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1738 lines
61 KiB
1738 lines
61 KiB
/*
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**
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** Copyright 2008, The Android Open Source Project
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**
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** Licensed under the Apache License, Version 2.0 (the "License");
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** you may not use this file except in compliance with the License.
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** You may obtain a copy of the License at
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**
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** http://www.apache.org/licenses/LICENSE-2.0
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**
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** Unless required by applicable law or agreed to in writing, software
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** distributed under the License is distributed on an "AS IS" BASIS,
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** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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** See the License for the specific language governing permissions and
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** limitations under the License.
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*/
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//#define LOG_NDEBUG 0
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#define LOG_TAG "AudioRecord"
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#include <inttypes.h>
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#include <android-base/macros.h>
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#include <sys/resource.h>
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#include <audiomanager/AudioManager.h>
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#include <audiomanager/IAudioManager.h>
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#include <binder/Binder.h>
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#include <binder/IPCThreadState.h>
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#include <binder/IServiceManager.h>
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#include <media/AudioRecord.h>
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#include <utils/Log.h>
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#include <private/media/AudioTrackShared.h>
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#include <processgroup/sched_policy.h>
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#include <media/IAudioFlinger.h>
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#include <media/MediaMetricsItem.h>
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#include <media/TypeConverter.h>
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#define WAIT_PERIOD_MS 10
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namespace android {
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using android::content::AttributionSourceState;
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using aidl_utils::statusTFromBinderStatus;
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// ---------------------------------------------------------------------------
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// static
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status_t AudioRecord::getMinFrameCount(
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size_t* frameCount,
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uint32_t sampleRate,
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audio_format_t format,
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audio_channel_mask_t channelMask)
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{
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if (frameCount == NULL) {
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return BAD_VALUE;
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}
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size_t size;
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status_t status = AudioSystem::getInputBufferSize(sampleRate, format, channelMask, &size);
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if (status != NO_ERROR) {
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ALOGE("%s(): AudioSystem could not query the input buffer size for"
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" sampleRate %u, format %#x, channelMask %#x; status %d",
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__func__, sampleRate, format, channelMask, status);
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return status;
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}
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// We double the size of input buffer for ping pong use of record buffer.
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// Assumes audio_is_linear_pcm(format)
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if ((*frameCount = (size * 2) / (audio_channel_count_from_in_mask(channelMask) *
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audio_bytes_per_sample(format))) == 0) {
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ALOGE("%s(): Unsupported configuration: sampleRate %u, format %#x, channelMask %#x",
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__func__, sampleRate, format, channelMask);
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return BAD_VALUE;
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}
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return NO_ERROR;
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}
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// ---------------------------------------------------------------------------
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void AudioRecord::MediaMetrics::gather(const AudioRecord *record)
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{
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#define MM_PREFIX "android.media.audiorecord." // avoid cut-n-paste errors.
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// Java API 28 entries, do not change.
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mMetricsItem->setCString(MM_PREFIX "encoding", toString(record->mFormat).c_str());
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mMetricsItem->setCString(MM_PREFIX "source", toString(record->mAttributes.source).c_str());
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mMetricsItem->setInt32(MM_PREFIX "latency", (int32_t)record->mLatency); // bad estimate.
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mMetricsItem->setInt32(MM_PREFIX "samplerate", (int32_t)record->mSampleRate);
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mMetricsItem->setInt32(MM_PREFIX "channels", (int32_t)record->mChannelCount);
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// Non-API entries, these can change.
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mMetricsItem->setInt32(MM_PREFIX "portId", (int32_t)record->mPortId);
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mMetricsItem->setInt32(MM_PREFIX "frameCount", (int32_t)record->mFrameCount);
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mMetricsItem->setCString(MM_PREFIX "attributes", toString(record->mAttributes).c_str());
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mMetricsItem->setInt64(MM_PREFIX "channelMask", (int64_t)record->mChannelMask);
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// log total duration recording, including anything currently running.
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int64_t activeNs = 0;
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if (mStartedNs != 0) {
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activeNs = systemTime() - mStartedNs;
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}
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mMetricsItem->setDouble(MM_PREFIX "durationMs", (mDurationNs + activeNs) * 1e-6);
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mMetricsItem->setInt64(MM_PREFIX "startCount", (int64_t)mCount);
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if (mLastError != NO_ERROR) {
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mMetricsItem->setInt32(MM_PREFIX "lastError.code", (int32_t)mLastError);
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mMetricsItem->setCString(MM_PREFIX "lastError.at", mLastErrorFunc.c_str());
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}
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mMetricsItem->setCString(MM_PREFIX "logSessionId", record->mLogSessionId.c_str());
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}
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static const char *stateToString(bool active) {
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return active ? "ACTIVE" : "STOPPED";
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}
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// hand the user a snapshot of the metrics.
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status_t AudioRecord::getMetrics(mediametrics::Item * &item)
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{
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mMediaMetrics.gather(this);
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mediametrics::Item *tmp = mMediaMetrics.dup();
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if (tmp == nullptr) {
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return BAD_VALUE;
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}
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item = tmp;
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return NO_ERROR;
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}
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AudioRecord::AudioRecord(const AttributionSourceState &client)
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: mActive(false), mStatus(NO_INIT), mClientAttributionSource(client),
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mSessionId(AUDIO_SESSION_ALLOCATE), mPreviousPriority(ANDROID_PRIORITY_NORMAL),
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mPreviousSchedulingGroup(SP_DEFAULT), mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
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mRoutedDeviceId(AUDIO_PORT_HANDLE_NONE), mSelectedMicDirection(MIC_DIRECTION_UNSPECIFIED),
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mSelectedMicFieldDimension(MIC_FIELD_DIMENSION_DEFAULT)
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{
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}
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AudioRecord::AudioRecord(
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audio_source_t inputSource,
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uint32_t sampleRate,
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audio_format_t format,
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audio_channel_mask_t channelMask,
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const AttributionSourceState& client,
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size_t frameCount,
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callback_t cbf,
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void* user,
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uint32_t notificationFrames,
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audio_session_t sessionId,
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transfer_type transferType,
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audio_input_flags_t flags,
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const audio_attributes_t* pAttributes,
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audio_port_handle_t selectedDeviceId,
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audio_microphone_direction_t selectedMicDirection,
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float microphoneFieldDimension)
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: mActive(false),
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mStatus(NO_INIT),
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mClientAttributionSource(client),
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mSessionId(AUDIO_SESSION_ALLOCATE),
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mPreviousPriority(ANDROID_PRIORITY_NORMAL),
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mPreviousSchedulingGroup(SP_DEFAULT),
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mProxy(NULL)
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{
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uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(mClientAttributionSource.uid));
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pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mClientAttributionSource.pid));
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(void)set(inputSource, sampleRate, format, channelMask, frameCount, cbf, user,
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notificationFrames, false /*threadCanCallJava*/, sessionId, transferType, flags,
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uid, pid, pAttributes, selectedDeviceId, selectedMicDirection,
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microphoneFieldDimension);
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}
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AudioRecord::~AudioRecord()
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{
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mMediaMetrics.gather(this);
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mediametrics::LogItem(mMetricsId)
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.set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_DTOR)
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.set(AMEDIAMETRICS_PROP_CALLERNAME,
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mCallerName.empty()
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? AMEDIAMETRICS_PROP_CALLERNAME_VALUE_UNKNOWN
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: mCallerName.c_str())
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.set(AMEDIAMETRICS_PROP_STATUS, (int32_t)mStatus)
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.record();
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stopAndJoinCallbacks(); // checks mStatus
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if (mStatus == NO_ERROR) {
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IInterface::asBinder(mAudioRecord)->unlinkToDeath(mDeathNotifier, this);
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mAudioRecord.clear();
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mCblkMemory.clear();
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mBufferMemory.clear();
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IPCThreadState::self()->flushCommands();
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ALOGV("%s(%d): releasing session id %d",
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__func__, mPortId, mSessionId);
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pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mClientAttributionSource.pid));
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AudioSystem::releaseAudioSessionId(mSessionId, pid);
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}
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}
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void AudioRecord::stopAndJoinCallbacks() {
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// Prevent nullptr crash if it did not open properly.
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if (mStatus != NO_ERROR) return;
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// Make sure that callback function exits in the case where
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// it is looping on buffer empty condition in obtainBuffer().
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// Otherwise the callback thread will never exit.
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stop();
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if (mAudioRecordThread != 0) {
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mProxy->interrupt();
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mAudioRecordThread->requestExit(); // see comment in AudioRecord.h
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mAudioRecordThread->requestExitAndWait();
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mAudioRecordThread.clear();
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}
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// No lock here: worst case we remove a NULL callback which will be a nop
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if (mDeviceCallback != 0 && mInput != AUDIO_IO_HANDLE_NONE) {
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// This may not stop all of these device callbacks!
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// TODO: Add some sort of protection.
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AudioSystem::removeAudioDeviceCallback(this, mInput, mPortId);
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}
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}
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status_t AudioRecord::set(
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audio_source_t inputSource,
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uint32_t sampleRate,
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audio_format_t format,
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audio_channel_mask_t channelMask,
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size_t frameCount,
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callback_t cbf,
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void* user,
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uint32_t notificationFrames,
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bool threadCanCallJava,
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audio_session_t sessionId,
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transfer_type transferType,
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audio_input_flags_t flags,
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uid_t uid,
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pid_t pid,
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const audio_attributes_t* pAttributes,
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audio_port_handle_t selectedDeviceId,
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audio_microphone_direction_t selectedMicDirection,
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float microphoneFieldDimension,
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int32_t maxSharedAudioHistoryMs)
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{
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status_t status = NO_ERROR;
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uint32_t channelCount;
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// Note mPortId is not valid until the track is created, so omit mPortId in ALOG for set.
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ALOGV("%s(): inputSource %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
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"notificationFrames %u, sessionId %d, transferType %d, flags %#x, attributionSource %s"
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"uid %d, pid %d",
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__func__,
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inputSource, sampleRate, format, channelMask, frameCount, notificationFrames,
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sessionId, transferType, flags, mClientAttributionSource.toString().c_str(), uid, pid);
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// TODO b/182392553: refactor or remove
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pid_t callingPid = IPCThreadState::self()->getCallingPid();
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pid_t myPid = getpid();
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pid_t adjPid = pid;
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if (pid == -1 || (callingPid != myPid)) {
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adjPid = callingPid;
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}
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mClientAttributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(adjPid));
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uid_t adjUid = uid;
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if (uid == -1 || (callingPid != myPid)) {
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adjUid = IPCThreadState::self()->getCallingUid();
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}
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mClientAttributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(adjUid));
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mTracker.reset(new RecordingActivityTracker());
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mSelectedDeviceId = selectedDeviceId;
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mSelectedMicDirection = selectedMicDirection;
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mSelectedMicFieldDimension = microphoneFieldDimension;
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mMaxSharedAudioHistoryMs = maxSharedAudioHistoryMs;
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switch (transferType) {
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case TRANSFER_DEFAULT:
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if (cbf == NULL || threadCanCallJava) {
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transferType = TRANSFER_SYNC;
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} else {
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transferType = TRANSFER_CALLBACK;
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}
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break;
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case TRANSFER_CALLBACK:
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if (cbf == NULL) {
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ALOGE("%s(): Transfer type TRANSFER_CALLBACK but cbf == NULL", __func__);
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status = BAD_VALUE;
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goto exit;
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}
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break;
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case TRANSFER_OBTAIN:
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case TRANSFER_SYNC:
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break;
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default:
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ALOGE("%s(): Invalid transfer type %d", __func__, transferType);
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status = BAD_VALUE;
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goto exit;
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}
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mTransfer = transferType;
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// invariant that mAudioRecord != 0 is true only after set() returns successfully
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if (mAudioRecord != 0) {
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ALOGE("%s(): Track already in use", __func__);
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status = INVALID_OPERATION;
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goto exit;
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}
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if (pAttributes == NULL) {
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mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
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mAttributes.source = inputSource;
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if (inputSource == AUDIO_SOURCE_VOICE_COMMUNICATION
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|| inputSource == AUDIO_SOURCE_CAMCORDER) {
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mAttributes.flags = static_cast<audio_flags_mask_t>(
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mAttributes.flags | AUDIO_FLAG_CAPTURE_PRIVATE);
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}
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} else {
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// stream type shouldn't be looked at, this track has audio attributes
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memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t));
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ALOGV("%s(): Building AudioRecord with attributes: source=%d flags=0x%x tags=[%s]",
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__func__, mAttributes.source, mAttributes.flags, mAttributes.tags);
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}
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|
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mSampleRate = sampleRate;
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|
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// these below should probably come from the audioFlinger too...
|
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if (format == AUDIO_FORMAT_DEFAULT) {
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format = AUDIO_FORMAT_PCM_16_BIT;
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}
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|
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// validate parameters
|
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// AudioFlinger capture only supports linear PCM
|
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if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) {
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ALOGE("%s(): Format %#x is not linear pcm", __func__, format);
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status = BAD_VALUE;
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goto exit;
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}
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mFormat = format;
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if (!audio_is_input_channel(channelMask)) {
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ALOGE("%s(): Invalid channel mask %#x", __func__, channelMask);
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status = BAD_VALUE;
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goto exit;
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}
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mChannelMask = channelMask;
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channelCount = audio_channel_count_from_in_mask(channelMask);
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mChannelCount = channelCount;
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if (audio_is_linear_pcm(format)) {
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mFrameSize = channelCount * audio_bytes_per_sample(format);
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} else {
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mFrameSize = sizeof(uint8_t);
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}
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// mFrameCount is initialized in createRecord_l
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mReqFrameCount = frameCount;
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mNotificationFramesReq = notificationFrames;
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// mNotificationFramesAct is initialized in createRecord_l
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mSessionId = sessionId;
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ALOGV("%s(): mSessionId %d", __func__, mSessionId);
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mOrigFlags = mFlags = flags;
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mCbf = cbf;
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if (cbf != NULL) {
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mAudioRecordThread = new AudioRecordThread(*this);
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mAudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO);
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// thread begins in paused state, and will not reference us until start()
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}
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// create the IAudioRecord
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{
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AutoMutex lock(mLock);
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status = createRecord_l(0 /*epoch*/);
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}
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ALOGV("%s(%d): status %d", __func__, mPortId, status);
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if (status != NO_ERROR) {
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if (mAudioRecordThread != 0) {
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mAudioRecordThread->requestExit(); // see comment in AudioRecord.h
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mAudioRecordThread->requestExitAndWait();
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mAudioRecordThread.clear();
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}
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goto exit;
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}
|
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mUserData = user;
|
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// TODO: add audio hardware input latency here
|
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mLatency = (1000LL * mFrameCount) / mSampleRate;
|
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mMarkerPosition = 0;
|
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mMarkerReached = false;
|
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mNewPosition = 0;
|
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mUpdatePeriod = 0;
|
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AudioSystem::acquireAudioSessionId(mSessionId, adjPid, adjUid);
|
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mSequence = 1;
|
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mObservedSequence = mSequence;
|
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mInOverrun = false;
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mFramesRead = 0;
|
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mFramesReadServerOffset = 0;
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|
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exit:
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mStatus = status;
|
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if (status != NO_ERROR) {
|
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mMediaMetrics.markError(status, __FUNCTION__);
|
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}
|
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return status;
|
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}
|
|
|
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// -------------------------------------------------------------------------
|
|
|
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status_t AudioRecord::start(AudioSystem::sync_event_t event, audio_session_t triggerSession)
|
|
{
|
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const int64_t beginNs = systemTime();
|
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ALOGV("%s(%d): sync event %d trigger session %d", __func__, mPortId, event, triggerSession);
|
|
AutoMutex lock(mLock);
|
|
|
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status_t status = NO_ERROR;
|
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mediametrics::Defer defer([&] {
|
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mediametrics::LogItem(mMetricsId)
|
|
.set(AMEDIAMETRICS_PROP_CALLERNAME,
|
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mCallerName.empty()
|
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? AMEDIAMETRICS_PROP_CALLERNAME_VALUE_UNKNOWN
|
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: mCallerName.c_str())
|
|
.set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_START)
|
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.set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
|
|
.set(AMEDIAMETRICS_PROP_STATE, stateToString(mActive))
|
|
.set(AMEDIAMETRICS_PROP_STATUS, (int32_t)status)
|
|
.record(); });
|
|
|
|
if (mActive) {
|
|
return status;
|
|
}
|
|
|
|
// discard data in buffer
|
|
const uint32_t framesFlushed = mProxy->flush();
|
|
mFramesReadServerOffset -= mFramesRead + framesFlushed;
|
|
mFramesRead = 0;
|
|
mProxy->clearTimestamp(); // timestamp is invalid until next server push
|
|
mPreviousTimestamp.clear();
|
|
mTimestampRetrogradePositionReported = false;
|
|
mTimestampRetrogradeTimeReported = false;
|
|
|
|
// reset current position as seen by client to 0
|
|
mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition());
|
|
// force refresh of remaining frames by processAudioBuffer() as last
|
|
// read before stop could be partial.
|
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mRefreshRemaining = true;
|
|
|
|
mNewPosition = mProxy->getPosition() + mUpdatePeriod;
|
|
int32_t flags = android_atomic_acquire_load(&mCblk->mFlags);
|
|
|
|
// we reactivate markers (mMarkerPosition != 0) as the position is reset to 0.
|
|
// This is legacy behavior. This is not done in stop() to avoid a race condition
|
|
// where the last marker event is issued twice.
|
|
mMarkerReached = false;
|
|
// mActive is checked by restoreRecord_l
|
|
mActive = true;
|
|
|
|
if (!(flags & CBLK_INVALID)) {
|
|
status = statusTFromBinderStatus(mAudioRecord->start(event, triggerSession));
|
|
if (status == DEAD_OBJECT) {
|
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flags |= CBLK_INVALID;
|
|
}
|
|
}
|
|
if (flags & CBLK_INVALID) {
|
|
status = restoreRecord_l("start");
|
|
}
|
|
|
|
// Call these directly because we are already holding the lock.
|
|
mAudioRecord->setPreferredMicrophoneDirection(mSelectedMicDirection);
|
|
mAudioRecord->setPreferredMicrophoneFieldDimension(mSelectedMicFieldDimension);
|
|
|
|
if (status != NO_ERROR) {
|
|
mActive = false;
|
|
ALOGE("%s(%d): status %d", __func__, mPortId, status);
|
|
mMediaMetrics.markError(status, __FUNCTION__);
|
|
} else {
|
|
mTracker->recordingStarted();
|
|
sp<AudioRecordThread> t = mAudioRecordThread;
|
|
if (t != 0) {
|
|
t->resume();
|
|
} else {
|
|
mPreviousPriority = getpriority(PRIO_PROCESS, 0);
|
|
get_sched_policy(0, &mPreviousSchedulingGroup);
|
|
androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
|
|
}
|
|
|
|
// we've successfully started, log that time
|
|
mMediaMetrics.logStart(systemTime());
|
|
}
|
|
return status;
|
|
}
|
|
|
|
void AudioRecord::stop()
|
|
{
|
|
const int64_t beginNs = systemTime();
|
|
AutoMutex lock(mLock);
|
|
mediametrics::Defer defer([&] {
|
|
mediametrics::LogItem(mMetricsId)
|
|
.set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_STOP)
|
|
.set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
|
|
.set(AMEDIAMETRICS_PROP_STATE, stateToString(mActive))
|
|
.record(); });
|
|
|
|
ALOGV("%s(%d): mActive:%d\n", __func__, mPortId, mActive);
|
|
if (!mActive) {
|
|
return;
|
|
}
|
|
|
|
mActive = false;
|
|
mProxy->interrupt();
|
|
mAudioRecord->stop();
|
|
mTracker->recordingStopped();
|
|
|
|
// Note: legacy handling - stop does not clear record marker and
|
|
// periodic update position; we update those on start().
|
|
|
|
sp<AudioRecordThread> t = mAudioRecordThread;
|
|
if (t != 0) {
|
|
t->pause();
|
|
} else {
|
|
setpriority(PRIO_PROCESS, 0, mPreviousPriority);
|
|
set_sched_policy(0, mPreviousSchedulingGroup);
|
|
}
|
|
|
|
// we've successfully started, log that time
|
|
mMediaMetrics.logStop(systemTime());
|
|
}
|
|
|
|
bool AudioRecord::stopped() const
|
|
{
|
|
AutoMutex lock(mLock);
|
|
return !mActive;
|
|
}
|
|
|
|
status_t AudioRecord::setMarkerPosition(uint32_t marker)
|
|
{
|
|
// The only purpose of setting marker position is to get a callback
|
|
if (mCbf == NULL) {
|
|
return INVALID_OPERATION;
|
|
}
|
|
|
|
AutoMutex lock(mLock);
|
|
mMarkerPosition = marker;
|
|
mMarkerReached = false;
|
|
|
|
sp<AudioRecordThread> t = mAudioRecordThread;
|
|
if (t != 0) {
|
|
t->wake();
|
|
}
|
|
return NO_ERROR;
|
|
}
|
|
|
|
status_t AudioRecord::getMarkerPosition(uint32_t *marker) const
|
|
{
|
|
if (marker == NULL) {
|
|
return BAD_VALUE;
|
|
}
|
|
|
|
AutoMutex lock(mLock);
|
|
mMarkerPosition.getValue(marker);
|
|
|
|
return NO_ERROR;
|
|
}
|
|
|
|
status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod)
|
|
{
|
|
// The only purpose of setting position update period is to get a callback
|
|
if (mCbf == NULL) {
|
|
return INVALID_OPERATION;
|
|
}
|
|
|
|
AutoMutex lock(mLock);
|
|
mNewPosition = mProxy->getPosition() + updatePeriod;
|
|
mUpdatePeriod = updatePeriod;
|
|
|
|
sp<AudioRecordThread> t = mAudioRecordThread;
|
|
if (t != 0) {
|
|
t->wake();
|
|
}
|
|
return NO_ERROR;
|
|
}
|
|
|
|
status_t AudioRecord::getPositionUpdatePeriod(uint32_t *updatePeriod) const
|
|
{
|
|
if (updatePeriod == NULL) {
|
|
return BAD_VALUE;
|
|
}
|
|
|
|
AutoMutex lock(mLock);
|
|
*updatePeriod = mUpdatePeriod;
|
|
|
|
return NO_ERROR;
|
|
}
|
|
|
|
status_t AudioRecord::getPosition(uint32_t *position) const
|
|
{
|
|
if (position == NULL) {
|
|
return BAD_VALUE;
|
|
}
|
|
|
|
AutoMutex lock(mLock);
|
|
mProxy->getPosition().getValue(position);
|
|
|
|
return NO_ERROR;
|
|
}
|
|
|
|
uint32_t AudioRecord::getInputFramesLost() const
|
|
{
|
|
// no need to check mActive, because if inactive this will return 0, which is what we want
|
|
return AudioSystem::getInputFramesLost(getInputPrivate());
|
|
}
|
|
|
|
status_t AudioRecord::getTimestamp(ExtendedTimestamp *timestamp)
|
|
{
|
|
if (timestamp == nullptr) {
|
|
return BAD_VALUE;
|
|
}
|
|
AutoMutex lock(mLock);
|
|
status_t status = mProxy->getTimestamp(timestamp);
|
|
if (status == OK) {
|
|
timestamp->mPosition[ExtendedTimestamp::LOCATION_CLIENT] = mFramesRead;
|
|
timestamp->mTimeNs[ExtendedTimestamp::LOCATION_CLIENT] = 0;
|
|
// server side frame offset in case AudioRecord has been restored.
|
|
for (int i = ExtendedTimestamp::LOCATION_SERVER;
|
|
i < ExtendedTimestamp::LOCATION_MAX; ++i) {
|
|
if (timestamp->mTimeNs[i] >= 0) {
|
|
timestamp->mPosition[i] += mFramesReadServerOffset;
|
|
}
|
|
}
|
|
|
|
bool timestampRetrogradeTimeReported = false;
|
|
bool timestampRetrogradePositionReported = false;
|
|
for (int i = 0; i < ExtendedTimestamp::LOCATION_MAX; ++i) {
|
|
if (timestamp->mTimeNs[i] >= 0 && mPreviousTimestamp.mTimeNs[i] >= 0) {
|
|
if (timestamp->mTimeNs[i] < mPreviousTimestamp.mTimeNs[i]) {
|
|
if (!mTimestampRetrogradeTimeReported) {
|
|
ALOGD("%s: retrograde time adjusting [%d] current:%lld to previous:%lld",
|
|
__func__, i, (long long)timestamp->mTimeNs[i],
|
|
(long long)mPreviousTimestamp.mTimeNs[i]);
|
|
timestampRetrogradeTimeReported = true;
|
|
}
|
|
timestamp->mTimeNs[i] = mPreviousTimestamp.mTimeNs[i];
|
|
}
|
|
if (timestamp->mPosition[i] < mPreviousTimestamp.mPosition[i]) {
|
|
if (!mTimestampRetrogradePositionReported) {
|
|
ALOGD("%s: retrograde position"
|
|
" adjusting [%d] current:%lld to previous:%lld",
|
|
__func__, i, (long long)timestamp->mPosition[i],
|
|
(long long)mPreviousTimestamp.mPosition[i]);
|
|
timestampRetrogradePositionReported = true;
|
|
}
|
|
timestamp->mPosition[i] = mPreviousTimestamp.mPosition[i];
|
|
}
|
|
}
|
|
}
|
|
mPreviousTimestamp = *timestamp;
|
|
if (timestampRetrogradeTimeReported) {
|
|
mTimestampRetrogradeTimeReported = true;
|
|
}
|
|
if (timestampRetrogradePositionReported) {
|
|
mTimestampRetrogradePositionReported = true;
|
|
}
|
|
}
|
|
return status;
|
|
}
|
|
|
|
// ---- Explicit Routing ---------------------------------------------------
|
|
status_t AudioRecord::setInputDevice(audio_port_handle_t deviceId) {
|
|
AutoMutex lock(mLock);
|
|
if (mSelectedDeviceId != deviceId) {
|
|
mSelectedDeviceId = deviceId;
|
|
if (mStatus == NO_ERROR) {
|
|
// stop capture so that audio policy manager does not reject the new instance start request
|
|
// as only one capture can be active at a time.
|
|
if (mAudioRecord != 0 && mActive) {
|
|
mAudioRecord->stop();
|
|
}
|
|
android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
|
|
mProxy->interrupt();
|
|
}
|
|
}
|
|
return NO_ERROR;
|
|
}
|
|
|
|
audio_port_handle_t AudioRecord::getInputDevice() {
|
|
AutoMutex lock(mLock);
|
|
return mSelectedDeviceId;
|
|
}
|
|
|
|
// must be called with mLock held
|
|
void AudioRecord::updateRoutedDeviceId_l()
|
|
{
|
|
// if the record is inactive, do not update actual device as the input stream maybe routed
|
|
// from a device not relevant to this client because of other active use cases.
|
|
if (!mActive) {
|
|
return;
|
|
}
|
|
if (mInput != AUDIO_IO_HANDLE_NONE) {
|
|
audio_port_handle_t deviceId = AudioSystem::getDeviceIdForIo(mInput);
|
|
if (deviceId != AUDIO_PORT_HANDLE_NONE) {
|
|
mRoutedDeviceId = deviceId;
|
|
}
|
|
}
|
|
}
|
|
|
|
audio_port_handle_t AudioRecord::getRoutedDeviceId() {
|
|
AutoMutex lock(mLock);
|
|
updateRoutedDeviceId_l();
|
|
return mRoutedDeviceId;
|
|
}
|
|
|
|
status_t AudioRecord::dump(int fd, const Vector<String16>& args __unused) const
|
|
{
|
|
String8 result;
|
|
|
|
result.append(" AudioRecord::dump\n");
|
|
result.appendFormat(" id(%d) status(%d), active(%d), session Id(%d)\n",
|
|
mPortId, mStatus, mActive, mSessionId);
|
|
result.appendFormat(" flags(%#x), req. flags(%#x), audio source(%d)\n",
|
|
mFlags, mOrigFlags, mAttributes.source);
|
|
result.appendFormat(" format(%#x), channel mask(%#x), channel count(%u), sample rate(%u)\n",
|
|
mFormat, mChannelMask, mChannelCount, mSampleRate);
|
|
result.appendFormat(" frame count(%zu), req. frame count(%zu)\n",
|
|
mFrameCount, mReqFrameCount);
|
|
result.appendFormat(" notif. frame count(%u), req. notif. frame count(%u)\n",
|
|
mNotificationFramesAct, mNotificationFramesReq);
|
|
result.appendFormat(" input(%d), latency(%u), selected device Id(%d), routed device Id(%d)\n",
|
|
mInput, mLatency, mSelectedDeviceId, mRoutedDeviceId);
|
|
result.appendFormat(" mic direction(%d) mic field dimension(%f)",
|
|
mSelectedMicDirection, mSelectedMicFieldDimension);
|
|
::write(fd, result.string(), result.size());
|
|
return NO_ERROR;
|
|
}
|
|
|
|
// -------------------------------------------------------------------------
|
|
// TODO Move this macro to a common header file for enum to string conversion in audio framework.
|
|
#define MEDIA_CASE_ENUM(name) case name: return #name
|
|
const char * AudioRecord::convertTransferToText(transfer_type transferType) {
|
|
switch (transferType) {
|
|
MEDIA_CASE_ENUM(TRANSFER_DEFAULT);
|
|
MEDIA_CASE_ENUM(TRANSFER_CALLBACK);
|
|
MEDIA_CASE_ENUM(TRANSFER_OBTAIN);
|
|
MEDIA_CASE_ENUM(TRANSFER_SYNC);
|
|
default:
|
|
return "UNRECOGNIZED";
|
|
}
|
|
}
|
|
|
|
// must be called with mLock held
|
|
status_t AudioRecord::createRecord_l(const Modulo<uint32_t> &epoch)
|
|
{
|
|
const int64_t beginNs = systemTime();
|
|
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
|
|
IAudioFlinger::CreateRecordInput input;
|
|
IAudioFlinger::CreateRecordOutput output;
|
|
audio_session_t originalSessionId;
|
|
void *iMemPointer;
|
|
audio_track_cblk_t* cblk;
|
|
status_t status;
|
|
static const int32_t kMaxCreateAttempts = 3;
|
|
int32_t remainingAttempts = kMaxCreateAttempts;
|
|
|
|
if (audioFlinger == 0) {
|
|
ALOGE("%s(%d): Could not get audioflinger", __func__, mPortId);
|
|
status = NO_INIT;
|
|
goto exit;
|
|
}
|
|
|
|
// mFlags (not mOrigFlags) is modified depending on whether fast request is accepted.
|
|
// After fast request is denied, we will request again if IAudioRecord is re-created.
|
|
|
|
// Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger,
|
|
// we must release it ourselves if anything goes wrong.
|
|
|
|
// Client can only express a preference for FAST. Server will perform additional tests.
|
|
if (mFlags & AUDIO_INPUT_FLAG_FAST) {
|
|
bool useCaseAllowed =
|
|
// any of these use cases:
|
|
// use case 1: callback transfer mode
|
|
(mTransfer == TRANSFER_CALLBACK) ||
|
|
// use case 2: blocking read mode
|
|
// The default buffer capacity at 48 kHz is 2048 frames, or ~42.6 ms.
|
|
// That's enough for double-buffering with our standard 20 ms rule of thumb for
|
|
// the minimum period of a non-SCHED_FIFO thread.
|
|
// This is needed so that AAudio apps can do a low latency non-blocking read from a
|
|
// callback running with SCHED_FIFO.
|
|
(mTransfer == TRANSFER_SYNC) ||
|
|
// use case 3: obtain/release mode
|
|
(mTransfer == TRANSFER_OBTAIN);
|
|
if (!useCaseAllowed) {
|
|
ALOGD("%s(%d): AUDIO_INPUT_FLAG_FAST denied, incompatible transfer = %s",
|
|
__func__, mPortId,
|
|
convertTransferToText(mTransfer));
|
|
mFlags = (audio_input_flags_t) (mFlags & ~(AUDIO_INPUT_FLAG_FAST |
|
|
AUDIO_INPUT_FLAG_RAW));
|
|
}
|
|
}
|
|
|
|
input.attr = mAttributes;
|
|
input.config.sample_rate = mSampleRate;
|
|
input.config.channel_mask = mChannelMask;
|
|
input.config.format = mFormat;
|
|
input.clientInfo.attributionSource = mClientAttributionSource;
|
|
input.clientInfo.clientTid = -1;
|
|
if (mFlags & AUDIO_INPUT_FLAG_FAST) {
|
|
if (mAudioRecordThread != 0) {
|
|
input.clientInfo.clientTid = mAudioRecordThread->getTid();
|
|
}
|
|
}
|
|
input.riid = mTracker->getRiid();
|
|
|
|
input.flags = mFlags;
|
|
// The notification frame count is the period between callbacks, as suggested by the client
|
|
// but moderated by the server. For record, the calculations are done entirely on server side.
|
|
input.frameCount = mReqFrameCount;
|
|
input.notificationFrameCount = mNotificationFramesReq;
|
|
input.selectedDeviceId = mSelectedDeviceId;
|
|
input.sessionId = mSessionId;
|
|
originalSessionId = mSessionId;
|
|
input.maxSharedAudioHistoryMs = mMaxSharedAudioHistoryMs;
|
|
|
|
do {
|
|
media::CreateRecordResponse response;
|
|
status = audioFlinger->createRecord(VALUE_OR_FATAL(input.toAidl()), response);
|
|
output = VALUE_OR_FATAL(IAudioFlinger::CreateRecordOutput::fromAidl(response));
|
|
if (status == NO_ERROR) {
|
|
break;
|
|
}
|
|
if (status != FAILED_TRANSACTION || --remainingAttempts <= 0) {
|
|
ALOGE("%s(%d): AudioFlinger could not create record track, status: %d",
|
|
__func__, mPortId, status);
|
|
goto exit;
|
|
}
|
|
// FAILED_TRANSACTION happens under very specific conditions causing a state mismatch
|
|
// between audio policy manager and audio flinger during the input stream open sequence
|
|
// and can be recovered by retrying.
|
|
// Leave time for race condition to clear before retrying and randomize delay
|
|
// to reduce the probability of concurrent retries in locked steps.
|
|
usleep((20 + rand() % 30) * 10000);
|
|
} while (1);
|
|
|
|
ALOG_ASSERT(output.audioRecord != 0);
|
|
|
|
// AudioFlinger now owns the reference to the I/O handle,
|
|
// so we are no longer responsible for releasing it.
|
|
|
|
mAwaitBoost = false;
|
|
if (output.flags & AUDIO_INPUT_FLAG_FAST) {
|
|
ALOGI("%s(%d): AUDIO_INPUT_FLAG_FAST successful; frameCount %zu -> %zu",
|
|
__func__, mPortId,
|
|
mReqFrameCount, output.frameCount);
|
|
mAwaitBoost = true;
|
|
}
|
|
mFlags = output.flags;
|
|
mRoutedDeviceId = output.selectedDeviceId;
|
|
mSessionId = output.sessionId;
|
|
mSampleRate = output.sampleRate;
|
|
|
|
if (output.cblk == 0) {
|
|
ALOGE("%s(%d): Could not get control block", __func__, mPortId);
|
|
status = NO_INIT;
|
|
goto exit;
|
|
}
|
|
// TODO: Using unsecurePointer() has some associated security pitfalls
|
|
// (see declaration for details).
|
|
// Either document why it is safe in this case or address the
|
|
// issue (e.g. by copying).
|
|
iMemPointer = output.cblk ->unsecurePointer();
|
|
if (iMemPointer == NULL) {
|
|
ALOGE("%s(%d): Could not get control block pointer", __func__, mPortId);
|
|
status = NO_INIT;
|
|
goto exit;
|
|
}
|
|
cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
|
|
|
|
// Starting address of buffers in shared memory.
|
|
// The buffers are either immediately after the control block,
|
|
// or in a separate area at discretion of server.
|
|
void *buffers;
|
|
if (output.buffers == 0) {
|
|
buffers = cblk + 1;
|
|
} else {
|
|
// TODO: Using unsecurePointer() has some associated security pitfalls
|
|
// (see declaration for details).
|
|
// Either document why it is safe in this case or address the
|
|
// issue (e.g. by copying).
|
|
buffers = output.buffers->unsecurePointer();
|
|
if (buffers == NULL) {
|
|
ALOGE("%s(%d): Could not get buffer pointer", __func__, mPortId);
|
|
status = NO_INIT;
|
|
goto exit;
|
|
}
|
|
}
|
|
|
|
// invariant that mAudioRecord != 0 is true only after set() returns successfully
|
|
if (mAudioRecord != 0) {
|
|
IInterface::asBinder(mAudioRecord)->unlinkToDeath(mDeathNotifier, this);
|
|
mDeathNotifier.clear();
|
|
}
|
|
mAudioRecord = output.audioRecord;
|
|
mCblkMemory = output.cblk;
|
|
mBufferMemory = output.buffers;
|
|
IPCThreadState::self()->flushCommands();
|
|
|
|
mCblk = cblk;
|
|
// note that output.frameCount is the (possibly revised) value of mReqFrameCount
|
|
if (output.frameCount < mReqFrameCount || (mReqFrameCount == 0 && output.frameCount == 0)) {
|
|
ALOGW("%s(%d): Requested frameCount %zu but received frameCount %zu",
|
|
__func__, output.portId,
|
|
mReqFrameCount, output.frameCount);
|
|
}
|
|
|
|
// Make sure that application is notified with sufficient margin before overrun.
|
|
// The computation is done on server side.
|
|
if (mNotificationFramesReq > 0 && output.notificationFrameCount != mNotificationFramesReq) {
|
|
ALOGW("%s(%d): Server adjusted notificationFrames from %u to %zu for frameCount %zu",
|
|
__func__, output.portId,
|
|
mNotificationFramesReq, output.notificationFrameCount, output.frameCount);
|
|
}
|
|
mNotificationFramesAct = (uint32_t)output.notificationFrameCount;
|
|
|
|
//mInput != input includes the case where mInput == AUDIO_IO_HANDLE_NONE for first creation
|
|
if (mDeviceCallback != 0) {
|
|
if (mInput != AUDIO_IO_HANDLE_NONE) {
|
|
AudioSystem::removeAudioDeviceCallback(this, mInput, mPortId);
|
|
}
|
|
AudioSystem::addAudioDeviceCallback(this, output.inputId, output.portId);
|
|
}
|
|
|
|
if (!mSharedAudioPackageName.empty()) {
|
|
mAudioRecord->shareAudioHistory(mSharedAudioPackageName, mSharedAudioStartMs);
|
|
}
|
|
|
|
mPortId = output.portId;
|
|
// We retain a copy of the I/O handle, but don't own the reference
|
|
mInput = output.inputId;
|
|
mRefreshRemaining = true;
|
|
|
|
mFrameCount = output.frameCount;
|
|
// If IAudioRecord is re-created, don't let the requested frameCount
|
|
// decrease. This can confuse clients that cache frameCount().
|
|
if (mFrameCount > mReqFrameCount) {
|
|
mReqFrameCount = mFrameCount;
|
|
}
|
|
|
|
// update proxy
|
|
mProxy = new AudioRecordClientProxy(cblk, buffers, mFrameCount, mFrameSize);
|
|
mProxy->setEpoch(epoch);
|
|
mProxy->setMinimum(mNotificationFramesAct);
|
|
|
|
mDeathNotifier = new DeathNotifier(this);
|
|
IInterface::asBinder(mAudioRecord)->linkToDeath(mDeathNotifier, this);
|
|
|
|
mMetricsId = std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_RECORD) + std::to_string(mPortId);
|
|
mediametrics::LogItem(mMetricsId)
|
|
.set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE)
|
|
.set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
|
|
// the following are immutable (at least until restore)
|
|
.set(AMEDIAMETRICS_PROP_FLAGS, toString(mFlags).c_str())
|
|
.set(AMEDIAMETRICS_PROP_ORIGINALFLAGS, toString(mOrigFlags).c_str())
|
|
.set(AMEDIAMETRICS_PROP_SESSIONID, (int32_t)mSessionId)
|
|
.set(AMEDIAMETRICS_PROP_TRACKID, mPortId)
|
|
.set(AMEDIAMETRICS_PROP_LOGSESSIONID, mLogSessionId)
|
|
.set(AMEDIAMETRICS_PROP_SOURCE, toString(mAttributes.source).c_str())
|
|
.set(AMEDIAMETRICS_PROP_THREADID, (int32_t)output.inputId)
|
|
.set(AMEDIAMETRICS_PROP_SELECTEDDEVICEID, (int32_t)mSelectedDeviceId)
|
|
.set(AMEDIAMETRICS_PROP_ROUTEDDEVICEID, (int32_t)mRoutedDeviceId)
|
|
.set(AMEDIAMETRICS_PROP_ENCODING, toString(mFormat).c_str())
|
|
.set(AMEDIAMETRICS_PROP_CHANNELMASK, (int32_t)mChannelMask)
|
|
.set(AMEDIAMETRICS_PROP_FRAMECOUNT, (int32_t)mFrameCount)
|
|
.set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
|
|
// the following are NOT immutable
|
|
.set(AMEDIAMETRICS_PROP_STATE, stateToString(mActive))
|
|
.set(AMEDIAMETRICS_PROP_STATUS, (int32_t)status)
|
|
.set(AMEDIAMETRICS_PROP_SELECTEDMICDIRECTION, (int32_t)mSelectedMicDirection)
|
|
.set(AMEDIAMETRICS_PROP_SELECTEDMICFIELDDIRECTION, (double)mSelectedMicFieldDimension)
|
|
.record();
|
|
|
|
exit:
|
|
mStatus = status;
|
|
// sp<IAudioTrack> track destructor will cause releaseOutput() to be called by AudioFlinger
|
|
return status;
|
|
}
|
|
|
|
status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_t *nonContig)
|
|
{
|
|
if (audioBuffer == NULL) {
|
|
if (nonContig != NULL) {
|
|
*nonContig = 0;
|
|
}
|
|
return BAD_VALUE;
|
|
}
|
|
if (mTransfer != TRANSFER_OBTAIN) {
|
|
audioBuffer->frameCount = 0;
|
|
audioBuffer->size = 0;
|
|
audioBuffer->raw = NULL;
|
|
if (nonContig != NULL) {
|
|
*nonContig = 0;
|
|
}
|
|
return INVALID_OPERATION;
|
|
}
|
|
|
|
const struct timespec *requested;
|
|
struct timespec timeout;
|
|
if (waitCount == -1) {
|
|
requested = &ClientProxy::kForever;
|
|
} else if (waitCount == 0) {
|
|
requested = &ClientProxy::kNonBlocking;
|
|
} else if (waitCount > 0) {
|
|
time_t ms = WAIT_PERIOD_MS * (time_t) waitCount;
|
|
timeout.tv_sec = ms / 1000;
|
|
timeout.tv_nsec = (long) (ms % 1000) * 1000000;
|
|
requested = &timeout;
|
|
} else {
|
|
ALOGE("%s(%d): invalid waitCount %d", __func__, mPortId, waitCount);
|
|
requested = NULL;
|
|
}
|
|
return obtainBuffer(audioBuffer, requested, NULL /*elapsed*/, nonContig);
|
|
}
|
|
|
|
status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
|
|
struct timespec *elapsed, size_t *nonContig)
|
|
{
|
|
// previous and new IAudioRecord sequence numbers are used to detect track re-creation
|
|
uint32_t oldSequence = 0;
|
|
|
|
Proxy::Buffer buffer;
|
|
status_t status = NO_ERROR;
|
|
|
|
static const int32_t kMaxTries = 5;
|
|
int32_t tryCounter = kMaxTries;
|
|
|
|
do {
|
|
// obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to
|
|
// keep them from going away if another thread re-creates the track during obtainBuffer()
|
|
sp<AudioRecordClientProxy> proxy;
|
|
sp<IMemory> iMem;
|
|
sp<IMemory> bufferMem;
|
|
{
|
|
// start of lock scope
|
|
AutoMutex lock(mLock);
|
|
|
|
uint32_t newSequence = mSequence;
|
|
// did previous obtainBuffer() fail due to media server death or voluntary invalidation?
|
|
if (status == DEAD_OBJECT) {
|
|
// re-create track, unless someone else has already done so
|
|
if (newSequence == oldSequence) {
|
|
status = restoreRecord_l("obtainBuffer");
|
|
if (status != NO_ERROR) {
|
|
buffer.mFrameCount = 0;
|
|
buffer.mRaw = NULL;
|
|
buffer.mNonContig = 0;
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
oldSequence = newSequence;
|
|
|
|
// Keep the extra references
|
|
proxy = mProxy;
|
|
iMem = mCblkMemory;
|
|
bufferMem = mBufferMemory;
|
|
|
|
// Non-blocking if track is stopped
|
|
if (!mActive) {
|
|
requested = &ClientProxy::kNonBlocking;
|
|
}
|
|
|
|
} // end of lock scope
|
|
|
|
buffer.mFrameCount = audioBuffer->frameCount;
|
|
// FIXME starts the requested timeout and elapsed over from scratch
|
|
status = proxy->obtainBuffer(&buffer, requested, elapsed);
|
|
|
|
} while ((status == DEAD_OBJECT) && (tryCounter-- > 0));
|
|
|
|
audioBuffer->frameCount = buffer.mFrameCount;
|
|
audioBuffer->size = buffer.mFrameCount * mFrameSize;
|
|
audioBuffer->raw = buffer.mRaw;
|
|
audioBuffer->sequence = oldSequence;
|
|
if (nonContig != NULL) {
|
|
*nonContig = buffer.mNonContig;
|
|
}
|
|
return status;
|
|
}
|
|
|
|
void AudioRecord::releaseBuffer(const Buffer* audioBuffer)
|
|
{
|
|
// FIXME add error checking on mode, by adding an internal version
|
|
|
|
size_t stepCount = audioBuffer->size / mFrameSize;
|
|
if (stepCount == 0) {
|
|
return;
|
|
}
|
|
|
|
Proxy::Buffer buffer;
|
|
buffer.mFrameCount = stepCount;
|
|
buffer.mRaw = audioBuffer->raw;
|
|
|
|
AutoMutex lock(mLock);
|
|
if (audioBuffer->sequence != mSequence) {
|
|
// This Buffer came from a different IAudioRecord instance, so ignore the releaseBuffer
|
|
ALOGD("%s is no-op due to IAudioRecord sequence mismatch %u != %u",
|
|
__func__, audioBuffer->sequence, mSequence);
|
|
return;
|
|
}
|
|
mInOverrun = false;
|
|
mProxy->releaseBuffer(&buffer);
|
|
|
|
// the server does not automatically disable recorder on overrun, so no need to restart
|
|
}
|
|
|
|
audio_io_handle_t AudioRecord::getInputPrivate() const
|
|
{
|
|
AutoMutex lock(mLock);
|
|
return mInput;
|
|
}
|
|
|
|
// -------------------------------------------------------------------------
|
|
|
|
ssize_t AudioRecord::read(void* buffer, size_t userSize, bool blocking)
|
|
{
|
|
if (mTransfer != TRANSFER_SYNC) {
|
|
return INVALID_OPERATION;
|
|
}
|
|
|
|
if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
|
|
// Validation. user is most-likely passing an error code, and it would
|
|
// make the return value ambiguous (actualSize vs error).
|
|
ALOGE("%s(%d) (buffer=%p, size=%zu (%zu)",
|
|
__func__, mPortId, buffer, userSize, userSize);
|
|
return BAD_VALUE;
|
|
}
|
|
|
|
ssize_t read = 0;
|
|
Buffer audioBuffer;
|
|
|
|
while (userSize >= mFrameSize) {
|
|
audioBuffer.frameCount = userSize / mFrameSize;
|
|
|
|
status_t err = obtainBuffer(&audioBuffer,
|
|
blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking);
|
|
if (err < 0) {
|
|
if (read > 0) {
|
|
break;
|
|
}
|
|
if (err == TIMED_OUT || err == -EINTR) {
|
|
err = WOULD_BLOCK;
|
|
}
|
|
return ssize_t(err);
|
|
}
|
|
|
|
size_t bytesRead = audioBuffer.size;
|
|
memcpy(buffer, audioBuffer.i8, bytesRead);
|
|
buffer = ((char *) buffer) + bytesRead;
|
|
userSize -= bytesRead;
|
|
read += bytesRead;
|
|
|
|
releaseBuffer(&audioBuffer);
|
|
}
|
|
if (read > 0) {
|
|
mFramesRead += read / mFrameSize;
|
|
// mFramesReadTime = systemTime(SYSTEM_TIME_MONOTONIC); // not provided at this time.
|
|
}
|
|
return read;
|
|
}
|
|
|
|
// -------------------------------------------------------------------------
|
|
|
|
nsecs_t AudioRecord::processAudioBuffer()
|
|
{
|
|
mLock.lock();
|
|
if (mAwaitBoost) {
|
|
mAwaitBoost = false;
|
|
mLock.unlock();
|
|
static const int32_t kMaxTries = 5;
|
|
int32_t tryCounter = kMaxTries;
|
|
uint32_t pollUs = 10000;
|
|
do {
|
|
int policy = sched_getscheduler(0) & ~SCHED_RESET_ON_FORK;
|
|
if (policy == SCHED_FIFO || policy == SCHED_RR) {
|
|
break;
|
|
}
|
|
usleep(pollUs);
|
|
pollUs <<= 1;
|
|
} while (tryCounter-- > 0);
|
|
if (tryCounter < 0) {
|
|
ALOGE("%s(%d): did not receive expected priority boost on time", __func__, mPortId);
|
|
}
|
|
// Run again immediately
|
|
return 0;
|
|
}
|
|
|
|
// Can only reference mCblk while locked
|
|
int32_t flags = android_atomic_and(~CBLK_OVERRUN, &mCblk->mFlags);
|
|
|
|
// Check for track invalidation
|
|
if (flags & CBLK_INVALID) {
|
|
(void) restoreRecord_l("processAudioBuffer");
|
|
mLock.unlock();
|
|
// Run again immediately, but with a new IAudioRecord
|
|
return 0;
|
|
}
|
|
|
|
bool active = mActive;
|
|
|
|
// Manage overrun callback, must be done under lock to avoid race with releaseBuffer()
|
|
bool newOverrun = false;
|
|
if (flags & CBLK_OVERRUN) {
|
|
if (!mInOverrun) {
|
|
mInOverrun = true;
|
|
newOverrun = true;
|
|
}
|
|
}
|
|
|
|
// Get current position of server
|
|
Modulo<uint32_t> position(mProxy->getPosition());
|
|
|
|
// Manage marker callback
|
|
bool markerReached = false;
|
|
Modulo<uint32_t> markerPosition(mMarkerPosition);
|
|
// FIXME fails for wraparound, need 64 bits
|
|
if (!mMarkerReached && markerPosition.value() > 0 && position >= markerPosition) {
|
|
mMarkerReached = markerReached = true;
|
|
}
|
|
|
|
// Determine the number of new position callback(s) that will be needed, while locked
|
|
size_t newPosCount = 0;
|
|
Modulo<uint32_t> newPosition(mNewPosition);
|
|
uint32_t updatePeriod = mUpdatePeriod;
|
|
// FIXME fails for wraparound, need 64 bits
|
|
if (updatePeriod > 0 && position >= newPosition) {
|
|
newPosCount = ((position - newPosition).value() / updatePeriod) + 1;
|
|
mNewPosition += updatePeriod * newPosCount;
|
|
}
|
|
|
|
// Cache other fields that will be needed soon
|
|
uint32_t notificationFrames = mNotificationFramesAct;
|
|
if (mRefreshRemaining) {
|
|
mRefreshRemaining = false;
|
|
mRemainingFrames = notificationFrames;
|
|
mRetryOnPartialBuffer = false;
|
|
}
|
|
size_t misalignment = mProxy->getMisalignment();
|
|
uint32_t sequence = mSequence;
|
|
|
|
// These fields don't need to be cached, because they are assigned only by set():
|
|
// mTransfer, mCbf, mUserData, mSampleRate, mFrameSize
|
|
|
|
mLock.unlock();
|
|
|
|
// perform callbacks while unlocked
|
|
if (newOverrun) {
|
|
mCbf(EVENT_OVERRUN, mUserData, NULL);
|
|
}
|
|
if (markerReached) {
|
|
mCbf(EVENT_MARKER, mUserData, &markerPosition);
|
|
}
|
|
while (newPosCount > 0) {
|
|
size_t temp = newPosition.value(); // FIXME size_t != uint32_t
|
|
mCbf(EVENT_NEW_POS, mUserData, &temp);
|
|
newPosition += updatePeriod;
|
|
newPosCount--;
|
|
}
|
|
if (mObservedSequence != sequence) {
|
|
mObservedSequence = sequence;
|
|
mCbf(EVENT_NEW_IAUDIORECORD, mUserData, NULL);
|
|
}
|
|
|
|
// if inactive, then don't run me again until re-started
|
|
if (!active) {
|
|
return NS_INACTIVE;
|
|
}
|
|
|
|
// Compute the estimated time until the next timed event (position, markers)
|
|
uint32_t minFrames = ~0;
|
|
if (!markerReached && position < markerPosition) {
|
|
minFrames = (markerPosition - position).value();
|
|
}
|
|
if (updatePeriod > 0) {
|
|
uint32_t remaining = (newPosition - position).value();
|
|
if (remaining < minFrames) {
|
|
minFrames = remaining;
|
|
}
|
|
}
|
|
|
|
// If > 0, poll periodically to recover from a stuck server. A good value is 2.
|
|
static const uint32_t kPoll = 0;
|
|
if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) {
|
|
minFrames = kPoll * notificationFrames;
|
|
}
|
|
|
|
// Convert frame units to time units
|
|
nsecs_t ns = NS_WHENEVER;
|
|
if (minFrames != (uint32_t) ~0) {
|
|
// This "fudge factor" avoids soaking CPU, and compensates for late progress by server
|
|
static const nsecs_t kFudgeNs = 10000000LL; // 10 ms
|
|
ns = ((minFrames * 1000000000LL) / mSampleRate) + kFudgeNs;
|
|
}
|
|
|
|
// If not supplying data by EVENT_MORE_DATA, then we're done
|
|
if (mTransfer != TRANSFER_CALLBACK) {
|
|
return ns;
|
|
}
|
|
|
|
struct timespec timeout;
|
|
const struct timespec *requested = &ClientProxy::kForever;
|
|
if (ns != NS_WHENEVER) {
|
|
timeout.tv_sec = ns / 1000000000LL;
|
|
timeout.tv_nsec = ns % 1000000000LL;
|
|
ALOGV("%s(%d): timeout %ld.%03d",
|
|
__func__, mPortId, timeout.tv_sec, (int) timeout.tv_nsec / 1000000);
|
|
requested = &timeout;
|
|
}
|
|
|
|
size_t readFrames = 0;
|
|
while (mRemainingFrames > 0) {
|
|
|
|
Buffer audioBuffer;
|
|
audioBuffer.frameCount = mRemainingFrames;
|
|
size_t nonContig;
|
|
status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig);
|
|
LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0),
|
|
"%s(%d): obtainBuffer() err=%d frameCount=%zu",
|
|
__func__, mPortId, err, audioBuffer.frameCount);
|
|
requested = &ClientProxy::kNonBlocking;
|
|
size_t avail = audioBuffer.frameCount + nonContig;
|
|
ALOGV("%s(%d): obtainBuffer(%u) returned %zu = %zu + %zu err %d",
|
|
__func__, mPortId, mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
|
|
if (err != NO_ERROR) {
|
|
if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) {
|
|
break;
|
|
}
|
|
ALOGE("%s(%d): Error %d obtaining an audio buffer, giving up.",
|
|
__func__, mPortId, err);
|
|
return NS_NEVER;
|
|
}
|
|
|
|
if (mRetryOnPartialBuffer) {
|
|
mRetryOnPartialBuffer = false;
|
|
if (avail < mRemainingFrames) {
|
|
int64_t myns = ((mRemainingFrames - avail) *
|
|
1100000000LL) / mSampleRate;
|
|
if (ns < 0 || myns < ns) {
|
|
ns = myns;
|
|
}
|
|
return ns;
|
|
}
|
|
}
|
|
|
|
size_t reqSize = audioBuffer.size;
|
|
mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
|
|
size_t readSize = audioBuffer.size;
|
|
|
|
// Validate on returned size
|
|
if (ssize_t(readSize) < 0 || readSize > reqSize) {
|
|
ALOGE("%s(%d): EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes",
|
|
__func__, mPortId, reqSize, ssize_t(readSize));
|
|
return NS_NEVER;
|
|
}
|
|
|
|
if (readSize == 0) {
|
|
// The callback is done consuming buffers
|
|
// Keep this thread going to handle timed events and
|
|
// still try to provide more data in intervals of WAIT_PERIOD_MS
|
|
// but don't just loop and block the CPU, so wait
|
|
return WAIT_PERIOD_MS * 1000000LL;
|
|
}
|
|
|
|
size_t releasedFrames = readSize / mFrameSize;
|
|
audioBuffer.frameCount = releasedFrames;
|
|
mRemainingFrames -= releasedFrames;
|
|
if (misalignment >= releasedFrames) {
|
|
misalignment -= releasedFrames;
|
|
} else {
|
|
misalignment = 0;
|
|
}
|
|
|
|
releaseBuffer(&audioBuffer);
|
|
readFrames += releasedFrames;
|
|
|
|
// FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer
|
|
// if callback doesn't like to accept the full chunk
|
|
if (readSize < reqSize) {
|
|
continue;
|
|
}
|
|
|
|
// There could be enough non-contiguous frames available to satisfy the remaining request
|
|
if (mRemainingFrames <= nonContig) {
|
|
continue;
|
|
}
|
|
|
|
#if 0
|
|
// This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a
|
|
// sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA
|
|
// that total to a sum == notificationFrames.
|
|
if (0 < misalignment && misalignment <= mRemainingFrames) {
|
|
mRemainingFrames = misalignment;
|
|
return (mRemainingFrames * 1100000000LL) / mSampleRate;
|
|
}
|
|
#endif
|
|
|
|
}
|
|
if (readFrames > 0) {
|
|
AutoMutex lock(mLock);
|
|
mFramesRead += readFrames;
|
|
// mFramesReadTime = systemTime(SYSTEM_TIME_MONOTONIC); // not provided at this time.
|
|
}
|
|
mRemainingFrames = notificationFrames;
|
|
mRetryOnPartialBuffer = true;
|
|
|
|
// A lot has transpired since ns was calculated, so run again immediately and re-calculate
|
|
return 0;
|
|
}
|
|
|
|
status_t AudioRecord::restoreRecord_l(const char *from)
|
|
{
|
|
status_t result = NO_ERROR; // logged: make sure to set this before returning.
|
|
const int64_t beginNs = systemTime();
|
|
mediametrics::Defer defer([&] {
|
|
mediametrics::LogItem(mMetricsId)
|
|
.set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_RESTORE)
|
|
.set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
|
|
.set(AMEDIAMETRICS_PROP_STATE, stateToString(mActive))
|
|
.set(AMEDIAMETRICS_PROP_STATUS, (int32_t)result)
|
|
.set(AMEDIAMETRICS_PROP_WHERE, from)
|
|
.record(); });
|
|
|
|
ALOGW("%s(%d): dead IAudioRecord, creating a new one from %s()", __func__, mPortId, from);
|
|
++mSequence;
|
|
|
|
const int INITIAL_RETRIES = 3;
|
|
int retries = INITIAL_RETRIES;
|
|
retry:
|
|
if (retries < INITIAL_RETRIES) {
|
|
// refresh the audio configuration cache in this process to make sure we get new
|
|
// input parameters and new IAudioRecord in createRecord_l()
|
|
AudioSystem::clearAudioConfigCache();
|
|
}
|
|
mFlags = mOrigFlags;
|
|
|
|
// if the new IAudioRecord is created, createRecord_l() will modify the
|
|
// following member variables: mAudioRecord, mCblkMemory, mCblk, mBufferMemory.
|
|
// It will also delete the strong references on previous IAudioRecord and IMemory
|
|
Modulo<uint32_t> position(mProxy->getPosition());
|
|
mNewPosition = position + mUpdatePeriod;
|
|
result = createRecord_l(position);
|
|
|
|
if (result == NO_ERROR) {
|
|
if (mActive) {
|
|
// callback thread or sync event hasn't changed
|
|
// FIXME this fails if we have a new AudioFlinger instance
|
|
result = statusTFromBinderStatus(mAudioRecord->start(
|
|
AudioSystem::SYNC_EVENT_SAME, AUDIO_SESSION_NONE));
|
|
}
|
|
mFramesReadServerOffset = mFramesRead; // server resets to zero so we need an offset.
|
|
}
|
|
|
|
if (result != NO_ERROR) {
|
|
ALOGW("%s(%d): failed status %d, retries %d", __func__, mPortId, result, retries);
|
|
if (--retries > 0) {
|
|
// leave time for an eventual race condition to clear before retrying
|
|
usleep(500000);
|
|
goto retry;
|
|
}
|
|
// if no retries left, set invalid bit to force restoring at next occasion
|
|
// and avoid inconsistent active state on client and server sides
|
|
if (mCblk != nullptr) {
|
|
android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
|
|
}
|
|
}
|
|
|
|
return result;
|
|
}
|
|
|
|
status_t AudioRecord::addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback)
|
|
{
|
|
if (callback == 0) {
|
|
ALOGW("%s(%d): adding NULL callback!", __func__, mPortId);
|
|
return BAD_VALUE;
|
|
}
|
|
AutoMutex lock(mLock);
|
|
if (mDeviceCallback.unsafe_get() == callback.get()) {
|
|
ALOGW("%s(%d): adding same callback!", __func__, mPortId);
|
|
return INVALID_OPERATION;
|
|
}
|
|
status_t status = NO_ERROR;
|
|
if (mInput != AUDIO_IO_HANDLE_NONE) {
|
|
if (mDeviceCallback != 0) {
|
|
ALOGW("%s(%d): callback already present!", __func__, mPortId);
|
|
AudioSystem::removeAudioDeviceCallback(this, mInput, mPortId);
|
|
}
|
|
status = AudioSystem::addAudioDeviceCallback(this, mInput, mPortId);
|
|
}
|
|
mDeviceCallback = callback;
|
|
return status;
|
|
}
|
|
|
|
status_t AudioRecord::removeAudioDeviceCallback(
|
|
const sp<AudioSystem::AudioDeviceCallback>& callback)
|
|
{
|
|
if (callback == 0) {
|
|
ALOGW("%s(%d): removing NULL callback!", __func__, mPortId);
|
|
return BAD_VALUE;
|
|
}
|
|
AutoMutex lock(mLock);
|
|
if (mDeviceCallback.unsafe_get() != callback.get()) {
|
|
ALOGW("%s(%d): removing different callback!", __func__, mPortId);
|
|
return INVALID_OPERATION;
|
|
}
|
|
mDeviceCallback.clear();
|
|
if (mInput != AUDIO_IO_HANDLE_NONE) {
|
|
AudioSystem::removeAudioDeviceCallback(this, mInput, mPortId);
|
|
}
|
|
return NO_ERROR;
|
|
}
|
|
|
|
void AudioRecord::onAudioDeviceUpdate(audio_io_handle_t audioIo,
|
|
audio_port_handle_t deviceId)
|
|
{
|
|
sp<AudioSystem::AudioDeviceCallback> callback;
|
|
{
|
|
AutoMutex lock(mLock);
|
|
if (audioIo != mInput) {
|
|
return;
|
|
}
|
|
callback = mDeviceCallback.promote();
|
|
// only update device if the record is active as route changes due to other use cases are
|
|
// irrelevant for this client
|
|
if (mActive) {
|
|
mRoutedDeviceId = deviceId;
|
|
}
|
|
}
|
|
if (callback.get() != nullptr) {
|
|
callback->onAudioDeviceUpdate(mInput, mRoutedDeviceId);
|
|
}
|
|
}
|
|
|
|
// -------------------------------------------------------------------------
|
|
|
|
status_t AudioRecord::getActiveMicrophones(std::vector<media::MicrophoneInfo>* activeMicrophones)
|
|
{
|
|
AutoMutex lock(mLock);
|
|
std::vector<media::MicrophoneInfoData> mics;
|
|
status_t status = statusTFromBinderStatus(mAudioRecord->getActiveMicrophones(&mics));
|
|
activeMicrophones->resize(mics.size());
|
|
for (size_t i = 0; status == OK && i < mics.size(); ++i) {
|
|
status = activeMicrophones->at(i).readFromParcelable(mics[i]);
|
|
}
|
|
return status;
|
|
}
|
|
|
|
status_t AudioRecord::setPreferredMicrophoneDirection(audio_microphone_direction_t direction)
|
|
{
|
|
AutoMutex lock(mLock);
|
|
if (mSelectedMicDirection == direction) {
|
|
// NOP
|
|
return OK;
|
|
}
|
|
|
|
mSelectedMicDirection = direction;
|
|
if (mAudioRecord == 0) {
|
|
// the internal AudioRecord hasn't be created yet, so just stash the attribute.
|
|
return OK;
|
|
} else {
|
|
return statusTFromBinderStatus(mAudioRecord->setPreferredMicrophoneDirection(direction));
|
|
}
|
|
}
|
|
|
|
status_t AudioRecord::setPreferredMicrophoneFieldDimension(float zoom) {
|
|
AutoMutex lock(mLock);
|
|
if (mSelectedMicFieldDimension == zoom) {
|
|
// NOP
|
|
return OK;
|
|
}
|
|
|
|
mSelectedMicFieldDimension = zoom;
|
|
if (mAudioRecord == 0) {
|
|
// the internal AudioRecord hasn't be created yet, so just stash the attribute.
|
|
return OK;
|
|
} else {
|
|
return statusTFromBinderStatus(mAudioRecord->setPreferredMicrophoneFieldDimension(zoom));
|
|
}
|
|
}
|
|
|
|
void AudioRecord::setLogSessionId(const char *logSessionId)
|
|
{
|
|
AutoMutex lock(mLock);
|
|
if (logSessionId == nullptr) logSessionId = ""; // an empty string is an unset session id.
|
|
if (mLogSessionId == logSessionId) return;
|
|
|
|
mLogSessionId = logSessionId;
|
|
mediametrics::LogItem(mMetricsId)
|
|
.set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETLOGSESSIONID)
|
|
.set(AMEDIAMETRICS_PROP_LOGSESSIONID, logSessionId)
|
|
.record();
|
|
}
|
|
|
|
status_t AudioRecord::shareAudioHistory(const std::string& sharedPackageName,
|
|
int64_t sharedStartMs)
|
|
{
|
|
AutoMutex lock(mLock);
|
|
if (mAudioRecord == 0) {
|
|
return NO_INIT;
|
|
}
|
|
status_t status = statusTFromBinderStatus(
|
|
mAudioRecord->shareAudioHistory(sharedPackageName, sharedStartMs));
|
|
if (status == NO_ERROR) {
|
|
mSharedAudioPackageName = sharedPackageName;
|
|
mSharedAudioStartMs = sharedStartMs;
|
|
}
|
|
return status;
|
|
}
|
|
|
|
// =========================================================================
|
|
|
|
void AudioRecord::DeathNotifier::binderDied(const wp<IBinder>& who __unused)
|
|
{
|
|
sp<AudioRecord> audioRecord = mAudioRecord.promote();
|
|
if (audioRecord != 0) {
|
|
AutoMutex lock(audioRecord->mLock);
|
|
audioRecord->mProxy->binderDied();
|
|
}
|
|
}
|
|
|
|
// =========================================================================
|
|
|
|
AudioRecord::AudioRecordThread::AudioRecordThread(AudioRecord& receiver)
|
|
: Thread(true /* bCanCallJava */) // binder recursion on restoreRecord_l() may call Java.
|
|
, mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL),
|
|
mIgnoreNextPausedInt(false)
|
|
{
|
|
}
|
|
|
|
AudioRecord::AudioRecordThread::~AudioRecordThread()
|
|
{
|
|
}
|
|
|
|
bool AudioRecord::AudioRecordThread::threadLoop()
|
|
{
|
|
{
|
|
AutoMutex _l(mMyLock);
|
|
if (mPaused) {
|
|
// TODO check return value and handle or log
|
|
mMyCond.wait(mMyLock);
|
|
// caller will check for exitPending()
|
|
return true;
|
|
}
|
|
if (mIgnoreNextPausedInt) {
|
|
mIgnoreNextPausedInt = false;
|
|
mPausedInt = false;
|
|
}
|
|
if (mPausedInt) {
|
|
if (mPausedNs > 0) {
|
|
// TODO check return value and handle or log
|
|
(void) mMyCond.waitRelative(mMyLock, mPausedNs);
|
|
} else {
|
|
// TODO check return value and handle or log
|
|
mMyCond.wait(mMyLock);
|
|
}
|
|
mPausedInt = false;
|
|
return true;
|
|
}
|
|
}
|
|
if (exitPending()) {
|
|
return false;
|
|
}
|
|
nsecs_t ns = mReceiver.processAudioBuffer();
|
|
switch (ns) {
|
|
case 0:
|
|
return true;
|
|
case NS_INACTIVE:
|
|
pauseInternal();
|
|
return true;
|
|
case NS_NEVER:
|
|
return false;
|
|
case NS_WHENEVER:
|
|
// Event driven: call wake() when callback notifications conditions change.
|
|
ns = INT64_MAX;
|
|
FALLTHROUGH_INTENDED;
|
|
default:
|
|
LOG_ALWAYS_FATAL_IF(ns < 0, "%s() returned %lld", __func__, (long long)ns);
|
|
pauseInternal(ns);
|
|
return true;
|
|
}
|
|
}
|
|
|
|
void AudioRecord::AudioRecordThread::requestExit()
|
|
{
|
|
// must be in this order to avoid a race condition
|
|
Thread::requestExit();
|
|
resume();
|
|
}
|
|
|
|
void AudioRecord::AudioRecordThread::pause()
|
|
{
|
|
AutoMutex _l(mMyLock);
|
|
mPaused = true;
|
|
}
|
|
|
|
void AudioRecord::AudioRecordThread::resume()
|
|
{
|
|
AutoMutex _l(mMyLock);
|
|
mIgnoreNextPausedInt = true;
|
|
if (mPaused || mPausedInt) {
|
|
mPaused = false;
|
|
mPausedInt = false;
|
|
mMyCond.signal();
|
|
}
|
|
}
|
|
|
|
void AudioRecord::AudioRecordThread::wake()
|
|
{
|
|
AutoMutex _l(mMyLock);
|
|
if (!mPaused) {
|
|
// wake() might be called while servicing a callback - ignore the next
|
|
// pause time and call processAudioBuffer.
|
|
mIgnoreNextPausedInt = true;
|
|
if (mPausedInt && mPausedNs > 0) {
|
|
// audio record is active and internally paused with timeout.
|
|
mPausedInt = false;
|
|
mMyCond.signal();
|
|
}
|
|
}
|
|
}
|
|
|
|
void AudioRecord::AudioRecordThread::pauseInternal(nsecs_t ns)
|
|
{
|
|
AutoMutex _l(mMyLock);
|
|
mPausedInt = true;
|
|
mPausedNs = ns;
|
|
}
|
|
|
|
// -------------------------------------------------------------------------
|
|
|
|
} // namespace android
|