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ara-mdk / platform / frameworks / av / 72c324efd5a0f632d8f2b57b9f587a540d809b82 / . / media / libmedia / AudioTrack.cpp
blob: aec8c4a6a9dfd3577a998d4f07c4b6b613d82516 [file] [log] [blame]
/*
**
** Copyright 2007, The Android Open Source Project
**
** Licensed under the Apache License, Version 2.0 (the "License");
** you may not use this file except in compliance with the License.
** You may obtain a copy of the License at
**
** http://www.apache.org/licenses/LICENSE-2.0
**
** Unless required by applicable law or agreed to in writing, software
** distributed under the License is distributed on an "AS IS" BASIS,
** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
** See the License for the specific language governing permissions and
** limitations under the License.
*/
//#define LOG_NDEBUG 0
#define LOG_TAG "AudioTrack"
#include <stdint.h>
#include <sys/types.h>
#include <limits.h>
#include <sched.h>
#include <sys/resource.h>
#include <private/media/AudioTrackShared.h>
#include <media/AudioSystem.h>
#include <media/AudioTrack.h>
#include <utils/Log.h>
#include <binder/Parcel.h>
#include <binder/IPCThreadState.h>
#include <utils/Timers.h>
#include <utils/Atomic.h>
#include <cutils/bitops.h>
#include <cutils/compiler.h>
#include <system/audio.h>
#include <system/audio_policy.h>
#include <audio_utils/primitives.h>
namespace android {
// ---------------------------------------------------------------------------
// static
status_t AudioTrack::getMinFrameCount(
int* frameCount,
audio_stream_type_t streamType,
uint32_t sampleRate)
{
if (frameCount == NULL) return BAD_VALUE;
// default to 0 in case of error
*frameCount = 0;
// FIXME merge with similar code in createTrack_l(), except we're missing
// some information here that is available in createTrack_l():
// audio_io_handle_t output
// audio_format_t format
// audio_channel_mask_t channelMask
// audio_output_flags_t flags
int afSampleRate;
if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
return NO_INIT;
}
int afFrameCount;
if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
return NO_INIT;
}
uint32_t afLatency;
if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
return NO_INIT;
}
// Ensure that buffer depth covers at least audio hardware latency
uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate);
if (minBufCount < 2) minBufCount = 2;
*frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
afFrameCount * minBufCount * sampleRate / afSampleRate;
ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d",
*frameCount, afFrameCount, minBufCount, afSampleRate, afLatency);
return NO_ERROR;
}
// ---------------------------------------------------------------------------
AudioTrack::AudioTrack()
: mStatus(NO_INIT),
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT)
{
}
AudioTrack::AudioTrack(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
int frameCount,
audio_output_flags_t flags,
callback_t cbf,
void* user,
int notificationFrames,
int sessionId)
: mStatus(NO_INIT),
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT)
{
mStatus = set(streamType, sampleRate, format, channelMask,
frameCount, flags, cbf, user, notificationFrames,
0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId);
}
// DEPRECATED
AudioTrack::AudioTrack(
int streamType,
uint32_t sampleRate,
int format,
int channelMask,
int frameCount,
uint32_t flags,
callback_t cbf,
void* user,
int notificationFrames,
int sessionId)
: mStatus(NO_INIT),
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT)
{
mStatus = set((audio_stream_type_t)streamType, sampleRate, (audio_format_t)format,
(audio_channel_mask_t) channelMask,
frameCount, (audio_output_flags_t)flags, cbf, user, notificationFrames,
0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId);
}
AudioTrack::AudioTrack(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
const sp<IMemory>& sharedBuffer,
audio_output_flags_t flags,
callback_t cbf,
void* user,
int notificationFrames,
int sessionId)
: mStatus(NO_INIT),
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT)
{
mStatus = set(streamType, sampleRate, format, channelMask,
0 /*frameCount*/, flags, cbf, user, notificationFrames,
sharedBuffer, false /*threadCanCallJava*/, sessionId);
}
AudioTrack::~AudioTrack()
{
ALOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer());
if (mStatus == NO_ERROR) {
// Make sure that callback function exits in the case where
// it is looping on buffer full condition in obtainBuffer().
// Otherwise the callback thread will never exit.
stop();
if (mAudioTrackThread != 0) {
mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
mAudioTrackThread->requestExitAndWait();
mAudioTrackThread.clear();
}
mAudioTrack.clear();
IPCThreadState::self()->flushCommands();
AudioSystem::releaseAudioSessionId(mSessionId);
}
}
status_t AudioTrack::set(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
int frameCount,
audio_output_flags_t flags,
callback_t cbf,
void* user,
int notificationFrames,
const sp<IMemory>& sharedBuffer,
bool threadCanCallJava,
int sessionId)
{
ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
ALOGV("set() streamType %d frameCount %d flags %04x", streamType, frameCount, flags);
AutoMutex lock(mLock);
if (mAudioTrack != 0) {
ALOGE("Track already in use");
return INVALID_OPERATION;
}
// handle default values first.
if (streamType == AUDIO_STREAM_DEFAULT) {
streamType = AUDIO_STREAM_MUSIC;
}
if (sampleRate == 0) {
int afSampleRate;
if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
return NO_INIT;
}
sampleRate = afSampleRate;
}
// these below should probably come from the audioFlinger too...
if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
}
if (channelMask == 0) {
channelMask = AUDIO_CHANNEL_OUT_STEREO;
}
// validate parameters
if (!audio_is_valid_format(format)) {
ALOGE("Invalid format");
return BAD_VALUE;
}
// AudioFlinger does not currently support 8-bit data in shared memory
if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) {
ALOGE("8-bit data in shared memory is not supported");
return BAD_VALUE;
}
// force direct flag if format is not linear PCM
if (!audio_is_linear_pcm(format)) {
flags = (audio_output_flags_t)
// FIXME why can't we allow direct AND fast?
((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST);
}
// only allow deep buffering for music stream type
if (streamType != AUDIO_STREAM_MUSIC) {
flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
}
if (!audio_is_output_channel(channelMask)) {
ALOGE("Invalid channel mask %#x", channelMask);
return BAD_VALUE;
}
uint32_t channelCount = popcount(channelMask);
audio_io_handle_t output = AudioSystem::getOutput(
streamType,
sampleRate, format, channelMask,
flags);
if (output == 0) {
ALOGE("Could not get audio output for stream type %d", streamType);
return BAD_VALUE;
}
mVolume[LEFT] = 1.0f;
mVolume[RIGHT] = 1.0f;
mSendLevel = 0.0f;
mFrameCount = frameCount;
mNotificationFramesReq = notificationFrames;
mSessionId = sessionId;
mAuxEffectId = 0;
mFlags = flags;
mCbf = cbf;
if (cbf != NULL) {
mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
}
// create the IAudioTrack
status_t status = createTrack_l(streamType,
sampleRate,
format,
channelMask,
frameCount,
flags,
sharedBuffer,
output);
if (status != NO_ERROR) {
if (mAudioTrackThread != 0) {
mAudioTrackThread->requestExit();
mAudioTrackThread.clear();
}
return status;
}
mStatus = NO_ERROR;
mStreamType = streamType;
mFormat = format;
mChannelMask = channelMask;
mChannelCount = channelCount;
mSharedBuffer = sharedBuffer;
mMuted = false;
mActive = false;
mUserData = user;
mLoopCount = 0;
mMarkerPosition = 0;
mMarkerReached = false;
mNewPosition = 0;
mUpdatePeriod = 0;
mFlushed = false;
AudioSystem::acquireAudioSessionId(mSessionId);
mRestoreStatus = NO_ERROR;
return NO_ERROR;
}
status_t AudioTrack::initCheck() const
{
return mStatus;
}
// -------------------------------------------------------------------------
uint32_t AudioTrack::latency() const
{
return mLatency;
}
audio_stream_type_t AudioTrack::streamType() const
{
return mStreamType;
}
audio_format_t AudioTrack::format() const
{
return mFormat;
}
int AudioTrack::channelCount() const
{
return mChannelCount;
}
uint32_t AudioTrack::frameCount() const
{
return mCblk->frameCount;
}
size_t AudioTrack::frameSize() const
{
if (audio_is_linear_pcm(mFormat)) {
return channelCount()*audio_bytes_per_sample(mFormat);
} else {
return sizeof(uint8_t);
}
}
sp<IMemory>& AudioTrack::sharedBuffer()
{
return mSharedBuffer;
}
// -------------------------------------------------------------------------
void AudioTrack::start()
{
sp<AudioTrackThread> t = mAudioTrackThread;
ALOGV("start %p", this);
AutoMutex lock(mLock);
// acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
// while we are accessing the cblk
sp<IAudioTrack> audioTrack = mAudioTrack;
sp<IMemory> iMem = mCblkMemory;
audio_track_cblk_t* cblk = mCblk;
if (!mActive) {
mFlushed = false;
mActive = true;
mNewPosition = cblk->server + mUpdatePeriod;
cblk->lock.lock();
cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
cblk->waitTimeMs = 0;
android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags);
if (t != 0) {
t->resume();
} else {
mPreviousPriority = getpriority(PRIO_PROCESS, 0);
get_sched_policy(0, &mPreviousSchedulingGroup);
androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
}
ALOGV("start %p before lock cblk %p", this, mCblk);
status_t status = NO_ERROR;
if (!(cblk->flags & CBLK_INVALID_MSK)) {
cblk->lock.unlock();
ALOGV("mAudioTrack->start()");
status = mAudioTrack->start();
cblk->lock.lock();
if (status == DEAD_OBJECT) {
android_atomic_or(CBLK_INVALID_ON, &cblk->flags);
}
}
if (cblk->flags & CBLK_INVALID_MSK) {
status = restoreTrack_l(cblk, true);
}
cblk->lock.unlock();
if (status != NO_ERROR) {
ALOGV("start() failed");
mActive = false;
if (t != 0) {
t->pause();
} else {
setpriority(PRIO_PROCESS, 0, mPreviousPriority);
set_sched_policy(0, mPreviousSchedulingGroup);
}
}
}
}
void AudioTrack::stop()
{
sp<AudioTrackThread> t = mAudioTrackThread;
ALOGV("stop %p", this);
AutoMutex lock(mLock);
if (mActive) {
mActive = false;
mCblk->cv.signal();
mAudioTrack->stop();
// Cancel loops (If we are in the middle of a loop, playback
// would not stop until loopCount reaches 0).
setLoop_l(0, 0, 0);
// the playback head position will reset to 0, so if a marker is set, we need
// to activate it again
mMarkerReached = false;
// Force flush if a shared buffer is used otherwise audioflinger
// will not stop before end of buffer is reached.
if (mSharedBuffer != 0) {
flush_l();
}
if (t != 0) {
t->pause();
} else {
setpriority(PRIO_PROCESS, 0, mPreviousPriority);
set_sched_policy(0, mPreviousSchedulingGroup);
}
}
}
bool AudioTrack::stopped() const
{
AutoMutex lock(mLock);
return stopped_l();
}
void AudioTrack::flush()
{
AutoMutex lock(mLock);
flush_l();
}
// must be called with mLock held
void AudioTrack::flush_l()
{
ALOGV("flush");
// clear playback marker and periodic update counter
mMarkerPosition = 0;
mMarkerReached = false;
mUpdatePeriod = 0;
if (!mActive) {
mFlushed = true;
mAudioTrack->flush();
// Release AudioTrack callback thread in case it was waiting for new buffers
// in AudioTrack::obtainBuffer()
mCblk->cv.signal();
}
}
void AudioTrack::pause()
{
ALOGV("pause");
AutoMutex lock(mLock);
if (mActive) {
mActive = false;
mCblk->cv.signal();
mAudioTrack->pause();
}
}
void AudioTrack::mute(bool e)
{
mAudioTrack->mute(e);
mMuted = e;
}
bool AudioTrack::muted() const
{
return mMuted;
}
status_t AudioTrack::setVolume(float left, float right)
{
if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) {
return BAD_VALUE;
}
AutoMutex lock(mLock);
mVolume[LEFT] = left;
mVolume[RIGHT] = right;
mCblk->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000));
return NO_ERROR;
}
void AudioTrack::getVolume(float* left, float* right) const
{
if (left != NULL) {
*left = mVolume[LEFT];
}
if (right != NULL) {
*right = mVolume[RIGHT];
}
}
status_t AudioTrack::setAuxEffectSendLevel(float level)
{
ALOGV("setAuxEffectSendLevel(%f)", level);
if (level < 0.0f || level > 1.0f) {
return BAD_VALUE;
}
AutoMutex lock(mLock);
mSendLevel = level;
mCblk->setSendLevel(level);
return NO_ERROR;
}
void AudioTrack::getAuxEffectSendLevel(float* level) const
{
if (level != NULL) {
*level = mSendLevel;
}
}
status_t AudioTrack::setSampleRate(int rate)
{
int afSamplingRate;
if (mIsTimed) {
return INVALID_OPERATION;
}
if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {
return NO_INIT;
}
// Resampler implementation limits input sampling rate to 2 x output sampling rate.
if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE;
AutoMutex lock(mLock);
mCblk->sampleRate = rate;
return NO_ERROR;
}
uint32_t AudioTrack::getSampleRate() const
{
if (mIsTimed) {
return INVALID_OPERATION;
}
AutoMutex lock(mLock);
return mCblk->sampleRate;
}
status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
{
AutoMutex lock(mLock);
return setLoop_l(loopStart, loopEnd, loopCount);
}
// must be called with mLock held
status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
{
audio_track_cblk_t* cblk = mCblk;
Mutex::Autolock _l(cblk->lock);
if (loopCount == 0) {
cblk->loopStart = UINT_MAX;
cblk->loopEnd = UINT_MAX;
cblk->loopCount = 0;
mLoopCount = 0;
return NO_ERROR;
}
if (mIsTimed) {
return INVALID_OPERATION;
}
if (loopStart >= loopEnd ||
loopEnd - loopStart > cblk->frameCount ||
cblk->server > loopStart) {
ALOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user);
return BAD_VALUE;
}
if ((mSharedBuffer != 0) && (loopEnd > cblk->frameCount)) {
ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d",
loopStart, loopEnd, cblk->frameCount);
return BAD_VALUE;
}
cblk->loopStart = loopStart;
cblk->loopEnd = loopEnd;
cblk->loopCount = loopCount;
mLoopCount = loopCount;
return NO_ERROR;
}
status_t AudioTrack::setMarkerPosition(uint32_t marker)
{
if (mCbf == NULL) return INVALID_OPERATION;
mMarkerPosition = marker;
mMarkerReached = false;
return NO_ERROR;
}
status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
{
if (marker == NULL) return BAD_VALUE;
*marker = mMarkerPosition;
return NO_ERROR;
}
status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
{
if (mCbf == NULL) return INVALID_OPERATION;
uint32_t curPosition;
getPosition(&curPosition);
mNewPosition = curPosition + updatePeriod;
mUpdatePeriod = updatePeriod;
return NO_ERROR;
}
status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const
{
if (updatePeriod == NULL) return BAD_VALUE;
*updatePeriod = mUpdatePeriod;
return NO_ERROR;
}
status_t AudioTrack::setPosition(uint32_t position)
{
if (mIsTimed) return INVALID_OPERATION;
AutoMutex lock(mLock);
if (!stopped_l()) return INVALID_OPERATION;
Mutex::Autolock _l(mCblk->lock);
if (position > mCblk->user) return BAD_VALUE;
mCblk->server = position;
android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags);
return NO_ERROR;
}
status_t AudioTrack::getPosition(uint32_t *position)
{
if (position == NULL) return BAD_VALUE;
AutoMutex lock(mLock);
*position = mFlushed ? 0 : mCblk->server;
return NO_ERROR;
}
status_t AudioTrack::reload()
{
AutoMutex lock(mLock);
if (!stopped_l()) return INVALID_OPERATION;
flush_l();
mCblk->stepUser(mCblk->frameCount);
return NO_ERROR;
}
audio_io_handle_t AudioTrack::getOutput()
{
AutoMutex lock(mLock);
return getOutput_l();
}
// must be called with mLock held
audio_io_handle_t AudioTrack::getOutput_l()
{
return AudioSystem::getOutput(mStreamType,
mCblk->sampleRate, mFormat, mChannelMask, mFlags);
}
int AudioTrack::getSessionId() const
{
return mSessionId;
}
status_t AudioTrack::attachAuxEffect(int effectId)
{
ALOGV("attachAuxEffect(%d)", effectId);
status_t status = mAudioTrack->attachAuxEffect(effectId);
if (status == NO_ERROR) {
mAuxEffectId = effectId;
}
return status;
}
// -------------------------------------------------------------------------
// must be called with mLock held
status_t AudioTrack::createTrack_l(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
int frameCount,
audio_output_flags_t flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output)
{
status_t status;
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
if (audioFlinger == 0) {
ALOGE("Could not get audioflinger");
return NO_INIT;
}
uint32_t afLatency;
if (AudioSystem::getLatency(output, streamType, &afLatency) != NO_ERROR) {
return NO_INIT;
}
// Client decides whether the track is TIMED (see below), but can only express a preference
// for FAST. Server will perform additional tests.
if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !(
// either of these use cases:
// use case 1: shared buffer
(sharedBuffer != 0) ||
// use case 2: callback handler
(mCbf != NULL))) {
ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client");
// once denied, do not request again if IAudioTrack is re-created
flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
mFlags = flags;
}
ALOGV("createTrack_l() output %d afLatency %d", output, afLatency);
mNotificationFramesAct = mNotificationFramesReq;
if (!audio_is_linear_pcm(format)) {
if (sharedBuffer != 0) {
// Same comment as below about ignoring frameCount parameter for set()
frameCount = sharedBuffer->size();
} else if (frameCount == 0) {
int afFrameCount;
if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) {
return NO_INIT;
}
frameCount = afFrameCount;
}
} else if (sharedBuffer != 0) {
// Ensure that buffer alignment matches channelCount
int channelCount = popcount(channelMask);
// 8-bit data in shared memory is not currently supported by AudioFlinger
size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2;
if (channelCount > 1) {
// More than 2 channels does not require stronger alignment than stereo
alignment <<= 1;
}
if (((uint32_t)sharedBuffer->pointer() & (alignment - 1)) != 0) {
ALOGE("Invalid buffer alignment: address %p, channelCount %d",
sharedBuffer->pointer(), channelCount);
return BAD_VALUE;
}
// 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 = sharedBuffer->size()/channelCount/sizeof(int16_t);
} else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
// FIXME move these calculations and associated checks to server
int afSampleRate;
if (AudioSystem::getSamplingRate(output, streamType, &afSampleRate) != NO_ERROR) {
return NO_INIT;
}
int afFrameCount;
if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) {
return NO_INIT;
}
// Ensure that buffer depth covers at least audio hardware latency
uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
if (minBufCount < 2) minBufCount = 2;
int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
ALOGV("minFrameCount: %d, afFrameCount=%d, minBufCount=%d, sampleRate=%d, afSampleRate=%d"
", afLatency=%d",
minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency);
if (frameCount == 0) {
frameCount = minFrameCount;
}
if (mNotificationFramesAct == 0) {
mNotificationFramesAct = frameCount/2;
}
// Make sure that application is notified with sufficient margin
// before underrun
if (mNotificationFramesAct > (uint32_t)frameCount/2) {
mNotificationFramesAct = frameCount/2;
}
if (frameCount < minFrameCount) {
// not ALOGW because it happens all the time when playing key clicks over A2DP
ALOGV("Minimum buffer size corrected from %d to %d",
frameCount, minFrameCount);
frameCount = minFrameCount;
}
} else {
// For fast tracks, the frame count calculations and checks are done by server
}
IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT;
if (mIsTimed) {
trackFlags |= IAudioFlinger::TRACK_TIMED;
}
pid_t tid = -1;
if (flags & AUDIO_OUTPUT_FLAG_FAST) {
trackFlags |= IAudioFlinger::TRACK_FAST;
if (mAudioTrackThread != 0) {
tid = mAudioTrackThread->getTid();
}
}
sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
streamType,
sampleRate,
format,
channelMask,
frameCount,
trackFlags,
sharedBuffer,
output,
tid,
&mSessionId,
&status);
if (track == 0) {
ALOGE("AudioFlinger could not create track, status: %d", status);
return status;
}
sp<IMemory> cblk = track->getCblk();
if (cblk == 0) {
ALOGE("Could not get control block");
return NO_INIT;
}
mAudioTrack = track;
mCblkMemory = cblk;
mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
// old has the previous value of mCblk->flags before the "or" operation
int32_t old = android_atomic_or(CBLK_DIRECTION_OUT, &mCblk->flags);
if (flags & AUDIO_OUTPUT_FLAG_FAST) {
if (old & CBLK_FAST) {
ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", mCblk->frameCount);
} else {
ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", mCblk->frameCount);
// once denied, do not request again if IAudioTrack is re-created
flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
mFlags = flags;
}
if (sharedBuffer == 0) {
mNotificationFramesAct = mCblk->frameCount/2;
}
}
if (sharedBuffer == 0) {
mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
} else {
mCblk->buffers = sharedBuffer->pointer();
// Force buffer full condition as data is already present in shared memory
mCblk->stepUser(mCblk->frameCount);
}
mCblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000));
mCblk->setSendLevel(mSendLevel);
mAudioTrack->attachAuxEffect(mAuxEffectId);
mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
mCblk->waitTimeMs = 0;
mRemainingFrames = mNotificationFramesAct;
// FIXME don't believe this lie
mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate;
// If IAudioTrack is re-created, don't let the requested frameCount
// decrease. This can confuse clients that cache frameCount().
if (mCblk->frameCount > mFrameCount) {
mFrameCount = mCblk->frameCount;
}
return NO_ERROR;
}
status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
{
AutoMutex lock(mLock);
bool active;
status_t result = NO_ERROR;
audio_track_cblk_t* cblk = mCblk;
uint32_t framesReq = audioBuffer->frameCount;
uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS;
audioBuffer->frameCount = 0;
audioBuffer->size = 0;
uint32_t framesAvail = cblk->framesAvailable();
cblk->lock.lock();
if (cblk->flags & CBLK_INVALID_MSK) {
goto create_new_track;
}
cblk->lock.unlock();
if (framesAvail == 0) {
cblk->lock.lock();
goto start_loop_here;
while (framesAvail == 0) {
active = mActive;
if (CC_UNLIKELY(!active)) {
ALOGV("Not active and NO_MORE_BUFFERS");
cblk->lock.unlock();
return NO_MORE_BUFFERS;
}
if (CC_UNLIKELY(!waitCount)) {
cblk->lock.unlock();
return WOULD_BLOCK;
}
if (!(cblk->flags & CBLK_INVALID_MSK)) {
mLock.unlock();
result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
cblk->lock.unlock();
mLock.lock();
if (!mActive) {
return status_t(STOPPED);
}
cblk->lock.lock();
}
if (cblk->flags & CBLK_INVALID_MSK) {
goto create_new_track;
}
if (CC_UNLIKELY(result != NO_ERROR)) {
cblk->waitTimeMs += waitTimeMs;
if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
// timing out when a loop has been set and we have already written upto loop end
// is a normal condition: no need to wake AudioFlinger up.
if (cblk->user < cblk->loopEnd) {
ALOGW( "obtainBuffer timed out (is the CPU pegged?) %p name=%#x"
"user=%08x, server=%08x", this, cblk->mName, cblk->user, cblk->server);
//unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140)
cblk->lock.unlock();
result = mAudioTrack->start();
cblk->lock.lock();
if (result == DEAD_OBJECT) {
android_atomic_or(CBLK_INVALID_ON, &cblk->flags);
create_new_track:
result = restoreTrack_l(cblk, false);
}
if (result != NO_ERROR) {
ALOGW("obtainBuffer create Track error %d", result);
cblk->lock.unlock();
return result;
}
}
cblk->waitTimeMs = 0;
}
if (--waitCount == 0) {
cblk->lock.unlock();
return TIMED_OUT;
}
}
// read the server count again
start_loop_here:
framesAvail = cblk->framesAvailable_l();
}
cblk->lock.unlock();
}
cblk->waitTimeMs = 0;
if (framesReq > framesAvail) {
framesReq = framesAvail;
}
uint32_t u = cblk->user;
uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
if (framesReq > bufferEnd - u) {
framesReq = bufferEnd - u;
}
audioBuffer->flags = mMuted ? Buffer::MUTE : 0;
audioBuffer->channelCount = mChannelCount;
audioBuffer->frameCount = framesReq;
audioBuffer->size = framesReq * cblk->frameSize;
if (audio_is_linear_pcm(mFormat)) {
audioBuffer->format = AUDIO_FORMAT_PCM_16_BIT;
} else {
audioBuffer->format = mFormat;
}
audioBuffer->raw = (int8_t *)cblk->buffer(u);
active = mActive;
return active ? status_t(NO_ERROR) : status_t(STOPPED);
}
void AudioTrack::releaseBuffer(Buffer* audioBuffer)
{
AutoMutex lock(mLock);
mCblk->stepUser(audioBuffer->frameCount);
if (audioBuffer->frameCount > 0) {
// restart track if it was disabled by audioflinger due to previous underrun
if (mActive && (mCblk->flags & CBLK_DISABLED_MSK)) {
android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags);
ALOGW("releaseBuffer() track %p name=%#x disabled, restarting", this, mCblk->mName);
mAudioTrack->start();
}
}
}
// -------------------------------------------------------------------------
ssize_t AudioTrack::write(const void* buffer, size_t userSize)
{
if (mSharedBuffer != 0) return INVALID_OPERATION;
if (mIsTimed) return INVALID_OPERATION;
if (ssize_t(userSize) < 0) {
// Sanity-check: user is most-likely passing an error code, and it would
// make the return value ambiguous (actualSize vs error).
ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)",
buffer, userSize, userSize);
return BAD_VALUE;
}
ALOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive);
if (userSize == 0) {
return 0;
}
// acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
// while we are accessing the cblk
mLock.lock();
sp<IAudioTrack> audioTrack = mAudioTrack;
sp<IMemory> iMem = mCblkMemory;
mLock.unlock();
ssize_t written = 0;
const int8_t *src = (const int8_t *)buffer;
Buffer audioBuffer;
size_t frameSz = frameSize();
do {
audioBuffer.frameCount = userSize/frameSz;
status_t err = obtainBuffer(&audioBuffer, -1);
if (err < 0) {
// out of buffers, return #bytes written
if (err == status_t(NO_MORE_BUFFERS))
break;
return ssize_t(err);
}
size_t toWrite;
if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
// Divide capacity by 2 to take expansion into account
toWrite = audioBuffer.size>>1;
memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) src, toWrite);
} else {
toWrite = audioBuffer.size;
memcpy(audioBuffer.i8, src, toWrite);
src += toWrite;
}
userSize -= toWrite;
written += toWrite;
releaseBuffer(&audioBuffer);
} while (userSize >= frameSz);
return written;
}
// -------------------------------------------------------------------------
TimedAudioTrack::TimedAudioTrack() {
mIsTimed = true;
}
status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer)
{
status_t result = UNKNOWN_ERROR;
// If the track is not invalid already, try to allocate a buffer. alloc
// fails indicating that the server is dead, flag the track as invalid so
// we can attempt to restore in just a bit.
if (!(mCblk->flags & CBLK_INVALID_MSK)) {
result = mAudioTrack->allocateTimedBuffer(size, buffer);
if (result == DEAD_OBJECT) {
android_atomic_or(CBLK_INVALID_ON, &mCblk->flags);
}
}
// If the track is invalid at this point, attempt to restore it. and try the
// allocation one more time.
if (mCblk->flags & CBLK_INVALID_MSK) {
mCblk->lock.lock();
result = restoreTrack_l(mCblk, false);
mCblk->lock.unlock();
if (result == OK)
result = mAudioTrack->allocateTimedBuffer(size, buffer);
}
return result;
}
status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer,
int64_t pts)
{
status_t status = mAudioTrack->queueTimedBuffer(buffer, pts);
{
AutoMutex lock(mLock);
// restart track if it was disabled by audioflinger due to previous underrun
if (buffer->size() != 0 && status == NO_ERROR &&
mActive && (mCblk->flags & CBLK_DISABLED_MSK)) {
android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags);
ALOGW("queueTimedBuffer() track %p disabled, restarting", this);
mAudioTrack->start();
}
}
return status;
}
status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform,
TargetTimeline target)
{
return mAudioTrack->setMediaTimeTransform(xform, target);
}
// -------------------------------------------------------------------------
bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
{
Buffer audioBuffer;
uint32_t frames;
size_t writtenSize;
mLock.lock();
// acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
// while we are accessing the cblk
sp<IAudioTrack> audioTrack = mAudioTrack;
sp<IMemory> iMem = mCblkMemory;
audio_track_cblk_t* cblk = mCblk;
bool active = mActive;
mLock.unlock();
// Manage underrun callback
if (active && (cblk->framesAvailable() == cblk->frameCount)) {
ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags);
if (!(android_atomic_or(CBLK_UNDERRUN_ON, &cblk->flags) & CBLK_UNDERRUN_MSK)) {
mCbf(EVENT_UNDERRUN, mUserData, 0);
if (cblk->server == cblk->frameCount) {
mCbf(EVENT_BUFFER_END, mUserData, 0);
}
if (mSharedBuffer != 0) return false;
}
}
// Manage loop end callback
while (mLoopCount > cblk->loopCount) {
int loopCount = -1;
mLoopCount--;
if (mLoopCount >= 0) loopCount = mLoopCount;
mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount);
}
// Manage marker callback
if (!mMarkerReached && (mMarkerPosition > 0)) {
if (cblk->server >= mMarkerPosition) {
mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition);
mMarkerReached = true;
}
}
// Manage new position callback
if (mUpdatePeriod > 0) {
while (cblk->server >= mNewPosition) {
mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition);
mNewPosition += mUpdatePeriod;
}
}
// If Shared buffer is used, no data is requested from client.
if (mSharedBuffer != 0) {
frames = 0;
} else {
frames = mRemainingFrames;
}
// See description of waitCount parameter at declaration of obtainBuffer().
// The logic below prevents us from being stuck below at obtainBuffer()
// not being able to handle timed events (position, markers, loops).
int32_t waitCount = -1;
if (mUpdatePeriod || (!mMarkerReached && mMarkerPosition) || mLoopCount) {
waitCount = 1;
}
do {
audioBuffer.frameCount = frames;
status_t err = obtainBuffer(&audioBuffer, waitCount);
if (err < NO_ERROR) {
if (err != TIMED_OUT) {
ALOGE_IF(err != status_t(NO_MORE_BUFFERS), "Error obtaining an audio buffer, giving up.");
return false;
}
break;
}
if (err == status_t(STOPPED)) return false;
// Divide buffer size by 2 to take into account the expansion
// due to 8 to 16 bit conversion: the callback must fill only half
// of the destination buffer
if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
audioBuffer.size >>= 1;
}
size_t reqSize = audioBuffer.size;
mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
writtenSize = audioBuffer.size;
// Sanity check on returned size
if (ssize_t(writtenSize) <= 0) {
// The callback is done filling buffers
// Keep this thread going to handle timed events and
// still try to get more data in intervals of WAIT_PERIOD_MS
// but don't just loop and block the CPU, so wait
usleep(WAIT_PERIOD_MS*1000);
break;
}
if (writtenSize > reqSize) writtenSize = reqSize;
if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
// 8 to 16 bit conversion, note that source and destination are the same address
memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize);
writtenSize <<= 1;
}
audioBuffer.size = writtenSize;
// NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for
// 8 bit PCM data: in this case, mCblk->frameSize is based on a sample size of
// 16 bit.
audioBuffer.frameCount = writtenSize/mCblk->frameSize;
frames -= audioBuffer.frameCount;
releaseBuffer(&audioBuffer);
}
while (frames);
if (frames == 0) {
mRemainingFrames = mNotificationFramesAct;
} else {
mRemainingFrames = frames;
}
return true;
}
// must be called with mLock and cblk.lock held. Callers must also hold strong references on
// the IAudioTrack and IMemory in case they are recreated here.
// If the IAudioTrack is successfully restored, the cblk pointer is updated
status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart)
{
status_t result;
if (!(android_atomic_or(CBLK_RESTORING_ON, &cblk->flags) & CBLK_RESTORING_MSK)) {
ALOGW("dead IAudioTrack, creating a new one from %s TID %d",
fromStart ? "start()" : "obtainBuffer()", gettid());
// signal old cblk condition so that other threads waiting for available buffers stop
// waiting now
cblk->cv.broadcast();
cblk->lock.unlock();
// refresh the audio configuration cache in this process to make sure we get new
// output parameters in getOutput_l() and createTrack_l()
AudioSystem::clearAudioConfigCache();
// if the new IAudioTrack is created, createTrack_l() will modify the
// following member variables: mAudioTrack, mCblkMemory and mCblk.
// It will also delete the strong references on previous IAudioTrack and IMemory
result = createTrack_l(mStreamType,
cblk->sampleRate,
mFormat,
mChannelMask,
mFrameCount,
mFlags,
mSharedBuffer,
getOutput_l());
if (result == NO_ERROR) {
uint32_t user = cblk->user;
uint32_t server = cblk->server;
// restore write index and set other indexes to reflect empty buffer status
mCblk->user = user;
mCblk->server = user;
mCblk->userBase = user;
mCblk->serverBase = user;
// restore loop: this is not guaranteed to succeed if new frame count is not
// compatible with loop length
setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount);
if (!fromStart) {
mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
// Make sure that a client relying on callback events indicating underrun or
// the actual amount of audio frames played (e.g SoundPool) receives them.
if (mSharedBuffer == 0) {
uint32_t frames = 0;
if (user > server) {
frames = ((user - server) > mCblk->frameCount) ?
mCblk->frameCount : (user - server);
memset(mCblk->buffers, 0, frames * mCblk->frameSize);
}
// restart playback even if buffer is not completely filled.
android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags);
// stepUser() clears CBLK_UNDERRUN_ON flag enabling underrun callbacks to
// the client
mCblk->stepUser(frames);
}
}
if (mSharedBuffer != 0) {
mCblk->stepUser(mCblk->frameCount);
}
if (mActive) {
result = mAudioTrack->start();
ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result);
}
if (fromStart && result == NO_ERROR) {
mNewPosition = mCblk->server + mUpdatePeriod;
}
}
if (result != NO_ERROR) {
android_atomic_and(~CBLK_RESTORING_ON, &cblk->flags);
ALOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result);
}
mRestoreStatus = result;
// signal old cblk condition for other threads waiting for restore completion
android_atomic_or(CBLK_RESTORED_ON, &cblk->flags);
cblk->cv.broadcast();
} else {
if (!(cblk->flags & CBLK_RESTORED_MSK)) {
ALOGW("dead IAudioTrack, waiting for a new one TID %d", gettid());
mLock.unlock();
result = cblk->cv.waitRelative(cblk->lock, milliseconds(RESTORE_TIMEOUT_MS));
if (result == NO_ERROR) {
result = mRestoreStatus;
}
cblk->lock.unlock();
mLock.lock();
} else {
ALOGW("dead IAudioTrack, already restored TID %d", gettid());
result = mRestoreStatus;
cblk->lock.unlock();
}
}
ALOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x",
result, mActive, mCblk, cblk, mCblk->flags, cblk->flags);
if (result == NO_ERROR) {
// from now on we switch to the newly created cblk
cblk = mCblk;
}
cblk->lock.lock();
ALOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d TID %d", result, gettid());
return result;
}
status_t AudioTrack::dump(int fd, const Vector<String16>& args) const
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
result.append(" AudioTrack::dump\n");
snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]);
result.append(buffer);
snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, (mCblk == 0) ? 0 : mCblk->frameCount);
result.append(buffer);
snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d)\n", (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted);
result.append(buffer);
snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency);
result.append(buffer);
::write(fd, result.string(), result.size());
return NO_ERROR;
}
// =========================================================================
AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava)
: Thread(bCanCallJava), mReceiver(receiver), mPaused(true)
{
}
AudioTrack::AudioTrackThread::~AudioTrackThread()
{
}
bool AudioTrack::AudioTrackThread::threadLoop()
{
{
AutoMutex _l(mMyLock);
if (mPaused) {
mMyCond.wait(mMyLock);
// caller will check for exitPending()
return true;
}
}
if (!mReceiver.processAudioBuffer(this)) {
pause();
}
return true;
}
void AudioTrack::AudioTrackThread::requestExit()
{
// must be in this order to avoid a race condition
Thread::requestExit();
resume();
}
void AudioTrack::AudioTrackThread::pause()
{
AutoMutex _l(mMyLock);
mPaused = true;
}
void AudioTrack::AudioTrackThread::resume()
{
AutoMutex _l(mMyLock);
if (mPaused) {
mPaused = false;
mMyCond.signal();
}
}
// =========================================================================
audio_track_cblk_t::audio_track_cblk_t()
: lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0),
userBase(0), serverBase(0), buffers(NULL), frameCount(0),
loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), mVolumeLR(0x10001000),
mSendLevel(0), flags(0)
{
}
uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount)
{
ALOGV("stepuser %08x %08x %d", user, server, frameCount);
uint32_t u = user;
u += frameCount;
// Ensure that user is never ahead of server for AudioRecord
if (flags & CBLK_DIRECTION_MSK) {
// If stepServer() has been called once, switch to normal obtainBuffer() timeout period
if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) {
bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
}
} else if (u > server) {
ALOGW("stepUser occurred after track reset");
u = server;
}
uint32_t fc = this->frameCount;
if (u >= fc) {
// common case, user didn't just wrap
if (u - fc >= userBase ) {
userBase += fc;
}
} else if (u >= userBase + fc) {
// user just wrapped
userBase += fc;
}
user = u;
// Clear flow control error condition as new data has been written/read to/from buffer.
if (flags & CBLK_UNDERRUN_MSK) {
android_atomic_and(~CBLK_UNDERRUN_MSK, &flags);
}
return u;
}
bool audio_track_cblk_t::stepServer(uint32_t frameCount)
{
ALOGV("stepserver %08x %08x %d", user, server, frameCount);
if (!tryLock()) {
ALOGW("stepServer() could not lock cblk");
return false;
}
uint32_t s = server;
bool flushed = (s == user);
s += frameCount;
if (flags & CBLK_DIRECTION_MSK) {
// Mark that we have read the first buffer so that next time stepUser() is called
// we switch to normal obtainBuffer() timeout period
if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) {
bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1;
}
// It is possible that we receive a flush()
// while the mixer is processing a block: in this case,
// stepServer() is called After the flush() has reset u & s and
// we have s > u
if (flushed) {
ALOGW("stepServer occurred after track reset");
s = user;
}
}
if (s >= loopEnd) {
ALOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd);
s = loopStart;
if (--loopCount == 0) {
loopEnd = UINT_MAX;
loopStart = UINT_MAX;
}
}
uint32_t fc = this->frameCount;
if (s >= fc) {
// common case, server didn't just wrap
if (s - fc >= serverBase ) {
serverBase += fc;
}
} else if (s >= serverBase + fc) {
// server just wrapped
serverBase += fc;
}
server = s;
if (!(flags & CBLK_INVALID_MSK)) {
cv.signal();
}
lock.unlock();
return true;
}
void* audio_track_cblk_t::buffer(uint32_t offset) const
{
return (int8_t *)buffers + (offset - userBase) * frameSize;
}
uint32_t audio_track_cblk_t::framesAvailable()
{
Mutex::Autolock _l(lock);
return framesAvailable_l();
}
uint32_t audio_track_cblk_t::framesAvailable_l()
{
uint32_t u = user;
uint32_t s = server;
if (flags & CBLK_DIRECTION_MSK) {
uint32_t limit = (s < loopStart) ? s : loopStart;
return limit + frameCount - u;
} else {
return frameCount + u - s;
}
}
uint32_t audio_track_cblk_t::framesReady()
{
uint32_t u = user;
uint32_t s = server;
if (flags & CBLK_DIRECTION_MSK) {
if (u < loopEnd) {
return u - s;
} else {
// do not block on mutex shared with client on AudioFlinger side
if (!tryLock()) {
ALOGW("framesReady() could not lock cblk");
return 0;
}
uint32_t frames = UINT_MAX;
if (loopCount >= 0) {
frames = (loopEnd - loopStart)*loopCount + u - s;
}
lock.unlock();
return frames;
}
} else {
return s - u;
}
}
bool audio_track_cblk_t::tryLock()
{
// the code below simulates lock-with-timeout
// we MUST do this to protect the AudioFlinger server
// as this lock is shared with the client.
status_t err;
err = lock.tryLock();
if (err == -EBUSY) { // just wait a bit
usleep(1000);
err = lock.tryLock();
}
if (err != NO_ERROR) {
// probably, the client just died.
return false;
}
return true;
}
// -------------------------------------------------------------------------
}; // namespace android