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Commit b826fd09 authored by Mikhail Naganov's avatar Mikhail Naganov Committed by Android (Google) Code Review
Browse files

Merge "audioflinger: Extend debug logs in RecordThread"

parents c4e4425c b3cf7e65

services/audioflinger/Threads.cpp

+23 −11
Original line number Diff line number Diff line
@@ -5970,12 +5970,17 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 
    switch (kUseFastCapture) {

    case FastCapture_Never:

        initFastCapture = false;

        ALOGV("%p kUseFastCapture = Never, initFastCapture = false", this);

        break;

    case FastCapture_Always:

        initFastCapture = true;

        ALOGV("%p kUseFastCapture = Always, initFastCapture = true", this);

        break;

    case FastCapture_Static:

        initFastCapture = (mFrameCount * 1000) / mSampleRate < kMinNormalCaptureBufferSizeMs;

        ALOGV("%p kUseFastCapture = Static, (%lld * 1000) / %u vs %u, initFastCapture = %d",

                this, (long long)mFrameCount, mSampleRate, kMinNormalCaptureBufferSizeMs,

                initFastCapture);

        break;

    // case FastCapture_Dynamic:

    }
@@ -5986,13 +5991,16 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 
        // quadruple-buffering of 20 ms each; this ensures we can sleep for 20ms in RecordThread

        size_t pipeFramesP2 = roundup(4 * FMS_20 * mSampleRate / 1000);

        size_t pipeSize = pipeFramesP2 * Format_frameSize(format);

        void *pipeBuffer;

        void *pipeBuffer = nullptr;

        const sp<MemoryDealer> roHeap(readOnlyHeap());

        sp<IMemory> pipeMemory;

        if ((roHeap == 0) ||

                (pipeMemory = roHeap->allocate(pipeSize)) == 0 ||

                (pipeBuffer = pipeMemory->pointer()) == NULL) {

            ALOGE("not enough memory for pipe buffer size=%zu", pipeSize);

                (pipeBuffer = pipeMemory->pointer()) == nullptr) {

            ALOGE("not enough memory for pipe buffer size=%zu; "

                    "roHeap=%p, pipeMemory=%p, pipeBuffer=%p; roHeapSize: %lld",

                    pipeSize, roHeap.get(), pipeMemory.get(), pipeBuffer,

                    (long long)kRecordThreadReadOnlyHeapSize);

            goto failed;

        }

        // pipe will be shared directly with fast clients, so clear to avoid leaking old information
@@ -6634,19 +6642,19 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRe 
              audio_input_flags_t old = *flags;

              chain->checkInputFlagCompatibility(flags);

              if (old != *flags) {

                  ALOGV("AUDIO_INPUT_FLAGS denied by effect old=%#x new=%#x",

                          (int)old, (int)*flags);

                  ALOGV("%p AUDIO_INPUT_FLAGS denied by effect old=%#x new=%#x",

                          this, (int)old, (int)*flags);

              }

          }

          ALOGV_IF((*flags & AUDIO_INPUT_FLAG_FAST) != 0,

                   "AUDIO_INPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu",

                   frameCount, mFrameCount);

                   "%p AUDIO_INPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu",

                   this, frameCount, mFrameCount);

      } else {

        ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%zu mFrameCount=%zu mPipeFramesP2=%zu "

                "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u "

        ALOGV("%p AUDIO_INPUT_FLAG_FAST denied: frameCount=%zu mFrameCount=%zu mPipeFramesP2=%zu "

                "format=%#x isLinear=%d mFormat=%#x channelMask=%#x sampleRate=%u mSampleRate=%u "

                "hasFastCapture=%d tid=%d mFastTrackAvail=%d",

                frameCount, mFrameCount, mPipeFramesP2,

                format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate,

                this, frameCount, mFrameCount, mPipeFramesP2,

                format, audio_is_linear_pcm(format), mFormat, channelMask, sampleRate, mSampleRate,

                hasFastCapture(), tid, mFastTrackAvail);

        *flags = (audio_input_flags_t)(*flags & ~AUDIO_INPUT_FLAG_FAST);

      }
@@ -7243,6 +7251,10 @@ void AudioFlinger::RecordThread::readInputParameters_l() 
    result = mInput->stream->getBufferSize(&mBufferSize);

    LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving buffer size from HAL: %d", result);

    mFrameCount = mBufferSize / mFrameSize;

    ALOGV("%p RecordThread params: mChannelCount=%u, mFormat=%#x, mFrameSize=%lld, "

            "mBufferSize=%lld, mFrameCount=%lld",

            this, mChannelCount, mFormat, (long long)mFrameSize, (long long)mBufferSize,

            (long long)mFrameCount);

    // This is the formula for calculating the temporary buffer size.

    // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to

    // 1 full output buffer, regardless of the alignment of the available input.