/* * Copyright (C) 2017 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 "AAudioServiceStreamShared" //#define LOG_NDEBUG 0 #include #include #include #include #include #include "binding/IAAudioService.h" #include "binding/AAudioServiceMessage.h" #include "AAudioServiceStreamBase.h" #include "AAudioServiceStreamShared.h" #include "AAudioEndpointManager.h" #include "AAudioService.h" #include "AAudioServiceEndpoint.h" using namespace android; using namespace aaudio; #define MIN_BURSTS_PER_BUFFER 2 #define DEFAULT_BURSTS_PER_BUFFER 16 // This is an arbitrary range. TODO review. #define MAX_FRAMES_PER_BUFFER (32 * 1024) AAudioServiceStreamShared::AAudioServiceStreamShared(AAudioService &audioService) : AAudioServiceStreamBase(audioService) , mTimestampPositionOffset(0) , mXRunCount(0) { } std::string AAudioServiceStreamShared::dumpHeader() { std::stringstream result; result << AAudioServiceStreamBase::dumpHeader(); result << " Write# Read# Avail XRuns"; return result.str(); } std::string AAudioServiceStreamShared::dump() const { std::stringstream result; result << AAudioServiceStreamBase::dump(); auto fifo = mAudioDataQueue->getFifoBuffer(); int32_t readCounter = fifo->getReadCounter(); int32_t writeCounter = fifo->getWriteCounter(); result << std::setw(10) << writeCounter; result << std::setw(10) << readCounter; result << std::setw(8) << (writeCounter - readCounter); result << std::setw(8) << getXRunCount(); return result.str(); } int32_t AAudioServiceStreamShared::calculateBufferCapacity(int32_t requestedCapacityFrames, int32_t framesPerBurst) { if (requestedCapacityFrames > MAX_FRAMES_PER_BUFFER) { ALOGE("calculateBufferCapacity() requested capacity %d > max %d", requestedCapacityFrames, MAX_FRAMES_PER_BUFFER); return AAUDIO_ERROR_OUT_OF_RANGE; } // Determine how many bursts will fit in the buffer. int32_t numBursts; if (requestedCapacityFrames == AAUDIO_UNSPECIFIED) { // Use fewer bursts if default is too many. if ((DEFAULT_BURSTS_PER_BUFFER * framesPerBurst) > MAX_FRAMES_PER_BUFFER) { numBursts = MAX_FRAMES_PER_BUFFER / framesPerBurst; } else { numBursts = DEFAULT_BURSTS_PER_BUFFER; } } else { // round up to nearest burst boundary numBursts = (requestedCapacityFrames + framesPerBurst - 1) / framesPerBurst; } // Clip to bare minimum. if (numBursts < MIN_BURSTS_PER_BUFFER) { numBursts = MIN_BURSTS_PER_BUFFER; } // Check for numeric overflow. if (numBursts > 0x8000 || framesPerBurst > 0x8000) { ALOGE("calculateBufferCapacity() overflow, capacity = %d * %d", numBursts, framesPerBurst); return AAUDIO_ERROR_OUT_OF_RANGE; } int32_t capacityInFrames = numBursts * framesPerBurst; // Final range check. if (capacityInFrames > MAX_FRAMES_PER_BUFFER) { ALOGE("calculateBufferCapacity() calc capacity %d > max %d", capacityInFrames, MAX_FRAMES_PER_BUFFER); return AAUDIO_ERROR_OUT_OF_RANGE; } ALOGV("calculateBufferCapacity() requested %d frames, actual = %d", requestedCapacityFrames, capacityInFrames); return capacityInFrames; } aaudio_result_t AAudioServiceStreamShared::open(const aaudio::AAudioStreamRequest &request) { sp keep(this); if (request.getConstantConfiguration().getSharingMode() != AAUDIO_SHARING_MODE_SHARED) { ALOGE("%s() sharingMode mismatch %d", __func__, request.getConstantConfiguration().getSharingMode()); return AAUDIO_ERROR_INTERNAL; } aaudio_result_t result = AAudioServiceStreamBase::open(request); if (result != AAUDIO_OK) { return result; } const AAudioStreamConfiguration &configurationInput = request.getConstantConfiguration(); sp endpoint = mServiceEndpointWeak.promote(); if (endpoint == nullptr) { result = AAUDIO_ERROR_INVALID_STATE; goto error; } // Is the request compatible with the shared endpoint? setFormat(configurationInput.getFormat()); if (getFormat() == AUDIO_FORMAT_DEFAULT) { setFormat(AUDIO_FORMAT_PCM_FLOAT); } else if (getFormat() != AUDIO_FORMAT_PCM_FLOAT) { ALOGD("%s() audio_format_t mAudioFormat = %d, need FLOAT", __func__, getFormat()); result = AAUDIO_ERROR_INVALID_FORMAT; goto error; } setSampleRate(configurationInput.getSampleRate()); if (getSampleRate() == AAUDIO_UNSPECIFIED) { setSampleRate(endpoint->getSampleRate()); } else if (getSampleRate() != endpoint->getSampleRate()) { ALOGD("%s() mSampleRate = %d, need %d", __func__, getSampleRate(), endpoint->getSampleRate()); result = AAUDIO_ERROR_INVALID_RATE; goto error; } setSamplesPerFrame(configurationInput.getSamplesPerFrame()); if (getSamplesPerFrame() == AAUDIO_UNSPECIFIED) { setSamplesPerFrame(endpoint->getSamplesPerFrame()); } else if (getSamplesPerFrame() != endpoint->getSamplesPerFrame()) { ALOGD("%s() mSamplesPerFrame = %d, need %d", __func__, getSamplesPerFrame(), endpoint->getSamplesPerFrame()); result = AAUDIO_ERROR_OUT_OF_RANGE; goto error; } setBufferCapacity(calculateBufferCapacity(configurationInput.getBufferCapacity(), mFramesPerBurst)); if (getBufferCapacity() < 0) { result = getBufferCapacity(); // negative error code setBufferCapacity(0); goto error; } { std::lock_guard lock(mAudioDataQueueLock); // Create audio data shared memory buffer for client. mAudioDataQueue = new SharedRingBuffer(); result = mAudioDataQueue->allocate(calculateBytesPerFrame(), getBufferCapacity()); if (result != AAUDIO_OK) { ALOGE("%s() could not allocate FIFO with %d frames", __func__, getBufferCapacity()); result = AAUDIO_ERROR_NO_MEMORY; goto error; } } result = endpoint->registerStream(keep); if (result != AAUDIO_OK) { goto error; } setState(AAUDIO_STREAM_STATE_OPEN); return AAUDIO_OK; error: close(); return result; } aaudio_result_t AAudioServiceStreamShared::close() { aaudio_result_t result = AAudioServiceStreamBase::close(); { std::lock_guard lock(mAudioDataQueueLock); delete mAudioDataQueue; mAudioDataQueue = nullptr; } return result; } /** * Get an immutable description of the data queue created by this service. */ aaudio_result_t AAudioServiceStreamShared::getAudioDataDescription( AudioEndpointParcelable &parcelable) { std::lock_guard lock(mAudioDataQueueLock); if (mAudioDataQueue == nullptr) { ALOGW("%s(): mUpMessageQueue null! - stream not open", __func__); return AAUDIO_ERROR_NULL; } // Gather information on the data queue. mAudioDataQueue->fillParcelable(parcelable, parcelable.mDownDataQueueParcelable); parcelable.mDownDataQueueParcelable.setFramesPerBurst(getFramesPerBurst()); return AAUDIO_OK; } void AAudioServiceStreamShared::markTransferTime(Timestamp ×tamp) { mAtomicStreamTimestamp.write(timestamp); } // Get timestamp that was written by mixer or distributor. aaudio_result_t AAudioServiceStreamShared::getFreeRunningPosition(int64_t *positionFrames, int64_t *timeNanos) { // TODO Get presentation timestamp from the HAL if (mAtomicStreamTimestamp.isValid()) { Timestamp timestamp = mAtomicStreamTimestamp.read(); *positionFrames = timestamp.getPosition(); *timeNanos = timestamp.getNanoseconds(); return AAUDIO_OK; } else { return AAUDIO_ERROR_UNAVAILABLE; } } // Get timestamp from lower level service. aaudio_result_t AAudioServiceStreamShared::getHardwareTimestamp(int64_t *positionFrames, int64_t *timeNanos) { int64_t position = 0; sp endpoint = mServiceEndpointWeak.promote(); if (endpoint == nullptr) { ALOGW("%s() has no endpoint", __func__); return AAUDIO_ERROR_INVALID_STATE; } aaudio_result_t result = endpoint->getTimestamp(&position, timeNanos); if (result == AAUDIO_OK) { int64_t offset = mTimestampPositionOffset.load(); // TODO, do not go below starting value position -= offset; // Offset from shared MMAP stream ALOGV("%s() %8lld = %8lld - %8lld", __func__, (long long) position, (long long) (position + offset), (long long) offset); } *positionFrames = position; return result; }