audioflinger: Extend debug logs in RecordThread
This seems to be helpful to figure out why requesting
low latency recording fails. E.g. it can fail because
RecordThread just doesn't enable fast capture, or because
pipe memory allocation has failed, or due to parameters
incompatibility.
Change-Id: Ia5122b0591836c4999e73f3460c761484f5c2859
Test: manual
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 8596817..df10d23 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -5970,12 +5970,17 @@
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 @@
// 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 @@
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 @@
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.