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      1 /*
      2 **
      3 ** Copyright 2008, The Android Open Source Project
      4 **
      5 ** Licensed under the Apache License, Version 2.0 (the "License");
      6 ** you may not use this file except in compliance with the License.
      7 ** You may obtain a copy of the License at
      8 **
      9 **     http://www.apache.org/licenses/LICENSE-2.0
     10 **
     11 ** Unless required by applicable law or agreed to in writing, software
     12 ** distributed under the License is distributed on an "AS IS" BASIS,
     13 ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
     14 ** See the License for the specific language governing permissions and
     15 ** limitations under the License.
     16 */
     17 
     18 //#define LOG_NDEBUG 0
     19 #define LOG_TAG "AudioRecord"
     20 
     21 #include <inttypes.h>
     22 #include <sys/resource.h>
     23 
     24 #include <binder/IPCThreadState.h>
     25 #include <media/AudioRecord.h>
     26 #include <utils/Log.h>
     27 #include <private/media/AudioTrackShared.h>
     28 #include <media/IAudioFlinger.h>
     29 
     30 #define WAIT_PERIOD_MS          10
     31 
     32 namespace android {
     33 // ---------------------------------------------------------------------------
     34 
     35 // static
     36 status_t AudioRecord::getMinFrameCount(
     37         size_t* frameCount,
     38         uint32_t sampleRate,
     39         audio_format_t format,
     40         audio_channel_mask_t channelMask)
     41 {
     42     if (frameCount == NULL) {
     43         return BAD_VALUE;
     44     }
     45 
     46     size_t size;
     47     status_t status = AudioSystem::getInputBufferSize(sampleRate, format, channelMask, &size);
     48     if (status != NO_ERROR) {
     49         ALOGE("AudioSystem could not query the input buffer size for sampleRate %u, format %#x, "
     50               "channelMask %#x; status %d", sampleRate, format, channelMask, status);
     51         return status;
     52     }
     53 
     54     // We double the size of input buffer for ping pong use of record buffer.
     55     // Assumes audio_is_linear_pcm(format)
     56     if ((*frameCount = (size * 2) / (audio_channel_count_from_in_mask(channelMask) *
     57             audio_bytes_per_sample(format))) == 0) {
     58         ALOGE("Unsupported configuration: sampleRate %u, format %#x, channelMask %#x",
     59             sampleRate, format, channelMask);
     60         return BAD_VALUE;
     61     }
     62 
     63     return NO_ERROR;
     64 }
     65 
     66 // ---------------------------------------------------------------------------
     67 
     68 AudioRecord::AudioRecord()
     69     : mStatus(NO_INIT), mSessionId(AUDIO_SESSION_ALLOCATE),
     70       mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT)
     71 {
     72 }
     73 
     74 AudioRecord::AudioRecord(
     75         audio_source_t inputSource,
     76         uint32_t sampleRate,
     77         audio_format_t format,
     78         audio_channel_mask_t channelMask,
     79         size_t frameCount,
     80         callback_t cbf,
     81         void* user,
     82         uint32_t notificationFrames,
     83         int sessionId,
     84         transfer_type transferType,
     85         audio_input_flags_t flags,
     86         const audio_attributes_t* pAttributes)
     87     : mStatus(NO_INIT), mSessionId(AUDIO_SESSION_ALLOCATE),
     88       mPreviousPriority(ANDROID_PRIORITY_NORMAL),
     89       mPreviousSchedulingGroup(SP_DEFAULT),
     90       mProxy(NULL)
     91 {
     92     mStatus = set(inputSource, sampleRate, format, channelMask, frameCount, cbf, user,
     93             notificationFrames, false /*threadCanCallJava*/, sessionId, transferType, flags,
     94             pAttributes);
     95 }
     96 
     97 AudioRecord::~AudioRecord()
     98 {
     99     if (mStatus == NO_ERROR) {
    100         // Make sure that callback function exits in the case where
    101         // it is looping on buffer empty condition in obtainBuffer().
    102         // Otherwise the callback thread will never exit.
    103         stop();
    104         if (mAudioRecordThread != 0) {
    105             mProxy->interrupt();
    106             mAudioRecordThread->requestExit();  // see comment in AudioRecord.h
    107             mAudioRecordThread->requestExitAndWait();
    108             mAudioRecordThread.clear();
    109         }
    110         mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this);
    111         mAudioRecord.clear();
    112         mCblkMemory.clear();
    113         mBufferMemory.clear();
    114         IPCThreadState::self()->flushCommands();
    115         AudioSystem::releaseAudioSessionId(mSessionId, -1);
    116     }
    117 }
    118 
    119 status_t AudioRecord::set(
    120         audio_source_t inputSource,
    121         uint32_t sampleRate,
    122         audio_format_t format,
    123         audio_channel_mask_t channelMask,
    124         size_t frameCount,
    125         callback_t cbf,
    126         void* user,
    127         uint32_t notificationFrames,
    128         bool threadCanCallJava,
    129         int sessionId,
    130         transfer_type transferType,
    131         audio_input_flags_t flags,
    132         const audio_attributes_t* pAttributes)
    133 {
    134     ALOGV("set(): inputSource %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
    135           "notificationFrames %u, sessionId %d, transferType %d, flags %#x",
    136           inputSource, sampleRate, format, channelMask, frameCount, notificationFrames,
    137           sessionId, transferType, flags);
    138 
    139     switch (transferType) {
    140     case TRANSFER_DEFAULT:
    141         if (cbf == NULL || threadCanCallJava) {
    142             transferType = TRANSFER_SYNC;
    143         } else {
    144             transferType = TRANSFER_CALLBACK;
    145         }
    146         break;
    147     case TRANSFER_CALLBACK:
    148         if (cbf == NULL) {
    149             ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL");
    150             return BAD_VALUE;
    151         }
    152         break;
    153     case TRANSFER_OBTAIN:
    154     case TRANSFER_SYNC:
    155         break;
    156     default:
    157         ALOGE("Invalid transfer type %d", transferType);
    158         return BAD_VALUE;
    159     }
    160     mTransfer = transferType;
    161 
    162     AutoMutex lock(mLock);
    163 
    164     // invariant that mAudioRecord != 0 is true only after set() returns successfully
    165     if (mAudioRecord != 0) {
    166         ALOGE("Track already in use");
    167         return INVALID_OPERATION;
    168     }
    169 
    170     if (pAttributes == NULL) {
    171         memset(&mAttributes, 0, sizeof(audio_attributes_t));
    172         mAttributes.source = inputSource;
    173     } else {
    174         // stream type shouldn't be looked at, this track has audio attributes
    175         memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t));
    176         ALOGV("Building AudioRecord with attributes: source=%d flags=0x%x tags=[%s]",
    177               mAttributes.source, mAttributes.flags, mAttributes.tags);
    178     }
    179 
    180     if (sampleRate == 0) {
    181         ALOGE("Invalid sample rate %u", sampleRate);
    182         return BAD_VALUE;
    183     }
    184     mSampleRate = sampleRate;
    185 
    186     // these below should probably come from the audioFlinger too...
    187     if (format == AUDIO_FORMAT_DEFAULT) {
    188         format = AUDIO_FORMAT_PCM_16_BIT;
    189     }
    190 
    191     // validate parameters
    192     if (!audio_is_valid_format(format)) {
    193         ALOGE("Invalid format %#x", format);
    194         return BAD_VALUE;
    195     }
    196     // Temporary restriction: AudioFlinger currently supports 16-bit PCM only
    197     if (format != AUDIO_FORMAT_PCM_16_BIT) {
    198         ALOGE("Format %#x is not supported", format);
    199         return BAD_VALUE;
    200     }
    201     mFormat = format;
    202 
    203     if (!audio_is_input_channel(channelMask)) {
    204         ALOGE("Invalid channel mask %#x", channelMask);
    205         return BAD_VALUE;
    206     }
    207     mChannelMask = channelMask;
    208     uint32_t channelCount = audio_channel_count_from_in_mask(channelMask);
    209     mChannelCount = channelCount;
    210 
    211     if (audio_is_linear_pcm(format)) {
    212         mFrameSize = channelCount * audio_bytes_per_sample(format);
    213     } else {
    214         mFrameSize = sizeof(uint8_t);
    215     }
    216 
    217     // mFrameCount is initialized in openRecord_l
    218     mReqFrameCount = frameCount;
    219 
    220     mNotificationFramesReq = notificationFrames;
    221     // mNotificationFramesAct is initialized in openRecord_l
    222 
    223     if (sessionId == AUDIO_SESSION_ALLOCATE) {
    224         mSessionId = AudioSystem::newAudioUniqueId();
    225     } else {
    226         mSessionId = sessionId;
    227     }
    228     ALOGV("set(): mSessionId %d", mSessionId);
    229 
    230     mFlags = flags;
    231     mCbf = cbf;
    232 
    233     if (cbf != NULL) {
    234         mAudioRecordThread = new AudioRecordThread(*this, threadCanCallJava);
    235         mAudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO);
    236     }
    237 
    238     // create the IAudioRecord
    239     status_t status = openRecord_l(0 /*epoch*/);
    240 
    241     if (status != NO_ERROR) {
    242         if (mAudioRecordThread != 0) {
    243             mAudioRecordThread->requestExit();   // see comment in AudioRecord.h
    244             mAudioRecordThread->requestExitAndWait();
    245             mAudioRecordThread.clear();
    246         }
    247         return status;
    248     }
    249 
    250     mStatus = NO_ERROR;
    251     mActive = false;
    252     mUserData = user;
    253     // TODO: add audio hardware input latency here
    254     mLatency = (1000*mFrameCount) / sampleRate;
    255     mMarkerPosition = 0;
    256     mMarkerReached = false;
    257     mNewPosition = 0;
    258     mUpdatePeriod = 0;
    259     AudioSystem::acquireAudioSessionId(mSessionId, -1);
    260     mSequence = 1;
    261     mObservedSequence = mSequence;
    262     mInOverrun = false;
    263 
    264     return NO_ERROR;
    265 }
    266 
    267 // -------------------------------------------------------------------------
    268 
    269 status_t AudioRecord::start(AudioSystem::sync_event_t event, int triggerSession)
    270 {
    271     ALOGV("start, sync event %d trigger session %d", event, triggerSession);
    272 
    273     AutoMutex lock(mLock);
    274     if (mActive) {
    275         return NO_ERROR;
    276     }
    277 
    278     // reset current position as seen by client to 0
    279     mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition());
    280     // force refresh of remaining frames by processAudioBuffer() as last
    281     // read before stop could be partial.
    282     mRefreshRemaining = true;
    283 
    284     mNewPosition = mProxy->getPosition() + mUpdatePeriod;
    285     int32_t flags = android_atomic_acquire_load(&mCblk->mFlags);
    286 
    287     status_t status = NO_ERROR;
    288     if (!(flags & CBLK_INVALID)) {
    289         ALOGV("mAudioRecord->start()");
    290         status = mAudioRecord->start(event, triggerSession);
    291         if (status == DEAD_OBJECT) {
    292             flags |= CBLK_INVALID;
    293         }
    294     }
    295     if (flags & CBLK_INVALID) {
    296         status = restoreRecord_l("start");
    297     }
    298 
    299     if (status != NO_ERROR) {
    300         ALOGE("start() status %d", status);
    301     } else {
    302         mActive = true;
    303         sp<AudioRecordThread> t = mAudioRecordThread;
    304         if (t != 0) {
    305             t->resume();
    306         } else {
    307             mPreviousPriority = getpriority(PRIO_PROCESS, 0);
    308             get_sched_policy(0, &mPreviousSchedulingGroup);
    309             androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
    310         }
    311     }
    312 
    313     return status;
    314 }
    315 
    316 void AudioRecord::stop()
    317 {
    318     AutoMutex lock(mLock);
    319     if (!mActive) {
    320         return;
    321     }
    322 
    323     mActive = false;
    324     mProxy->interrupt();
    325     mAudioRecord->stop();
    326     // the record head position will reset to 0, so if a marker is set, we need
    327     // to activate it again
    328     mMarkerReached = false;
    329     sp<AudioRecordThread> t = mAudioRecordThread;
    330     if (t != 0) {
    331         t->pause();
    332     } else {
    333         setpriority(PRIO_PROCESS, 0, mPreviousPriority);
    334         set_sched_policy(0, mPreviousSchedulingGroup);
    335     }
    336 }
    337 
    338 bool AudioRecord::stopped() const
    339 {
    340     AutoMutex lock(mLock);
    341     return !mActive;
    342 }
    343 
    344 status_t AudioRecord::setMarkerPosition(uint32_t marker)
    345 {
    346     // The only purpose of setting marker position is to get a callback
    347     if (mCbf == NULL) {
    348         return INVALID_OPERATION;
    349     }
    350 
    351     AutoMutex lock(mLock);
    352     mMarkerPosition = marker;
    353     mMarkerReached = false;
    354 
    355     return NO_ERROR;
    356 }
    357 
    358 status_t AudioRecord::getMarkerPosition(uint32_t *marker) const
    359 {
    360     if (marker == NULL) {
    361         return BAD_VALUE;
    362     }
    363 
    364     AutoMutex lock(mLock);
    365     *marker = mMarkerPosition;
    366 
    367     return NO_ERROR;
    368 }
    369 
    370 status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod)
    371 {
    372     // The only purpose of setting position update period is to get a callback
    373     if (mCbf == NULL) {
    374         return INVALID_OPERATION;
    375     }
    376 
    377     AutoMutex lock(mLock);
    378     mNewPosition = mProxy->getPosition() + updatePeriod;
    379     mUpdatePeriod = updatePeriod;
    380 
    381     return NO_ERROR;
    382 }
    383 
    384 status_t AudioRecord::getPositionUpdatePeriod(uint32_t *updatePeriod) const
    385 {
    386     if (updatePeriod == NULL) {
    387         return BAD_VALUE;
    388     }
    389 
    390     AutoMutex lock(mLock);
    391     *updatePeriod = mUpdatePeriod;
    392 
    393     return NO_ERROR;
    394 }
    395 
    396 status_t AudioRecord::getPosition(uint32_t *position) const
    397 {
    398     if (position == NULL) {
    399         return BAD_VALUE;
    400     }
    401 
    402     AutoMutex lock(mLock);
    403     *position = mProxy->getPosition();
    404 
    405     return NO_ERROR;
    406 }
    407 
    408 uint32_t AudioRecord::getInputFramesLost() const
    409 {
    410     // no need to check mActive, because if inactive this will return 0, which is what we want
    411     return AudioSystem::getInputFramesLost(getInput());
    412 }
    413 
    414 // -------------------------------------------------------------------------
    415 
    416 // must be called with mLock held
    417 status_t AudioRecord::openRecord_l(size_t epoch)
    418 {
    419     status_t status;
    420     const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
    421     if (audioFlinger == 0) {
    422         ALOGE("Could not get audioflinger");
    423         return NO_INIT;
    424     }
    425 
    426     // Fast tracks must be at the primary _output_ [sic] sampling rate,
    427     // because there is currently no concept of a primary input sampling rate
    428     uint32_t afSampleRate = AudioSystem::getPrimaryOutputSamplingRate();
    429     if (afSampleRate == 0) {
    430         ALOGW("getPrimaryOutputSamplingRate failed");
    431     }
    432 
    433     // Client can only express a preference for FAST.  Server will perform additional tests.
    434     if ((mFlags & AUDIO_INPUT_FLAG_FAST) && !(
    435             // use case: callback transfer mode
    436             (mTransfer == TRANSFER_CALLBACK) &&
    437             // matching sample rate
    438             (mSampleRate == afSampleRate))) {
    439         ALOGW("AUDIO_INPUT_FLAG_FAST denied by client");
    440         // once denied, do not request again if IAudioRecord is re-created
    441         mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST);
    442     }
    443 
    444     IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT;
    445 
    446     pid_t tid = -1;
    447     if (mFlags & AUDIO_INPUT_FLAG_FAST) {
    448         trackFlags |= IAudioFlinger::TRACK_FAST;
    449         if (mAudioRecordThread != 0) {
    450             tid = mAudioRecordThread->getTid();
    451         }
    452     }
    453 
    454     audio_io_handle_t input;
    455     status = AudioSystem::getInputForAttr(&mAttributes, &input, (audio_session_t)mSessionId,
    456                                         mSampleRate, mFormat, mChannelMask, mFlags);
    457 
    458     if (status != NO_ERROR) {
    459         ALOGE("Could not get audio input for record source %d, sample rate %u, format %#x, "
    460               "channel mask %#x, session %d, flags %#x",
    461               mAttributes.source, mSampleRate, mFormat, mChannelMask, mSessionId, mFlags);
    462         return BAD_VALUE;
    463     }
    464     {
    465     // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger,
    466     // we must release it ourselves if anything goes wrong.
    467 
    468     size_t frameCount = mReqFrameCount;
    469     size_t temp = frameCount;   // temp may be replaced by a revised value of frameCount,
    470                                 // but we will still need the original value also
    471     int originalSessionId = mSessionId;
    472 
    473     // The notification frame count is the period between callbacks, as suggested by the server.
    474     size_t notificationFrames = mNotificationFramesReq;
    475 
    476     sp<IMemory> iMem;           // for cblk
    477     sp<IMemory> bufferMem;
    478     sp<IAudioRecord> record = audioFlinger->openRecord(input,
    479                                                        mSampleRate, mFormat,
    480                                                        mChannelMask,
    481                                                        &temp,
    482                                                        &trackFlags,
    483                                                        tid,
    484                                                        &mSessionId,
    485                                                        &notificationFrames,
    486                                                        iMem,
    487                                                        bufferMem,
    488                                                        &status);
    489     ALOGE_IF(originalSessionId != AUDIO_SESSION_ALLOCATE && mSessionId != originalSessionId,
    490             "session ID changed from %d to %d", originalSessionId, mSessionId);
    491 
    492     if (status != NO_ERROR) {
    493         ALOGE("AudioFlinger could not create record track, status: %d", status);
    494         goto release;
    495     }
    496     ALOG_ASSERT(record != 0);
    497 
    498     // AudioFlinger now owns the reference to the I/O handle,
    499     // so we are no longer responsible for releasing it.
    500 
    501     if (iMem == 0) {
    502         ALOGE("Could not get control block");
    503         return NO_INIT;
    504     }
    505     void *iMemPointer = iMem->pointer();
    506     if (iMemPointer == NULL) {
    507         ALOGE("Could not get control block pointer");
    508         return NO_INIT;
    509     }
    510     audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
    511 
    512     // Starting address of buffers in shared memory.
    513     // The buffers are either immediately after the control block,
    514     // or in a separate area at discretion of server.
    515     void *buffers;
    516     if (bufferMem == 0) {
    517         buffers = cblk + 1;
    518     } else {
    519         buffers = bufferMem->pointer();
    520         if (buffers == NULL) {
    521             ALOGE("Could not get buffer pointer");
    522             return NO_INIT;
    523         }
    524     }
    525 
    526     // invariant that mAudioRecord != 0 is true only after set() returns successfully
    527     if (mAudioRecord != 0) {
    528         mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this);
    529         mDeathNotifier.clear();
    530     }
    531     mAudioRecord = record;
    532     mCblkMemory = iMem;
    533     mBufferMemory = bufferMem;
    534     IPCThreadState::self()->flushCommands();
    535 
    536     mCblk = cblk;
    537     // note that temp is the (possibly revised) value of frameCount
    538     if (temp < frameCount || (frameCount == 0 && temp == 0)) {
    539         ALOGW("Requested frameCount %zu but received frameCount %zu", frameCount, temp);
    540     }
    541     frameCount = temp;
    542 
    543     mAwaitBoost = false;
    544     if (mFlags & AUDIO_INPUT_FLAG_FAST) {
    545         if (trackFlags & IAudioFlinger::TRACK_FAST) {
    546             ALOGV("AUDIO_INPUT_FLAG_FAST successful; frameCount %zu", frameCount);
    547             mAwaitBoost = true;
    548         } else {
    549             ALOGV("AUDIO_INPUT_FLAG_FAST denied by server; frameCount %zu", frameCount);
    550             // once denied, do not request again if IAudioRecord is re-created
    551             mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST);
    552         }
    553     }
    554 
    555     // Make sure that application is notified with sufficient margin before overrun
    556     if (notificationFrames == 0 || notificationFrames > frameCount) {
    557         ALOGW("Received notificationFrames %zu for frameCount %zu", notificationFrames, frameCount);
    558     }
    559     mNotificationFramesAct = notificationFrames;
    560 
    561     // We retain a copy of the I/O handle, but don't own the reference
    562     mInput = input;
    563     mRefreshRemaining = true;
    564 
    565     mFrameCount = frameCount;
    566     // If IAudioRecord is re-created, don't let the requested frameCount
    567     // decrease.  This can confuse clients that cache frameCount().
    568     if (frameCount > mReqFrameCount) {
    569         mReqFrameCount = frameCount;
    570     }
    571 
    572     // update proxy
    573     mProxy = new AudioRecordClientProxy(cblk, buffers, mFrameCount, mFrameSize);
    574     mProxy->setEpoch(epoch);
    575     mProxy->setMinimum(mNotificationFramesAct);
    576 
    577     mDeathNotifier = new DeathNotifier(this);
    578     mAudioRecord->asBinder()->linkToDeath(mDeathNotifier, this);
    579 
    580     return NO_ERROR;
    581     }
    582 
    583 release:
    584     AudioSystem::releaseInput(input, (audio_session_t)mSessionId);
    585     if (status == NO_ERROR) {
    586         status = NO_INIT;
    587     }
    588     return status;
    589 }
    590 
    591 status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
    592 {
    593     if (audioBuffer == NULL) {
    594         return BAD_VALUE;
    595     }
    596     if (mTransfer != TRANSFER_OBTAIN) {
    597         audioBuffer->frameCount = 0;
    598         audioBuffer->size = 0;
    599         audioBuffer->raw = NULL;
    600         return INVALID_OPERATION;
    601     }
    602 
    603     const struct timespec *requested;
    604     struct timespec timeout;
    605     if (waitCount == -1) {
    606         requested = &ClientProxy::kForever;
    607     } else if (waitCount == 0) {
    608         requested = &ClientProxy::kNonBlocking;
    609     } else if (waitCount > 0) {
    610         long long ms = WAIT_PERIOD_MS * (long long) waitCount;
    611         timeout.tv_sec = ms / 1000;
    612         timeout.tv_nsec = (int) (ms % 1000) * 1000000;
    613         requested = &timeout;
    614     } else {
    615         ALOGE("%s invalid waitCount %d", __func__, waitCount);
    616         requested = NULL;
    617     }
    618     return obtainBuffer(audioBuffer, requested);
    619 }
    620 
    621 status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
    622         struct timespec *elapsed, size_t *nonContig)
    623 {
    624     // previous and new IAudioRecord sequence numbers are used to detect track re-creation
    625     uint32_t oldSequence = 0;
    626     uint32_t newSequence;
    627 
    628     Proxy::Buffer buffer;
    629     status_t status = NO_ERROR;
    630 
    631     static const int32_t kMaxTries = 5;
    632     int32_t tryCounter = kMaxTries;
    633 
    634     do {
    635         // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to
    636         // keep them from going away if another thread re-creates the track during obtainBuffer()
    637         sp<AudioRecordClientProxy> proxy;
    638         sp<IMemory> iMem;
    639         sp<IMemory> bufferMem;
    640         {
    641             // start of lock scope
    642             AutoMutex lock(mLock);
    643 
    644             newSequence = mSequence;
    645             // did previous obtainBuffer() fail due to media server death or voluntary invalidation?
    646             if (status == DEAD_OBJECT) {
    647                 // re-create track, unless someone else has already done so
    648                 if (newSequence == oldSequence) {
    649                     status = restoreRecord_l("obtainBuffer");
    650                     if (status != NO_ERROR) {
    651                         buffer.mFrameCount = 0;
    652                         buffer.mRaw = NULL;
    653                         buffer.mNonContig = 0;
    654                         break;
    655                     }
    656                 }
    657             }
    658             oldSequence = newSequence;
    659 
    660             // Keep the extra references
    661             proxy = mProxy;
    662             iMem = mCblkMemory;
    663             bufferMem = mBufferMemory;
    664 
    665             // Non-blocking if track is stopped
    666             if (!mActive) {
    667                 requested = &ClientProxy::kNonBlocking;
    668             }
    669 
    670         }   // end of lock scope
    671 
    672         buffer.mFrameCount = audioBuffer->frameCount;
    673         // FIXME starts the requested timeout and elapsed over from scratch
    674         status = proxy->obtainBuffer(&buffer, requested, elapsed);
    675 
    676     } while ((status == DEAD_OBJECT) && (tryCounter-- > 0));
    677 
    678     audioBuffer->frameCount = buffer.mFrameCount;
    679     audioBuffer->size = buffer.mFrameCount * mFrameSize;
    680     audioBuffer->raw = buffer.mRaw;
    681     if (nonContig != NULL) {
    682         *nonContig = buffer.mNonContig;
    683     }
    684     return status;
    685 }
    686 
    687 void AudioRecord::releaseBuffer(Buffer* audioBuffer)
    688 {
    689     // all TRANSFER_* are valid
    690 
    691     size_t stepCount = audioBuffer->size / mFrameSize;
    692     if (stepCount == 0) {
    693         return;
    694     }
    695 
    696     Proxy::Buffer buffer;
    697     buffer.mFrameCount = stepCount;
    698     buffer.mRaw = audioBuffer->raw;
    699 
    700     AutoMutex lock(mLock);
    701     mInOverrun = false;
    702     mProxy->releaseBuffer(&buffer);
    703 
    704     // the server does not automatically disable recorder on overrun, so no need to restart
    705 }
    706 
    707 audio_io_handle_t AudioRecord::getInput() const
    708 {
    709     AutoMutex lock(mLock);
    710     return mInput;
    711 }
    712 
    713 // -------------------------------------------------------------------------
    714 
    715 ssize_t AudioRecord::read(void* buffer, size_t userSize)
    716 {
    717     if (mTransfer != TRANSFER_SYNC) {
    718         return INVALID_OPERATION;
    719     }
    720 
    721     if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
    722         // sanity-check. user is most-likely passing an error code, and it would
    723         // make the return value ambiguous (actualSize vs error).
    724         ALOGE("AudioRecord::read(buffer=%p, size=%zu (%zu)", buffer, userSize, userSize);
    725         return BAD_VALUE;
    726     }
    727 
    728     ssize_t read = 0;
    729     Buffer audioBuffer;
    730 
    731     while (userSize >= mFrameSize) {
    732         audioBuffer.frameCount = userSize / mFrameSize;
    733 
    734         status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever);
    735         if (err < 0) {
    736             if (read > 0) {
    737                 break;
    738             }
    739             return ssize_t(err);
    740         }
    741 
    742         size_t bytesRead = audioBuffer.size;
    743         memcpy(buffer, audioBuffer.i8, bytesRead);
    744         buffer = ((char *) buffer) + bytesRead;
    745         userSize -= bytesRead;
    746         read += bytesRead;
    747 
    748         releaseBuffer(&audioBuffer);
    749     }
    750 
    751     return read;
    752 }
    753 
    754 // -------------------------------------------------------------------------
    755 
    756 nsecs_t AudioRecord::processAudioBuffer()
    757 {
    758     mLock.lock();
    759     if (mAwaitBoost) {
    760         mAwaitBoost = false;
    761         mLock.unlock();
    762         static const int32_t kMaxTries = 5;
    763         int32_t tryCounter = kMaxTries;
    764         uint32_t pollUs = 10000;
    765         do {
    766             int policy = sched_getscheduler(0);
    767             if (policy == SCHED_FIFO || policy == SCHED_RR) {
    768                 break;
    769             }
    770             usleep(pollUs);
    771             pollUs <<= 1;
    772         } while (tryCounter-- > 0);
    773         if (tryCounter < 0) {
    774             ALOGE("did not receive expected priority boost on time");
    775         }
    776         // Run again immediately
    777         return 0;
    778     }
    779 
    780     // Can only reference mCblk while locked
    781     int32_t flags = android_atomic_and(~CBLK_OVERRUN, &mCblk->mFlags);
    782 
    783     // Check for track invalidation
    784     if (flags & CBLK_INVALID) {
    785         (void) restoreRecord_l("processAudioBuffer");
    786         mLock.unlock();
    787         // Run again immediately, but with a new IAudioRecord
    788         return 0;
    789     }
    790 
    791     bool active = mActive;
    792 
    793     // Manage overrun callback, must be done under lock to avoid race with releaseBuffer()
    794     bool newOverrun = false;
    795     if (flags & CBLK_OVERRUN) {
    796         if (!mInOverrun) {
    797             mInOverrun = true;
    798             newOverrun = true;
    799         }
    800     }
    801 
    802     // Get current position of server
    803     size_t position = mProxy->getPosition();
    804 
    805     // Manage marker callback
    806     bool markerReached = false;
    807     size_t markerPosition = mMarkerPosition;
    808     // FIXME fails for wraparound, need 64 bits
    809     if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) {
    810         mMarkerReached = markerReached = true;
    811     }
    812 
    813     // Determine the number of new position callback(s) that will be needed, while locked
    814     size_t newPosCount = 0;
    815     size_t newPosition = mNewPosition;
    816     uint32_t updatePeriod = mUpdatePeriod;
    817     // FIXME fails for wraparound, need 64 bits
    818     if (updatePeriod > 0 && position >= newPosition) {
    819         newPosCount = ((position - newPosition) / updatePeriod) + 1;
    820         mNewPosition += updatePeriod * newPosCount;
    821     }
    822 
    823     // Cache other fields that will be needed soon
    824     uint32_t notificationFrames = mNotificationFramesAct;
    825     if (mRefreshRemaining) {
    826         mRefreshRemaining = false;
    827         mRemainingFrames = notificationFrames;
    828         mRetryOnPartialBuffer = false;
    829     }
    830     size_t misalignment = mProxy->getMisalignment();
    831     uint32_t sequence = mSequence;
    832 
    833     // These fields don't need to be cached, because they are assigned only by set():
    834     //      mTransfer, mCbf, mUserData, mSampleRate, mFrameSize
    835 
    836     mLock.unlock();
    837 
    838     // perform callbacks while unlocked
    839     if (newOverrun) {
    840         mCbf(EVENT_OVERRUN, mUserData, NULL);
    841     }
    842     if (markerReached) {
    843         mCbf(EVENT_MARKER, mUserData, &markerPosition);
    844     }
    845     while (newPosCount > 0) {
    846         size_t temp = newPosition;
    847         mCbf(EVENT_NEW_POS, mUserData, &temp);
    848         newPosition += updatePeriod;
    849         newPosCount--;
    850     }
    851     if (mObservedSequence != sequence) {
    852         mObservedSequence = sequence;
    853         mCbf(EVENT_NEW_IAUDIORECORD, mUserData, NULL);
    854     }
    855 
    856     // if inactive, then don't run me again until re-started
    857     if (!active) {
    858         return NS_INACTIVE;
    859     }
    860 
    861     // Compute the estimated time until the next timed event (position, markers)
    862     uint32_t minFrames = ~0;
    863     if (!markerReached && position < markerPosition) {
    864         minFrames = markerPosition - position;
    865     }
    866     if (updatePeriod > 0 && updatePeriod < minFrames) {
    867         minFrames = updatePeriod;
    868     }
    869 
    870     // If > 0, poll periodically to recover from a stuck server.  A good value is 2.
    871     static const uint32_t kPoll = 0;
    872     if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) {
    873         minFrames = kPoll * notificationFrames;
    874     }
    875 
    876     // Convert frame units to time units
    877     nsecs_t ns = NS_WHENEVER;
    878     if (minFrames != (uint32_t) ~0) {
    879         // This "fudge factor" avoids soaking CPU, and compensates for late progress by server
    880         static const nsecs_t kFudgeNs = 10000000LL; // 10 ms
    881         ns = ((minFrames * 1000000000LL) / mSampleRate) + kFudgeNs;
    882     }
    883 
    884     // If not supplying data by EVENT_MORE_DATA, then we're done
    885     if (mTransfer != TRANSFER_CALLBACK) {
    886         return ns;
    887     }
    888 
    889     struct timespec timeout;
    890     const struct timespec *requested = &ClientProxy::kForever;
    891     if (ns != NS_WHENEVER) {
    892         timeout.tv_sec = ns / 1000000000LL;
    893         timeout.tv_nsec = ns % 1000000000LL;
    894         ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000);
    895         requested = &timeout;
    896     }
    897 
    898     while (mRemainingFrames > 0) {
    899 
    900         Buffer audioBuffer;
    901         audioBuffer.frameCount = mRemainingFrames;
    902         size_t nonContig;
    903         status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig);
    904         LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0),
    905                 "obtainBuffer() err=%d frameCount=%zu", err, audioBuffer.frameCount);
    906         requested = &ClientProxy::kNonBlocking;
    907         size_t avail = audioBuffer.frameCount + nonContig;
    908         ALOGV("obtainBuffer(%u) returned %zu = %zu + %zu err %d",
    909                 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
    910         if (err != NO_ERROR) {
    911             if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) {
    912                 break;
    913             }
    914             ALOGE("Error %d obtaining an audio buffer, giving up.", err);
    915             return NS_NEVER;
    916         }
    917 
    918         if (mRetryOnPartialBuffer) {
    919             mRetryOnPartialBuffer = false;
    920             if (avail < mRemainingFrames) {
    921                 int64_t myns = ((mRemainingFrames - avail) *
    922                         1100000000LL) / mSampleRate;
    923                 if (ns < 0 || myns < ns) {
    924                     ns = myns;
    925                 }
    926                 return ns;
    927             }
    928         }
    929 
    930         size_t reqSize = audioBuffer.size;
    931         mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
    932         size_t readSize = audioBuffer.size;
    933 
    934         // Sanity check on returned size
    935         if (ssize_t(readSize) < 0 || readSize > reqSize) {
    936             ALOGE("EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes",
    937                     reqSize, ssize_t(readSize));
    938             return NS_NEVER;
    939         }
    940 
    941         if (readSize == 0) {
    942             // The callback is done consuming buffers
    943             // Keep this thread going to handle timed events and
    944             // still try to provide more data in intervals of WAIT_PERIOD_MS
    945             // but don't just loop and block the CPU, so wait
    946             return WAIT_PERIOD_MS * 1000000LL;
    947         }
    948 
    949         size_t releasedFrames = readSize / mFrameSize;
    950         audioBuffer.frameCount = releasedFrames;
    951         mRemainingFrames -= releasedFrames;
    952         if (misalignment >= releasedFrames) {
    953             misalignment -= releasedFrames;
    954         } else {
    955             misalignment = 0;
    956         }
    957 
    958         releaseBuffer(&audioBuffer);
    959 
    960         // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer
    961         // if callback doesn't like to accept the full chunk
    962         if (readSize < reqSize) {
    963             continue;
    964         }
    965 
    966         // There could be enough non-contiguous frames available to satisfy the remaining request
    967         if (mRemainingFrames <= nonContig) {
    968             continue;
    969         }
    970 
    971 #if 0
    972         // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a
    973         // sum <= notificationFrames.  It replaces that series by at most two EVENT_MORE_DATA
    974         // that total to a sum == notificationFrames.
    975         if (0 < misalignment && misalignment <= mRemainingFrames) {
    976             mRemainingFrames = misalignment;
    977             return (mRemainingFrames * 1100000000LL) / mSampleRate;
    978         }
    979 #endif
    980 
    981     }
    982     mRemainingFrames = notificationFrames;
    983     mRetryOnPartialBuffer = true;
    984 
    985     // A lot has transpired since ns was calculated, so run again immediately and re-calculate
    986     return 0;
    987 }
    988 
    989 status_t AudioRecord::restoreRecord_l(const char *from)
    990 {
    991     ALOGW("dead IAudioRecord, creating a new one from %s()", from);
    992     ++mSequence;
    993     status_t result;
    994 
    995     // if the new IAudioRecord is created, openRecord_l() will modify the
    996     // following member variables: mAudioRecord, mCblkMemory, mCblk, mBufferMemory.
    997     // It will also delete the strong references on previous IAudioRecord and IMemory
    998     size_t position = mProxy->getPosition();
    999     mNewPosition = position + mUpdatePeriod;
   1000     result = openRecord_l(position);
   1001     if (result == NO_ERROR) {
   1002         if (mActive) {
   1003             // callback thread or sync event hasn't changed
   1004             // FIXME this fails if we have a new AudioFlinger instance
   1005             result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0);
   1006         }
   1007     }
   1008     if (result != NO_ERROR) {
   1009         ALOGW("restoreRecord_l() failed status %d", result);
   1010         mActive = false;
   1011     }
   1012 
   1013     return result;
   1014 }
   1015 
   1016 // =========================================================================
   1017 
   1018 void AudioRecord::DeathNotifier::binderDied(const wp<IBinder>& who __unused)
   1019 {
   1020     sp<AudioRecord> audioRecord = mAudioRecord.promote();
   1021     if (audioRecord != 0) {
   1022         AutoMutex lock(audioRecord->mLock);
   1023         audioRecord->mProxy->binderDied();
   1024     }
   1025 }
   1026 
   1027 // =========================================================================
   1028 
   1029 AudioRecord::AudioRecordThread::AudioRecordThread(AudioRecord& receiver, bool bCanCallJava)
   1030     : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL),
   1031       mIgnoreNextPausedInt(false)
   1032 {
   1033 }
   1034 
   1035 AudioRecord::AudioRecordThread::~AudioRecordThread()
   1036 {
   1037 }
   1038 
   1039 bool AudioRecord::AudioRecordThread::threadLoop()
   1040 {
   1041     {
   1042         AutoMutex _l(mMyLock);
   1043         if (mPaused) {
   1044             mMyCond.wait(mMyLock);
   1045             // caller will check for exitPending()
   1046             return true;
   1047         }
   1048         if (mIgnoreNextPausedInt) {
   1049             mIgnoreNextPausedInt = false;
   1050             mPausedInt = false;
   1051         }
   1052         if (mPausedInt) {
   1053             if (mPausedNs > 0) {
   1054                 (void) mMyCond.waitRelative(mMyLock, mPausedNs);
   1055             } else {
   1056                 mMyCond.wait(mMyLock);
   1057             }
   1058             mPausedInt = false;
   1059             return true;
   1060         }
   1061     }
   1062     nsecs_t ns =  mReceiver.processAudioBuffer();
   1063     switch (ns) {
   1064     case 0:
   1065         return true;
   1066     case NS_INACTIVE:
   1067         pauseInternal();
   1068         return true;
   1069     case NS_NEVER:
   1070         return false;
   1071     case NS_WHENEVER:
   1072         // FIXME increase poll interval, or make event-driven
   1073         ns = 1000000000LL;
   1074         // fall through
   1075     default:
   1076         LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns);
   1077         pauseInternal(ns);
   1078         return true;
   1079     }
   1080 }
   1081 
   1082 void AudioRecord::AudioRecordThread::requestExit()
   1083 {
   1084     // must be in this order to avoid a race condition
   1085     Thread::requestExit();
   1086     resume();
   1087 }
   1088 
   1089 void AudioRecord::AudioRecordThread::pause()
   1090 {
   1091     AutoMutex _l(mMyLock);
   1092     mPaused = true;
   1093 }
   1094 
   1095 void AudioRecord::AudioRecordThread::resume()
   1096 {
   1097     AutoMutex _l(mMyLock);
   1098     mIgnoreNextPausedInt = true;
   1099     if (mPaused || mPausedInt) {
   1100         mPaused = false;
   1101         mPausedInt = false;
   1102         mMyCond.signal();
   1103     }
   1104 }
   1105 
   1106 void AudioRecord::AudioRecordThread::pauseInternal(nsecs_t ns)
   1107 {
   1108     AutoMutex _l(mMyLock);
   1109     mPausedInt = true;
   1110     mPausedNs = ns;
   1111 }
   1112 
   1113 // -------------------------------------------------------------------------
   1114 
   1115 }; // namespace android
   1116