| /* |
| * 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. |
| */ |
| |
| #define LOG_TAG "AudioStreamInternal" |
| //#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 <media/AudioParameter.h> |
| #include <media/AudioSystem.h> |
| #include <media/MediaMetricsItem.h> |
| #include <utils/Trace.h> |
| |
| #include "AudioEndpointParcelable.h" |
| #include "binding/AAudioStreamRequest.h" |
| #include "binding/AAudioStreamConfiguration.h" |
| #include "binding/AAudioServiceMessage.h" |
| #include "core/AudioGlobal.h" |
| #include "core/AudioStreamBuilder.h" |
| #include "fifo/FifoBuffer.h" |
| #include "utility/AudioClock.h" |
| #include <media/AidlConversion.h> |
| |
| #include "AudioStreamInternal.h" |
| |
| // We do this after the #includes because if a header uses ALOG. |
| // it would fail on the reference to mInService. |
| #undef LOG_TAG |
| // 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") |
| |
| using android::content::AttributionSourceState; |
| |
| 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() |
| , 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() { |
| ALOGD("%s() %p called", __func__, this); |
| } |
| |
| aaudio_result_t AudioStreamInternal::open(const AudioStreamBuilder &builder) { |
| |
| aaudio_result_t result = AAUDIO_OK; |
| 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 audio_format_t requestedFormat = getFormat(); |
| // We have to do volume scaling. So we prefer FLOAT format. |
| if (requestedFormat == AUDIO_FORMAT_DEFAULT) { |
| setFormat(AUDIO_FORMAT_PCM_FLOAT); |
| } |
| // Request FLOAT for the shared mixer or the device. |
| request.getConfiguration().setFormat(AUDIO_FORMAT_PCM_FLOAT); |
| |
| // TODO b/182392769: use attribution source util |
| AttributionSourceState attributionSource; |
| attributionSource.uid = VALUE_OR_FATAL(android::legacy2aidl_uid_t_int32_t(getuid())); |
| attributionSource.pid = VALUE_OR_FATAL(android::legacy2aidl_pid_t_int32_t(getpid())); |
| attributionSource.packageName = builder.getOpPackageName(); |
| attributionSource.attributionTag = builder.getAttributionTag(); |
| attributionSource.token = sp<android::BBinder>::make(); |
| |
| // Build the request to send to the server. |
| request.setAttributionSource(attributionSource); |
| request.setSharingModeMatchRequired(isSharingModeMatchRequired()); |
| request.setInService(isInService()); |
| |
| request.getConfiguration().setDeviceId(getDeviceId()); |
| request.getConfiguration().setSampleRate(getSampleRate()); |
| request.getConfiguration().setDirection(getDirection()); |
| request.getConfiguration().setSharingMode(getSharingMode()); |
| request.getConfiguration().setChannelMask(getChannelMask()); |
| |
| request.getConfiguration().setUsage(getUsage()); |
| request.getConfiguration().setContentType(getContentType()); |
| request.getConfiguration().setSpatializationBehavior(getSpatializationBehavior()); |
| request.getConfiguration().setIsContentSpatialized(isContentSpatialized()); |
| request.getConfiguration().setInputPreset(getInputPreset()); |
| request.getConfiguration().setPrivacySensitive(isPrivacySensitive()); |
| |
| 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 |
| || request.getConfiguration().getChannelMask() == AAUDIO_CHANNEL_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().setChannelMask(AAUDIO_CHANNEL_STEREO); |
| mServiceStreamHandle = mServiceInterface.openStream(request, configurationOutput); |
| } |
| if (mServiceStreamHandle < 0) { |
| return mServiceStreamHandle; |
| } |
| |
| // This must match the key generated in oboeservice/AAudioServiceStreamBase.cpp |
| // so the client can have permission to log. |
| if (!mInService) { |
| // No need to log if it is from service side. |
| mMetricsId = std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_STREAM) |
| + std::to_string(mServiceStreamHandle); |
| } |
| |
| android::mediametrics::LogItem(mMetricsId) |
| .set(AMEDIAMETRICS_PROP_PERFORMANCEMODE, |
| AudioGlobal_convertPerformanceModeToText(builder.getPerformanceMode())) |
| .set(AMEDIAMETRICS_PROP_SHARINGMODE, |
| AudioGlobal_convertSharingModeToText(builder.getSharingMode())) |
| .set(AMEDIAMETRICS_PROP_ENCODINGCLIENT, |
| android::toString(requestedFormat).c_str()).record(); |
| |
| result = configurationOutput.validate(); |
| if (result != AAUDIO_OK) { |
| goto error; |
| } |
| // Save results of the open. |
| if (getChannelMask() == AAUDIO_UNSPECIFIED) { |
| setChannelMask(configurationOutput.getChannelMask()); |
| } |
| |
| mDeviceChannelCount = configurationOutput.getSamplesPerFrame(); |
| |
| setSampleRate(configurationOutput.getSampleRate()); |
| setDeviceId(configurationOutput.getDeviceId()); |
| setSessionId(configurationOutput.getSessionId()); |
| setSharingMode(configurationOutput.getSharingMode()); |
| |
| setUsage(configurationOutput.getUsage()); |
| setContentType(configurationOutput.getContentType()); |
| setSpatializationBehavior(configurationOutput.getSpatializationBehavior()); |
| setIsContentSpatialized(configurationOutput.isContentSpatialized()); |
| 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. |
| mAudioEndpoint = std::make_unique<AudioEndpoint>(); |
| result = mAudioEndpoint->configure(&mEndpointDescriptor, getDirection()); |
| if (result != AAUDIO_OK) { |
| goto error; |
| } |
| |
| if ((result = configureDataInformation(builder.getFramesPerDataCallback())) != AAUDIO_OK) { |
| goto error; |
| } |
| |
| setState(AAUDIO_STREAM_STATE_OPEN); |
| |
| return result; |
| |
| error: |
| safeReleaseClose(); |
| return result; |
| } |
| |
| aaudio_result_t AudioStreamInternal::configureDataInformation(int32_t callbackFrames) { |
| int32_t 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. |
| int32_t framesPerBurst = framesPerHardwareBurst; |
| int32_t burstMicros = 0; |
| const int32_t burstMinMicros = android::AudioSystem::getAAudioHardwareBurstMinUsec(); |
| 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); |
| return AAUDIO_ERROR_OUT_OF_RANGE; |
| } |
| setFramesPerBurst(framesPerBurst); // only save good value |
| |
| mBufferCapacityInFrames = mEndpointDescriptor.dataQueueDescriptor.capacityInFrames; |
| if (mBufferCapacityInFrames < getFramesPerBurst() |
| || mBufferCapacityInFrames > MAX_BUFFER_CAPACITY_IN_FRAMES) { |
| ALOGE("%s - bufferCapacity out of range = %d", __func__, mBufferCapacityInFrames); |
| return AAUDIO_ERROR_OUT_OF_RANGE; |
| } |
| |
| mClockModel.setSampleRate(getSampleRate()); |
| mClockModel.setFramesPerBurst(framesPerHardwareBurst); |
| |
| if (isDataCallbackSet()) { |
| mCallbackFrames = callbackFrames; |
| if (mCallbackFrames > getBufferCapacity() / 2) { |
| ALOGW("%s - framesPerCallback too big = %d, capacity = %d", |
| __func__, mCallbackFrames, getBufferCapacity()); |
| return AAUDIO_ERROR_OUT_OF_RANGE; |
| } else if (mCallbackFrames < 0) { |
| ALOGW("%s - framesPerCallback negative", __func__); |
| return AAUDIO_ERROR_OUT_OF_RANGE; |
| } |
| if (mCallbackFrames == AAUDIO_UNSPECIFIED) { |
| mCallbackFrames = getFramesPerBurst(); |
| } |
| |
| const int32_t callbackBufferSize = mCallbackFrames * getBytesPerFrame(); |
| mCallbackBuffer = std::make_unique<uint8_t[]>(callbackBufferSize); |
| } |
| |
| // Exclusive output streams should combine channels when mono audio adjustment |
| // is enabled. They should also adjust for audio balance. |
| if ((getDirection() == AAUDIO_DIRECTION_OUTPUT) && |
| (getSharingMode() == AAUDIO_SHARING_MODE_EXCLUSIVE)) { |
| bool isMasterMono = false; |
| android::AudioSystem::getMasterMono(&isMasterMono); |
| setRequireMonoBlend(isMasterMono); |
| float audioBalance = 0; |
| android::AudioSystem::getMasterBalance(&audioBalance); |
| setAudioBalance(audioBalance); |
| } |
| |
| // For debugging and analyzing the distribution of MMAP timestamps. |
| // For OUTPUT, use a NEGATIVE offset to move the CPU writes further BEFORE the HW reads. |
| // For INPUT, use a POSITIVE offset to move the CPU reads further AFTER the HW writes. |
| // You can use this offset to reduce glitching. |
| // You can also use this offset to force glitching. By iterating over multiple |
| // values you can reveal the distribution of the hardware timing jitter. |
| if (mAudioEndpoint->isFreeRunning()) { // MMAP? |
| int32_t offsetMicros = (getDirection() == AAUDIO_DIRECTION_OUTPUT) |
| ? AAudioProperty_getOutputMMapOffsetMicros() |
| : AAudioProperty_getInputMMapOffsetMicros(); |
| // This log is used to debug some tricky glitch issues. Please leave. |
| ALOGD_IF(offsetMicros, "%s() - %s mmap offset = %d micros", |
| __func__, |
| (getDirection() == AAUDIO_DIRECTION_OUTPUT) ? "output" : "input", |
| offsetMicros); |
| mTimeOffsetNanos = offsetMicros * AAUDIO_NANOS_PER_MICROSECOND; |
| } |
| |
| setBufferSize(mBufferCapacityInFrames / 2); // Default buffer size to match Q |
| return AAUDIO_OK; |
| } |
| |
| // This must be called under mStreamLock. |
| aaudio_result_t AudioStreamInternal::release_l() { |
| aaudio_result_t result = AAUDIO_OK; |
| ALOGD("%s(): mServiceStreamHandle = 0x%08X", __func__, mServiceStreamHandle); |
| if (mServiceStreamHandle != AAUDIO_HANDLE_INVALID) { |
| aaudio_stream_state_t currentState = getState(); |
| // Don't release 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_l(); |
| } |
| |
| logReleaseBufferState(); |
| |
| setState(AAUDIO_STREAM_STATE_CLOSING); |
| aaudio_handle_t serviceStreamHandle = mServiceStreamHandle; |
| mServiceStreamHandle = AAUDIO_HANDLE_INVALID; |
| |
| mServiceInterface.closeStream(serviceStreamHandle); |
| mCallbackBuffer.reset(); |
| |
| // Update local frame counters so we can query them after releasing the endpoint. |
| getFramesRead(); |
| getFramesWritten(); |
| mAudioEndpoint.reset(); |
| result = mEndPointParcelable.close(); |
| aaudio_result_t result2 = AudioStream::release_l(); |
| 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 != nullptr) { |
| return stream->callbackLoop(); |
| } else { |
| return nullptr; |
| } |
| } |
| |
| aaudio_result_t AudioStreamInternal::exitStandby_l() { |
| AudioEndpointParcelable endpointParcelable; |
| // The stream is in standby mode, copy all available data and then close the duplicated |
| // shared file descriptor so that it won't cause issue when the HAL try to reallocate new |
| // shared file descriptor when exiting from standby. |
| // Cache current read counter, which will be reset to new read and write counter |
| // when the new data queue and endpoint are reconfigured. |
| const android::fifo_counter_t readCounter = mAudioEndpoint->getDataReadCounter(); |
| // Cache the buffer size which may be from client. |
| const int32_t previousBufferSize = mBufferSizeInFrames; |
| // Copy all available data from current data queue. |
| uint8_t buffer[getBufferCapacity() * getBytesPerFrame()]; |
| android::fifo_frames_t fullFramesAvailable = |
| mAudioEndpoint->read(buffer, getBufferCapacity()); |
| mEndPointParcelable.closeDataFileDescriptor(); |
| aaudio_result_t result = mServiceInterface.exitStandby( |
| mServiceStreamHandle, endpointParcelable); |
| if (result != AAUDIO_OK) { |
| ALOGE("Failed to exit standby, error=%d", result); |
| goto exit; |
| } |
| // Reconstruct data queue descriptor using new shared file descriptor. |
| mEndPointParcelable.updateDataFileDescriptor(&endpointParcelable); |
| result = mEndPointParcelable.resolveDataQueue(&mEndpointDescriptor.dataQueueDescriptor); |
| if (result != AAUDIO_OK) { |
| ALOGE("Failed to resolve data queue after exiting standby, error=%d", result); |
| goto exit; |
| } |
| // Reconfigure audio endpoint with new data queue descriptor. |
| mAudioEndpoint->configureDataQueue( |
| mEndpointDescriptor.dataQueueDescriptor, getDirection()); |
| // Set read and write counters with previous read counter, the later write action |
| // will make the counter at the correct place. |
| mAudioEndpoint->setDataReadCounter(readCounter); |
| mAudioEndpoint->setDataWriteCounter(readCounter); |
| result = configureDataInformation(mCallbackFrames); |
| if (result != AAUDIO_OK) { |
| ALOGE("Failed to configure data information after exiting standby, error=%d", result); |
| goto exit; |
| } |
| // Write data from previous data buffer to new endpoint. |
| if (android::fifo_frames_t framesWritten = |
| mAudioEndpoint->write(buffer, fullFramesAvailable); |
| framesWritten != fullFramesAvailable) { |
| ALOGW("Some data lost after exiting standby, frames written: %d, " |
| "frames to write: %d", framesWritten, fullFramesAvailable); |
| } |
| // Reset previous buffer size as it may be requested by the client. |
| setBufferSize(previousBufferSize); |
| |
| exit: |
| return result; |
| } |
| |
| /* |
| * 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_l() |
| { |
| 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(); |
| |
| prepareBuffersForStart(); // tell subclasses to get ready |
| |
| aaudio_result_t result = mServiceInterface.startStream(mServiceStreamHandle); |
| if (result == AAUDIO_ERROR_STANDBY) { |
| // The stream is at standby mode. Need to exit standby before starting the stream. |
| result = exitStandby_l(); |
| if (result == AAUDIO_OK) { |
| result = mServiceInterface.startStream(mServiceStreamHandle); |
| } |
| } |
| if (result != AAUDIO_OK) { |
| ALOGD("%s() error = %d, stream was probably stolen", __func__, result); |
| // Stealing was added in R. Coerce result to improve backward compatibility. |
| result = AAUDIO_ERROR_DISCONNECTED; |
| setState(AAUDIO_STREAM_STATE_DISCONNECTED); |
| } |
| |
| 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_l(periodNanos, aaudio_callback_thread_proc, this); |
| } |
| if (result != AAUDIO_OK) { |
| // TODO(b/214607638): Do we want to roll back to original state or keep as disconnected? |
| 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_l() |
| { |
| if (isDataCallbackSet() |
| && (isActive() || getState() == AAUDIO_STREAM_STATE_DISCONNECTED)) { |
| mCallbackEnabled.store(false); |
| aaudio_result_t result = joinThread_l(nullptr); // may temporarily unlock mStreamLock |
| if (result == AAUDIO_ERROR_INVALID_HANDLE) { |
| ALOGD("%s() INVALID_HANDLE, stream was probably stolen", __func__); |
| result = AAUDIO_OK; |
| } |
| return result; |
| } else { |
| ALOGD("%s() skipped, isDataCallbackSet() = %d, isActive() = %d, getState() = %d", __func__, |
| isDataCallbackSet(), isActive(), getState()); |
| return AAUDIO_OK; |
| } |
| } |
| |
| aaudio_result_t AudioStreamInternal::requestStop_l() { |
| aaudio_result_t result = stopCallback_l(); |
| if (result != AAUDIO_OK) { |
| ALOGW("%s() stop callback returned %d, returning early", __func__, result); |
| 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_l(). |
| if (!(isActive() || getState() == AAUDIO_STREAM_STATE_DISCONNECTED)) { |
| ALOGD("%s() returning early, not active or disconnected", __func__); |
| 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(); |
| |
| result = mServiceInterface.stopStream(mServiceStreamHandle); |
| if (result == AAUDIO_ERROR_INVALID_HANDLE) { |
| ALOGD("%s() INVALID_HANDLE, stream was probably stolen", __func__); |
| result = AAUDIO_OK; |
| } |
| return result; |
| } |
| |
| 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, |
| const audio_attributes_t *attr, |
| 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, attr, 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; |
| } |
| |
| 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 + mTimeOffsetNanos); |
| 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) { |
| break; |
| } |
| 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) { |
| break; |
| } |
| 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 (wakeTimeNanos != 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; |
| const int32_t maximumSize = getBufferCapacity() - getFramesPerBurst(); |
| // Minimum size should be a multiple number of bursts. |
| const int32_t minimumSize = 1 * getFramesPerBurst(); |
| |
| // Clip to minimum size so that rounding up will work better. |
| adjustedFrames = std::max(minimumSize, adjustedFrames); |
| |
| // Prevent arithmetic overflow by clipping before we round. |
| if (adjustedFrames >= maximumSize) { |
| adjustedFrames = maximumSize; |
| } else { |
| // Round to the next highest burst size. |
| int32_t numBursts = (adjustedFrames + getFramesPerBurst() - 1) / getFramesPerBurst(); |
| adjustedFrames = numBursts * getFramesPerBurst(); |
| // Clip just in case maximumSize is not a multiple of getFramesPerBurst(). |
| adjustedFrames = std::min(maximumSize, adjustedFrames); |
| } |
| |
| if (mAudioEndpoint) { |
| // Clip against the actual size from the endpoint. |
| int32_t actualFrames = 0; |
| // Set to maximum size so we can write extra data when ready in order to reduce glitches. |
| // The amount we keep in the buffer is controlled by mBufferSizeInFrames. |
| mAudioEndpoint->setBufferSizeInFrames(maximumSize, &actualFrames); |
| // actualFrames should be <= actual maximum size of endpoint |
| adjustedFrames = std::min(actualFrames, adjustedFrames); |
| } |
| |
| if (adjustedFrames != mBufferSizeInFrames) { |
| android::mediametrics::LogItem(mMetricsId) |
| .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETBUFFERSIZE) |
| .set(AMEDIAMETRICS_PROP_BUFFERSIZEFRAMES, adjustedFrames) |
| .set(AMEDIAMETRICS_PROP_UNDERRUN, (int32_t) getXRunCount()) |
| .record(); |
| } |
| |
| mBufferSizeInFrames = adjustedFrames; |
| ALOGV("%s(%d) returns %d", __func__, requestedFrames, adjustedFrames); |
| return (aaudio_result_t) adjustedFrames; |
| } |
| |
| int32_t AudioStreamInternal::getBufferSize() const { |
| return mBufferSizeInFrames; |
| } |
| |
| int32_t AudioStreamInternal::getBufferCapacity() const { |
| return mBufferCapacityInFrames; |
| } |
| |
| bool AudioStreamInternal::isClockModelInControl() const { |
| return isActive() && mAudioEndpoint->isFreeRunning() && mClockModel.isRunning(); |
| } |
