Merge "Fix capture overruns at non-primary sample rate" into mnc-dev (5ce05966) · Commits · e / os / android_frameworks_av

services/audioflinger/AudioFlinger.cpp

+0 −3

Original line number	Diff line number	Diff line
		@@ -181,7 +181,6 @@ AudioFlinger::AudioFlinger()
		mIsLowRamDevice(true),
		mIsDeviceTypeKnown(false),
		mGlobalEffectEnableTime(0),
		mPrimaryOutputSampleRate(0),
		mSystemReady(false)
		{
		getpid_cached = getpid();
		@@ -1869,8 +1868,6 @@ status_t AudioFlinger::openOutput(audio_module_handle_t module,
		mHardwareStatus = AUDIO_HW_SET_MODE;
		mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode);
		mHardwareStatus = AUDIO_HW_IDLE;

		mPrimaryOutputSampleRate = config->sample_rate;
		}
		return NO_ERROR;
		}

services/audioflinger/AudioFlinger.h

+1 −3

Original line number	Diff line number	Diff line
		@@ -763,8 +763,6 @@ private:

		sp<PatchPanel> mPatchPanel;

		uint32_t mPrimaryOutputSampleRate; // sample rate of the primary output, or zero if none
		// protected by mHardwareLock
		bool mSystemReady;
		};

services/audioflinger/Threads.cpp

+6 −14

Original line number	Diff line number	Diff line
		@@ -125,10 +125,15 @@ static const uint32_t kMinThreadSleepTimeUs = 5000;
		static const uint32_t kMaxThreadSleepTimeShift = 2;

		// minimum normal sink buffer size, expressed in milliseconds rather than frames
		// FIXME This should be based on experimentally observed scheduling jitter
		static const uint32_t kMinNormalSinkBufferSizeMs = 20;
		// maximum normal sink buffer size
		static const uint32_t kMaxNormalSinkBufferSizeMs = 24;

		// minimum capture buffer size in milliseconds to _not_ need a fast capture thread
		// FIXME This should be based on experimentally observed scheduling jitter
		static const uint32_t kMinNormalCaptureBufferSizeMs = 12;

		// Offloaded output thread standby delay: allows track transition without going to standby
		static const nsecs_t kOffloadStandbyDelayNs = seconds(1);

		@@ -5490,20 +5495,7 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
		initFastCapture = true;
		break;
		case FastCapture_Static:
		uint32_t primaryOutputSampleRate;
		{
		AutoMutex _l(audioFlinger->mHardwareLock);
		primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate;
		}
		initFastCapture =
		// either capture sample rate is same as (a reasonable) primary output sample rate
		((isMusicRate(primaryOutputSampleRate) &&
		(mSampleRate == primaryOutputSampleRate)) \|\|
		// or primary output sample rate is unknown, and capture sample rate is reasonable
		((primaryOutputSampleRate == 0) &&
		isMusicRate(mSampleRate))) &&
		// and the buffer size is < 12 ms
		(mFrameCount * 1000) / mSampleRate < 12;
		initFastCapture = (mFrameCount * 1000) / mSampleRate < kMinNormalCaptureBufferSizeMs;
		break;
		// case FastCapture_Dynamic:
		}