/* * 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 #define ATRACE_TAG ATRACE_TAG_AUDIO #include #include #include #include #include #include #include #include #include "AudioEndpointParcelable.h" #include "binding/AAudioBinderClient.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 #include #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 // Minimum number of bursts to use when sample rate conversion is used. #define MIN_SAMPLE_RATE_CONVERSION_NUM_BURSTS 3 AudioStreamInternal::AudioStreamInternal(AAudioServiceInterface &serviceInterface, bool inService) : AudioStream() , mClockModel() , 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::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()); mServiceStreamHandleInfo = mServiceInterface.openStream(request, configurationOutput); if (getServiceHandle() < 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__, getServiceHandle()); request.getConfiguration().setChannelMask(AAUDIO_CHANNEL_STEREO); mServiceStreamHandleInfo = mServiceInterface.openStream(request, configurationOutput); } if (getServiceHandle() < 0) { return getServiceHandle(); } // 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(getServiceHandle()); } 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()); } setDeviceId(configurationOutput.getDeviceId()); setSessionId(configurationOutput.getSessionId()); setSharingMode(configurationOutput.getSharingMode()); setUsage(configurationOutput.getUsage()); setContentType(configurationOutput.getContentType()); setSpatializationBehavior(configurationOutput.getSpatializationBehavior()); setIsContentSpatialized(configurationOutput.isContentSpatialized()); setInputPreset(configurationOutput.getInputPreset()); setDeviceSampleRate(configurationOutput.getSampleRate()); if (getSampleRate() == AAUDIO_UNSPECIFIED) { setSampleRate(configurationOutput.getSampleRate()); } if (!com::android::media::aaudio::sample_rate_conversion()) { if (getSampleRate() != getDeviceSampleRate()) { ALOGD("%s - skipping sample rate converter. SR = %d, Device SR = %d", __func__, getSampleRate(), getDeviceSampleRate()); result = AAUDIO_ERROR_INVALID_RATE; goto error; } } // Save device format so we can do format conversion and volume scaling together. setDeviceFormat(configurationOutput.getFormat()); setDeviceSamplesPerFrame(configurationOutput.getSamplesPerFrame()); setHardwareSamplesPerFrame(configurationOutput.getHardwareSamplesPerFrame()); setHardwareSampleRate(configurationOutput.getHardwareSampleRate()); setHardwareFormat(configurationOutput.getHardwareFormat()); result = mServiceInterface.getStreamDescription(mServiceStreamHandleInfo, 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(); 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 originalFramesPerBurst = mEndpointDescriptor.dataQueueDescriptor.framesPerBurst; int32_t deviceFramesPerBurst = originalFramesPerBurst; // 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. The actual number of frames that we call back to // the app with will be 0 < N <= framesPerBurst so round up the division. int32_t burstMicros = 0; const int32_t burstMinMicros = android::AudioSystem::getAAudioHardwareBurstMinUsec(); do { if (burstMicros > 0) { // skip first loop deviceFramesPerBurst *= 2; } burstMicros = deviceFramesPerBurst * static_cast(1000000) / getDeviceSampleRate(); } while (burstMicros < burstMinMicros); ALOGD("%s() original HW burst = %d, minMicros = %d => SW burst = %d\n", __func__, originalFramesPerBurst, burstMinMicros, deviceFramesPerBurst); // Validate final burst size. if (deviceFramesPerBurst < MIN_FRAMES_PER_BURST || deviceFramesPerBurst > MAX_FRAMES_PER_BURST) { ALOGE("%s - deviceFramesPerBurst out of range = %d", __func__, deviceFramesPerBurst); return AAUDIO_ERROR_OUT_OF_RANGE; } // Calculate the application framesPerBurst from the deviceFramesPerBurst int32_t framesPerBurst = (static_cast(deviceFramesPerBurst) * getSampleRate() + getDeviceSampleRate() - 1) / getDeviceSampleRate(); setDeviceFramesPerBurst(deviceFramesPerBurst); setFramesPerBurst(framesPerBurst); // only save good value mDeviceBufferCapacityInFrames = mEndpointDescriptor.dataQueueDescriptor.capacityInFrames; mBufferCapacityInFrames = static_cast(mDeviceBufferCapacityInFrames) * getSampleRate() / getDeviceSampleRate(); 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(getDeviceSampleRate()); mClockModel.setFramesPerBurst(deviceFramesPerBurst); 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(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; } // Default buffer size to match Q setBufferSize(mBufferCapacityInFrames / 2); 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__, getServiceHandle()); if (getServiceHandle() != AAUDIO_HANDLE_INVALID) { // 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() || isDisconnected()) { requestStop_l(); } logReleaseBufferState(); setState(AAUDIO_STREAM_STATE_CLOSING); auto serviceStreamHandleInfo = mServiceStreamHandleInfo; mServiceStreamHandleInfo = AAudioHandleInfo(); mServiceInterface.closeStream(serviceStreamHandleInfo); 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[getDeviceBufferCapacity() * getBytesPerFrame()]; android::fifo_frames_t fullFramesAvailable = mAudioEndpoint->read(buffer, getDeviceBufferCapacity()); // Before releasing the data queue, update the frames read and written. getFramesRead(); getFramesWritten(); // Call freeDataQueue() here because the following call to // closeDataFileDescriptor() will invalidate the pointers used by the data queue. mAudioEndpoint->freeDataQueue(); mEndPointParcelable.closeDataFileDescriptor(); aaudio_result_t result = mServiceInterface.exitStandby( mServiceStreamHandleInfo, endpointParcelable); if (result != AAUDIO_OK) { ALOGE("Failed to exit standby, error=%d", result); goto exit; } // Reconstruct data queue descriptor using new shared file descriptor. result = mEndPointParcelable.updateDataFileDescriptor(&endpointParcelable); if (result != AAUDIO_OK) { ALOGE("%s failed to update data file descriptor, error=%d", __func__, result); goto exit; } 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 (const 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 (getServiceHandle() == AAUDIO_HANDLE_INVALID) { ALOGD("requestStart() mServiceStreamHandle invalid"); return AAUDIO_ERROR_INVALID_STATE; } if (isActive()) { ALOGD("requestStart() already active"); return AAUDIO_ERROR_INVALID_STATE; } if (isDisconnected()) { ALOGD("requestStart() but DISCONNECTED"); return AAUDIO_ERROR_DISCONNECTED; } const aaudio_stream_state_t originalState = getState(); 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(mServiceStreamHandleInfo); 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(mServiceStreamHandleInfo); } } 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; setDisconnected(); } 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) { 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() || isDisconnected())) { 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() || isDisconnected())) { ALOGD("%s() returning early, not active or disconnected", __func__); return AAUDIO_OK; } if (getServiceHandle() == AAUDIO_HANDLE_INVALID) { ALOGW("%s() mServiceStreamHandle invalid = 0x%08X", __func__, getServiceHandle()); return AAUDIO_ERROR_INVALID_STATE; } // For playback, sleep until all the audio data has played. // Then clear the buffer to prevent noise. prepareBuffersForStop(); mClockModel.stop(AudioClock::getNanoseconds()); setState(AAUDIO_STREAM_STATE_STOPPING); mAtomicInternalTimestamp.clear(); #if 0 // Simulate very slow CPU, force race condition where the // DSP keeps playing after we stop writing. AudioClock::sleepForNanos(800 * AAUDIO_NANOS_PER_MILLISECOND); #endif result = mServiceInterface.stopStream(mServiceStreamHandleInfo); 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 (getServiceHandle() == AAUDIO_HANDLE_INVALID) { ALOGW("%s() mServiceStreamHandle invalid", __func__); return AAUDIO_ERROR_INVALID_STATE; } return mServiceInterface.registerAudioThread(mServiceStreamHandleInfo, gettid(), getPeriodNanoseconds()); } aaudio_result_t AudioStreamInternal::unregisterThread() { if (getServiceHandle() == AAUDIO_HANDLE_INVALID) { ALOGW("%s() mServiceStreamHandle invalid", __func__); return AAUDIO_ERROR_INVALID_STATE; } return mServiceInterface.unregisterAudioThread(mServiceStreamHandleInfo, 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 (getServiceHandle() == AAUDIO_HANDLE_INVALID) { ALOGE("%s() getServiceHandle() is invalid", __func__); return AAUDIO_ERROR_INVALID_STATE; } aaudio_result_t result = mServiceInterface.startClient(mServiceStreamHandleInfo, client, attr, portHandle); ALOGV("%s(), got %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 (getServiceHandle() == AAUDIO_HANDLE_INVALID) { ALOGE("%s(%d) getServiceHandle() is invalid", __func__, portHandle); return AAUDIO_ERROR_INVALID_STATE; } aaudio_result_t result = mServiceInterface.stopClient(mServiceStreamHandleInfo, 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(); // This should not overflow as timestamp.getPosition() should be a position in a buffer and // not the actual timestamp. timestamp.getNanoseconds() below uses the actual timestamp. // At 48000 Hz we can run for over 100 years before overflowing the int64_t. int64_t position = (timestamp.getPosition() + mFramesOffsetFromService) * getSampleRate() / getDeviceSampleRate(); 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); } mPlayerBase->triggerPortIdUpdate(static_cast( message->event.dataLong)); 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; setDisconnected(); 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(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) { if (isDisconnected()) { return AAUDIO_ERROR_DISCONNECTED; } if (!mInService && AAudioBinderClient::getInstance().getServiceLifetimeId() != getServiceLifetimeId()) { // The service lifetime id will be changed whenever the binder died. In that case, if // the service lifetime id from AAudioBinderClient is different from the cached one, // returns AAUDIO_ERROR_DISCONNECTED. // Note that only compare the service lifetime id if it is not in service as the streams // in service will all be gone when aaudio service dies. mClockModel.stop(AudioClock::getNanoseconds()); // Set the stream as disconnected as the service lifetime id will only change when // the binder dies. setDisconnected(); return AAUDIO_ERROR_DISCONNECTED; } 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. 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) { const int32_t maximumSize = getBufferCapacity() - getFramesPerBurst(); int32_t adjustedFrames = std::min(requestedFrames, maximumSize); // Buffer sizes should always be a multiple of framesPerBurst. int32_t numBursts = (static_cast(adjustedFrames) + getFramesPerBurst() - 1) / getFramesPerBurst(); // Use at least one burst if (numBursts == 0) { numBursts = 1; } // Set a minimum number of bursts if sample rate conversion is used. if ((getSampleRate() != getDeviceSampleRate()) && (numBursts < MIN_SAMPLE_RATE_CONVERSION_NUM_BURSTS)) { numBursts = MIN_SAMPLE_RATE_CONVERSION_NUM_BURSTS; } if (mAudioEndpoint) { // Clip against the actual size from the endpoint. int32_t actualFramesDevice = 0; int32_t maximumFramesDevice = getDeviceBufferCapacity() - getDeviceFramesPerBurst(); // 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(maximumFramesDevice, &actualFramesDevice); int32_t actualNumBursts = actualFramesDevice / getDeviceFramesPerBurst(); numBursts = std::min(numBursts, actualNumBursts); } const int32_t bufferSizeInFrames = numBursts * getFramesPerBurst(); const int32_t deviceBufferSizeInFrames = numBursts * getDeviceFramesPerBurst(); if (deviceBufferSizeInFrames != mDeviceBufferSizeInFrames) { android::mediametrics::LogItem(mMetricsId) .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETBUFFERSIZE) .set(AMEDIAMETRICS_PROP_BUFFERSIZEFRAMES, deviceBufferSizeInFrames) .set(AMEDIAMETRICS_PROP_UNDERRUN, (int32_t) getXRunCount()) .record(); } mBufferSizeInFrames = bufferSizeInFrames; mDeviceBufferSizeInFrames = deviceBufferSizeInFrames; ALOGV("%s(%d) returns %d", __func__, requestedFrames, adjustedFrames); return (aaudio_result_t) adjustedFrames; } int32_t AudioStreamInternal::getBufferSize() const { return mBufferSizeInFrames; } int32_t AudioStreamInternal::getDeviceBufferSize() const { return mDeviceBufferSizeInFrames; } int32_t AudioStreamInternal::getBufferCapacity() const { return mBufferCapacityInFrames; } int32_t AudioStreamInternal::getDeviceBufferCapacity() const { return mDeviceBufferCapacityInFrames; } bool AudioStreamInternal::isClockModelInControl() const { return isActive() && mAudioEndpoint->isFreeRunning() && mClockModel.isRunning(); }