| /* |
| * 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 ? "AudioStreamInternal_Service" : "AudioStreamInternal_Client") |
| //#define LOG_NDEBUG 0 |
| #include <utils/Log.h> |
| |
| #define ATRACE_TAG ATRACE_TAG_AUDIO |
| |
| #include <stdint.h> |
| |
| #include <binder/IServiceManager.h> |
| |
| #include <aaudio/AAudio.h> |
| #include <cutils/properties.h> |
| #include <utils/String16.h> |
| #include <utils/Trace.h> |
| |
| #include "AudioEndpointParcelable.h" |
| #include "binding/AAudioStreamRequest.h" |
| #include "binding/AAudioStreamConfiguration.h" |
| #include "binding/IAAudioService.h" |
| #include "binding/AAudioServiceMessage.h" |
| #include "core/AudioGlobal.h" |
| #include "core/AudioStreamBuilder.h" |
| #include "fifo/FifoBuffer.h" |
| #include "utility/AudioClock.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) |
| , mInService(inService) |
| , mServiceInterface(serviceInterface) |
| , mAtomicInternalTimestamp() |
| , mWakeupDelayNanos(AAudioProperty_getWakeupDelayMicros() * AAUDIO_NANOS_PER_MICROSECOND) |
| , mMinimumSleepNanos(AAudioProperty_getMinimumSleepMicros() * AAUDIO_NANOS_PER_MICROSECOND) |
| { |
| } |
| |
| AudioStreamInternal::~AudioStreamInternal() { |
| } |
| |
| aaudio_result_t AudioStreamInternal::open(const AudioStreamBuilder &builder) { |
| |
| aaudio_result_t result = AAUDIO_OK; |
| int32_t capacity; |
| int32_t framesPerBurst; |
| int32_t framesPerHardwareBurst; |
| AAudioStreamRequest request; |
| AAudioStreamConfiguration configurationOutput; |
| |
| if (getState() != AAUDIO_STREAM_STATE_UNINITIALIZED) { |
| ALOGE("%s - already open! state = %d", __func__, getState()); |
| return AAUDIO_ERROR_INVALID_STATE; |
| } |
| |
| // Copy requested parameters to the stream. |
| result = AudioStream::open(builder); |
| if (result < 0) { |
| return result; |
| } |
| |
| const int32_t burstMinMicros = AAudioProperty_getHardwareBurstMinMicros(); |
| int32_t burstMicros = 0; |
| |
| // We have to do volume scaling. So we prefer FLOAT format. |
| if (getFormat() == AUDIO_FORMAT_DEFAULT) { |
| setFormat(AUDIO_FORMAT_PCM_FLOAT); |
| } |
| // Request FLOAT for the shared mixer. |
| request.getConfiguration().setFormat(AUDIO_FORMAT_PCM_FLOAT); |
| |
| // Build the request to send to the server. |
| request.setUserId(getuid()); |
| request.setProcessId(getpid()); |
| request.setSharingModeMatchRequired(isSharingModeMatchRequired()); |
| request.setInService(isInService()); |
| |
| request.getConfiguration().setDeviceId(getDeviceId()); |
| request.getConfiguration().setSampleRate(getSampleRate()); |
| request.getConfiguration().setSamplesPerFrame(getSamplesPerFrame()); |
| request.getConfiguration().setDirection(getDirection()); |
| request.getConfiguration().setSharingMode(getSharingMode()); |
| |
| request.getConfiguration().setUsage(getUsage()); |
| request.getConfiguration().setContentType(getContentType()); |
| request.getConfiguration().setInputPreset(getInputPreset()); |
| |
| request.getConfiguration().setBufferCapacity(builder.getBufferCapacity()); |
| |
| mDeviceChannelCount = getSamplesPerFrame(); // Assume it will be the same. Update if not. |
| |
| mServiceStreamHandle = mServiceInterface.openStream(request, configurationOutput); |
| if (mServiceStreamHandle < 0 |
| && request.getConfiguration().getSamplesPerFrame() == 1 // mono? |
| && getDirection() == AAUDIO_DIRECTION_OUTPUT |
| && !isInService()) { |
| // if that failed then try switching from mono to stereo if OUTPUT. |
| // Only do this in the client. Otherwise we end up with a mono mixer in the service |
| // that writes to a stereo MMAP stream. |
| ALOGD("%s() - openStream() returned %d, try switching from MONO to STEREO", |
| __func__, mServiceStreamHandle); |
| request.getConfiguration().setSamplesPerFrame(2); // stereo |
| mServiceStreamHandle = mServiceInterface.openStream(request, configurationOutput); |
| } |
| if (mServiceStreamHandle < 0) { |
| return mServiceStreamHandle; |
| } |
| |
| result = configurationOutput.validate(); |
| if (result != AAUDIO_OK) { |
| goto error; |
| } |
| // Save results of the open. |
| if (getSamplesPerFrame() == AAUDIO_UNSPECIFIED) { |
| setSamplesPerFrame(configurationOutput.getSamplesPerFrame()); |
| } |
| mDeviceChannelCount = configurationOutput.getSamplesPerFrame(); |
| |
| setSampleRate(configurationOutput.getSampleRate()); |
| setDeviceId(configurationOutput.getDeviceId()); |
| setSessionId(configurationOutput.getSessionId()); |
| setSharingMode(configurationOutput.getSharingMode()); |
| |
| setUsage(configurationOutput.getUsage()); |
| setContentType(configurationOutput.getContentType()); |
| setInputPreset(configurationOutput.getInputPreset()); |
| |
| // Save device format so we can do format conversion and volume scaling together. |
| setDeviceFormat(configurationOutput.getFormat()); |
| |
| result = mServiceInterface.getStreamDescription(mServiceStreamHandle, mEndPointParcelable); |
| if (result != AAUDIO_OK) { |
| goto error; |
| } |
| |
| // Resolve parcelable into a descriptor. |
| result = mEndPointParcelable.resolve(&mEndpointDescriptor); |
| if (result != AAUDIO_OK) { |
| goto error; |
| } |
| |
| // Configure endpoint based on descriptor. |
| result = mAudioEndpoint.configure(&mEndpointDescriptor, getDirection()); |
| if (result != AAUDIO_OK) { |
| goto error; |
| } |
| |
| framesPerHardwareBurst = mEndpointDescriptor.dataQueueDescriptor.framesPerBurst; |
| |
| // Scale up the burst size to meet the minimum equivalent in microseconds. |
| // This is to avoid waking the CPU too often when the HW burst is very small |
| // or at high sample rates. |
| framesPerBurst = framesPerHardwareBurst; |
| do { |
| if (burstMicros > 0) { // skip first loop |
| framesPerBurst *= 2; |
| } |
| burstMicros = framesPerBurst * static_cast<int64_t>(1000000) / getSampleRate(); |
| } while (burstMicros < burstMinMicros); |
| ALOGD("%s() original HW burst = %d, minMicros = %d => SW burst = %d\n", |
| __func__, framesPerHardwareBurst, burstMinMicros, framesPerBurst); |
| |
| // Validate final burst size. |
| if (framesPerBurst < MIN_FRAMES_PER_BURST || framesPerBurst > MAX_FRAMES_PER_BURST) { |
| ALOGE("%s - framesPerBurst out of range = %d", __func__, framesPerBurst); |
| result = AAUDIO_ERROR_OUT_OF_RANGE; |
| goto error; |
| } |
| mFramesPerBurst = framesPerBurst; // only save good value |
| |
| capacity = mEndpointDescriptor.dataQueueDescriptor.capacityInFrames; |
| if (capacity < mFramesPerBurst || capacity > MAX_BUFFER_CAPACITY_IN_FRAMES) { |
| ALOGE("%s - bufferCapacity out of range = %d", __func__, capacity); |
| result = AAUDIO_ERROR_OUT_OF_RANGE; |
| goto error; |
| } |
| |
| mClockModel.setSampleRate(getSampleRate()); |
| mClockModel.setFramesPerBurst(framesPerHardwareBurst); |
| |
| if (isDataCallbackSet()) { |
| mCallbackFrames = builder.getFramesPerDataCallback(); |
| if (mCallbackFrames > getBufferCapacity() / 2) { |
| ALOGW("%s - framesPerCallback too big = %d, capacity = %d", |
| __func__, mCallbackFrames, getBufferCapacity()); |
| result = AAUDIO_ERROR_OUT_OF_RANGE; |
| goto error; |
| |
| } else if (mCallbackFrames < 0) { |
| ALOGW("%s - framesPerCallback negative", __func__); |
| result = AAUDIO_ERROR_OUT_OF_RANGE; |
| goto error; |
| |
| } |
| if (mCallbackFrames == AAUDIO_UNSPECIFIED) { |
| mCallbackFrames = mFramesPerBurst; |
| } |
| |
| const int32_t callbackBufferSize = mCallbackFrames * getBytesPerFrame(); |
| mCallbackBuffer = new uint8_t[callbackBufferSize]; |
| } |
| |
| setState(AAUDIO_STREAM_STATE_OPEN); |
| |
| return result; |
| |
| error: |
| close(); |
| return result; |
| } |
| |
| // This must be called under mStreamLock. |
| aaudio_result_t AudioStreamInternal::close() { |
| aaudio_result_t result = AAUDIO_OK; |
| ALOGV("%s(): mServiceStreamHandle = 0x%08X", __func__, mServiceStreamHandle); |
| if (mServiceStreamHandle != AAUDIO_HANDLE_INVALID) { |
| // Don't close a stream while it is running. |
| aaudio_stream_state_t currentState = getState(); |
| // Don't close a stream while it is running. Stop it first. |
| // If DISCONNECTED then we should still try to stop in case the |
| // error callback is still running. |
| if (isActive() || currentState == AAUDIO_STREAM_STATE_DISCONNECTED) { |
| requestStop(); |
| } |
| setState(AAUDIO_STREAM_STATE_CLOSING); |
| aaudio_handle_t serviceStreamHandle = mServiceStreamHandle; |
| mServiceStreamHandle = AAUDIO_HANDLE_INVALID; |
| |
| mServiceInterface.closeStream(serviceStreamHandle); |
| delete[] mCallbackBuffer; |
| mCallbackBuffer = nullptr; |
| |
| setState(AAUDIO_STREAM_STATE_CLOSED); |
| result = mEndPointParcelable.close(); |
| aaudio_result_t result2 = AudioStream::close(); |
| return (result != AAUDIO_OK) ? result : result2; |
| } else { |
| return AAUDIO_ERROR_INVALID_HANDLE; |
| } |
| } |
| |
| static void *aaudio_callback_thread_proc(void *context) |
| { |
| AudioStreamInternal *stream = (AudioStreamInternal *)context; |
| //LOGD("oboe_callback_thread, stream = %p", stream); |
| if (stream != NULL) { |
| return stream->callbackLoop(); |
| } else { |
| return NULL; |
| } |
| } |
| |
| /* |
| * It normally takes about 20-30 msec to start a stream on the server. |
| * But the first time can take as much as 200-300 msec. The HW |
| * starts right away so by the time the client gets a chance to write into |
| * the buffer, it is already in a deep underflow state. That can cause the |
| * XRunCount to be non-zero, which could lead an app to tune its latency higher. |
| * To avoid this problem, we set a request for the processing code to start the |
| * client stream at the same position as the server stream. |
| * The processing code will then save the current offset |
| * between client and server and apply that to any position given to the app. |
| */ |
| aaudio_result_t AudioStreamInternal::requestStart() |
| { |
| int64_t startTime; |
| if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) { |
| ALOGD("requestStart() mServiceStreamHandle invalid"); |
| return AAUDIO_ERROR_INVALID_STATE; |
| } |
| if (isActive()) { |
| ALOGD("requestStart() already active"); |
| return AAUDIO_ERROR_INVALID_STATE; |
| } |
| |
| aaudio_stream_state_t originalState = getState(); |
| if (originalState == AAUDIO_STREAM_STATE_DISCONNECTED) { |
| ALOGD("requestStart() but DISCONNECTED"); |
| return AAUDIO_ERROR_DISCONNECTED; |
| } |
| setState(AAUDIO_STREAM_STATE_STARTING); |
| |
| // Clear any stale timestamps from the previous run. |
| drainTimestampsFromService(); |
| |
| aaudio_result_t result = mServiceInterface.startStream(mServiceStreamHandle); |
| |
| startTime = AudioClock::getNanoseconds(); |
| mClockModel.start(startTime); |
| mNeedCatchUp.request(); // Ask data processing code to catch up when first timestamp received. |
| |
| // Start data callback thread. |
| if (result == AAUDIO_OK && isDataCallbackSet()) { |
| // 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); |
| } |
| if (result != AAUDIO_OK) { |
| setState(originalState); |
| } |
| 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()); |
| } |
| |
| // This must be called under mStreamLock. |
| aaudio_result_t AudioStreamInternal::stopCallback() |
| { |
| if (isDataCallbackSet() |
| && (isActive() || getState() == AAUDIO_STREAM_STATE_DISCONNECTED)) { |
| mCallbackEnabled.store(false); |
| return joinThread(NULL); // may temporarily unlock mStreamLock |
| } else { |
| return AAUDIO_OK; |
| } |
| } |
| |
| // This must be called under mStreamLock. |
| aaudio_result_t AudioStreamInternal::requestStop() { |
| aaudio_result_t result = stopCallback(); |
| if (result != AAUDIO_OK) { |
| return result; |
| } |
| // The stream may have been unlocked temporarily to let a callback finish |
| // and the callback may have stopped the stream. |
| // Check to make sure the stream still needs to be stopped. |
| // See also AudioStream::safeStop(). |
| if (!(isActive() || getState() == AAUDIO_STREAM_STATE_DISCONNECTED)) { |
| return AAUDIO_OK; |
| } |
| |
| if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) { |
| ALOGW("%s() mServiceStreamHandle invalid = 0x%08X", |
| __func__, mServiceStreamHandle); |
| return AAUDIO_ERROR_INVALID_STATE; |
| } |
| |
| mClockModel.stop(AudioClock::getNanoseconds()); |
| setState(AAUDIO_STREAM_STATE_STOPPING); |
| mAtomicInternalTimestamp.clear(); |
| |
| return mServiceInterface.stopStream(mServiceStreamHandle); |
| } |
| |
| aaudio_result_t AudioStreamInternal::registerThread() { |
| if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) { |
| ALOGW("%s() mServiceStreamHandle invalid", __func__); |
| return AAUDIO_ERROR_INVALID_STATE; |
| } |
| return mServiceInterface.registerAudioThread(mServiceStreamHandle, |
| gettid(), |
| getPeriodNanoseconds()); |
| } |
| |
| aaudio_result_t AudioStreamInternal::unregisterThread() { |
| if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) { |
| ALOGW("%s() mServiceStreamHandle invalid", __func__); |
| return AAUDIO_ERROR_INVALID_STATE; |
| } |
| return mServiceInterface.unregisterAudioThread(mServiceStreamHandle, gettid()); |
| } |
| |
| aaudio_result_t AudioStreamInternal::startClient(const android::AudioClient& client, |
| audio_port_handle_t *portHandle) { |
| ALOGV("%s() called", __func__); |
| if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) { |
| return AAUDIO_ERROR_INVALID_STATE; |
| } |
| aaudio_result_t result = mServiceInterface.startClient(mServiceStreamHandle, |
| client, portHandle); |
| ALOGV("%s(%d) returning %d", __func__, *portHandle, result); |
| return result; |
| } |
| |
| aaudio_result_t AudioStreamInternal::stopClient(audio_port_handle_t portHandle) { |
| ALOGV("%s(%d) called", __func__, portHandle); |
| if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) { |
| return AAUDIO_ERROR_INVALID_STATE; |
| } |
| aaudio_result_t result = mServiceInterface.stopClient(mServiceStreamHandle, portHandle); |
| ALOGV("%s(%d) returning %d", __func__, portHandle, result); |
| return result; |
| } |
| |
| aaudio_result_t AudioStreamInternal::getTimestamp(clockid_t clockId, |
| int64_t *framePosition, |
| int64_t *timeNanoseconds) { |
| // Generated in server and passed to client. Return latest. |
| if (mAtomicInternalTimestamp.isValid()) { |
| Timestamp timestamp = mAtomicInternalTimestamp.read(); |
| int64_t position = timestamp.getPosition() + mFramesOffsetFromService; |
| if (position >= 0) { |
| *framePosition = position; |
| *timeNanoseconds = timestamp.getNanoseconds(); |
| return AAUDIO_OK; |
| } |
| } |
| return AAUDIO_ERROR_INVALID_STATE; |
| } |
| |
| aaudio_result_t AudioStreamInternal::updateStateMachine() { |
| if (isDataCallbackActive()) { |
| return AAUDIO_OK; // state is getting updated by the callback thread read/write call |
| } |
| return processCommands(); |
| } |
| |
| 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("logTimestamp: timestamp says framePosition = %8lld 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("logTimestamp: framesDelta = %8lld, nanosDelta = %8lld, rate = %lld", |
| (long long) framesDelta, (long long) nanosDelta, (long long) rate); |
| } |
| oldPosition = framePosition; |
| oldTime = nanoTime; |
| } |
| |
| aaudio_result_t AudioStreamInternal::onTimestampService(AAudioServiceMessage *message) { |
| #if LOG_TIMESTAMPS |
| logTimestamp(*message); |
| #endif |
| processTimestamp(message->timestamp.position, message->timestamp.timestamp); |
| return AAUDIO_OK; |
| } |
| |
| aaudio_result_t AudioStreamInternal::onTimestampHardware(AAudioServiceMessage *message) { |
| Timestamp timestamp(message->timestamp.position, message->timestamp.timestamp); |
| mAtomicInternalTimestamp.write(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("%s - got AAUDIO_SERVICE_EVENT_STARTED", __func__); |
| if (getState() == AAUDIO_STREAM_STATE_STARTING) { |
| setState(AAUDIO_STREAM_STATE_STARTED); |
| } |
| break; |
| case AAUDIO_SERVICE_EVENT_PAUSED: |
| ALOGD("%s - got AAUDIO_SERVICE_EVENT_PAUSED", __func__); |
| if (getState() == AAUDIO_STREAM_STATE_PAUSING) { |
| setState(AAUDIO_STREAM_STATE_PAUSED); |
| } |
| break; |
| case AAUDIO_SERVICE_EVENT_STOPPED: |
| ALOGD("%s - got AAUDIO_SERVICE_EVENT_STOPPED", __func__); |
| if (getState() == AAUDIO_STREAM_STATE_STOPPING) { |
| setState(AAUDIO_STREAM_STATE_STOPPED); |
| } |
| break; |
| case AAUDIO_SERVICE_EVENT_FLUSHED: |
| ALOGD("%s - got AAUDIO_SERVICE_EVENT_FLUSHED", __func__); |
| if (getState() == AAUDIO_STREAM_STATE_FLUSHING) { |
| setState(AAUDIO_STREAM_STATE_FLUSHED); |
| onFlushFromServer(); |
| } |
| break; |
| case AAUDIO_SERVICE_EVENT_DISCONNECTED: |
| // Prevent hardware from looping on old data and making buzzing sounds. |
| if (getDirection() == AAUDIO_DIRECTION_OUTPUT) { |
| mAudioEndpoint.eraseDataMemory(); |
| } |
| result = AAUDIO_ERROR_DISCONNECTED; |
| setState(AAUDIO_STREAM_STATE_DISCONNECTED); |
| ALOGW("%s - AAUDIO_SERVICE_EVENT_DISCONNECTED - FIFO cleared", __func__); |
| break; |
| case AAUDIO_SERVICE_EVENT_VOLUME: |
| ALOGD("%s - AAUDIO_SERVICE_EVENT_VOLUME %lf", __func__, message->event.dataDouble); |
| mStreamVolume = (float)message->event.dataDouble; |
| doSetVolume(); |
| break; |
| case AAUDIO_SERVICE_EVENT_XRUN: |
| mXRunCount = static_cast<int32_t>(message->event.dataLong); |
| break; |
| default: |
| ALOGE("%s - Unrecognized event = %d", __func__, (int) message->event.event); |
| break; |
| } |
| return result; |
| } |
| |
| aaudio_result_t AudioStreamInternal::drainTimestampsFromService() { |
| aaudio_result_t result = AAUDIO_OK; |
| |
| while (result == AAUDIO_OK) { |
| AAudioServiceMessage message; |
| if (mAudioEndpoint.readUpCommand(&message) != 1) { |
| break; // no command this time, no problem |
| } |
| switch (message.what) { |
| // ignore most messages |
| case AAudioServiceMessage::code::TIMESTAMP_SERVICE: |
| case AAudioServiceMessage::code::TIMESTAMP_HARDWARE: |
| break; |
| |
| case AAudioServiceMessage::code::EVENT: |
| result = onEventFromServer(&message); |
| break; |
| |
| default: |
| ALOGE("%s - unrecognized message.what = %d", __func__, (int) message.what); |
| result = AAUDIO_ERROR_INTERNAL; |
| 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) { |
| AAudioServiceMessage message; |
| if (mAudioEndpoint.readUpCommand(&message) != 1) { |
| break; // no command this time, no problem |
| } |
| switch (message.what) { |
| case AAudioServiceMessage::code::TIMESTAMP_SERVICE: |
| result = onTimestampService(&message); |
| break; |
| |
| case AAudioServiceMessage::code::TIMESTAMP_HARDWARE: |
| result = onTimestampHardware(&message); |
| break; |
| |
| case AAudioServiceMessage::code::EVENT: |
| result = onEventFromServer(&message); |
| break; |
| |
| default: |
| ALOGE("%s - unrecognized message.what = %d", __func__, (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 = "aaProc"; |
| const char * fifoName = "aaRdy"; |
| ATRACE_BEGIN(traceName); |
| if (ATRACE_ENABLED()) { |
| int32_t fullFrames = mAudioEndpoint.getFullFramesAvailable(); |
| ATRACE_INT(fifoName, fullFrames); |
| } |
| |
| aaudio_result_t result = AAUDIO_OK; |
| int32_t loopCount = 0; |
| uint8_t* audioData = (uint8_t*)buffer; |
| int64_t currentTimeNanos = AudioClock::getNanoseconds(); |
| const int64_t entryTimeNanos = currentTimeNanos; |
| const int64_t deadlineNanos = currentTimeNanos + timeoutNanoseconds; |
| int32_t framesLeft = numFrames; |
| |
| // 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 process as much as it can. |
| int64_t wakeTimeNanos = 0; |
| aaudio_result_t framesProcessed = processDataNow(audioData, framesLeft, |
| currentTimeNanos, &wakeTimeNanos); |
| if (framesProcessed < 0) { |
| result = framesProcessed; |
| break; |
| } |
| framesLeft -= (int32_t) framesProcessed; |
| audioData += framesProcessed * getBytesPerFrame(); |
| |
| // Should we block? |
| if (timeoutNanoseconds == 0) { |
| break; // don't block |
| } else if (framesLeft > 0) { |
| if (!mAudioEndpoint.isFreeRunning()) { |
| // If there is software on the other end of the FIFO then it may get delayed. |
| // So wake up just a little after we expect it to be ready. |
| wakeTimeNanos += mWakeupDelayNanos; |
| } |
| |
| currentTimeNanos = AudioClock::getNanoseconds(); |
| int64_t earliestWakeTime = currentTimeNanos + mMinimumSleepNanos; |
| // Guarantee a minimum sleep time. |
| if (wakeTimeNanos < earliestWakeTime) { |
| wakeTimeNanos = earliestWakeTime; |
| } |
| |
| if (wakeTimeNanos > deadlineNanos) { |
| // If we time out, just return the framesWritten so far. |
| // TODO remove after we fix the deadline bug |
| ALOGW("processData(): entered at %lld nanos, currently %lld", |
| (long long) entryTimeNanos, (long long) currentTimeNanos); |
| ALOGW("processData(): TIMEOUT after %lld nanos", |
| (long long) timeoutNanoseconds); |
| ALOGW("processData(): wakeTime = %lld, deadline = %lld nanos", |
| (long long) wakeTimeNanos, (long long) deadlineNanos); |
| ALOGW("processData(): past deadline by %d micros", |
| (int)((wakeTimeNanos - deadlineNanos) / AAUDIO_NANOS_PER_MICROSECOND)); |
| mClockModel.dump(); |
| mAudioEndpoint.dump(); |
| break; |
| } |
| |
| if (ATRACE_ENABLED()) { |
| int32_t fullFrames = mAudioEndpoint.getFullFramesAvailable(); |
| ATRACE_INT(fifoName, fullFrames); |
| int64_t sleepForNanos = wakeTimeNanos - currentTimeNanos; |
| ATRACE_INT("aaSlpNs", (int32_t)sleepForNanos); |
| } |
| |
| AudioClock::sleepUntilNanoTime(wakeTimeNanos); |
| currentTimeNanos = AudioClock::getNanoseconds(); |
| } |
| } |
| |
| if (ATRACE_ENABLED()) { |
| int32_t fullFrames = mAudioEndpoint.getFullFramesAvailable(); |
| ATRACE_INT(fifoName, fullFrames); |
| } |
| |
| // 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 adjustedFrames = requestedFrames; |
| int32_t actualFrames = 0; |
| int32_t maximumSize = getBufferCapacity(); |
| |
| // Clip to minimum size so that rounding up will work better. |
| if (adjustedFrames < 1) { |
| adjustedFrames = 1; |
| } |
| |
| if (adjustedFrames > maximumSize) { |
| // Clip to maximum size. |
| adjustedFrames = maximumSize; |
| } else { |
| // Round to the next highest burst size. |
| int32_t numBursts = (adjustedFrames + mFramesPerBurst - 1) / mFramesPerBurst; |
| adjustedFrames = numBursts * mFramesPerBurst; |
| // Rounding may have gone above maximum. |
| if (adjustedFrames > maximumSize) { |
| adjustedFrames = maximumSize; |
| } |
| } |
| |
| aaudio_result_t result = mAudioEndpoint.setBufferSizeInFrames(adjustedFrames, &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 mFramesPerBurst; |
| } |
| |
| // This must be called under mStreamLock. |
| aaudio_result_t AudioStreamInternal::joinThread(void** returnArg) { |
| return AudioStream::joinThread(returnArg, calculateReasonableTimeout(getFramesPerBurst())); |
| } |
| |
| bool AudioStreamInternal::isClockModelInControl() const { |
| return isActive() && mAudioEndpoint.isFreeRunning() && mClockModel.isRunning(); |
| } |
