refactor AudioTrack and AudioFlinger createTrack()
Refactor the mechanism used by audio tracks to query and attach
to an output mixer/stream in audio flinger. This will:
- reduce the number of binder transactions needed to create a track
- move sample rate, framecount and flags validations to audio server
side
- move audio session allocation to audio server side
- prepare restriction of certain binder transactions to audioserver only
Test: CTS tests for AudioTrack
Change-Id: If4369aad6c080a56c0b42fbfcc97c8ade17a7439
diff --git a/media/libaudioclient/AudioSystem.cpp b/media/libaudioclient/AudioSystem.cpp
index 58330ae..c284f73 100644
--- a/media/libaudioclient/AudioSystem.cpp
+++ b/media/libaudioclient/AudioSystem.cpp
@@ -20,6 +20,7 @@
#include <utils/Log.h>
#include <binder/IServiceManager.h>
#include <binder/ProcessState.h>
+#include <media/AudioResamplerPublic.h>
#include <media/AudioSystem.h>
#include <media/IAudioFlinger.h>
#include <media/IAudioPolicyService.h>
@@ -253,6 +254,31 @@
return volume ? 100 - int(dBConvertInverse * log(volume) + 0.5) : 0;
}
+/* static */ size_t AudioSystem::calculateMinFrameCount(
+ uint32_t afLatencyMs, uint32_t afFrameCount, uint32_t afSampleRate,
+ uint32_t sampleRate, float speed /*, uint32_t notificationsPerBufferReq*/)
+{
+ // Ensure that buffer depth covers at least audio hardware latency
+ uint32_t minBufCount = afLatencyMs / ((1000 * afFrameCount) / afSampleRate);
+ if (minBufCount < 2) {
+ minBufCount = 2;
+ }
+#if 0
+ // The notificationsPerBufferReq parameter is not yet used for non-fast tracks,
+ // but keeping the code here to make it easier to add later.
+ if (minBufCount < notificationsPerBufferReq) {
+ minBufCount = notificationsPerBufferReq;
+ }
+#endif
+ ALOGV("calculateMinFrameCount afLatency %u afFrameCount %u afSampleRate %u "
+ "sampleRate %u speed %f minBufCount: %u" /*" notificationsPerBufferReq %u"*/,
+ afLatencyMs, afFrameCount, afSampleRate, sampleRate, speed, minBufCount
+ /*, notificationsPerBufferReq*/);
+ return minBufCount * sourceFramesNeededWithTimestretch(
+ sampleRate, afFrameCount, afSampleRate, speed);
+}
+
+
status_t AudioSystem::getOutputSamplingRate(uint32_t* samplingRate, audio_stream_type_t streamType)
{
audio_io_handle_t output;
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index 356b321..36961d6 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -99,32 +99,6 @@
return kFixPitch ? AUDIO_TIMESTRETCH_PITCH_NORMAL : pitch;
}
-// Must match similar computation in createTrack_l in Threads.cpp.
-// TODO: Move to a common library
-static size_t calculateMinFrameCount(
- uint32_t afLatencyMs, uint32_t afFrameCount, uint32_t afSampleRate,
- uint32_t sampleRate, float speed /*, uint32_t notificationsPerBufferReq*/)
-{
- // Ensure that buffer depth covers at least audio hardware latency
- uint32_t minBufCount = afLatencyMs / ((1000 * afFrameCount) / afSampleRate);
- if (minBufCount < 2) {
- minBufCount = 2;
- }
-#if 0
- // The notificationsPerBufferReq parameter is not yet used for non-fast tracks,
- // but keeping the code here to make it easier to add later.
- if (minBufCount < notificationsPerBufferReq) {
- minBufCount = notificationsPerBufferReq;
- }
-#endif
- ALOGV("calculateMinFrameCount afLatency %u afFrameCount %u afSampleRate %u "
- "sampleRate %u speed %f minBufCount: %u" /*" notificationsPerBufferReq %u"*/,
- afLatencyMs, afFrameCount, afSampleRate, sampleRate, speed, minBufCount
- /*, notificationsPerBufferReq*/);
- return minBufCount * sourceFramesNeededWithTimestretch(
- sampleRate, afFrameCount, afSampleRate, speed);
-}
-
// static
status_t AudioTrack::getMinFrameCount(
size_t* frameCount,
@@ -165,8 +139,8 @@
// When called from createTrack, speed is 1.0f (normal speed).
// This is rechecked again on setting playback rate (TODO: on setting sample rate, too).
- *frameCount = calculateMinFrameCount(afLatency, afFrameCount, afSampleRate, sampleRate, 1.0f
- /*, 0 notificationsPerBufferReq*/);
+ *frameCount = AudioSystem::calculateMinFrameCount(afLatency, afFrameCount, afSampleRate,
+ sampleRate, 1.0f /*, 0 notificationsPerBufferReq*/);
// The formula above should always produce a non-zero value under normal circumstances:
// AudioTrack.SAMPLE_RATE_HZ_MIN <= sampleRate <= AudioTrack.SAMPLE_RATE_HZ_MAX.
@@ -190,8 +164,7 @@
mPreviousSchedulingGroup(SP_DEFAULT),
mPausedPosition(0),
mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
- mRoutedDeviceId(AUDIO_PORT_HANDLE_NONE),
- mPortId(AUDIO_PORT_HANDLE_NONE)
+ mRoutedDeviceId(AUDIO_PORT_HANDLE_NONE)
{
mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN;
mAttributes.usage = AUDIO_USAGE_UNKNOWN;
@@ -222,8 +195,7 @@
mState(STATE_STOPPED),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT),
- mPausedPosition(0),
- mPortId(AUDIO_PORT_HANDLE_NONE)
+ mPausedPosition(0)
{
mStatus = set(streamType, sampleRate, format, channelMask,
frameCount, flags, cbf, user, notificationFrames,
@@ -254,8 +226,7 @@
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT),
mPausedPosition(0),
- mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
- mPortId(AUDIO_PORT_HANDLE_NONE)
+ mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE)
{
mStatus = set(streamType, sampleRate, format, channelMask,
0 /*frameCount*/, flags, cbf, user, notificationFrames,
@@ -320,6 +291,7 @@
mThreadCanCallJava = threadCanCallJava;
mSelectedDeviceId = selectedDeviceId;
+ mSessionId = sessionId;
switch (transferType) {
case TRANSFER_DEFAULT:
@@ -500,11 +472,6 @@
notificationFrames, minNotificationsPerBuffer, maxNotificationsPerBuffer);
}
mNotificationFramesAct = 0;
- if (sessionId == AUDIO_SESSION_ALLOCATE) {
- mSessionId = (audio_session_t) AudioSystem::newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
- } else {
- mSessionId = sessionId;
- }
int callingpid = IPCThreadState::self()->getCallingPid();
int mypid = getpid();
if (uid == AUDIO_UID_INVALID || (callingpid != mypid)) {
@@ -1317,70 +1284,12 @@
return NO_INIT;
}
- audio_io_handle_t output;
- audio_stream_type_t streamType = mStreamType;
- audio_attributes_t *attr = (mStreamType == AUDIO_STREAM_DEFAULT) ? &mAttributes : NULL;
+ status_t status;
bool callbackAdded = false;
+ {
// mFlags (not mOrigFlags) is modified depending on whether fast request is accepted.
// After fast request is denied, we will request again if IAudioTrack is re-created.
-
- status_t status;
- audio_config_t config = AUDIO_CONFIG_INITIALIZER;
- config.sample_rate = mSampleRate;
- config.channel_mask = mChannelMask;
- config.format = mFormat;
- config.offload_info = mOffloadInfoCopy;
- mRoutedDeviceId = mSelectedDeviceId;
- status = AudioSystem::getOutputForAttr(attr, &output,
- mSessionId, &streamType, mClientUid,
- &config,
- mFlags, &mRoutedDeviceId, &mPortId);
-
- if (status != NO_ERROR || output == AUDIO_IO_HANDLE_NONE) {
- ALOGE("Could not get audio output for session %d, stream type %d, usage %d, sample rate %u,"
- " format %#x, channel mask %#x, flags %#x",
- mSessionId, streamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask,
- mFlags);
- return BAD_VALUE;
- }
- {
- // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger,
- // we must release it ourselves if anything goes wrong.
-
- // Not all of these values are needed under all conditions, but it is easier to get them all
- status = AudioSystem::getLatency(output, &mAfLatency);
- if (status != NO_ERROR) {
- ALOGE("getLatency(%d) failed status %d", output, status);
- goto release;
- }
- ALOGV("createTrack_l() output %d afLatency %u", output, mAfLatency);
-
- status = AudioSystem::getFrameCount(output, &mAfFrameCount);
- if (status != NO_ERROR) {
- ALOGE("getFrameCount(output=%d) status %d", output, status);
- goto release;
- }
-
- // TODO consider making this a member variable if there are other uses for it later
- size_t afFrameCountHAL;
- status = AudioSystem::getFrameCountHAL(output, &afFrameCountHAL);
- if (status != NO_ERROR) {
- ALOGE("getFrameCountHAL(output=%d) status %d", output, status);
- goto release;
- }
- ALOG_ASSERT(afFrameCountHAL > 0);
-
- status = AudioSystem::getSamplingRate(output, &mAfSampleRate);
- if (status != NO_ERROR) {
- ALOGE("getSamplingRate(output=%d) status %d", output, status);
- goto release;
- }
- if (mSampleRate == 0) {
- mSampleRate = mAfSampleRate;
- mOriginalSampleRate = mAfSampleRate;
- }
-
// Client can only express a preference for FAST. Server will perform additional tests.
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
// either of these use cases:
@@ -1394,130 +1303,78 @@
// use case 4: synchronous write
((mTransfer == TRANSFER_SYNC) && mThreadCanCallJava);
- bool useCaseAllowed = sharedBuffer || transferAllowed;
- if (!useCaseAllowed) {
+ bool fastAllowed = sharedBuffer || transferAllowed;
+ if (!fastAllowed) {
ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client, not shared buffer and transfer = %s",
convertTransferToText(mTransfer));
- }
-
- // sample rates must also match
- bool sampleRateAllowed = mSampleRate == mAfSampleRate;
- if (!sampleRateAllowed) {
- ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client, sample rate %u Hz but HAL needs %u Hz",
- mSampleRate, mAfSampleRate);
- }
-
- bool fastAllowed = useCaseAllowed && sampleRateAllowed;
- if (!fastAllowed) {
mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
}
}
- mNotificationFramesAct = mNotificationFramesReq;
-
- size_t frameCount = mReqFrameCount;
- if (!audio_has_proportional_frames(mFormat)) {
-
- if (mSharedBuffer != 0) {
- // Same comment as below about ignoring frameCount parameter for set()
- frameCount = mSharedBuffer->size();
- } else if (frameCount == 0) {
- frameCount = mAfFrameCount;
- }
- if (mNotificationFramesAct != frameCount) {
- mNotificationFramesAct = frameCount;
- }
- } else if (mSharedBuffer != 0) {
- // FIXME: Ensure client side memory buffers need
- // not have additional alignment beyond sample
- // (e.g. 16 bit stereo accessed as 32 bit frame).
- size_t alignment = audio_bytes_per_sample(mFormat);
- if (alignment & 1) {
- // for AUDIO_FORMAT_PCM_24_BIT_PACKED (not exposed through Java).
- alignment = 1;
- }
- if (mChannelCount > 1) {
- // More than 2 channels does not require stronger alignment than stereo
- alignment <<= 1;
- }
- if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) {
- ALOGE("Invalid buffer alignment: address %p, channel count %u",
- mSharedBuffer->pointer(), mChannelCount);
- status = BAD_VALUE;
- goto release;
- }
-
- // When initializing a shared buffer AudioTrack via constructors,
- // there's no frameCount parameter.
- // But when initializing a shared buffer AudioTrack via set(),
- // there _is_ a frameCount parameter. We silently ignore it.
- frameCount = mSharedBuffer->size() / mFrameSize;
+ IAudioFlinger::CreateTrackInput input;
+ if (mStreamType != AUDIO_STREAM_DEFAULT) {
+ stream_type_to_audio_attributes(mStreamType, &input.attr);
} else {
- size_t minFrameCount = 0;
- // For fast tracks the frame count calculations and checks are mostly done by server,
- // but we try to respect the application's request for notifications per buffer.
- if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
- if (mNotificationsPerBufferReq > 0) {
- // Avoid possible arithmetic overflow during multiplication.
- // mNotificationsPerBuffer is clamped to a small integer earlier, so it is unlikely.
- if (mNotificationsPerBufferReq > SIZE_MAX / afFrameCountHAL) {
- ALOGE("Requested notificationPerBuffer=%u ignored for HAL frameCount=%zu",
- mNotificationsPerBufferReq, afFrameCountHAL);
- } else {
- minFrameCount = afFrameCountHAL * mNotificationsPerBufferReq;
- }
- }
- } else {
- // for normal tracks precompute the frame count based on speed.
- const float speed = !isPurePcmData_l() || isOffloadedOrDirect_l() ? 1.0f :
- max(mMaxRequiredSpeed, mPlaybackRate.mSpeed);
- minFrameCount = calculateMinFrameCount(
- mAfLatency, mAfFrameCount, mAfSampleRate, mSampleRate,
- speed /*, 0 mNotificationsPerBufferReq*/);
- }
- if (frameCount < minFrameCount) {
- frameCount = minFrameCount;
- }
+ input.attr = mAttributes;
}
-
- audio_output_flags_t flags = mFlags;
-
- pid_t tid = -1;
+ input.config = AUDIO_CONFIG_INITIALIZER;
+ input.config.sample_rate = mSampleRate;
+ input.config.channel_mask = mChannelMask;
+ input.config.format = mFormat;
+ input.config.offload_info = mOffloadInfoCopy;
+ input.clientInfo.clientUid = mClientUid;
+ input.clientInfo.clientPid = mClientPid;
+ input.clientInfo.clientTid = -1;
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
// It is currently meaningless to request SCHED_FIFO for a Java thread. Even if the
// application-level code follows all non-blocking design rules, the language runtime
// doesn't also follow those rules, so the thread will not benefit overall.
if (mAudioTrackThread != 0 && !mThreadCanCallJava) {
- tid = mAudioTrackThread->getTid();
+ input.clientInfo.clientTid = mAudioTrackThread->getTid();
}
}
+ input.sharedBuffer = mSharedBuffer;
+ input.notificationsPerBuffer = mNotificationsPerBufferReq;
+ input.speed = 1.0;
+ if (audio_has_proportional_frames(mFormat) && mSharedBuffer == 0 &&
+ (mFlags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
+ input.speed = !isPurePcmData_l() || isOffloadedOrDirect_l() ? 1.0f :
+ max(mMaxRequiredSpeed, mPlaybackRate.mSpeed);
+ }
+ input.flags = mFlags;
+ input.frameCount = mReqFrameCount;
+ input.notificationFrameCount = mNotificationFramesReq;
+ input.selectedDeviceId = mSelectedDeviceId;
+ input.sessionId = mSessionId;
- size_t temp = frameCount; // temp may be replaced by a revised value of frameCount,
- // but we will still need the original value also
- audio_session_t originalSessionId = mSessionId;
- sp<IAudioTrack> track = audioFlinger->createTrack(streamType,
- mSampleRate,
- mFormat,
- mChannelMask,
- &temp,
- &flags,
- mSharedBuffer,
+ IAudioFlinger::CreateTrackOutput output;
+
+ sp<IAudioTrack> track = audioFlinger->createTrack(input,
output,
- mClientPid,
- tid,
- &mSessionId,
- mClientUid,
- &status,
- mPortId);
- ALOGE_IF(originalSessionId != AUDIO_SESSION_ALLOCATE && mSessionId != originalSessionId,
- "session ID changed from %d to %d", originalSessionId, mSessionId);
+ &status);
- if (status != NO_ERROR) {
- ALOGE("AudioFlinger could not create track, status: %d", status);
- goto release;
+ if (status != NO_ERROR || output.outputId == AUDIO_IO_HANDLE_NONE) {
+ ALOGE("AudioFlinger could not create track, status: %d output %d", status, output.outputId);
+ goto error;
}
ALOG_ASSERT(track != 0);
+ mFrameCount = output.frameCount;
+ mNotificationFramesAct = (uint32_t)output.notificationFrameCount;
+ mRoutedDeviceId = output.selectedDeviceId;
+ mSessionId = output.sessionId;
+
+ mSampleRate = output.sampleRate;
+ if (mOriginalSampleRate == 0) {
+ mOriginalSampleRate = mSampleRate;
+ }
+
+ mAfFrameCount = output.afFrameCount;
+ mAfSampleRate = output.afSampleRate;
+ mAfLatency = output.afLatencyMs;
+
+ mLatency = mAfLatency + (1000LL * mFrameCount) / mSampleRate;
+
// AudioFlinger now owns the reference to the I/O handle,
// so we are no longer responsible for releasing it.
@@ -1526,13 +1383,13 @@
if (iMem == 0) {
ALOGE("Could not get control block");
status = NO_INIT;
- goto release;
+ goto error;
}
void *iMemPointer = iMem->pointer();
if (iMemPointer == NULL) {
ALOGE("Could not get control block pointer");
status = NO_INIT;
- goto release;
+ goto error;
}
// invariant that mAudioTrack != 0 is true only after set() returns successfully
if (mAudioTrack != 0) {
@@ -1545,75 +1402,33 @@
audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
mCblk = cblk;
- // note that temp is the (possibly revised) value of frameCount
- if (temp < frameCount || (frameCount == 0 && temp == 0)) {
- // In current design, AudioTrack client checks and ensures frame count validity before
- // passing it to AudioFlinger so AudioFlinger should not return a different value except
- // for fast track as it uses a special method of assigning frame count.
- ALOGW("Requested frameCount %zu but received frameCount %zu", frameCount, temp);
- }
- frameCount = temp;
mAwaitBoost = false;
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
- if (flags & AUDIO_OUTPUT_FLAG_FAST) {
- ALOGI("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu -> %zu", frameCount, temp);
+ if (output.flags & AUDIO_OUTPUT_FLAG_FAST) {
+ ALOGI("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu -> %zu",
+ mReqFrameCount, mFrameCount);
if (!mThreadCanCallJava) {
mAwaitBoost = true;
}
} else {
- ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu -> %zu", frameCount,
- temp);
+ ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu -> %zu", mReqFrameCount,
+ mFrameCount);
}
}
- mFlags = flags;
-
- // Make sure that application is notified with sufficient margin before underrun.
- // The client can divide the AudioTrack buffer into sub-buffers,
- // and expresses its desire to server as the notification frame count.
- if (mSharedBuffer == 0 && audio_is_linear_pcm(mFormat)) {
- size_t maxNotificationFrames;
- if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
- // notify every HAL buffer, regardless of the size of the track buffer
- maxNotificationFrames = afFrameCountHAL;
- } else {
- // For normal tracks, use at least double-buffering if no sample rate conversion,
- // or at least triple-buffering if there is sample rate conversion
- const int nBuffering = mOriginalSampleRate == mAfSampleRate ? 2 : 3;
- maxNotificationFrames = frameCount / nBuffering;
- // If client requested a fast track but this was denied, then use the smaller maximum.
- // FMS_20 is the minimum task wakeup period in ms for which CFS operates reliably.
-#define FMS_20 20 // FIXME share a common declaration with the same symbol in Threads.cpp
- if (mOrigFlags & AUDIO_OUTPUT_FLAG_FAST) {
- size_t maxNotificationFramesFastDenied = FMS_20 * mSampleRate / 1000;
- if (maxNotificationFrames > maxNotificationFramesFastDenied) {
- maxNotificationFrames = maxNotificationFramesFastDenied;
- }
- }
- }
- if (mNotificationFramesAct == 0 || mNotificationFramesAct > maxNotificationFrames) {
- if (mNotificationFramesAct == 0) {
- ALOGD("Client defaulted notificationFrames to %zu for frameCount %zu",
- maxNotificationFrames, frameCount);
- } else {
- ALOGW("Client adjusted notificationFrames from %u to %zu for frameCount %zu",
- mNotificationFramesAct, maxNotificationFrames, frameCount);
- }
- mNotificationFramesAct = (uint32_t) maxNotificationFrames;
- }
- }
+ mFlags = output.flags;
//mOutput != output includes the case where mOutput == AUDIO_IO_HANDLE_NONE for first creation
- if (mDeviceCallback != 0 && mOutput != output) {
+ if (mDeviceCallback != 0 && mOutput != output.outputId) {
if (mOutput != AUDIO_IO_HANDLE_NONE) {
AudioSystem::removeAudioDeviceCallback(this, mOutput);
}
- AudioSystem::addAudioDeviceCallback(this, output);
+ AudioSystem::addAudioDeviceCallback(this, output.outputId);
callbackAdded = true;
}
// We retain a copy of the I/O handle, but don't own the reference
- mOutput = output;
+ mOutput = output.outputId;
mRefreshRemaining = true;
// Starting address of buffers in shared memory. If there is a shared buffer, buffers
@@ -1628,18 +1443,16 @@
if (buffers == NULL) {
ALOGE("Could not get buffer pointer");
status = NO_INIT;
- goto release;
+ goto error;
}
}
mAudioTrack->attachAuxEffect(mAuxEffectId);
- mFrameCount = frameCount;
- updateLatency_l(); // this refetches mAfLatency and sets mLatency
// If IAudioTrack is re-created, don't let the requested frameCount
// decrease. This can confuse clients that cache frameCount().
- if (frameCount > mReqFrameCount) {
- mReqFrameCount = frameCount;
+ if (mFrameCount > mReqFrameCount) {
+ mReqFrameCount = mFrameCount;
}
// reset server position to 0 as we have new cblk.
@@ -1648,9 +1461,9 @@
// update proxy
if (mSharedBuffer == 0) {
mStaticProxy.clear();
- mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize);
+ mProxy = new AudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
} else {
- mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize);
+ mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
mProxy = mStaticProxy;
}
@@ -1676,8 +1489,7 @@
return NO_ERROR;
}
-release:
- AudioSystem::releaseOutput(output, streamType, mSessionId);
+error:
if (callbackAdded) {
// note: mOutput is always valid is callbackAdded is true
AudioSystem::removeAudioDeviceCallback(this, mOutput);
@@ -1685,6 +1497,8 @@
if (status == NO_ERROR) {
status = NO_INIT;
}
+
+ // sp<IAudioTrack> track destructor will cause releaseOutput() to be called by AudioFlinger
return status;
}
@@ -2420,8 +2234,8 @@
return true; // static tracks do not have issues with buffer sizing.
}
const size_t minFrameCount =
- calculateMinFrameCount(mAfLatency, mAfFrameCount, mAfSampleRate, sampleRate, speed
- /*, 0 mNotificationsPerBufferReq*/);
+ AudioSystem::calculateMinFrameCount(mAfLatency, mAfFrameCount, mAfSampleRate,
+ sampleRate, speed /*, 0 mNotificationsPerBufferReq*/);
const bool allowed = mFrameCount >= minFrameCount;
ALOGD_IF(!allowed,
"isSampleRateSpeedAllowed_l denied "
diff --git a/media/libaudioclient/IAudioFlinger.cpp b/media/libaudioclient/IAudioFlinger.cpp
index fc8c11a..5cf2bdb 100644
--- a/media/libaudioclient/IAudioFlinger.cpp
+++ b/media/libaudioclient/IAudioFlinger.cpp
@@ -95,83 +95,38 @@
{
}
- virtual sp<IAudioTrack> createTrack(
- audio_stream_type_t streamType,
- uint32_t sampleRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- size_t *pFrameCount,
- audio_output_flags_t *flags,
- const sp<IMemory>& sharedBuffer,
- audio_io_handle_t output,
- pid_t pid,
- pid_t tid,
- audio_session_t *sessionId,
- int clientUid,
- status_t *status,
- audio_port_handle_t portId)
+ virtual sp<IAudioTrack> createTrack(const CreateTrackInput& input,
+ CreateTrackOutput& output,
+ status_t *status)
{
Parcel data, reply;
sp<IAudioTrack> track;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32((int32_t) streamType);
- data.writeInt32(sampleRate);
- data.writeInt32(format);
- data.writeInt32(channelMask);
- size_t frameCount = pFrameCount != NULL ? *pFrameCount : 0;
- data.writeInt64(frameCount);
- audio_output_flags_t lFlags = flags != NULL ? *flags : AUDIO_OUTPUT_FLAG_NONE;
- data.writeInt32(lFlags);
- // haveSharedBuffer
- if (sharedBuffer != 0) {
- data.writeInt32(true);
- data.writeStrongBinder(IInterface::asBinder(sharedBuffer));
- } else {
- data.writeInt32(false);
+
+ if (status == nullptr) {
+ return track;
}
- data.writeInt32((int32_t) output);
- data.writeInt32((int32_t) pid);
- data.writeInt32((int32_t) tid);
- audio_session_t lSessionId = AUDIO_SESSION_ALLOCATE;
- if (sessionId != NULL) {
- lSessionId = *sessionId;
- }
- data.writeInt32(lSessionId);
- data.writeInt32(clientUid);
- data.writeInt32(portId);
+
+ input.writeToParcel(&data);
+
status_t lStatus = remote()->transact(CREATE_TRACK, data, &reply);
if (lStatus != NO_ERROR) {
- ALOGE("createTrack error: %s", strerror(-lStatus));
- } else {
- frameCount = reply.readInt64();
- if (pFrameCount != NULL) {
- *pFrameCount = frameCount;
- }
- lFlags = (audio_output_flags_t)reply.readInt32();
- if (flags != NULL) {
- *flags = lFlags;
- }
- lSessionId = (audio_session_t) reply.readInt32();
- if (sessionId != NULL) {
- *sessionId = lSessionId;
- }
- lStatus = reply.readInt32();
- track = interface_cast<IAudioTrack>(reply.readStrongBinder());
- if (lStatus == NO_ERROR) {
- if (track == 0) {
- ALOGE("createTrack should have returned an IAudioTrack");
- lStatus = UNKNOWN_ERROR;
- }
- } else {
- if (track != 0) {
- ALOGE("createTrack returned an IAudioTrack but with status %d", lStatus);
- track.clear();
- }
- }
+ ALOGE("createTrack transaction error %d", lStatus);
+ *status = DEAD_OBJECT;
+ return track;
}
- if (status != NULL) {
- *status = lStatus;
+ *status = reply.readInt32();
+ if (*status != NO_ERROR) {
+ ALOGE("createTrack returned error %d", *status);
+ return track;
}
+ track = interface_cast<IAudioTrack>(reply.readStrongBinder());
+ if (track == 0) {
+ ALOGE("createTrack returned an NULL IAudioTrack with status OK");
+ *status = DEAD_OBJECT;
+ return track;
+ }
+ output.readFromParcel(&reply);
return track;
}
@@ -970,41 +925,27 @@
switch (code) {
case CREATE_TRACK: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- int streamType = data.readInt32();
- uint32_t sampleRate = data.readInt32();
- audio_format_t format = (audio_format_t) data.readInt32();
- audio_channel_mask_t channelMask = data.readInt32();
- size_t frameCount = data.readInt64();
- audio_output_flags_t flags = (audio_output_flags_t) data.readInt32();
- bool haveSharedBuffer = data.readInt32() != 0;
- sp<IMemory> buffer;
- if (haveSharedBuffer) {
- buffer = interface_cast<IMemory>(data.readStrongBinder());
+
+ CreateTrackInput input;
+ if (input.readFromParcel((Parcel*)&data) != NO_ERROR) {
+ reply->writeInt32(DEAD_OBJECT);
+ return NO_ERROR;
}
- audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
- pid_t pid = (pid_t) data.readInt32();
- pid_t tid = (pid_t) data.readInt32();
- audio_session_t sessionId = (audio_session_t) data.readInt32();
- int clientUid = data.readInt32();
- audio_port_handle_t portId = (audio_port_handle_t) data.readInt32();
- status_t status = NO_ERROR;
- sp<IAudioTrack> track;
- if ((haveSharedBuffer && (buffer == 0)) ||
- ((buffer != 0) && (buffer->pointer() == NULL))) {
- ALOGW("CREATE_TRACK: cannot retrieve shared memory");
- status = DEAD_OBJECT;
- } else {
- track = createTrack(
- (audio_stream_type_t) streamType, sampleRate, format,
- channelMask, &frameCount, &flags, buffer, output, pid, tid,
- &sessionId, clientUid, &status, portId);
- LOG_ALWAYS_FATAL_IF((track != 0) != (status == NO_ERROR));
- }
- reply->writeInt64(frameCount);
- reply->writeInt32(flags);
- reply->writeInt32(sessionId);
+
+ status_t status;
+ CreateTrackOutput output;
+
+ sp<IAudioTrack> track= createTrack(input,
+ output,
+ &status);
+
+ LOG_ALWAYS_FATAL_IF((track != 0) != (status == NO_ERROR));
reply->writeInt32(status);
+ if (status != NO_ERROR) {
+ return NO_ERROR;
+ }
reply->writeStrongBinder(IInterface::asBinder(track));
+ output.writeToParcel(reply);
return NO_ERROR;
} break;
case OPEN_RECORD: {
diff --git a/media/libaudioclient/include/media/AudioClient.h b/media/libaudioclient/include/media/AudioClient.h
index 9efd76d..108e326 100644
--- a/media/libaudioclient/include/media/AudioClient.h
+++ b/media/libaudioclient/include/media/AudioClient.h
@@ -18,6 +18,7 @@
#ifndef ANDROID_AUDIO_CLIENT_H
#define ANDROID_AUDIO_CLIENT_H
+#include <binder/Parcel.h>
#include <system/audio.h>
#include <utils/String16.h>
@@ -26,11 +27,28 @@
class AudioClient {
public:
AudioClient() :
- clientUid(-1), clientPid(-1), packageName("") {}
+ clientUid(-1), clientPid(-1), clientTid(-1), packageName("") {}
uid_t clientUid;
pid_t clientPid;
+ pid_t clientTid;
String16 packageName;
+
+ status_t readFromParcel(Parcel *parcel) {
+ clientUid = parcel->readInt32();
+ clientPid = parcel->readInt32();
+ clientTid = parcel->readInt32();
+ packageName = parcel->readString16();
+ return NO_ERROR;
+ }
+
+ status_t writeToParcel(Parcel *parcel) const {
+ parcel->writeInt32(clientUid);
+ parcel->writeInt32(clientPid);
+ parcel->writeInt32(clientTid);
+ parcel->writeString16(packageName);
+ return NO_ERROR;
+ }
};
}; // namespace android
diff --git a/media/libaudioclient/include/media/AudioSystem.h b/media/libaudioclient/include/media/AudioSystem.h
index 327eba8..66601da 100644
--- a/media/libaudioclient/include/media/AudioSystem.h
+++ b/media/libaudioclient/include/media/AudioSystem.h
@@ -106,6 +106,9 @@
static float linearToLog(int volume);
static int logToLinear(float volume);
+ static size_t calculateMinFrameCount(
+ uint32_t afLatencyMs, uint32_t afFrameCount, uint32_t afSampleRate,
+ uint32_t sampleRate, float speed /*, uint32_t notificationsPerBufferReq*/);
// Returned samplingRate and frameCount output values are guaranteed
// to be non-zero if status == NO_ERROR
@@ -209,8 +212,6 @@
static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config);
static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
- // Client must successfully hand off the handle reference to AudioFlinger via createTrack(),
- // or release it with releaseOutput().
static status_t getOutputForAttr(const audio_attributes_t *attr,
audio_io_handle_t *output,
audio_session_t session,
diff --git a/media/libaudioclient/include/media/AudioTrack.h b/media/libaudioclient/include/media/AudioTrack.h
index 8973133..9fbd04b 100644
--- a/media/libaudioclient/include/media/AudioTrack.h
+++ b/media/libaudioclient/include/media/AudioTrack.h
@@ -1182,7 +1182,6 @@
pid_t mClientPid;
wp<AudioSystem::AudioDeviceCallback> mDeviceCallback;
- audio_port_handle_t mPortId; // unique ID allocated by audio policy
};
}; // namespace android
diff --git a/media/libaudioclient/include/media/IAudioFlinger.h b/media/libaudioclient/include/media/IAudioFlinger.h
index 133d6c9..9061c26 100644
--- a/media/libaudioclient/include/media/IAudioFlinger.h
+++ b/media/libaudioclient/include/media/IAudioFlinger.h
@@ -24,6 +24,8 @@
#include <utils/RefBase.h>
#include <utils/Errors.h>
#include <binder/IInterface.h>
+#include <binder/Parcel.h>
+#include <media/AudioClient.h>
#include <media/IAudioTrack.h>
#include <media/IAudioFlingerClient.h>
#include <system/audio.h>
@@ -44,6 +46,135 @@
public:
DECLARE_META_INTERFACE(AudioFlinger);
+ /* CreateTrackInput contains all input arguments sent by AudioTrack to AudioFlinger
+ * when calling createTrack() including arguments that will be updated by AudioFlinger
+ * and returned in CreateTrackOutput object
+ */
+ class CreateTrackInput {
+ public:
+ status_t readFromParcel(Parcel *parcel) {
+ /* input arguments*/
+ memset(&attr, 0, sizeof(audio_attributes_t));
+ if (parcel->read(&attr, sizeof(audio_attributes_t)) != NO_ERROR) {
+ return DEAD_OBJECT;
+ }
+ attr.tags[AUDIO_ATTRIBUTES_TAGS_MAX_SIZE -1] = '\0';
+ memset(&config, 0, sizeof(audio_config_t));
+ if (parcel->read(&config, sizeof(audio_config_t)) != NO_ERROR) {
+ return DEAD_OBJECT;
+ }
+ (void)clientInfo.readFromParcel(parcel);
+ if (parcel->readInt32() != 0) {
+ sharedBuffer = interface_cast<IMemory>(parcel->readStrongBinder());
+ if (sharedBuffer == 0 || sharedBuffer->pointer() == NULL) {
+ return BAD_VALUE;
+ }
+ }
+ notificationsPerBuffer = parcel->readInt32();
+ speed = parcel->readFloat();
+
+ /* input/output arguments*/
+ (void)parcel->read(&flags, sizeof(audio_output_flags_t));
+ frameCount = parcel->readInt64();
+ notificationFrameCount = parcel->readInt64();
+ (void)parcel->read(&selectedDeviceId, sizeof(audio_port_handle_t));
+ (void)parcel->read(&sessionId, sizeof(audio_session_t));
+ return NO_ERROR;
+ }
+
+ status_t writeToParcel(Parcel *parcel) const {
+ /* input arguments*/
+ (void)parcel->write(&attr, sizeof(audio_attributes_t));
+ (void)parcel->write(&config, sizeof(audio_config_t));
+ (void)clientInfo.writeToParcel(parcel);
+ if (sharedBuffer != 0) {
+ (void)parcel->writeInt32(1);
+ (void)parcel->writeStrongBinder(IInterface::asBinder(sharedBuffer));
+ } else {
+ (void)parcel->writeInt32(0);
+ }
+ (void)parcel->writeInt32(notificationsPerBuffer);
+ (void)parcel->writeFloat(speed);
+
+ /* input/output arguments*/
+ (void)parcel->write(&flags, sizeof(audio_output_flags_t));
+ (void)parcel->writeInt64(frameCount);
+ (void)parcel->writeInt64(notificationFrameCount);
+ (void)parcel->write(&selectedDeviceId, sizeof(audio_port_handle_t));
+ (void)parcel->write(&sessionId, sizeof(audio_session_t));
+ return NO_ERROR;
+ }
+
+ /* input */
+ audio_attributes_t attr;
+ audio_config_t config;
+ AudioClient clientInfo;
+ sp<IMemory> sharedBuffer;
+ uint32_t notificationsPerBuffer;
+ float speed;
+
+ /* input/output */
+ audio_output_flags_t flags;
+ size_t frameCount;
+ size_t notificationFrameCount;
+ audio_port_handle_t selectedDeviceId;
+ audio_session_t sessionId;
+ };
+
+ /* CreateTrackOutput contains all output arguments returned by AudioFlinger to AudioTrack
+ * when calling createTrack() including arguments that were passed as I/O for update by
+ * CreateTrackInput.
+ */
+ class CreateTrackOutput {
+ public:
+ status_t readFromParcel(Parcel *parcel) {
+ /* input/output arguments*/
+ (void)parcel->read(&flags, sizeof(audio_output_flags_t));
+ frameCount = parcel->readInt64();
+ notificationFrameCount = parcel->readInt64();
+ (void)parcel->read(&selectedDeviceId, sizeof(audio_port_handle_t));
+ (void)parcel->read(&sessionId, sizeof(audio_session_t));
+
+ /* output arguments*/
+ sampleRate = parcel->readUint32();
+ afFrameCount = parcel->readInt64();
+ afSampleRate = parcel->readInt64();
+ afLatencyMs = parcel->readInt32();
+ (void)parcel->read(&outputId, sizeof(audio_io_handle_t));
+ return NO_ERROR;
+ }
+
+ status_t writeToParcel(Parcel *parcel) const {
+ /* input/output arguments*/
+ (void)parcel->write(&flags, sizeof(audio_output_flags_t));
+ (void)parcel->writeInt64(frameCount);
+ (void)parcel->writeInt64(notificationFrameCount);
+ (void)parcel->write(&selectedDeviceId, sizeof(audio_port_handle_t));
+ (void)parcel->write(&sessionId, sizeof(audio_session_t));
+
+ /* output arguments*/
+ (void)parcel->writeUint32(sampleRate);
+ (void)parcel->writeInt64(afFrameCount);
+ (void)parcel->writeInt64(afSampleRate);
+ (void)parcel->writeInt32(afLatencyMs);
+ (void)parcel->write(&outputId, sizeof(audio_io_handle_t));
+ return NO_ERROR;
+ }
+
+ /* input/output */
+ audio_output_flags_t flags;
+ size_t frameCount;
+ size_t notificationFrameCount;
+ audio_port_handle_t selectedDeviceId;
+ audio_session_t sessionId;
+
+ /* output */
+ uint32_t sampleRate;
+ size_t afFrameCount;
+ uint32_t afSampleRate;
+ uint32_t afLatencyMs;
+ audio_io_handle_t outputId;
+ };
// invariant on exit for all APIs that return an sp<>:
// (return value != 0) == (*status == NO_ERROR)
@@ -51,24 +182,9 @@
/* create an audio track and registers it with AudioFlinger.
* return null if the track cannot be created.
*/
- virtual sp<IAudioTrack> createTrack(
- audio_stream_type_t streamType,
- uint32_t sampleRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- size_t *pFrameCount,
- audio_output_flags_t *flags,
- const sp<IMemory>& sharedBuffer,
- // On successful return, AudioFlinger takes over the handle
- // reference and will release it when the track is destroyed.
- // However on failure, the client is responsible for release.
- audio_io_handle_t output,
- pid_t pid,
- pid_t tid, // -1 means unused, otherwise must be valid non-0
- audio_session_t *sessionId,
- int clientUid,
- status_t *status,
- audio_port_handle_t portId) = 0;
+ virtual sp<IAudioTrack> createTrack(const CreateTrackInput& input,
+ CreateTrackOutput& output,
+ status_t *status) = 0;
virtual sp<media::IAudioRecord> openRecord(
// On successful return, AudioFlinger takes over the handle