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Commit 34f1d8ec authored by Eric Laurent's avatar Eric Laurent
Browse files

Fix issue 2203561: Sholes: audio playing out of earpiece. 

Create a new IAudioTrack interface to AudioFlinger when start() fails due to a broken pipe error.
Do the same if start fails due to the same error after time out in obtainBuffer().
Do not indicate that the AudioTrack is started to AudioPolicyManager if IAudioTrack start fails.
This avoids that an AudioTrack keeps a dead IAudioTrack after a media server crash.

Same modifications for AudioRecord.

Add a flag to ToneGenerator indicating that the callback thread can call Java. Without it, when the media server crashes and restarts, the AudioSystem error callback will crash in JNI if the IAudiotrack is created from AudioTrack callback thread.
parent c6c6237a

include/media/AudioRecord.h

+6 −0
Original line number Diff line number Diff line
@@ -314,6 +314,11 @@ private: 
    };



            bool processAudioBuffer(const sp<ClientRecordThread>& thread);

            status_t openRecord(uint32_t sampleRate,

                                int format,

                                int channelCount,

                                int frameCount,

                                uint32_t flags);



    sp<IAudioRecord>        mAudioRecord;

    sp<IMemory>             mCblkMemory;
@@ -341,6 +346,7 @@ private: 
    uint32_t                mNewPosition;

    uint32_t                mUpdatePeriod;

    audio_io_handle_t       mInput;

    uint32_t                mFlags;

};



}; // namespace android

include/media/AudioTrack.h

+8 −0
Original line number Diff line number Diff line
@@ -391,6 +391,14 @@ private: 
    };



            bool processAudioBuffer(const sp<AudioTrackThread>& thread);

            status_t createTrack(int streamType,

                                 uint32_t sampleRate,

                                 int format,

                                 int channelCount,

                                 int frameCount,

                                 uint32_t flags,

                                 const sp<IMemory>& sharedBuffer,

                                 audio_io_handle_t output);



    sp<IAudioTrack>         mAudioTrack;

    sp<IMemory>             mCblkMemory;

include/media/ToneGenerator.h

+2 −1
Original line number Diff line number Diff line
@@ -151,7 +151,7 @@ public: 
        NUM_SUP_TONES = LAST_SUP_TONE-FIRST_SUP_TONE+1

    };



    ToneGenerator(int streamType, float volume);

    ToneGenerator(int streamType, float volume, bool threadCanCallJava = false);

    ~ToneGenerator();



    bool startTone(int toneType, int durationMs = -1);
@@ -242,6 +242,7 @@ private: 


    static const ToneDescriptor sToneDescriptors[];



    bool mThreadCanCallJava;

    unsigned int mTotalSmp;  // Total number of audio samples played (gives current time)

    unsigned int mNextSegSmp;  // Position of next segment transition expressed in samples

    // NOTE: because mTotalSmp, mNextSegSmp are stored on 32 bit, current design will operate properly

media/libmedia/AudioRecord.cpp

+85 −42
Original line number Diff line number Diff line
@@ -101,11 +101,6 @@ status_t AudioRecord::set( 
        return INVALID_OPERATION;

    }



    const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();

    if (audioFlinger == 0) {

        return NO_INIT;

    }



    if (inputSource == AUDIO_SOURCE_DEFAULT) {

        inputSource = AUDIO_SOURCE_MIC;

    }
@@ -171,22 +166,14 @@ status_t AudioRecord::set( 
        notificationFrames = frameCount/2;

    }



    // open record channel

    status_t status;

    sp<IAudioRecord> record = audioFlinger->openRecord(getpid(), mInput,

                                                       sampleRate, format,

                                                       channelCount,

                                                       frameCount,

                                                       ((uint16_t)flags) << 16,

                                                       &status);

    if (record == 0) {

        LOGE("AudioFlinger could not create record track, status: %d", status);

    // create the IAudioRecord

    status_t status = openRecord(sampleRate, format, channelCount,

                                 frameCount, flags);



    if (status != NO_ERROR) {

        return status;

    }

    sp<IMemory> cblk = record->getCblk();

    if (cblk == 0) {

        return NO_INIT;

    }



    if (cbf != 0) {

        mClientRecordThread = new ClientRecordThread(*this, threadCanCallJava);

        if (mClientRecordThread == 0) {
@@ -196,11 +183,6 @@ status_t AudioRecord::set( 


    mStatus = NO_ERROR;



    mAudioRecord = record;

    mCblkMemory = cblk;

    mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());

    mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);

    mCblk->out = 0;

    mFormat = format;

    // Update buffer size in case it has been limited by AudioFlinger during track creation

    mFrameCount = mCblk->frameCount;
@@ -217,6 +199,7 @@ status_t AudioRecord::set( 
    mNewPosition = 0;

    mUpdatePeriod = 0;

    mInputSource = (uint8_t)inputSource;

    mFlags = flags;



    return NO_ERROR;

}
@@ -283,6 +266,13 @@ status_t AudioRecord::start() 


    if (android_atomic_or(1, &mActive) == 0) {

        ret = AudioSystem::startInput(mInput);

        if (ret == NO_ERROR) {

            ret = mAudioRecord->start();

            if (ret == DEAD_OBJECT) {

                LOGV("start() dead IAudioRecord: creating a new one");

                ret = openRecord(mCblk->sampleRate, mFormat, mChannelCount,

                        mFrameCount, mFlags);

            }

            if (ret == NO_ERROR) {

                mNewPosition = mCblk->user + mUpdatePeriod;

                mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
@@ -292,7 +282,11 @@ status_t AudioRecord::start() 
                } else {

                    setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT);

                }

            ret = mAudioRecord->start();

            } else {

                LOGV("start() failed");

                AudioSystem::stopInput(mInput);

                android_atomic_and(~1, &mActive);

            }

        }

    }
@@ -396,10 +390,48 @@ status_t AudioRecord::getPosition(uint32_t *position) 


// -------------------------------------------------------------------------



status_t AudioRecord::openRecord(

        uint32_t sampleRate,

        int format,

        int channelCount,

        int frameCount,

        uint32_t flags)

{

    status_t status;

    const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();

    if (audioFlinger == 0) {

        return NO_INIT;

    }



    sp<IAudioRecord> record = audioFlinger->openRecord(getpid(), mInput,

                                                       sampleRate, format,

                                                       channelCount,

                                                       frameCount,

                                                       ((uint16_t)flags) << 16,

                                                       &status);

    if (record == 0) {

        LOGE("AudioFlinger could not create record track, status: %d", status);

        return status;

    }

    sp<IMemory> cblk = record->getCblk();

    if (cblk == 0) {

        LOGE("Could not get control block");

        return NO_INIT;

    }

    mAudioRecord.clear();

    mAudioRecord = record;

    mCblkMemory.clear();

    mCblkMemory = cblk;

    mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());

    mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);

    mCblk->out = 0;



    return NO_ERROR;

}



status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)

{

    int active;

    int timeout = 0;

    status_t result;

    audio_track_cblk_t* cblk = mCblk;

    uint32_t framesReq = audioBuffer->frameCount;
@@ -411,25 +443,40 @@ status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 
    uint32_t framesReady = cblk->framesReady();



    if (framesReady == 0) {

        Mutex::Autolock _l(cblk->lock);

        cblk->lock.lock();

        goto start_loop_here;

        while (framesReady == 0) {

            active = mActive;

            if (UNLIKELY(!active))

            if (UNLIKELY(!active)) {

                cblk->lock.unlock();

                return NO_MORE_BUFFERS;

            if (UNLIKELY(!waitCount))

            }

            if (UNLIKELY(!waitCount)) {

                cblk->lock.unlock();

                return WOULD_BLOCK;

            timeout = 0;

            }

            result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));

            if (__builtin_expect(result!=NO_ERROR, false)) {

                cblk->waitTimeMs += waitTimeMs;

                if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {

                    LOGW(   "obtainBuffer timed out (is the CPU pegged?) "

                            "user=%08x, server=%08x", cblk->user, cblk->server);

                    timeout = 1;

                    cblk->lock.unlock();

                    result = mAudioRecord->start();

                    if (result == DEAD_OBJECT) {

                        LOGW("obtainBuffer() dead IAudioRecord: creating a new one");

                        result = openRecord(cblk->sampleRate, mFormat, mChannelCount,

                                            mFrameCount, mFlags);

                        if (result == NO_ERROR) {

                            cblk = mCblk;

                            cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;

                        }

                    }

                    cblk->lock.lock();

                    cblk->waitTimeMs = 0;

                }

                if (--waitCount == 0) {

                    cblk->lock.unlock();

                    return TIMED_OUT;

                }

            }
@@ -437,13 +484,9 @@ status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 
        start_loop_here:

            framesReady = cblk->framesReady();

        }

        cblk->lock.unlock();

    }



    LOGW_IF(timeout,

        "*** SERIOUS WARNING *** obtainBuffer() timed out "

        "but didn't need to be locked. We recovered, but "

        "this shouldn't happen (user=%08x, server=%08x)", cblk->user, cblk->server);



    cblk->waitTimeMs = 0;



    if (framesReq > framesReady) {

media/libmedia/AudioTrack.cpp

+103 −60
Original line number Diff line number Diff line
@@ -122,11 +122,6 @@ status_t AudioTrack::set( 
        return INVALID_OPERATION;

    }



    const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();

    if (audioFlinger == 0) {

       LOGE("Could not get audioflinger");

       return NO_INIT;

    }

    int afSampleRate;

    if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {

        return NO_INIT;
@@ -217,28 +212,16 @@ status_t AudioTrack::set( 
        }

    }



    // create the track

    status_t status;

    sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),

                                                      streamType,

                                                      sampleRate,

                                                      format,

                                                      channelCount,

                                                      frameCount,

                                                      ((uint16_t)flags) << 16,

                                                      sharedBuffer,

                                                      output,

                                                      &status);

    mVolume[LEFT] = 1.0f;

    mVolume[RIGHT] = 1.0f;

    // create the IAudioTrack

    status_t status = createTrack(streamType, sampleRate, format, channelCount,

                                  frameCount, flags, sharedBuffer, output);



    if (track == 0) {

        LOGE("AudioFlinger could not create track, status: %d", status);

    if (status != NO_ERROR) {

        return status;

    }

    sp<IMemory> cblk = track->getCblk();

    if (cblk == 0) {

        LOGE("Could not get control block");

        return NO_INIT;

    }



    if (cbf != 0) {

        mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);

        if (mAudioTrackThread == 0) {
@@ -249,22 +232,6 @@ status_t AudioTrack::set( 


    mStatus = NO_ERROR;



    mAudioTrack = track;

    mCblkMemory = cblk;

    mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());

    mCblk->out = 1;

    // Update buffer size in case it has been limited by AudioFlinger during track creation

    mFrameCount = mCblk->frameCount;

    if (sharedBuffer == 0) {

        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);

    } else {

        mCblk->buffers = sharedBuffer->pointer();

         // Force buffer full condition as data is already present in shared memory

        mCblk->stepUser(mFrameCount);

    }

    mCblk->volume[0] = mCblk->volume[1] = 0x1000;

    mVolume[LEFT] = 1.0f;

    mVolume[RIGHT] = 1.0f;

    mStreamType = streamType;

    mFormat = format;

    mChannels = channels;
@@ -351,7 +318,15 @@ void AudioTrack::start() 
     }



    if (android_atomic_or(1, &mActive) == 0) {

        AudioSystem::startOutput(getOutput(), (AudioSystem::stream_type)mStreamType);

        audio_io_handle_t output = AudioTrack::getOutput();

        status_t status = mAudioTrack->start();

        if (status == DEAD_OBJECT) {

            LOGV("start() dead IAudioTrack: creating a new one");

            status = createTrack(mStreamType, mCblk->sampleRate, mFormat, mChannelCount,

                                 mFrameCount, mFlags, mSharedBuffer, output);

        }

        if (status == NO_ERROR) {

            AudioSystem::startOutput(output, (AudioSystem::stream_type)mStreamType);

            mNewPosition = mCblk->server + mUpdatePeriod;

            mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;

            mCblk->waitTimeMs = 0;
@@ -360,7 +335,10 @@ void AudioTrack::start() 
            } else {

                setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT);

            }

        mAudioTrack->start();

        } else {

            LOGV("start() failed");

            android_atomic_and(~1, &mActive);

        }

    }



    if (t != 0) {
@@ -617,10 +595,67 @@ audio_io_handle_t AudioTrack::getOutput() 


// -------------------------------------------------------------------------



status_t AudioTrack::createTrack(

        int streamType,

        uint32_t sampleRate,

        int format,

        int channelCount,

        int frameCount,

        uint32_t flags,

        const sp<IMemory>& sharedBuffer,

        audio_io_handle_t output)

{

    status_t status;

    const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();

    if (audioFlinger == 0) {

       LOGE("Could not get audioflinger");

       return NO_INIT;

    }



    sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),

                                                      streamType,

                                                      sampleRate,

                                                      format,

                                                      channelCount,

                                                      frameCount,

                                                      ((uint16_t)flags) << 16,

                                                      sharedBuffer,

                                                      output,

                                                      &status);



    if (track == 0) {

        LOGE("AudioFlinger could not create track, status: %d", status);

        return status;

    }

    sp<IMemory> cblk = track->getCblk();

    if (cblk == 0) {

        LOGE("Could not get control block");

        return NO_INIT;

    }

    mAudioTrack.clear();

    mAudioTrack = track;

    mCblkMemory.clear();

    mCblkMemory = cblk;

    mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());

    mCblk->out = 1;

    // Update buffer size in case it has been limited by AudioFlinger during track creation

    mFrameCount = mCblk->frameCount;

    if (sharedBuffer == 0) {

        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);

    } else {

        mCblk->buffers = sharedBuffer->pointer();

         // Force buffer full condition as data is already present in shared memory

        mCblk->stepUser(mFrameCount);

    }



    mCblk->volumeLR = (int32_t(int16_t(mVolume[LEFT] * 0x1000)) << 16) | int16_t(mVolume[RIGHT] * 0x1000);



    return NO_ERROR;

}



status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)

{

    int active;

    int timeout = 0;

    status_t result;

    audio_track_cblk_t* cblk = mCblk;

    uint32_t framesReq = audioBuffer->frameCount;
@@ -632,17 +667,20 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 
    uint32_t framesAvail = cblk->framesAvailable();



    if (framesAvail == 0) {

        Mutex::Autolock _l(cblk->lock);

        cblk->lock.lock();

        goto start_loop_here;

        while (framesAvail == 0) {

            active = mActive;

            if (UNLIKELY(!active)) {

                LOGV("Not active and NO_MORE_BUFFERS");

                cblk->lock.unlock();

                return NO_MORE_BUFFERS;

            }

            if (UNLIKELY(!waitCount))

            if (UNLIKELY(!waitCount)) {

                cblk->lock.unlock();

                return WOULD_BLOCK;

            timeout = 0;

            }



            result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));

            if (__builtin_expect(result!=NO_ERROR, false)) {

                cblk->waitTimeMs += waitTimeMs;
@@ -654,14 +692,23 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 
                                "user=%08x, server=%08x", this, cblk->user, cblk->server);

                        //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140)

                        cblk->lock.unlock();

                        mAudioTrack->start();

                        result = mAudioTrack->start();

                        if (result == DEAD_OBJECT) {

                            LOGW("obtainBuffer() dead IAudioTrack: creating a new one");

                            result = createTrack(mStreamType, cblk->sampleRate, mFormat, mChannelCount,

                                                 mFrameCount, mFlags, mSharedBuffer, getOutput());

                            if (result == NO_ERROR) {

                                cblk = mCblk;

                                cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;

                            }

                        }

                        cblk->lock.lock();

                        timeout = 1;

                    }

                    cblk->waitTimeMs = 0;

                }



                if (--waitCount == 0) {

                    cblk->lock.unlock();

                    return TIMED_OUT;

                }

            }
@@ -669,6 +716,7 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 
        start_loop_here:

            framesAvail = cblk->framesAvailable_l();

        }

        cblk->lock.unlock();

    }



    cblk->waitTimeMs = 0;
@@ -684,11 +732,6 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 
        framesReq = bufferEnd - u;

    }



    LOGW_IF(timeout,

        "*** SERIOUS WARNING *** obtainBuffer() timed out "

        "but didn't need to be locked. We recovered, but "

        "this shouldn't happen (user=%08x, server=%08x)", cblk->user, cblk->server);



    audioBuffer->flags = mMuted ? Buffer::MUTE : 0;

    audioBuffer->channelCount = mChannelCount;

    audioBuffer->frameCount = framesReq;
@@ -991,7 +1034,7 @@ bool audio_track_cblk_t::stepServer(uint32_t frameCount) 
        // Mark that we have read the first buffer so that next time stepUser() is called

        // we switch to normal obtainBuffer() timeout period

        if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) {

            bufferTimeoutMs = MAX_RUN_TIMEOUT_MS - 1;

            bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1;

        }

        // It is possible that we receive a flush()

        // while the mixer is processing a block: in this case,