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Commit bbda5ce4 authored by Eric Laurent's avatar Eric Laurent Committed by Android (Google) Code Review
Browse files

Merge "AudioSystem: Add mutex for output cache" into lmp-mr1-dev

parents 408dba5f f6778fd0

include/media/AudioSystem.h

+5 −4
Original line number Diff line number Diff line
@@ -373,8 +373,9 @@ private: 
    friend class AudioFlingerClient;

    friend class AudioPolicyServiceClient;



    static Mutex gLock;     // protects all members except gAudioPolicyService,

                            // gAudioPolicyServiceClient, and gAudioPortCallback

    static Mutex gLock;      // protects gAudioFlinger and gAudioErrorCallback,

    static Mutex gLockCache; // protects gOutputs, gPrevInSamplingRate, gPrevInFormat,

                             // gPrevInChannelMask and gInBuffSize

    static Mutex gLockAPS;   // protects gAudioPolicyService and gAudioPolicyServiceClient

    static Mutex gLockAPC;   // protects gAudioPortCallback

    static sp<IAudioFlinger> gAudioFlinger;

media/libmedia/AudioSystem.cpp

+47 −37
Original line number Diff line number Diff line
@@ -32,6 +32,7 @@ namespace android { 


// client singleton for AudioFlinger binder interface

Mutex AudioSystem::gLock;

Mutex AudioSystem::gLockCache;

Mutex AudioSystem::gLockAPS;

Mutex AudioSystem::gLockAPC;

sp<IAudioFlinger> AudioSystem::gAudioFlinger;
@@ -250,20 +251,20 @@ status_t AudioSystem::getOutputSamplingRate(uint32_t* samplingRate, audio_stream 
status_t AudioSystem::getSamplingRate(audio_io_handle_t output,

                                      uint32_t* samplingRate)

{

    OutputDescriptor *outputDesc;

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

    if (af == 0) return PERMISSION_DENIED;



    Mutex::Autolock _l(gLockCache);



    gLock.lock();

    outputDesc = AudioSystem::gOutputs.valueFor(output);

    OutputDescriptor *outputDesc = AudioSystem::gOutputs.valueFor(output);

    if (outputDesc == NULL) {

        ALOGV("getOutputSamplingRate() no output descriptor for output %d in gOutputs", output);

        gLock.unlock();

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

        if (af == 0) return PERMISSION_DENIED;

        gLockCache.unlock();

        *samplingRate = af->sampleRate(output);

        gLockCache.lock();

    } else {

        ALOGV("getOutputSamplingRate() reading from output desc");

        *samplingRate = outputDesc->samplingRate;

        gLock.unlock();

    }

    if (*samplingRate == 0) {

        ALOGE("AudioSystem::getSamplingRate failed for output %d", output);
@@ -294,18 +295,18 @@ status_t AudioSystem::getOutputFrameCount(size_t* frameCount, audio_stream_type_ 
status_t AudioSystem::getFrameCount(audio_io_handle_t output,

                                    size_t* frameCount)

{

    OutputDescriptor *outputDesc;



    gLock.lock();

    outputDesc = AudioSystem::gOutputs.valueFor(output);

    if (outputDesc == NULL) {

        gLock.unlock();

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

    if (af == 0) return PERMISSION_DENIED;



    Mutex::Autolock _l(gLockCache);



    OutputDescriptor *outputDesc = AudioSystem::gOutputs.valueFor(output);

    if (outputDesc == NULL) {

        gLockCache.unlock();

        *frameCount = af->frameCount(output);

        gLockCache.lock();

    } else {

        *frameCount = outputDesc->frameCount;

        gLock.unlock();

    }

    if (*frameCount == 0) {

        ALOGE("AudioSystem::getFrameCount failed for output %d", output);
@@ -336,18 +337,18 @@ status_t AudioSystem::getOutputLatency(uint32_t* latency, audio_stream_type_t st 
status_t AudioSystem::getLatency(audio_io_handle_t output,

                                 uint32_t* latency)

{

    OutputDescriptor *outputDesc;



    gLock.lock();

    outputDesc = AudioSystem::gOutputs.valueFor(output);

    if (outputDesc == NULL) {

        gLock.unlock();

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

    if (af == 0) return PERMISSION_DENIED;



    Mutex::Autolock _l(gLockCache);



    OutputDescriptor *outputDesc = AudioSystem::gOutputs.valueFor(output);

    if (outputDesc == NULL) {

        gLockCache.unlock();

        *latency = af->latency(output);

        gLockCache.lock();

    } else {

        *latency = outputDesc->latency;

        gLock.unlock();

    }



    ALOGV("getLatency() output %d, latency %d", output, *latency);
@@ -358,24 +359,24 @@ status_t AudioSystem::getLatency(audio_io_handle_t output, 
status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, audio_format_t format,

        audio_channel_mask_t channelMask, size_t* buffSize)

{

    gLock.lock();

    // Do we have a stale gInBufferSize or are we requesting the input buffer size for new values

    size_t inBuffSize = gInBuffSize;

    if ((inBuffSize == 0) || (sampleRate != gPrevInSamplingRate) || (format != gPrevInFormat)

        || (channelMask != gPrevInChannelMask)) {

        gLock.unlock();

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

    if (af == 0) {

        return PERMISSION_DENIED;

    }

    Mutex::Autolock _l(gLockCache);

    // Do we have a stale gInBufferSize or are we requesting the input buffer size for new values

    size_t inBuffSize = gInBuffSize;

    if ((inBuffSize == 0) || (sampleRate != gPrevInSamplingRate) || (format != gPrevInFormat)

        || (channelMask != gPrevInChannelMask)) {

        gLockCache.unlock();

        inBuffSize = af->getInputBufferSize(sampleRate, format, channelMask);

        gLockCache.lock();

        if (inBuffSize == 0) {

            ALOGE("AudioSystem::getInputBufferSize failed sampleRate %d format %#x channelMask %x",

                    sampleRate, format, channelMask);

            return BAD_VALUE;

        }

        // A benign race is possible here: we could overwrite a fresher cache entry

        gLock.lock();

        // save the request params

        gPrevInSamplingRate = sampleRate;

        gPrevInFormat = format;
@@ -383,7 +384,6 @@ status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, audio_format_t for 


        gInBuffSize = inBuffSize;

    }

    gLock.unlock();

    *buffSize = inBuffSize;



    return NO_ERROR;
@@ -449,15 +449,22 @@ audio_hw_sync_t AudioSystem::getAudioHwSyncForSession(audio_session_t sessionId) 
// ---------------------------------------------------------------------------



void AudioSystem::AudioFlingerClient::binderDied(const wp<IBinder>& who __unused)

{

    audio_error_callback cb = NULL;

    {

        Mutex::Autolock _l(AudioSystem::gLock);



        AudioSystem::gAudioFlinger.clear();

        cb = gAudioErrorCallback;

    }



    {

        // clear output handles and stream to output map caches

        Mutex::Autolock _l(gLockCache);

        AudioSystem::gOutputs.clear();

    }



    if (gAudioErrorCallback) {

        gAudioErrorCallback(DEAD_OBJECT);

    if (cb) {

        cb(DEAD_OBJECT);

    }

    ALOGW("AudioFlinger server died!");

}
@@ -470,7 +477,7 @@ void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, audio_io_handle 


    if (ioHandle == AUDIO_IO_HANDLE_NONE) return;



    Mutex::Autolock _l(AudioSystem::gLock);

    Mutex::Autolock _l(AudioSystem::gLockCache);



    switch (event) {

    case STREAM_CONFIG_CHANGED:
@@ -829,8 +836,11 @@ void AudioSystem::clearAudioConfigCache() 
    // called by restoreTrack_l(), which needs new IAudioFlinger and IAudioPolicyService instances

    ALOGV("clearAudioConfigCache()");

    {

        Mutex::Autolock _l(gLock);

        Mutex::Autolock _l(gLockCache);

        gOutputs.clear();

    }

    {

        Mutex::Autolock _l(gLock);

        gAudioFlinger.clear();

    }

    {