency. |
| if (OboeGlobals::areWorkaroundsEnabled() |
| && mSessionId != SessionId::None |
| && mDirection == oboe::Direction::Output |
| && mPerformanceMode == PerformanceMode::LowLatency) { |
| mPerformanceMode = PerformanceMode::None; |
| LOGD("AudioStreamAAudio.open() performance mode changed to None when session " |
| "id is requested"); |
| } |
| } |
| |
| // Channel mask was added in SC_V2. Given the corresponding channel count of selected channel |
| // mask may be different from selected channel count, the last set value will be respected. |
| // If channel count is set after channel mask, the previously set channel mask will be cleared. |
| // If channel mask is set after channel count, the channel count will be automatically |
| // calculated from selected channel mask. In that case, only set channel mask when the API |
| // is available and the channel mask is specified. |
| if (mLibLoader->builder_setChannelMask != nullptr && mChannelMask != ChannelMask::Unspecified) { |
| mLibLoader->builder_setChannelMask(aaudioBuilder, |
| static_cast<aaudio_channel_mask_t>(mChannelMask)); |
| } else { |
| mLibLoader->builder_setChannelCount(aaudioBuilder, mChannelCount); |
| } |
| mLibLoader->builder_setDeviceId(aaudioBuilder, mDeviceId); |
| mLibLoader->builder_setDirection(aaudioBuilder, static_cast<aaudio_direction_t>(mDirection)); |
| mLibLoader->builder_setFormat(aaudioBuilder, static_cast<aaudio_format_t>(mFormat)); |
| mLibLoader->builder_setSampleRate(aaudioBuilder, mSampleRate); |
| mLibLoader->builder_setSharingMode(aaudioBuilder, |
| static_cast<aaudio_sharing_mode_t>(mSharingMode)); |
| mLibLoader->builder_setPerformanceMode(aaudioBuilder, |
| static_cast<aaudio_performance_mode_t>(mPerformanceMode)); |
| |
| // These were added in P so we have to check for the function pointer. |
| if (mLibLoader->builder_setUsage != nullptr) { |
| mLibLoader->builder_setUsage(aaudioBuilder, |
| static_cast<aaudio_usage_t>(mUsage)); |
| } |
| |
| if (mLibLoader->builder_setContentType != nullptr) { |
| mLibLoader->builder_setContentType(aaudioBuilder, |
| static_cast<aaudio_content_type_t>(mContentType)); |
| } |
| |
| if (mLibLoader->builder_setInputPreset != nullptr) { |
| aaudio_input_preset_t inputPreset = mInputPreset; |
| if (getSdkVersion() <= __ANDROID_API_P__ && inputPreset == InputPreset::VoicePerformance) { |
| LOGD("InputPreset::VoicePerformance not supported before Q. Using VoiceRecognition."); |
| inputPreset = InputPreset::VoiceRecognition; // most similar preset |
| } |
| mLibLoader->builder_setInputPreset(aaudioBuilder, |
| static_cast<aaudio_input_preset_t>(inputPreset)); |
| } |
| |
| // These were added in S so we have to check for the function pointer. |
| if (mLibLoader->builder_setPackageName != nullptr && !mPackageName.empty()) { |
| mLibLoader->builder_setPackageName(aaudioBuilder, |
| mPackageName.c_str()); |
| } |
| |
| if (mLibLoader->builder_setAttributionTag != nullptr && !mAttributionTag.empty()) { |
| mLibLoader->builder_setAttributionTag(aaudioBuilder, |
| mAttributionTag.c_str()); |
| } |
| |
| // This was added in Q so we have to check for the function pointer. |
| if (mLibLoader->builder_setAllowedCapturePolicy != nullptr && mDirection == oboe::Direction::Output) { |
| mLibLoader->builder_setAllowedCapturePolicy(aaudioBuilder, |
| static_cast<aaudio_allowed_capture_policy_t>(mAllowedCapturePolicy)); |
| } |
| |
| if (mLibLoader->builder_setPrivacySensitive != nullptr && mDirection == oboe::Direction::Input |
| && mPrivacySensitiveMode != PrivacySensitiveMode::Unspecified) { |
| mLibLoader->builder_setPrivacySensitive(aaudioBuilder, |
| mPrivacySensitiveMode == PrivacySensitiveMode::Enabled); |
| } |
| |
| if (mLibLoader->builder_setIsContentSpatialized != nullptr) { |
| mLibLoader->builder_setIsContentSpatialized(aaudioBuilder, mIsContentSpatialized); |
| } |
| |
| if (mLibLoader->builder_setSpatializationBehavior != nullptr) { |
| // Override Unspecified as Never to reduce latency. |
| if (mSpatializationBehavior == SpatializationBehavior::Unspecified) { |
| mSpatializationBehavior = SpatializationBehavior::Never; |
| } |
| mLibLoader->builder_setSpatializationBehavior(aaudioBuilder, |
| static_cast<aaudio_spatialization_behavior_t>(mSpatializationBehavior)); |
| } else { |
| mSpatializationBehavior = SpatializationBehavior::Never; |
| } |
| |
| if (isDataCallbackSpecified()) { |
| mLibLoader->builder_setDataCallback(aaudioBuilder, oboe_aaudio_data_callback_proc, this); |
| mLibLoader->builder_setFramesPerDataCallback(aaudioBuilder, getFramesPerDataCallback()); |
| |
| if (!isErrorCallbackSpecified()) { |
| // The app did not specify a callback so we should specify |
| // our own so the stream gets closed and stopped. |
| mErrorCallback = &mDefaultErrorCallback; |
| } |
| mLibLoader->builder_setErrorCallback(aaudioBuilder, internalErrorCallback, this); |
| } |
| // Else if the data callback is not being used then the write method will return an error |
| // and the app can stop and close the stream. |
| |
| // ============= OPEN THE STREAM ================ |
| { |
| AAudioStream *stream = nullptr; |
| result = static_cast<Result>(mLibLoader->builder_openStream(aaudioBuilder, &stream)); |
| mAAudioStream.store(stream); |
| } |
| if (result != Result::OK) { |
| // Warn developer because ErrorInternal is not very informative. |
| if (result == Result::ErrorInternal && mDirection == Direction::Input) { |
| LOGW("AudioStreamAAudio.open() may have failed due to lack of " |
| "audio recording permission."); |
| } |
| goto error2; |
| } |
| |
| // Query and cache the stream properties |
| mDeviceId = mLibLoader->stream_getDeviceId(mAAudioStream); |
| mChannelCount = mLibLoader->stream_getChannelCount(mAAudioStream); |
| mSampleRate = mLibLoader->stream_getSampleRate(mAAudioStream); |
| mFormat = static_cast<AudioFormat>(mLibLoader->stream_getFormat(mAAudioStream)); |
| mSharingMode = static_cast<SharingMode>(mLibLoader->stream_getSharingMode(mAAudioStream)); |
| mPerformanceMode = static_cast<PerformanceMode>( |
| mLibLoader->stream_getPerformanceMode(mAAudioStream)); |
| mBufferCapacityInFrames = mLibLoader->stream_getBufferCapacity(mAAudioStream); |
| mBufferSizeInFrames = mLibLoader->stream_getBufferSize(mAAudioStream); |
| mFramesPerBurst = mLibLoader->stream_getFramesPerBurst(mAAudioStream); |
| |
| // These were added in P so we have to check for the function pointer. |
| if (mLibLoader->stream_getUsage != nullptr) { |
| mUsage = static_cast<Usage>(mLibLoader->stream_getUsage(mAAudioStream)); |
| } |
| if (mLibLoader->stream_getContentType != nullptr) { |
| mContentType = static_cast<ContentType>(mLibLoader->stream_getContentType(mAAudioStream)); |
| } |
| if (mLibLoader->stream_getInputPreset != nullptr) { |
| mInputPreset = static_cast<InputPreset>(mLibLoader->stream_getInputPreset(mAAudioStream)); |
| } |
| if (mLibLoader->stream_getSessionId != nullptr) { |
| mSessionId = static_cast<SessionId>(mLibLoader->stream_getSessionId(mAAudioStream)); |
| } else { |
| mSessionId = SessionId::None; |
| } |
| |
| // This was added in Q so we have to check for the function pointer. |
| if (mLibLoader->stream_getAllowedCapturePolicy != nullptr && mDirection == oboe::Direction::Output) { |
| mAllowedCapturePolicy = static_cast<AllowedCapturePolicy>(mLibLoader->stream_getAllowedCapturePolicy(mAAudioStream)); |
| } else { |
| mAllowedCapturePolicy = AllowedCapturePolicy::Unspecified; |
| } |
| |
| if (mLibLoader->stream_isPrivacySensitive != nullptr && mDirection == oboe::Direction::Input) { |
| bool isPrivacySensitive = mLibLoader->stream_isPrivacySensitive(mAAudioStream); |
| mPrivacySensitiveMode = isPrivacySensitive ? PrivacySensitiveMode::Enabled : |
| PrivacySensitiveMode::Disabled; |
| } else { |
| mPrivacySensitiveMode = PrivacySensitiveMode::Unspecified; |
| } |
| |
| if (mLibLoader->stream_getChannelMask != nullptr) { |
| mChannelMask = static_cast<ChannelMask>(mLibLoader->stream_getChannelMask(mAAudioStream)); |
| } |
| |
| if (mLibLoader->stream_isContentSpatialized != nullptr) { |
| mIsContentSpatialized = mLibLoader->stream_isContentSpatialized(mAAudioStream); |
| } |
| |
| if (mLibLoader->stream_getSpatializationBehavior != nullptr) { |
| mSpatializationBehavior = static_cast<SpatializationBehavior>( |
| mLibLoader->stream_getSpatializationBehavior(mAAudioStream)); |
| } |
| |
| if (mLibLoader->stream_getHardwareChannelCount != nullptr) { |
| mHardwareChannelCount = mLibLoader->stream_getHardwareChannelCount(mAAudioStream); |
| } |
| if (mLibLoader->stream_getHardwareSampleRate != nullptr) { |
| mHardwareSampleRate = mLibLoader->stream_getHardwareSampleRate(mAAudioStream); |
| } |
| if (mLibLoader->stream_getHardwareFormat != nullptr) { |
| mHardwareFormat = static_cast<AudioFormat>(mLibLoader->stream_getHardwareFormat(mAAudioStream)); |
| } |
| |
| LOGD("AudioStreamAAudio.open() format=%d, sampleRate=%d, capacity = %d", |
| static_cast<int>(mFormat), static_cast<int>(mSampleRate), |
| static_cast<int>(mBufferCapacityInFrames)); |
| |
| calculateDefaultDelayBeforeCloseMillis(); |
| |
| error2: |
| mLibLoader->builder_delete(aaudioBuilder); |
| if (static_cast<int>(result) > 0) { |
| // Possibly due to b/267531411 |
| LOGW("AudioStreamAAudio.open: AAudioStream_Open() returned positive error = %d", |
| static_cast<int>(result)); |
| if (OboeGlobals::areWorkaroundsEnabled()) { |
| result = Result::ErrorInternal; // Coerce to negative error. |
| } |
| } else { |
| LOGD("AudioStreamAAudio.open: AAudioStream_Open() returned %s = %d", |
| mLibLoader->convertResultToText(static_cast<aaudio_result_t>(result)), |
| static_cast<int>(result)); |
| } |
| return result; |
| } |
| |
| Result AudioStreamAAudio::release() { |
| if (getSdkVersion() < __ANDROID_API_R__) { |
| return Result::ErrorUnimplemented; |
| } |
| |
| // AAudioStream_release() is buggy on Android R. |
| if (OboeGlobals::areWorkaroundsEnabled() && getSdkVersion() == __ANDROID_API_R__) { |
| LOGW("Skipping release() on Android R"); |
| return Result::ErrorUnimplemented; |
| } |
| |
| std::lock_guard<std::mutex> lock(mLock); |
| AAudioStream *stream = mAAudioStream.load(); |
| if (stream != nullptr) { |
| if (OboeGlobals::areWorkaroundsEnabled()) { |
| // Make sure we are really stopped. Do it under mLock |
| // so another thread cannot call requestStart() right before the close. |
| requestStop_l(stream); |
| } |
| return static_cast<Result>(mLibLoader->stream_release(stream)); |
| } else { |
| return Result::ErrorClosed; |
| } |
| } |
| |
| Result AudioStreamAAudio::close() { |
| // Prevent two threads from closing the stream at the same time and crashing. |
| // This could occur, for example, if an application called close() at the same |
| // time that an onError callback was being executed because of a disconnect. |
| std::lock_guard<std::mutex> lock(mLock); |
| |
| AudioStream::close(); |
| |
| AAudioStream *stream = nullptr; |
| { |
| // Wait for any methods using mAAudioStream to finish. |
| std::unique_lock<std::shared_mutex> lock2(mAAudioStreamLock); |
| // Closing will delete *mAAudioStream so we need to null out the pointer atomically. |
| stream = mAAudioStream.exchange(nullptr); |
| } |
| if (stream != nullptr) { |
| if (OboeGlobals::areWorkaroundsEnabled()) { |
| // Make sure we are really stopped. Do it under mLock |
| // so another thread cannot call requestStart() right before the close. |
| requestStop_l(stream); |
| sleepBeforeClose(); |
| } |
| return static_cast<Result>(mLibLoader->stream_close(stream)); |
| } else { |
| return Result::ErrorClosed; |
| } |
| } |
| |
| static void oboe_stop_thread_proc(AudioStream *oboeStream) { |
| if (oboeStream != nullptr) { |
| oboeStream->requestStop(); |
| } |
| } |
| |
| void AudioStreamAAudio::launchStopThread() { |
| // Prevent multiple stop threads from being launched. |
| if (mStopThreadAllowed.exchange(false)) { |
| // Stop this stream on a separate thread |
| std::thread t(oboe_stop_thread_proc, this); |
| t.detach(); |
| } |
| } |
| |
| DataCallbackResult AudioStreamAAudio::callOnAudioReady(AAudioStream * /*stream*/, |
| void *audioData, |
| int32_t numFrames) { |
| DataCallbackResult result = fireDataCallback(audioData, numFrames); |
| if (result == DataCallbackResult::Continue) { |
| return result; |
| } else { |
| if (result == DataCallbackResult::Stop) { |
| LOGD("Oboe callback returned DataCallbackResult::Stop"); |
| } else { |
| LOGE("Oboe callback returned unexpected value. Error: %d", static_cast<int>(result)); |
| } |
| |
| // Returning Stop caused various problems before S. See #1230 |
| if (OboeGlobals::areWorkaroundsEnabled() && getSdkVersion() <= __ANDROID_API_R__) { |
| launchStopThread(); |
| return DataCallbackResult::Continue; |
| } else { |
| return DataCallbackResult::Stop; // OK >= API_S |
| } |
| } |
| } |
| |
| Result AudioStreamAAudio::requestStart() { |
| std::lock_guard<std::mutex> lock(mLock); |
| AAudioStream *stream = mAAudioStream.load(); |
| if (stream != nullptr) { |
| // Avoid state machine errors in O_MR1. |
| if (getSdkVersion() <= __ANDROID_API_O_MR1__) { |
| StreamState state = static_cast<StreamState>(mLibLoader->stream_getState(stream)); |
| if (state == StreamState::Starting || state == StreamState::Started) { |
| // WARNING: On P, AAudio is returning ErrorInvalidState for Output and OK for Input. |
| return Result::OK; |
| } |
| } |
| if (isDataCallbackSpecified()) { |
| setDataCallbackEnabled(true); |
| } |
| mStopThreadAllowed = true; |
| closePerformanceHint(); |
| return static_cast<Result>(mLibLoader->stream_requestStart(stream)); |
| } else { |
| return Result::ErrorClosed; |
| } |
| } |
| |
| Result AudioStreamAAudio::requestPause() { |
| std::lock_guard<std::mutex> lock(mLock); |
| AAudioStream *stream = mAAudioStream.load(); |
| if (stream != nullptr) { |
| // Avoid state machine errors in O_MR1. |
| if (getSdkVersion() <= __ANDROID_API_O_MR1__) { |
| StreamState state = static_cast<StreamState>(mLibLoader->stream_getState(stream)); |
| if (state == StreamState::Pausing || state == StreamState::Paused) { |
| return Result::OK; |
| } |
| } |
| return static_cast<Result>(mLibLoader->stream_requestPause(stream)); |
| } else { |
| return Result::ErrorClosed; |
| } |
| } |
| |
| Result AudioStreamAAudio::requestFlush() { |
| std::lock_guard<std::mutex> lock(mLock); |
| AAudioStream *stream = mAAudioStream.load(); |
| if (stream != nullptr) { |
| // Avoid state machine errors in O_MR1. |
| if (getSdkVersion() <= __ANDROID_API_O_MR1__) { |
| StreamState state = static_cast<StreamState>(mLibLoader->stream_getState(stream)); |
| if (state == StreamState::Flushing || state == StreamState::Flushed) { |
| return Result::OK; |
| } |
| } |
| return static_cast<Result>(mLibLoader->stream_requestFlush(stream)); |
| } else { |
| return Result::ErrorClosed; |
| } |
| } |
| |
| Result AudioStreamAAudio::requestStop() { |
| std::lock_guard<std::mutex> lock(mLock); |
| AAudioStream *stream = mAAudioStream.load(); |
| if (stream != nullptr) { |
| return requestStop_l(stream); |
| } else { |
| return Result::ErrorClosed; |
| } |
| } |
| |
| // Call under mLock |
| Result AudioStreamAAudio::requestStop_l(AAudioStream *stream) { |
| // Avoid state machine errors in O_MR1. |
| if (getSdkVersion() <= __ANDROID_API_O_MR1__) { |
| StreamState state = static_cast<StreamState>(mLibLoader->stream_getState(stream)); |
| if (state == StreamState::Stopping || state == StreamState::Stopped) { |
| return Result::OK; |
| } |
| } |
| return static_cast<Result>(mLibLoader->stream_requestStop(stream)); |
| } |
| |
| ResultWithValue<int32_t> AudioStreamAAudio::write(const void *buffer, |
| int32_t numFrames, |
| int64_t timeoutNanoseconds) { |
| std::shared_lock<std::shared_mutex> lock(mAAudioStreamLock); |
| AAudioStream *stream = mAAudioStream.load(); |
| if (stream != nullptr) { |
| int32_t result = mLibLoader->stream_write(mAAudioStream, buffer, |
| numFrames, timeoutNanoseconds); |
| return ResultWithValue<int32_t>::createBasedOnSign(result); |
| } else { |
| return ResultWithValue<int32_t>(Result::ErrorClosed); |
| } |
| } |
| |
| ResultWithValue<int32_t> AudioStreamAAudio::read(void *buffer, |
| int32_t numFrames, |
| int64_t timeoutNanoseconds) { |
| std::shared_lock<std::shared_mutex> lock(mAAudioStreamLock); |
| AAudioStream *stream = mAAudioStream.load(); |
| if (stream != nullptr) { |
| int32_t result = mLibLoader->stream_read(mAAudioStream, buffer, |
| numFrames, timeoutNanoseconds); |
| return ResultWithValue<int32_t>::createBasedOnSign(result); |
| } else { |
| return ResultWithValue<int32_t>(Result::ErrorClosed); |
| } |
| } |
| |
| |
| // AAudioStream_waitForStateChange() can crash if it is waiting on a stream and that stream |
| // is closed from another thread. We do not want to lock the stream for the duration of the call. |
| // So we call AAudioStream_waitForStateChange() with a timeout of zero so that it will not block. |
| // Then we can do our own sleep with the lock unlocked. |
| Result AudioStreamAAudio::waitForStateChange(StreamState currentState, |
| StreamState *nextState, |
| int64_t timeoutNanoseconds) { |
| Result oboeResult = Result::ErrorTimeout; |
| int64_t sleepTimeNanos = 20 * kNanosPerMillisecond; // arbitrary |
| aaudio_stream_state_t currentAAudioState = static_cast<aaudio_stream_state_t>(currentState); |
| |
| aaudio_result_t result = AAUDIO_OK; |
| int64_t timeLeftNanos = timeoutNanoseconds; |
| |
| mLock.lock(); |
| while (true) { |
| // Do we still have an AAudio stream? If not then stream must have been closed. |
| AAudioStream *stream = mAAudioStream.load(); |
| if (stream == nullptr) { |
| if (nextState != nullptr) { |
| *nextState = StreamState::Closed; |
| } |
| oboeResult = Result::ErrorClosed; |
| break; |
| } |
| |
| // Update and query state change with no blocking. |
| aaudio_stream_state_t aaudioNextState; |
| result = mLibLoader->stream_waitForStateChange( |
| mAAudioStream, |
| currentAAudioState, |
| &aaudioNextState, |
| 0); // timeout=0 for non-blocking |
| // AAudio will return AAUDIO_ERROR_TIMEOUT if timeout=0 and the state does not change. |
| if (result != AAUDIO_OK && result != AAUDIO_ERROR_TIMEOUT) { |
| oboeResult = static_cast<Result>(result); |
| break; |
| } |
| #if OBOE_FIX_FORCE_STARTING_TO_STARTED |
| if (OboeGlobals::areWorkaroundsEnabled() |
| && aaudioNextState == static_cast<aaudio_stream_state_t >(StreamState::Starting)) { |
| aaudioNextState = static_cast<aaudio_stream_state_t >(StreamState::Started); |
| } |
| #endif // OBOE_FIX_FORCE_STARTING_TO_STARTED |
| if (nextState != nullptr) { |
| *nextState = static_cast<StreamState>(aaudioNextState); |
| } |
| if (currentAAudioState != aaudioNextState) { // state changed? |
| oboeResult = Result::OK; |
| break; |
| } |
| |
| // Did we timeout or did user ask for non-blocking? |
| if (timeLeftNanos <= 0) { |
| break; |
| } |
| |
| // No change yet so sleep. |
| mLock.unlock(); // Don't sleep while locked. |
| if (sleepTimeNanos > timeLeftNanos) { |
| sleepTimeNanos = timeLeftNanos; // last little bit |
| } |
| AudioClock::sleepForNanos(sleepTimeNanos); |
| timeLeftNanos -= sleepTimeNanos; |
| mLock.lock(); |
| } |
| |
| mLock.unlock(); |
| return oboeResult; |
| } |
| |
| ResultWithValue<int32_t> AudioStreamAAudio::setBufferSizeInFrames(int32_t requestedFrames) { |
| int32_t adjustedFrames = requestedFrames; |
| if (adjustedFrames > mBufferCapacityInFrames) { |
| adjustedFrames = mBufferCapacityInFrames; |
| } |
| // This calls getBufferSize() so avoid recursive lock. |
| adjustedFrames = QuirksManager::getInstance().clipBufferSize(*this, adjustedFrames); |
| |
| std::shared_lock<std::shared_mutex> lock(mAAudioStreamLock); |
| AAudioStream *stream = mAAudioStream.load(); |
| if (stream != nullptr) { |
| int32_t newBufferSize = mLibLoader->stream_setBufferSize(mAAudioStream, adjustedFrames); |
| // Cache the result if it's valid |
| if (newBufferSize > 0) mBufferSizeInFrames = newBufferSize; |
| return ResultWithValue<int32_t>::createBasedOnSign(newBufferSize); |
| } else { |
| return ResultWithValue<int32_t>(Result::ErrorClosed); |
| } |
| } |
| |
| StreamState AudioStreamAAudio::getState() { |
| std::shared_lock<std::shared_mutex> lock(mAAudioStreamLock); |
| AAudioStream *stream = mAAudioStream.load(); |
| if (stream != nullptr) { |
| aaudio_stream_state_t aaudioState = mLibLoader->stream_getState(stream); |
| #if OBOE_FIX_FORCE_STARTING_TO_STARTED |
| if (OboeGlobals::areWorkaroundsEnabled() |
| && aaudioState == AAUDIO_STREAM_STATE_STARTING) { |
| aaudioState = AAUDIO_STREAM_STATE_STARTED; |
| } |
| #endif // OBOE_FIX_FORCE_STARTING_TO_STARTED |
| return static_cast<StreamState>(aaudioState); |
| } else { |
| return StreamState::Closed; |
| } |
| } |
| |
| int32_t AudioStreamAAudio::getBufferSizeInFrames() { |
| std::shared_lock<std::shared_mutex> lock(mAAudioStreamLock); |
| AAudioStream *stream = mAAudioStream.load(); |
| if (stream != nullptr) { |
| mBufferSizeInFrames = mLibLoader->stream_getBufferSize(stream); |
| } |
| return mBufferSizeInFrames; |
| } |
| |
| void AudioStreamAAudio::updateFramesRead() { |
| std::shared_lock<std::shared_mutex> lock(mAAudioStreamLock); |
| AAudioStream *stream = mAAudioStream.load(); |
| // Set to 1 for debugging race condition #1180 with mAAudioStream. |
| // See also DEBUG_CLOSE_RACE in OboeTester. |
| // This was left in the code so that we could test the fix again easily in the future. |
| // We could not trigger the race condition without adding these get calls and the sleeps. |
| #define DEBUG_CLOSE_RACE 0 |
| #if DEBUG_CLOSE_RACE |
| // This is used when testing race conditions with close(). |
| // See DEBUG_CLOSE_RACE in OboeTester |
| AudioClock::sleepForNanos(400 * kNanosPerMillisecond); |
| #endif // DEBUG_CLOSE_RACE |
| if (stream != nullptr) { |
| mFramesRead = mLibLoader->stream_getFramesRead(stream); |
| } |
| } |
| |
| void AudioStreamAAudio::updateFramesWritten() { |
| std::shared_lock<std::shared_mutex> lock(mAAudioStreamLock); |
| AAudioStream *stream = mAAudioStream.load(); |
| if (stream != nullptr) { |
| mFramesWritten = mLibLoader->stream_getFramesWritten(stream); |
| } |
| } |
| |
| ResultWithValue<int32_t> AudioStreamAAudio::getXRunCount() { |
| std::shared_lock<std::shared_mutex> lock(mAAudioStreamLock); |
| AAudioStream *stream = mAAudioStream.load(); |
| if (stream != nullptr) { |
| return ResultWithValue<int32_t>::createBasedOnSign(mLibLoader->stream_getXRunCount(stream)); |
| } else { |
| return ResultWithValue<int32_t>(Result::ErrorNull); |
| } |
| } |
| |
| Result AudioStreamAAudio::getTimestamp(clockid_t clockId, |
| int64_t *framePosition, |
| int64_t *timeNanoseconds) { |
| if (getState() != StreamState::Started) { |
| return Result::ErrorInvalidState; |
| } |
| std::shared_lock<std::shared_mutex> lock(mAAudioStreamLock); |
| AAudioStream *stream = mAAudioStream.load(); |
| if (stream != nullptr) { |
| return static_cast<Result>(mLibLoader->stream_getTimestamp(stream, clockId, |
| framePosition, timeNanoseconds)); |
| } else { |
| return Result::ErrorNull; |
| } |
| } |
| |
| ResultWithValue<double> AudioStreamAAudio::calculateLatencyMillis() { |
| // Get the time that a known audio frame was presented. |
| int64_t hardwareFrameIndex; |
| int64_t hardwareFrameHardwareTime; |
| auto result = getTimestamp(CLOCK_MONOTONIC, |
| &hardwareFrameIndex, |
| &hardwareFrameHardwareTime); |
| if (result != oboe::Result::OK) { |
| return ResultWithValue<double>(static_cast<Result>(result)); |
| } |
| |
| // Get counter closest to the app. |
| bool isOutput = (getDirection() == oboe::Direction::Output); |
| int64_t appFrameIndex = isOutput ? getFramesWritten() : getFramesRead(); |
| |
| // Assume that the next frame will be processed at the current time |
| using namespace std::chrono; |
| int64_t appFrameAppTime = |
| duration_cast<nanoseconds>(steady_clock::now().time_since_epoch()).count(); |
| |
| // Calculate the number of frames between app and hardware |
| int64_t frameIndexDelta = appFrameIndex - hardwareFrameIndex; |
| |
| // Calculate the time which the next frame will be or was presented |
| int64_t frameTimeDelta = (frameIndexDelta * oboe::kNanosPerSecond) / getSampleRate(); |
| int64_t appFrameHardwareTime = hardwareFrameHardwareTime + frameTimeDelta; |
| |
| // The current latency is the difference in time between when the current frame is at |
| // the app and when it is at the hardware. |
| double latencyNanos = static_cast<double>(isOutput |
| ? (appFrameHardwareTime - appFrameAppTime) // hardware is later |
| : (appFrameAppTime - appFrameHardwareTime)); // hardware is earlier |
| double latencyMillis = latencyNanos / kNanosPerMillisecond; |
| |
| return ResultWithValue<double>(latencyMillis); |
| } |
| |
| bool AudioStreamAAudio::isMMapUsed() { |
| std::shared_lock<std::shared_mutex> lock(mAAudioStreamLock); |
| AAudioStream *stream = mAAudioStream.load(); |
| if (stream != nullptr) { |
| return AAudioExtensions::getInstance().isMMapUsed(stream); |
| } else { |
| return false; |
| } |
| } |
| |
| } // namespace oboe |