diff options
Diffstat (limited to 'services/audioflinger')
| -rw-r--r-- | services/audioflinger/Threads.cpp | 45 |
1 files changed, 30 insertions, 15 deletions
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp index 30cebf4..193f8e4 100644 --- a/services/audioflinger/Threads.cpp +++ b/services/audioflinger/Threads.cpp @@ -84,6 +84,8 @@ #define ALOGVV(a...) do { } while(0) #endif +#define max(a, b) ((a) > (b) ? (a) : (b)) + namespace android { // retry counts for buffer fill timeout @@ -4868,7 +4870,7 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, if (initFastCapture) { // create a Pipe for FastMixer to write to, and for us and fast tracks to read from NBAIO_Format format = mInputSource->format(); - size_t pipeFramesP2 = roundup(mFrameCount * 8); + size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each size_t pipeSize = pipeFramesP2 * Format_frameSize(format); void *pipeBuffer; const sp<MemoryDealer> roHeap(readOnlyHeap()); @@ -5537,20 +5539,25 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRe // ignore requested notificationFrames, and always notify exactly once every HAL buffer *notificationFrames = mFrameCount; } else { - // not fast track: frame count is at least 2 HAL buffers and at least 20 ms - size_t minFrameCount = ((int64_t) mFrameCount * 2 * sampleRate + mSampleRate - 1) / - mSampleRate; - if (frameCount < minFrameCount) { - frameCount = minFrameCount; - } - minFrameCount = (sampleRate * 20 / 1000 + 1) & ~1; - if (frameCount < minFrameCount) { - frameCount = minFrameCount; - } - // notification is forced to be at least double-buffering - size_t maxNotification = frameCount / 2; - if (*notificationFrames == 0 || *notificationFrames > maxNotification) { - *notificationFrames = maxNotification; + // not fast track: max notification period is resampled equivalent of one HAL buffer time + // or 20 ms if there is a fast capture + // TODO This could be a roundupRatio inline, and const + size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) + * sampleRate + mSampleRate - 1) / mSampleRate; + // minimum number of notification periods is at least kMinNotifications, + // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) + static const size_t kMinNotifications = 3; + static const uint32_t kMinMs = 30; + // TODO This could be a roundupRatio inline + const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; + // TODO This could be a roundupRatio inline + const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / + maxNotificationFrames; + const size_t minFrameCount = maxNotificationFrames * + max(kMinNotifications, minNotificationsByMs); + frameCount = max(frameCount, minFrameCount); + if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { + *notificationFrames = maxNotificationFrames; } } *pFrameCount = frameCount; @@ -6073,6 +6080,14 @@ void AudioFlinger::RecordThread::readInputParameters_l() mRsmpInFrames = mFrameCount * 7; mRsmpInFramesP2 = roundup(mRsmpInFrames); delete[] mRsmpInBuffer; + + // TODO optimize audio capture buffer sizes ... + // Here we calculate the size of the sliding buffer used as a source + // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). + // For current HAL frame counts, this is usually 2048 = 40 ms. It would + // be better to have it derived from the pipe depth in the long term. + // The current value is higher than necessary. However it should not add to latency. + // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; |