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