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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 <stdint.h>
     22 #include <sys/types.h>
     23 
     24 #include <sched.h>
     25 #include <sys/resource.h>
     26 
     27 #include <private/media/AudioTrackShared.h>
     28 
     29 #include <media/AudioSystem.h>
     30 #include <media/AudioRecord.h>
     31 #include <media/mediarecorder.h>
     32 
     33 #include <binder/IServiceManager.h>
     34 #include <utils/Log.h>
     35 #include <binder/Parcel.h>
     36 #include <binder/IPCThreadState.h>
     37 #include <utils/Timers.h>
     38 #include <cutils/atomic.h>
     39 
     40 #define LIKELY( exp )       (__builtin_expect( (exp) != 0, true  ))
     41 #define UNLIKELY( exp )     (__builtin_expect( (exp) != 0, false ))
     42 
     43 namespace android {
     44 // ---------------------------------------------------------------------------
     45 
     46 // static
     47 status_t AudioRecord::getMinFrameCount(
     48         int* frameCount,
     49         uint32_t sampleRate,
     50         int format,
     51         int channelCount)
     52 {
     53     size_t size = 0;
     54     if (AudioSystem::getInputBufferSize(sampleRate, format, channelCount, &size)
     55             != NO_ERROR) {
     56         LOGE("AudioSystem could not query the input buffer size.");
     57         return NO_INIT;
     58     }
     59 
     60     if (size == 0) {
     61         LOGE("Unsupported configuration: sampleRate %d, format %d, channelCount %d",
     62             sampleRate, format, channelCount);
     63         return BAD_VALUE;
     64     }
     65 
     66     // We double the size of input buffer for ping pong use of record buffer.
     67     size <<= 1;
     68 
     69     if (AudioSystem::isLinearPCM(format)) {
     70         size /= channelCount * (format == AudioSystem::PCM_16_BIT ? 2 : 1);
     71     }
     72 
     73     *frameCount = size;
     74     return NO_ERROR;
     75 }
     76 
     77 // ---------------------------------------------------------------------------
     78 
     79 AudioRecord::AudioRecord()
     80     : mStatus(NO_INIT), mSessionId(0)
     81 {
     82 }
     83 
     84 AudioRecord::AudioRecord(
     85         int inputSource,
     86         uint32_t sampleRate,
     87         int format,
     88         uint32_t channels,
     89         int frameCount,
     90         uint32_t flags,
     91         callback_t cbf,
     92         void* user,
     93         int notificationFrames,
     94         int sessionId)
     95     : mStatus(NO_INIT), mSessionId(0)
     96 {
     97     mStatus = set(inputSource, sampleRate, format, channels,
     98             frameCount, flags, cbf, user, notificationFrames, sessionId);
     99 }
    100 
    101 AudioRecord::~AudioRecord()
    102 {
    103     if (mStatus == NO_ERROR) {
    104         // Make sure that callback function exits in the case where
    105         // it is looping on buffer empty condition in obtainBuffer().
    106         // Otherwise the callback thread will never exit.
    107         stop();
    108         if (mClientRecordThread != 0) {
    109             mClientRecordThread->requestExitAndWait();
    110             mClientRecordThread.clear();
    111         }
    112         mAudioRecord.clear();
    113         IPCThreadState::self()->flushCommands();
    114     }
    115 }
    116 
    117 status_t AudioRecord::set(
    118         int inputSource,
    119         uint32_t sampleRate,
    120         int format,
    121         uint32_t channels,
    122         int frameCount,
    123         uint32_t flags,
    124         callback_t cbf,
    125         void* user,
    126         int notificationFrames,
    127         bool threadCanCallJava,
    128         int sessionId)
    129 {
    130 
    131     LOGV("set(): sampleRate %d, channels %d, frameCount %d",sampleRate, channels, frameCount);
    132     if (mAudioRecord != 0) {
    133         return INVALID_OPERATION;
    134     }
    135 
    136     if (inputSource == AUDIO_SOURCE_DEFAULT) {
    137         inputSource = AUDIO_SOURCE_MIC;
    138     }
    139 
    140     if (sampleRate == 0) {
    141         sampleRate = DEFAULT_SAMPLE_RATE;
    142     }
    143     // these below should probably come from the audioFlinger too...
    144     if (format == 0) {
    145         format = AudioSystem::PCM_16_BIT;
    146     }
    147     // validate parameters
    148     if (!AudioSystem::isValidFormat(format)) {
    149         LOGE("Invalid format");
    150         return BAD_VALUE;
    151     }
    152 
    153     if (!AudioSystem::isInputChannel(channels)) {
    154         return BAD_VALUE;
    155     }
    156 
    157     int channelCount = AudioSystem::popCount(channels);
    158 
    159     audio_io_handle_t input = AudioSystem::getInput(inputSource,
    160                                     sampleRate, format, channels, (AudioSystem::audio_in_acoustics)flags);
    161     if (input == 0) {
    162         LOGE("Could not get audio input for record source %d", inputSource);
    163         return BAD_VALUE;
    164     }
    165 
    166     // validate framecount
    167     int minFrameCount = 0;
    168     status_t status = getMinFrameCount(&minFrameCount, sampleRate, format, channelCount);
    169     if (status != NO_ERROR) {
    170         return status;
    171     }
    172     LOGV("AudioRecord::set() minFrameCount = %d", minFrameCount);
    173 
    174     if (frameCount == 0) {
    175         frameCount = minFrameCount;
    176     } else if (frameCount < minFrameCount) {
    177         return BAD_VALUE;
    178     }
    179 
    180     if (notificationFrames == 0) {
    181         notificationFrames = frameCount/2;
    182     }
    183 
    184     mSessionId = sessionId;
    185 
    186     // create the IAudioRecord
    187     status = openRecord(sampleRate, format, channelCount,
    188                         frameCount, flags, input);
    189     if (status != NO_ERROR) {
    190         return status;
    191     }
    192 
    193     if (cbf != 0) {
    194         mClientRecordThread = new ClientRecordThread(*this, threadCanCallJava);
    195         if (mClientRecordThread == 0) {
    196             return NO_INIT;
    197         }
    198     }
    199 
    200     mStatus = NO_ERROR;
    201 
    202     mFormat = format;
    203     // Update buffer size in case it has been limited by AudioFlinger during track creation
    204     mFrameCount = mCblk->frameCount;
    205     mChannelCount = (uint8_t)channelCount;
    206     mChannels = channels;
    207     mActive = 0;
    208     mCbf = cbf;
    209     mNotificationFrames = notificationFrames;
    210     mRemainingFrames = notificationFrames;
    211     mUserData = user;
    212     // TODO: add audio hardware input latency here
    213     mLatency = (1000*mFrameCount) / sampleRate;
    214     mMarkerPosition = 0;
    215     mMarkerReached = false;
    216     mNewPosition = 0;
    217     mUpdatePeriod = 0;
    218     mInputSource = (uint8_t)inputSource;
    219     mFlags = flags;
    220     mInput = input;
    221 
    222     return NO_ERROR;
    223 }
    224 
    225 status_t AudioRecord::initCheck() const
    226 {
    227     return mStatus;
    228 }
    229 
    230 // -------------------------------------------------------------------------
    231 
    232 uint32_t AudioRecord::latency() const
    233 {
    234     return mLatency;
    235 }
    236 
    237 int AudioRecord::format() const
    238 {
    239     return mFormat;
    240 }
    241 
    242 int AudioRecord::channelCount() const
    243 {
    244     return mChannelCount;
    245 }
    246 
    247 uint32_t AudioRecord::frameCount() const
    248 {
    249     return mFrameCount;
    250 }
    251 
    252 int AudioRecord::frameSize() const
    253 {
    254     if (AudioSystem::isLinearPCM(mFormat)) {
    255         return channelCount()*((format() == AudioSystem::PCM_8_BIT) ? sizeof(uint8_t) : sizeof(int16_t));
    256     } else {
    257         return sizeof(uint8_t);
    258     }
    259 }
    260 
    261 int AudioRecord::inputSource() const
    262 {
    263     return (int)mInputSource;
    264 }
    265 
    266 // -------------------------------------------------------------------------
    267 
    268 status_t AudioRecord::start()
    269 {
    270     status_t ret = NO_ERROR;
    271     sp<ClientRecordThread> t = mClientRecordThread;
    272 
    273     LOGV("start");
    274 
    275     if (t != 0) {
    276         if (t->exitPending()) {
    277             if (t->requestExitAndWait() == WOULD_BLOCK) {
    278                 LOGE("AudioRecord::start called from thread");
    279                 return WOULD_BLOCK;
    280             }
    281         }
    282         t->mLock.lock();
    283      }
    284 
    285     if (android_atomic_or(1, &mActive) == 0) {
    286         ret = mAudioRecord->start();
    287         if (ret == DEAD_OBJECT) {
    288             LOGV("start() dead IAudioRecord: creating a new one");
    289             ret = openRecord(mCblk->sampleRate, mFormat, mChannelCount,
    290                     mFrameCount, mFlags, getInput());
    291             if (ret == NO_ERROR) {
    292                 ret = mAudioRecord->start();
    293             }
    294         }
    295         if (ret == NO_ERROR) {
    296             mNewPosition = mCblk->user + mUpdatePeriod;
    297             mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
    298             mCblk->waitTimeMs = 0;
    299             if (t != 0) {
    300                t->run("ClientRecordThread", THREAD_PRIORITY_AUDIO_CLIENT);
    301             } else {
    302                 setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT);
    303             }
    304         } else {
    305             LOGV("start() failed");
    306             android_atomic_and(~1, &mActive);
    307         }
    308     }
    309 
    310     if (t != 0) {
    311         t->mLock.unlock();
    312     }
    313 
    314     return ret;
    315 }
    316 
    317 status_t AudioRecord::stop()
    318 {
    319     sp<ClientRecordThread> t = mClientRecordThread;
    320 
    321     LOGV("stop");
    322 
    323     if (t != 0) {
    324         t->mLock.lock();
    325      }
    326 
    327     if (android_atomic_and(~1, &mActive) == 1) {
    328         mCblk->cv.signal();
    329         mAudioRecord->stop();
    330         // the record head position will reset to 0, so if a marker is set, we need
    331         // to activate it again
    332         mMarkerReached = false;
    333         if (t != 0) {
    334             t->requestExit();
    335         } else {
    336             setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
    337         }
    338     }
    339 
    340     if (t != 0) {
    341         t->mLock.unlock();
    342     }
    343 
    344     return NO_ERROR;
    345 }
    346 
    347 bool AudioRecord::stopped() const
    348 {
    349     return !mActive;
    350 }
    351 
    352 uint32_t AudioRecord::getSampleRate()
    353 {
    354     return mCblk->sampleRate;
    355 }
    356 
    357 status_t AudioRecord::setMarkerPosition(uint32_t marker)
    358 {
    359     if (mCbf == 0) return INVALID_OPERATION;
    360 
    361     mMarkerPosition = marker;
    362     mMarkerReached = false;
    363 
    364     return NO_ERROR;
    365 }
    366 
    367 status_t AudioRecord::getMarkerPosition(uint32_t *marker)
    368 {
    369     if (marker == 0) return BAD_VALUE;
    370 
    371     *marker = mMarkerPosition;
    372 
    373     return NO_ERROR;
    374 }
    375 
    376 status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod)
    377 {
    378     if (mCbf == 0) return INVALID_OPERATION;
    379 
    380     uint32_t curPosition;
    381     getPosition(&curPosition);
    382     mNewPosition = curPosition + updatePeriod;
    383     mUpdatePeriod = updatePeriod;
    384 
    385     return NO_ERROR;
    386 }
    387 
    388 status_t AudioRecord::getPositionUpdatePeriod(uint32_t *updatePeriod)
    389 {
    390     if (updatePeriod == 0) return BAD_VALUE;
    391 
    392     *updatePeriod = mUpdatePeriod;
    393 
    394     return NO_ERROR;
    395 }
    396 
    397 status_t AudioRecord::getPosition(uint32_t *position)
    398 {
    399     if (position == 0) return BAD_VALUE;
    400 
    401     *position = mCblk->user;
    402 
    403     return NO_ERROR;
    404 }
    405 
    406 unsigned int AudioRecord::getInputFramesLost()
    407 {
    408     if (mActive)
    409         return AudioSystem::getInputFramesLost(mInput);
    410     else
    411         return 0;
    412 }
    413 
    414 // -------------------------------------------------------------------------
    415 
    416 status_t AudioRecord::openRecord(
    417         uint32_t sampleRate,
    418         int format,
    419         int channelCount,
    420         int frameCount,
    421         uint32_t flags,
    422         audio_io_handle_t input)
    423 {
    424     status_t status;
    425     const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
    426     if (audioFlinger == 0) {
    427         return NO_INIT;
    428     }
    429 
    430     sp<IAudioRecord> record = audioFlinger->openRecord(getpid(), input,
    431                                                        sampleRate, format,
    432                                                        channelCount,
    433                                                        frameCount,
    434                                                        ((uint16_t)flags) << 16,
    435                                                        &mSessionId,
    436                                                        &status);
    437     if (record == 0) {
    438         LOGE("AudioFlinger could not create record track, status: %d", status);
    439         return status;
    440     }
    441     sp<IMemory> cblk = record->getCblk();
    442     if (cblk == 0) {
    443         LOGE("Could not get control block");
    444         return NO_INIT;
    445     }
    446     mAudioRecord.clear();
    447     mAudioRecord = record;
    448     mCblkMemory.clear();
    449     mCblkMemory = cblk;
    450     mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
    451     mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
    452     mCblk->flags &= ~CBLK_DIRECTION_MSK;
    453     mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
    454     mCblk->waitTimeMs = 0;
    455     return NO_ERROR;
    456 }
    457 
    458 status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
    459 {
    460     int active;
    461     status_t result;
    462     audio_track_cblk_t* cblk = mCblk;
    463     uint32_t framesReq = audioBuffer->frameCount;
    464     uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS;
    465 
    466     audioBuffer->frameCount  = 0;
    467     audioBuffer->size        = 0;
    468 
    469     uint32_t framesReady = cblk->framesReady();
    470 
    471     if (framesReady == 0) {
    472         cblk->lock.lock();
    473         goto start_loop_here;
    474         while (framesReady == 0) {
    475             active = mActive;
    476             if (UNLIKELY(!active)) {
    477                 cblk->lock.unlock();
    478                 return NO_MORE_BUFFERS;
    479             }
    480             if (UNLIKELY(!waitCount)) {
    481                 cblk->lock.unlock();
    482                 return WOULD_BLOCK;
    483             }
    484             result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
    485             if (__builtin_expect(result!=NO_ERROR, false)) {
    486                 cblk->waitTimeMs += waitTimeMs;
    487                 if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
    488                     LOGW(   "obtainBuffer timed out (is the CPU pegged?) "
    489                             "user=%08x, server=%08x", cblk->user, cblk->server);
    490                     cblk->lock.unlock();
    491                     result = mAudioRecord->start();
    492                     if (result == DEAD_OBJECT) {
    493                         LOGW("obtainBuffer() dead IAudioRecord: creating a new one");
    494                         result = openRecord(cblk->sampleRate, mFormat, mChannelCount,
    495                                             mFrameCount, mFlags, getInput());
    496                         if (result == NO_ERROR) {
    497                             cblk = mCblk;
    498                             mAudioRecord->start();
    499                         }
    500                     }
    501                     cblk->lock.lock();
    502                     cblk->waitTimeMs = 0;
    503                 }
    504                 if (--waitCount == 0) {
    505                     cblk->lock.unlock();
    506                     return TIMED_OUT;
    507                 }
    508             }
    509             // read the server count again
    510         start_loop_here:
    511             framesReady = cblk->framesReady();
    512         }
    513         cblk->lock.unlock();
    514     }
    515 
    516     cblk->waitTimeMs = 0;
    517 
    518     if (framesReq > framesReady) {
    519         framesReq = framesReady;
    520     }
    521 
    522     uint32_t u = cblk->user;
    523     uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
    524 
    525     if (u + framesReq > bufferEnd) {
    526         framesReq = bufferEnd - u;
    527     }
    528 
    529     audioBuffer->flags       = 0;
    530     audioBuffer->channelCount= mChannelCount;
    531     audioBuffer->format      = mFormat;
    532     audioBuffer->frameCount  = framesReq;
    533     audioBuffer->size        = framesReq*cblk->frameSize;
    534     audioBuffer->raw         = (int8_t*)cblk->buffer(u);
    535     active = mActive;
    536     return active ? status_t(NO_ERROR) : status_t(STOPPED);
    537 }
    538 
    539 void AudioRecord::releaseBuffer(Buffer* audioBuffer)
    540 {
    541     audio_track_cblk_t* cblk = mCblk;
    542     cblk->stepUser(audioBuffer->frameCount);
    543 }
    544 
    545 audio_io_handle_t AudioRecord::getInput()
    546 {
    547     mInput = AudioSystem::getInput(mInputSource,
    548                                 mCblk->sampleRate,
    549                                 mFormat, mChannels,
    550                                 (AudioSystem::audio_in_acoustics)mFlags);
    551     return mInput;
    552 }
    553 
    554 int AudioRecord::getSessionId()
    555 {
    556     return mSessionId;
    557 }
    558 
    559 // -------------------------------------------------------------------------
    560 
    561 ssize_t AudioRecord::read(void* buffer, size_t userSize)
    562 {
    563     ssize_t read = 0;
    564     Buffer audioBuffer;
    565     int8_t *dst = static_cast<int8_t*>(buffer);
    566 
    567     if (ssize_t(userSize) < 0) {
    568         // sanity-check. user is most-likely passing an error code.
    569         LOGE("AudioRecord::read(buffer=%p, size=%u (%d)",
    570                 buffer, userSize, userSize);
    571         return BAD_VALUE;
    572     }
    573 
    574 
    575     do {
    576 
    577         audioBuffer.frameCount = userSize/frameSize();
    578 
    579         // By using a wait count corresponding to twice the timeout period in
    580         // obtainBuffer() we give a chance to recover once for a read timeout
    581         // (if media_server crashed for instance) before returning a length of
    582         // 0 bytes read to the client
    583         status_t err = obtainBuffer(&audioBuffer, ((2 * MAX_RUN_TIMEOUT_MS) / WAIT_PERIOD_MS));
    584         if (err < 0) {
    585             // out of buffers, return #bytes written
    586             if (err == status_t(NO_MORE_BUFFERS))
    587                 break;
    588             if (err == status_t(TIMED_OUT))
    589                 err = 0;
    590             return ssize_t(err);
    591         }
    592 
    593         size_t bytesRead = audioBuffer.size;
    594         memcpy(dst, audioBuffer.i8, bytesRead);
    595 
    596         dst += bytesRead;
    597         userSize -= bytesRead;
    598         read += bytesRead;
    599 
    600         releaseBuffer(&audioBuffer);
    601     } while (userSize);
    602 
    603     return read;
    604 }
    605 
    606 // -------------------------------------------------------------------------
    607 
    608 bool AudioRecord::processAudioBuffer(const sp<ClientRecordThread>& thread)
    609 {
    610     Buffer audioBuffer;
    611     uint32_t frames = mRemainingFrames;
    612     size_t readSize;
    613 
    614     // Manage marker callback
    615     if (!mMarkerReached && (mMarkerPosition > 0)) {
    616         if (mCblk->user >= mMarkerPosition) {
    617             mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition);
    618             mMarkerReached = true;
    619         }
    620     }
    621 
    622     // Manage new position callback
    623     if (mUpdatePeriod > 0) {
    624         while (mCblk->user >= mNewPosition) {
    625             mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition);
    626             mNewPosition += mUpdatePeriod;
    627         }
    628     }
    629 
    630     do {
    631         audioBuffer.frameCount = frames;
    632         // Calling obtainBuffer() with a wait count of 1
    633         // limits wait time to WAIT_PERIOD_MS. This prevents from being
    634         // stuck here not being able to handle timed events (position, markers).
    635         status_t err = obtainBuffer(&audioBuffer, 1);
    636         if (err < NO_ERROR) {
    637             if (err != TIMED_OUT) {
    638                 LOGE_IF(err != status_t(NO_MORE_BUFFERS), "Error obtaining an audio buffer, giving up.");
    639                 return false;
    640             }
    641             break;
    642         }
    643         if (err == status_t(STOPPED)) return false;
    644 
    645         size_t reqSize = audioBuffer.size;
    646         mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
    647         readSize = audioBuffer.size;
    648 
    649         // Sanity check on returned size
    650         if (ssize_t(readSize) <= 0) {
    651             // The callback is done filling buffers
    652             // Keep this thread going to handle timed events and
    653             // still try to get more data in intervals of WAIT_PERIOD_MS
    654             // but don't just loop and block the CPU, so wait
    655             usleep(WAIT_PERIOD_MS*1000);
    656             break;
    657         }
    658         if (readSize > reqSize) readSize = reqSize;
    659 
    660         audioBuffer.size = readSize;
    661         audioBuffer.frameCount = readSize/frameSize();
    662         frames -= audioBuffer.frameCount;
    663 
    664         releaseBuffer(&audioBuffer);
    665 
    666     } while (frames);
    667 
    668 
    669     // Manage overrun callback
    670     if (mActive && (mCblk->framesAvailable_l() == 0)) {
    671         LOGV("Overrun user: %x, server: %x, flags %04x", mCblk->user, mCblk->server, mCblk->flags);
    672         if ((mCblk->flags & CBLK_UNDERRUN_MSK) == CBLK_UNDERRUN_OFF) {
    673             mCbf(EVENT_OVERRUN, mUserData, 0);
    674             mCblk->flags |= CBLK_UNDERRUN_ON;
    675         }
    676     }
    677 
    678     if (frames == 0) {
    679         mRemainingFrames = mNotificationFrames;
    680     } else {
    681         mRemainingFrames = frames;
    682     }
    683     return true;
    684 }
    685 
    686 // =========================================================================
    687 
    688 AudioRecord::ClientRecordThread::ClientRecordThread(AudioRecord& receiver, bool bCanCallJava)
    689     : Thread(bCanCallJava), mReceiver(receiver)
    690 {
    691 }
    692 
    693 bool AudioRecord::ClientRecordThread::threadLoop()
    694 {
    695     return mReceiver.processAudioBuffer(this);
    696 }
    697 
    698 // -------------------------------------------------------------------------
    699 
    700 }; // namespace android
    701 
    702