| /* |
| * Copyright (C) 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. |
| */ |
| |
| // This file is used in both client and server processes. |
| // This is needed to make sense of the logs more easily. |
| #define LOG_TAG (mInService ? "AAudioService" : "AAudio") |
| //#define LOG_NDEBUG 0 |
| #include <utils/Log.h> |
| |
| #define ATRACE_TAG ATRACE_TAG_AUDIO |
| |
| #include <stdint.h> |
| #include <assert.h> |
| |
| #include <binder/IServiceManager.h> |
| |
| #include <aaudio/AAudio.h> |
| #include <utils/String16.h> |
| #include <utils/Trace.h> |
| |
| #include "AudioClock.h" |
| #include "AudioEndpointParcelable.h" |
| #include "binding/AAudioStreamRequest.h" |
| #include "binding/AAudioStreamConfiguration.h" |
| #include "binding/IAAudioService.h" |
| #include "binding/AAudioServiceMessage.h" |
| #include "core/AudioStreamBuilder.h" |
| #include "fifo/FifoBuffer.h" |
| #include "utility/LinearRamp.h" |
| |
| #include "AudioStreamInternal.h" |
| |
| using android::String16; |
| using android::Mutex; |
| using android::WrappingBuffer; |
| |
| using namespace aaudio; |
| |
| #define MIN_TIMEOUT_NANOS (1000 * AAUDIO_NANOS_PER_MILLISECOND) |
| |
| // Wait at least this many times longer than the operation should take. |
| #define MIN_TIMEOUT_OPERATIONS 4 |
| |
| #define LOG_TIMESTAMPS 0 |
| |
| AudioStreamInternal::AudioStreamInternal(AAudioServiceInterface &serviceInterface, bool inService) |
| : AudioStream() |
| , mClockModel() |
| , mAudioEndpoint() |
| , mServiceStreamHandle(AAUDIO_HANDLE_INVALID) |
| , mFramesPerBurst(16) |
| , mServiceInterface(serviceInterface) |
| , mInService(inService) { |
| } |
| |
| AudioStreamInternal::~AudioStreamInternal() { |
| } |
| |
| aaudio_result_t AudioStreamInternal::open(const AudioStreamBuilder &builder) { |
| |
| aaudio_result_t result = AAUDIO_OK; |
| AAudioStreamRequest request; |
| AAudioStreamConfiguration configuration; |
| |
| result = AudioStream::open(builder); |
| if (result < 0) { |
| return result; |
| } |
| |
| // We have to do volume scaling. So we prefer FLOAT format. |
| if (getFormat() == AAUDIO_FORMAT_UNSPECIFIED) { |
| setFormat(AAUDIO_FORMAT_PCM_FLOAT); |
| } |
| // Request FLOAT for the shared mixer. |
| request.getConfiguration().setAudioFormat(AAUDIO_FORMAT_PCM_FLOAT); |
| |
| // Build the request to send to the server. |
| request.setUserId(getuid()); |
| request.setProcessId(getpid()); |
| request.setDirection(getDirection()); |
| request.setSharingModeMatchRequired(isSharingModeMatchRequired()); |
| |
| request.getConfiguration().setDeviceId(getDeviceId()); |
| request.getConfiguration().setSampleRate(getSampleRate()); |
| request.getConfiguration().setSamplesPerFrame(getSamplesPerFrame()); |
| request.getConfiguration().setSharingMode(getSharingMode()); |
| |
| request.getConfiguration().setBufferCapacity(builder.getBufferCapacity()); |
| |
| mServiceStreamHandle = mServiceInterface.openStream(request, configuration); |
| if (mServiceStreamHandle < 0) { |
| result = mServiceStreamHandle; |
| ALOGE("AudioStreamInternal.open(): openStream() returned %d", result); |
| } else { |
| result = configuration.validate(); |
| if (result != AAUDIO_OK) { |
| close(); |
| return result; |
| } |
| // Save results of the open. |
| setSampleRate(configuration.getSampleRate()); |
| setSamplesPerFrame(configuration.getSamplesPerFrame()); |
| setDeviceId(configuration.getDeviceId()); |
| |
| // Save device format so we can do format conversion and volume scaling together. |
| mDeviceFormat = configuration.getAudioFormat(); |
| |
| result = mServiceInterface.getStreamDescription(mServiceStreamHandle, mEndPointParcelable); |
| if (result != AAUDIO_OK) { |
| mServiceInterface.closeStream(mServiceStreamHandle); |
| return result; |
| } |
| |
| // resolve parcelable into a descriptor |
| result = mEndPointParcelable.resolve(&mEndpointDescriptor); |
| if (result != AAUDIO_OK) { |
| mServiceInterface.closeStream(mServiceStreamHandle); |
| return result; |
| } |
| |
| // Configure endpoint based on descriptor. |
| mAudioEndpoint.configure(&mEndpointDescriptor); |
| |
| mFramesPerBurst = mEndpointDescriptor.dataQueueDescriptor.framesPerBurst; |
| int32_t capacity = mEndpointDescriptor.dataQueueDescriptor.capacityInFrames; |
| |
| // Validate result from server. |
| if (mFramesPerBurst < 16 || mFramesPerBurst > 16 * 1024) { |
| ALOGE("AudioStream::open(): framesPerBurst out of range = %d", mFramesPerBurst); |
| return AAUDIO_ERROR_OUT_OF_RANGE; |
| } |
| if (capacity < mFramesPerBurst || capacity > 32 * 1024) { |
| ALOGE("AudioStream::open(): bufferCapacity out of range = %d", capacity); |
| return AAUDIO_ERROR_OUT_OF_RANGE; |
| } |
| |
| mClockModel.setSampleRate(getSampleRate()); |
| mClockModel.setFramesPerBurst(mFramesPerBurst); |
| |
| if (getDataCallbackProc()) { |
| mCallbackFrames = builder.getFramesPerDataCallback(); |
| if (mCallbackFrames > getBufferCapacity() / 2) { |
| ALOGE("AudioStreamInternal.open(): framesPerCallback too large = %d, capacity = %d", |
| mCallbackFrames, getBufferCapacity()); |
| mServiceInterface.closeStream(mServiceStreamHandle); |
| return AAUDIO_ERROR_OUT_OF_RANGE; |
| |
| } else if (mCallbackFrames < 0) { |
| ALOGE("AudioStreamInternal.open(): framesPerCallback negative"); |
| mServiceInterface.closeStream(mServiceStreamHandle); |
| return AAUDIO_ERROR_OUT_OF_RANGE; |
| |
| } |
| if (mCallbackFrames == AAUDIO_UNSPECIFIED) { |
| mCallbackFrames = mFramesPerBurst; |
| } |
| |
| int32_t bytesPerFrame = getSamplesPerFrame() |
| * AAudioConvert_formatToSizeInBytes(getFormat()); |
| int32_t callbackBufferSize = mCallbackFrames * bytesPerFrame; |
| mCallbackBuffer = new uint8_t[callbackBufferSize]; |
| } |
| |
| setState(AAUDIO_STREAM_STATE_OPEN); |
| } |
| return result; |
| } |
| |
| aaudio_result_t AudioStreamInternal::close() { |
| ALOGD("AudioStreamInternal.close(): mServiceStreamHandle = 0x%08X", |
| mServiceStreamHandle); |
| if (mServiceStreamHandle != AAUDIO_HANDLE_INVALID) { |
| // Don't close a stream while it is running. |
| aaudio_stream_state_t currentState = getState(); |
| if (isActive()) { |
| requestStop(); |
| aaudio_stream_state_t nextState; |
| int64_t timeoutNanoseconds = MIN_TIMEOUT_NANOS; |
| aaudio_result_t result = waitForStateChange(currentState, &nextState, |
| timeoutNanoseconds); |
| if (result != AAUDIO_OK) { |
| ALOGE("AudioStreamInternal::close() waitForStateChange() returned %d %s", |
| result, AAudio_convertResultToText(result)); |
| } |
| } |
| aaudio_handle_t serviceStreamHandle = mServiceStreamHandle; |
| mServiceStreamHandle = AAUDIO_HANDLE_INVALID; |
| |
| mServiceInterface.closeStream(serviceStreamHandle); |
| delete[] mCallbackBuffer; |
| mCallbackBuffer = nullptr; |
| return mEndPointParcelable.close(); |
| } else { |
| return AAUDIO_ERROR_INVALID_HANDLE; |
| } |
| } |
| |
| |
| static void *aaudio_callback_thread_proc(void *context) |
| { |
| AudioStreamInternal *stream = (AudioStreamInternal *)context; |
| //LOGD("AudioStreamInternal(): oboe_callback_thread, stream = %p", stream); |
| if (stream != NULL) { |
| return stream->callbackLoop(); |
| } else { |
| return NULL; |
| } |
| } |
| |
| aaudio_result_t AudioStreamInternal::requestStart() |
| { |
| int64_t startTime; |
| ALOGD("AudioStreamInternal(): start()"); |
| if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) { |
| return AAUDIO_ERROR_INVALID_STATE; |
| } |
| |
| startTime = AudioClock::getNanoseconds(); |
| mClockModel.start(startTime); |
| setState(AAUDIO_STREAM_STATE_STARTING); |
| aaudio_result_t result = mServiceInterface.startStream(mServiceStreamHandle);; |
| |
| if (result == AAUDIO_OK && getDataCallbackProc() != nullptr) { |
| // Launch the callback loop thread. |
| int64_t periodNanos = mCallbackFrames |
| * AAUDIO_NANOS_PER_SECOND |
| / getSampleRate(); |
| mCallbackEnabled.store(true); |
| result = createThread(periodNanos, aaudio_callback_thread_proc, this); |
| } |
| return result; |
| } |
| |
| int64_t AudioStreamInternal::calculateReasonableTimeout(int32_t framesPerOperation) { |
| |
| // Wait for at least a second or some number of callbacks to join the thread. |
| int64_t timeoutNanoseconds = (MIN_TIMEOUT_OPERATIONS |
| * framesPerOperation |
| * AAUDIO_NANOS_PER_SECOND) |
| / getSampleRate(); |
| if (timeoutNanoseconds < MIN_TIMEOUT_NANOS) { // arbitrary number of seconds |
| timeoutNanoseconds = MIN_TIMEOUT_NANOS; |
| } |
| return timeoutNanoseconds; |
| } |
| |
| int64_t AudioStreamInternal::calculateReasonableTimeout() { |
| return calculateReasonableTimeout(getFramesPerBurst()); |
| } |
| |
| aaudio_result_t AudioStreamInternal::stopCallback() |
| { |
| if (isDataCallbackActive()) { |
| mCallbackEnabled.store(false); |
| return joinThread(NULL); |
| } else { |
| return AAUDIO_OK; |
| } |
| } |
| |
| aaudio_result_t AudioStreamInternal::requestPauseInternal() |
| { |
| if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) { |
| ALOGE("AudioStreamInternal(): requestPauseInternal() mServiceStreamHandle invalid = 0x%08X", |
| mServiceStreamHandle); |
| return AAUDIO_ERROR_INVALID_STATE; |
| } |
| |
| mClockModel.stop(AudioClock::getNanoseconds()); |
| setState(AAUDIO_STREAM_STATE_PAUSING); |
| return mServiceInterface.pauseStream(mServiceStreamHandle); |
| } |
| |
| aaudio_result_t AudioStreamInternal::requestPause() |
| { |
| aaudio_result_t result = stopCallback(); |
| if (result != AAUDIO_OK) { |
| return result; |
| } |
| result = requestPauseInternal(); |
| return result; |
| } |
| |
| aaudio_result_t AudioStreamInternal::requestFlush() { |
| if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) { |
| ALOGE("AudioStreamInternal(): requestFlush() mServiceStreamHandle invalid = 0x%08X", |
| mServiceStreamHandle); |
| return AAUDIO_ERROR_INVALID_STATE; |
| } |
| |
| setState(AAUDIO_STREAM_STATE_FLUSHING); |
| return mServiceInterface.flushStream(mServiceStreamHandle); |
| } |
| |
| // TODO for Play only |
| void AudioStreamInternal::onFlushFromServer() { |
| ALOGD("AudioStreamInternal(): onFlushFromServer()"); |
| int64_t readCounter = mAudioEndpoint.getDataReadCounter(); |
| int64_t writeCounter = mAudioEndpoint.getDataWriteCounter(); |
| |
| // Bump offset so caller does not see the retrograde motion in getFramesRead(). |
| int64_t framesFlushed = writeCounter - readCounter; |
| mFramesOffsetFromService += framesFlushed; |
| |
| // Flush written frames by forcing writeCounter to readCounter. |
| // This is because we cannot move the read counter in the hardware. |
| mAudioEndpoint.setDataWriteCounter(readCounter); |
| } |
| |
| aaudio_result_t AudioStreamInternal::requestStopInternal() |
| { |
| if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) { |
| ALOGE("AudioStreamInternal(): requestStopInternal() mServiceStreamHandle invalid = 0x%08X", |
| mServiceStreamHandle); |
| return AAUDIO_ERROR_INVALID_STATE; |
| } |
| |
| mClockModel.stop(AudioClock::getNanoseconds()); |
| setState(AAUDIO_STREAM_STATE_STOPPING); |
| return mServiceInterface.stopStream(mServiceStreamHandle); |
| } |
| |
| aaudio_result_t AudioStreamInternal::requestStop() |
| { |
| aaudio_result_t result = stopCallback(); |
| if (result != AAUDIO_OK) { |
| return result; |
| } |
| result = requestStopInternal(); |
| return result; |
| } |
| |
| aaudio_result_t AudioStreamInternal::registerThread() { |
| if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) { |
| return AAUDIO_ERROR_INVALID_STATE; |
| } |
| return mServiceInterface.registerAudioThread(mServiceStreamHandle, |
| getpid(), |
| gettid(), |
| getPeriodNanoseconds()); |
| } |
| |
| aaudio_result_t AudioStreamInternal::unregisterThread() { |
| if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) { |
| return AAUDIO_ERROR_INVALID_STATE; |
| } |
| return mServiceInterface.unregisterAudioThread(mServiceStreamHandle, getpid(), gettid()); |
| } |
| |
| aaudio_result_t AudioStreamInternal::getTimestamp(clockid_t clockId, |
| int64_t *framePosition, |
| int64_t *timeNanoseconds) { |
| // TODO Generate in server and pass to client. Return latest. |
| int64_t time = AudioClock::getNanoseconds(); |
| *framePosition = mClockModel.convertTimeToPosition(time); |
| // TODO Get a more accurate timestamp from the service. This code just adds a fudge factor. |
| *timeNanoseconds = time + (6 * AAUDIO_NANOS_PER_MILLISECOND); |
| return AAUDIO_OK; |
| } |
| |
| aaudio_result_t AudioStreamInternal::updateStateWhileWaiting() { |
| if (isDataCallbackActive()) { |
| return AAUDIO_OK; // state is getting updated by the callback thread read/write call |
| } |
| return processCommands(); |
| } |
| |
| #if LOG_TIMESTAMPS |
| static void AudioStreamInternal_logTimestamp(AAudioServiceMessage &command) { |
| static int64_t oldPosition = 0; |
| static int64_t oldTime = 0; |
| int64_t framePosition = command.timestamp.position; |
| int64_t nanoTime = command.timestamp.timestamp; |
| ALOGD("AudioStreamInternal() timestamp says framePosition = %08lld at nanoTime %lld", |
| (long long) framePosition, |
| (long long) nanoTime); |
| int64_t nanosDelta = nanoTime - oldTime; |
| if (nanosDelta > 0 && oldTime > 0) { |
| int64_t framesDelta = framePosition - oldPosition; |
| int64_t rate = (framesDelta * AAUDIO_NANOS_PER_SECOND) / nanosDelta; |
| ALOGD("AudioStreamInternal() - framesDelta = %08lld", (long long) framesDelta); |
| ALOGD("AudioStreamInternal() - nanosDelta = %08lld", (long long) nanosDelta); |
| ALOGD("AudioStreamInternal() - measured rate = %lld", (long long) rate); |
| } |
| oldPosition = framePosition; |
| oldTime = nanoTime; |
| } |
| #endif |
| |
| aaudio_result_t AudioStreamInternal::onTimestampFromServer(AAudioServiceMessage *message) { |
| #if LOG_TIMESTAMPS |
| AudioStreamInternal_logTimestamp(*message); |
| #endif |
| processTimestamp(message->timestamp.position, message->timestamp.timestamp); |
| return AAUDIO_OK; |
| } |
| |
| aaudio_result_t AudioStreamInternal::onEventFromServer(AAudioServiceMessage *message) { |
| aaudio_result_t result = AAUDIO_OK; |
| switch (message->event.event) { |
| case AAUDIO_SERVICE_EVENT_STARTED: |
| ALOGD("processCommands() got AAUDIO_SERVICE_EVENT_STARTED"); |
| if (getState() == AAUDIO_STREAM_STATE_STARTING) { |
| setState(AAUDIO_STREAM_STATE_STARTED); |
| } |
| break; |
| case AAUDIO_SERVICE_EVENT_PAUSED: |
| ALOGD("processCommands() got AAUDIO_SERVICE_EVENT_PAUSED"); |
| if (getState() == AAUDIO_STREAM_STATE_PAUSING) { |
| setState(AAUDIO_STREAM_STATE_PAUSED); |
| } |
| break; |
| case AAUDIO_SERVICE_EVENT_STOPPED: |
| ALOGD("processCommands() got AAUDIO_SERVICE_EVENT_STOPPED"); |
| if (getState() == AAUDIO_STREAM_STATE_STOPPING) { |
| setState(AAUDIO_STREAM_STATE_STOPPED); |
| } |
| break; |
| case AAUDIO_SERVICE_EVENT_FLUSHED: |
| ALOGD("processCommands() got AAUDIO_SERVICE_EVENT_FLUSHED"); |
| if (getState() == AAUDIO_STREAM_STATE_FLUSHING) { |
| setState(AAUDIO_STREAM_STATE_FLUSHED); |
| onFlushFromServer(); |
| } |
| break; |
| case AAUDIO_SERVICE_EVENT_CLOSED: |
| ALOGD("processCommands() got AAUDIO_SERVICE_EVENT_CLOSED"); |
| setState(AAUDIO_STREAM_STATE_CLOSED); |
| break; |
| case AAUDIO_SERVICE_EVENT_DISCONNECTED: |
| result = AAUDIO_ERROR_DISCONNECTED; |
| setState(AAUDIO_STREAM_STATE_DISCONNECTED); |
| ALOGW("WARNING - processCommands() AAUDIO_SERVICE_EVENT_DISCONNECTED"); |
| break; |
| case AAUDIO_SERVICE_EVENT_VOLUME: |
| mVolumeRamp.setTarget((float) message->event.dataDouble); |
| ALOGD("processCommands() AAUDIO_SERVICE_EVENT_VOLUME %lf", |
| message->event.dataDouble); |
| break; |
| default: |
| ALOGW("WARNING - processCommands() Unrecognized event = %d", |
| (int) message->event.event); |
| break; |
| } |
| return result; |
| } |
| |
| // Process all the commands coming from the server. |
| aaudio_result_t AudioStreamInternal::processCommands() { |
| aaudio_result_t result = AAUDIO_OK; |
| |
| while (result == AAUDIO_OK) { |
| //ALOGD("AudioStreamInternal::processCommands() - looping, %d", result); |
| AAudioServiceMessage message; |
| if (mAudioEndpoint.readUpCommand(&message) != 1) { |
| break; // no command this time, no problem |
| } |
| switch (message.what) { |
| case AAudioServiceMessage::code::TIMESTAMP: |
| result = onTimestampFromServer(&message); |
| break; |
| |
| case AAudioServiceMessage::code::EVENT: |
| result = onEventFromServer(&message); |
| break; |
| |
| default: |
| ALOGE("WARNING - AudioStreamInternal::processCommands() Unrecognized what = %d", |
| (int) message.what); |
| result = AAUDIO_ERROR_INTERNAL; |
| break; |
| } |
| } |
| return result; |
| } |
| |
| // Read or write the data, block if needed and timeoutMillis > 0 |
| aaudio_result_t AudioStreamInternal::processData(void *buffer, int32_t numFrames, |
| int64_t timeoutNanoseconds) |
| { |
| const char * traceName = (mInService) ? "aaWrtS" : "aaWrtC"; |
| ATRACE_BEGIN(traceName); |
| aaudio_result_t result = AAUDIO_OK; |
| int32_t loopCount = 0; |
| uint8_t* audioData = (uint8_t*)buffer; |
| int64_t currentTimeNanos = AudioClock::getNanoseconds(); |
| int64_t deadlineNanos = currentTimeNanos + timeoutNanoseconds; |
| int32_t framesLeft = numFrames; |
| |
| int32_t fullFrames = mAudioEndpoint.getFullFramesAvailable(); |
| if (ATRACE_ENABLED()) { |
| const char * traceName = (mInService) ? "aaFullS" : "aaFullC"; |
| ATRACE_INT(traceName, fullFrames); |
| } |
| |
| // Loop until all the data has been processed or until a timeout occurs. |
| while (framesLeft > 0) { |
| // The call to processDataNow() will not block. It will just read as much as it can. |
| int64_t wakeTimeNanos = 0; |
| aaudio_result_t framesProcessed = processDataNow(audioData, framesLeft, |
| currentTimeNanos, &wakeTimeNanos); |
| if (framesProcessed < 0) { |
| ALOGE("AudioStreamInternal::processData() loop: framesProcessed = %d", framesProcessed); |
| result = framesProcessed; |
| break; |
| } |
| framesLeft -= (int32_t) framesProcessed; |
| audioData += framesProcessed * getBytesPerFrame(); |
| |
| // Should we block? |
| if (timeoutNanoseconds == 0) { |
| break; // don't block |
| } else if (framesLeft > 0) { |
| // clip the wake time to something reasonable |
| if (wakeTimeNanos < currentTimeNanos) { |
| wakeTimeNanos = currentTimeNanos; |
| } |
| if (wakeTimeNanos > deadlineNanos) { |
| // If we time out, just return the framesWritten so far. |
| // TODO remove after we fix the deadline bug |
| ALOGE("AudioStreamInternal::processData(): timed out after %lld nanos", |
| (long long) timeoutNanoseconds); |
| ALOGE("AudioStreamInternal::processData(): wakeTime = %lld, deadline = %lld nanos", |
| (long long) wakeTimeNanos, (long long) deadlineNanos); |
| ALOGE("AudioStreamInternal::processData(): past deadline by %d micros", |
| (int)((wakeTimeNanos - deadlineNanos) / AAUDIO_NANOS_PER_MICROSECOND)); |
| break; |
| } |
| |
| int64_t sleepForNanos = wakeTimeNanos - currentTimeNanos; |
| AudioClock::sleepForNanos(sleepForNanos); |
| currentTimeNanos = AudioClock::getNanoseconds(); |
| } |
| } |
| |
| // return error or framesProcessed |
| (void) loopCount; |
| ATRACE_END(); |
| return (result < 0) ? result : numFrames - framesLeft; |
| } |
| |
| void AudioStreamInternal::processTimestamp(uint64_t position, int64_t time) { |
| mClockModel.processTimestamp(position, time); |
| } |
| |
| aaudio_result_t AudioStreamInternal::setBufferSize(int32_t requestedFrames) { |
| int32_t actualFrames = 0; |
| // Round to the next highest burst size. |
| if (getFramesPerBurst() > 0) { |
| int32_t numBursts = (requestedFrames + getFramesPerBurst() - 1) / getFramesPerBurst(); |
| requestedFrames = numBursts * getFramesPerBurst(); |
| } |
| |
| aaudio_result_t result = mAudioEndpoint.setBufferSizeInFrames(requestedFrames, &actualFrames); |
| ALOGD("AudioStreamInternal::setBufferSize() req = %d => %d", requestedFrames, actualFrames); |
| if (result < 0) { |
| return result; |
| } else { |
| return (aaudio_result_t) actualFrames; |
| } |
| } |
| |
| int32_t AudioStreamInternal::getBufferSize() const { |
| return mAudioEndpoint.getBufferSizeInFrames(); |
| } |
| |
| int32_t AudioStreamInternal::getBufferCapacity() const { |
| return mAudioEndpoint.getBufferCapacityInFrames(); |
| } |
| |
| int32_t AudioStreamInternal::getFramesPerBurst() const { |
| return mEndpointDescriptor.dataQueueDescriptor.framesPerBurst; |
| } |
| |
| aaudio_result_t AudioStreamInternal::joinThread(void** returnArg) { |
| return AudioStream::joinThread(returnArg, calculateReasonableTimeout(getFramesPerBurst())); |
| } |
