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Commit c1dc1cb1 authored by Steve Block's avatar Steve Block
Browse files

Rename LOG_ASSERT to ALOG_ASSERT DO NOT MERGE 

See https://android-git.corp.google.com/g/157519

Bug: 5449033
Change-Id: I8ceb2dba1b031a0fd68d15d146960d9ced62bbf3
parent 5f29ca38

media/libmediaplayerservice/TestPlayerStub.cpp

+1 −1
Original line number Diff line number Diff line
@@ -176,7 +176,7 @@ status_t TestPlayerStub::resetInternal() 
    mContentUrl = NULL;



    if (mPlayer) {

        LOG_ASSERT(mDeletePlayer != NULL, "mDeletePlayer is null");

        ALOG_ASSERT(mDeletePlayer != NULL, "mDeletePlayer is null");

        (*mDeletePlayer)(mPlayer);

        mPlayer = NULL;

    }

services/audioflinger/AudioFlinger.cpp

+1 −1
Original line number Diff line number Diff line
@@ -2127,7 +2127,7 @@ uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track 
                // the minimum track buffer size is normally twice the number of frames necessary

                // to fill one buffer and the resampler should not leave more than one buffer worth

                // of unreleased frames after each pass, but just in case...

                LOG_ASSERT(minFrames <= cblk->frameCount);

                ALOG_ASSERT(minFrames <= cblk->frameCount);

            }

        }

        if ((cblk->framesReady() >= minFrames) && track->isReady() &&

services/audioflinger/AudioResampler.cpp

+4 −4
Original line number Diff line number Diff line
@@ -123,7 +123,7 @@ AudioResampler::AudioResampler(int bitDepth, int inChannelCount, 
    if ((bitDepth != 16) ||(inChannelCount < 1) || (inChannelCount > 2)) {

        ALOGE("Unsupported sample format, %d bits, %d channels", bitDepth,

                inChannelCount);

        // LOG_ASSERT(0);

        // ALOG_ASSERT(0);

    }



    // initialize common members
@@ -164,7 +164,7 @@ void AudioResamplerOrder1::resample(int32_t* out, size_t outFrameCount, 
        AudioBufferProvider* provider) {



    // should never happen, but we overflow if it does

    // LOG_ASSERT(outFrameCount < 32767);

    // ALOG_ASSERT(outFrameCount < 32767);



    // select the appropriate resampler

    switch (mChannelCount) {
@@ -261,7 +261,7 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount, 
            provider->releaseBuffer(&mBuffer);



            // verify that the releaseBuffer resets the buffer frameCount

            // LOG_ASSERT(mBuffer.frameCount == 0);

            // ALOG_ASSERT(mBuffer.frameCount == 0);

        }

    }
@@ -355,7 +355,7 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount, 
            provider->releaseBuffer(&mBuffer);



            // verify that the releaseBuffer resets the buffer frameCount

            // LOG_ASSERT(mBuffer.frameCount == 0);

            // ALOG_ASSERT(mBuffer.frameCount == 0);

        }

    }

services/audioflinger/AudioResamplerCubic.cpp

+1 −1
Original line number Diff line number Diff line
@@ -36,7 +36,7 @@ void AudioResamplerCubic::resample(int32_t* out, size_t outFrameCount, 
        AudioBufferProvider* provider) {



    // should never happen, but we overflow if it does

    // LOG_ASSERT(outFrameCount < 32767);

    // ALOG_ASSERT(outFrameCount < 32767);



    // select the appropriate resampler

    switch (mChannelCount) {