Donate to e Foundation | Murena handsets with /e/OS | Own a part of Murena! Learn more

Commit a3db30f7 authored by Jean-Baptiste Queru's avatar Jean-Baptiste Queru Committed by Android Git Automerger
Browse files

am 9d25b82d: Merge "Rename LOG_ASSERT to ALOG_ASSERT"

* commit '9d25b82d280c2d979d500e7da4447148f32f820b':
  Rename LOG_ASSERT to ALOG_ASSERT
parents d7467379 39fd8eb9
Loading
Loading
Loading
Loading
+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;
    }
+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);
        }
    }

+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) {