Merge "Squashed commit of the following:" into lmp-preview-dev (1a392aa1) · Commits · e / os / android_frameworks_av

media/libmedia/AudioRecord.cpp

+24 −18

Original line number	Diff line number	Diff line
		@@ -203,23 +203,6 @@ status_t AudioRecord::set(
		mFrameSize = sizeof(uint8_t);
		}

		// validate framecount
		size_t minFrameCount;
		status_t status = AudioRecord::getMinFrameCount(&minFrameCount,
		sampleRate, format, channelMask);
		if (status != NO_ERROR) {
		ALOGE("getMinFrameCount() failed for sampleRate %u, format %#x, channelMask %#x; status %d",
		sampleRate, format, channelMask, status);
		return status;
		}
		ALOGV("AudioRecord::set() minFrameCount = %d", minFrameCount);

		if (frameCount == 0) {
		frameCount = minFrameCount;
		} else if (frameCount < minFrameCount) {
		ALOGE("frameCount %u < minFrameCount %u", frameCount, minFrameCount);
		return BAD_VALUE;
		}
		// mFrameCount is initialized in openRecord_l
		mReqFrameCount = frameCount;

		@@ -242,7 +225,7 @@ status_t AudioRecord::set(
		}

		// create the IAudioRecord
		status = openRecord_l(0 /epoch/);
		status_t status = openRecord_l(0 /epoch/);

		if (status != NO_ERROR) {
		if (mAudioRecordThread != 0) {
		@@ -464,6 +447,29 @@ status_t AudioRecord::openRecord_l(size_t epoch)
		size_t frameCount = mReqFrameCount;

		if (!(mFlags & AUDIO_INPUT_FLAG_FAST)) {
		// validate framecount
		// If fast track was not requested, this preserves
		// the old behavior of validating on client side.
		// FIXME Eventually the validation should be done on server side
		// regardless of whether it's a fast or normal track. It's debatable
		// whether to account for the input latency to provision buffers appropriately.
		size_t minFrameCount;
		status = AudioRecord::getMinFrameCount(&minFrameCount,
		mSampleRate, mFormat, mChannelMask);
		if (status != NO_ERROR) {
		ALOGE("getMinFrameCount() failed for sampleRate %u, format %#x, channelMask %#x; "
		"status %d",
		mSampleRate, mFormat, mChannelMask, status);
		return status;
		}

		if (frameCount == 0) {
		frameCount = minFrameCount;
		} else if (frameCount < minFrameCount) {
		ALOGE("frameCount %u < minFrameCount %u", frameCount, minFrameCount);
		return BAD_VALUE;
		}

		// Make sure that application is notified with sufficient margin before overrun
		if (mNotificationFramesAct == 0 \|\| mNotificationFramesAct > frameCount/2) {
		mNotificationFramesAct = frameCount/2;

media/libmedia/AudioTrackShared.cpp

+11 −4

Original line number	Diff line number	Diff line
		@@ -134,11 +134,18 @@ status_t ClientProxy::obtainBuffer(Buffer* buffer, const struct timespec *reques
		ssize_t filled = rear - front;
		// pipe should not be overfull
		if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
		ALOGE("Shared memory control block is corrupt (filled=%d); shutting down", filled);
		if (mIsOut) {
		ALOGE("Shared memory control block is corrupt (filled=%d, mFrameCount=%u); "
		"shutting down", filled, mFrameCount);
		mIsShutdown = true;
		status = NO_INIT;
		goto end;
		}
		// for input, sync up on overrun
		filled = 0;
		cblk->u.mStreaming.mFront = rear;
		(void) android_atomic_or(CBLK_OVERRUN, &cblk->mFlags);
		}
		// don't allow filling pipe beyond the nominal size
		size_t avail = mIsOut ? mFrameCount - filled : filled;
		if (avail > 0) {

services/audioflinger/Android.mk

+1 −0

Original line number	Diff line number	Diff line
		@@ -64,6 +64,7 @@ LOCAL_32_BIT_ONLY := true

		LOCAL_SRC_FILES += FastMixer.cpp FastMixerState.cpp AudioWatchdog.cpp
		LOCAL_SRC_FILES += FastThread.cpp FastThreadState.cpp
		LOCAL_SRC_FILES += FastCapture.cpp FastCaptureState.cpp

		LOCAL_CFLAGS += -DSTATE_QUEUE_INSTANTIATIONS='"StateQueueInstantiations.cpp"'

services/audioflinger/AudioFlinger.cpp

+4 −1

Original line number	Diff line number	Diff line
		@@ -169,7 +169,8 @@ AudioFlinger::AudioFlinger()
		mBtNrecIsOff(false),
		mIsLowRamDevice(true),
		mIsDeviceTypeKnown(false),
		mGlobalEffectEnableTime(0)
		mGlobalEffectEnableTime(0),
		mPrimaryOutputSampleRate(0)
		{
		getpid_cached = getpid();
		char value[PROPERTY_VALUE_MAX];
		@@ -1679,6 +1680,8 @@ audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module,
		mHardwareStatus = AUDIO_HW_SET_MODE;
		hwDevHal->set_mode(hwDevHal, mMode);
		mHardwareStatus = AUDIO_HW_IDLE;

		mPrimaryOutputSampleRate = config.sample_rate;
		}
		return id;
		}

services/audioflinger/AudioFlinger.h

+5 −0

Original line number	Diff line number	Diff line
		@@ -50,6 +50,8 @@

		#include <media/AudioBufferProvider.h>
		#include <media/ExtendedAudioBufferProvider.h>

		#include "FastCapture.h"
		#include "FastMixer.h"
		#include <media/nbaio/NBAIO.h>
		#include "AudioWatchdog.h"
		@@ -691,6 +693,9 @@ private:
		nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled

		sp<PatchPanel> mPatchPanel;

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

		#undef INCLUDING_FROM_AUDIOFLINGER_H