/*
**
** 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 <inttypes.h>
#include <math.h>
#include <sys/resource.h>
#include <audio_utils/clock.h>
#include <audio_utils/primitives.h>
#include <binder/IPCThreadState.h>
#include <media/AudioTrack.h>
#include <utils/Log.h>
#include <private/media/AudioTrackShared.h>
#include <media/IAudioFlinger.h>
#include <media/AudioParameter.h>
#include <media/AudioPolicyHelper.h>
#include <media/AudioResamplerPublic.h>
#include <media/MediaAnalyticsItem.h>
#include <media/TypeConverter.h>
#define WAIT_PERIOD_MS 10
#define WAIT_STREAM_END_TIMEOUT_SEC 120
static const int kMaxLoopCountNotifications = 32;
namespace android {
// ---------------------------------------------------------------------------
using media::VolumeShaper;
// TODO: Move to a separate .h
template <typename T>
static inline const T &min(const T &x, const T &y) {
return x < y ? x : y;
}
template <typename T>
static inline const T &max(const T &x, const T &y) {
return x > y ? x : y;
}
static inline nsecs_t framesToNanoseconds(ssize_t frames, uint32_t sampleRate, float speed)
{
return ((double)frames * 1000000000) / ((double)sampleRate * speed);
}
static int64_t convertTimespecToUs(const struct timespec &tv)
{
return tv.tv_sec * 1000000ll + tv.tv_nsec / 1000;
}
// TODO move to audio_utils.
static inline struct timespec convertNsToTimespec(int64_t ns) {
struct timespec tv;
tv.tv_sec = static_cast<time_t>(ns / NANOS_PER_SECOND);
tv.tv_nsec = static_cast<long>(ns % NANOS_PER_SECOND);
return tv;
}
// current monotonic time in microseconds.
static int64_t getNowUs()
{
struct timespec tv;
(void) clock_gettime(CLOCK_MONOTONIC, &tv);
return convertTimespecToUs(tv);
}
// FIXME: we don't use the pitch setting in the time stretcher (not working);
// instead we emulate it using our sample rate converter.
static const bool kFixPitch = true; // enable pitch fix
static inline uint32_t adjustSampleRate(uint32_t sampleRate, float pitch)
{
return kFixPitch ? (sampleRate * pitch + 0.5) : sampleRate;
}
static inline float adjustSpeed(float speed, float pitch)
{
return kFixPitch ? speed / max(pitch, AUDIO_TIMESTRETCH_PITCH_MIN_DELTA) : speed;
}
static inline float adjustPitch(float pitch)
{
return kFixPitch ? AUDIO_TIMESTRETCH_PITCH_NORMAL : pitch;
}
// static
status_t AudioTrack::getMinFrameCount(
size_t* frameCount,
audio_stream_type_t streamType,
uint32_t sampleRate)
{
if (frameCount == NULL) {
return BAD_VALUE;
}
// FIXME handle in server, like createTrack_l(), possible missing info:
// audio_io_handle_t output
// audio_format_t format
// audio_channel_mask_t channelMask
// audio_output_flags_t flags (FAST)
uint32_t afSampleRate;
status_t status;
status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType);
if (status != NO_ERROR) {
ALOGE("Unable to query output sample rate for stream type %d; status %d",
streamType, status);
return status;
}
size_t afFrameCount;
status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType);
if (status != NO_ERROR) {
ALOGE("Unable to query output frame count for stream type %d; status %d",
streamType, status);
return status;
}
uint32_t afLatency;
status = AudioSystem::getOutputLatency(&afLatency, streamType);
if (status != NO_ERROR) {
ALOGE("Unable to query output latency for stream type %d; status %d",
streamType, status);
return status;
}
// 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 = 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.
// Return error in the unlikely event that it does not, as that's part of the API contract.
if (*frameCount == 0) {
ALOGE("AudioTrack::getMinFrameCount failed for streamType %d, sampleRate %u",
streamType, sampleRate);
return BAD_VALUE;
}
ALOGV("getMinFrameCount=%zu: afFrameCount=%zu, afSampleRate=%u, afLatency=%u",
*frameCount, afFrameCount, afSampleRate, afLatency);
return NO_ERROR;
}
// ---------------------------------------------------------------------------
static std::string audioContentTypeString(audio_content_type_t value) {
std::string contentType;
if (AudioContentTypeConverter::toString(value, contentType)) {
return contentType;
}
char rawbuffer[16]; // room for "%d"
snprintf(rawbuffer, sizeof(rawbuffer), "%d", value);
return rawbuffer;
}
static std::string audioUsageString(audio_usage_t value) {
std::string usage;
if (UsageTypeConverter::toString(value, usage)) {
return usage;
}
char rawbuffer[16]; // room for "%d"
snprintf(rawbuffer, sizeof(rawbuffer), "%d", value);
return rawbuffer;
}
void AudioTrack::MediaMetrics::gather(const AudioTrack *track)
{
// key for media statistics is defined in the header
// attrs for media statistics
// NB: these are matched with public Java API constants defined
// in frameworks/base/media/java/android/media/AudioTrack.java
// These must be kept synchronized with the constants there.
static constexpr char kAudioTrackStreamType[] = "android.media.audiotrack.streamtype";
static constexpr char kAudioTrackContentType[] = "android.media.audiotrack.type";
static constexpr char kAudioTrackUsage[] = "android.media.audiotrack.usage";
static constexpr char kAudioTrackSampleRate[] = "android.media.audiotrack.samplerate";
static constexpr char kAudioTrackChannelMask[] = "android.media.audiotrack.channelmask";
// NB: These are not yet exposed as public Java API constants.
static constexpr char kAudioTrackUnderrunFrames[] = "android.media.audiotrack.underrunframes";
static constexpr char kAudioTrackStartupGlitch[] = "android.media.audiotrack.glitch.startup";
// only if we're in a good state...
// XXX: shall we gather alternative info if failing?
const status_t lstatus = track->initCheck();
if (lstatus != NO_ERROR) {
ALOGD("no metrics gathered, track status=%d", (int) lstatus);
return;
}
// constructor guarantees mAnalyticsItem is valid
const int32_t underrunFrames = track->getUnderrunFrames();
if (underrunFrames != 0) {
mAnalyticsItem->setInt32(kAudioTrackUnderrunFrames, underrunFrames);
}
if (track->mTimestampStartupGlitchReported) {
mAnalyticsItem->setInt32(kAudioTrackStartupGlitch, 1);
}
if (track->mStreamType != -1) {
// deprecated, but this will tell us who still uses it.
mAnalyticsItem->setInt32(kAudioTrackStreamType, track->mStreamType);
}
// XXX: consider including from mAttributes: source type
mAnalyticsItem->setCString(kAudioTrackContentType,
audioContentTypeString(track->mAttributes.content_type).c_str());
mAnalyticsItem->setCString(kAudioTrackUsage,
audioUsageString(track->mAttributes.usage).c_str());
mAnalyticsItem->setInt32(kAudioTrackSampleRate, track->mSampleRate);
mAnalyticsItem->setInt64(kAudioTrackChannelMask, track->mChannelMask);
}
// hand the user a snapshot of the metrics.
status_t AudioTrack::getMetrics(MediaAnalyticsItem * &item)
{
mMediaMetrics.gather(this);
MediaAnalyticsItem *tmp = mMediaMetrics.dup();
if (tmp == nullptr) {
return BAD_VALUE;
}
item = tmp;
return NO_ERROR;
}
AudioTrack::AudioTrack()
: mStatus(NO_INIT),
mState(STATE_STOPPED),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT),
mPausedPosition(0),
mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
mRoutedDeviceId(AUDIO_PORT_HANDLE_NONE)
{
mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN;
mAttributes.usage = AUDIO_USAGE_UNKNOWN;
mAttributes.flags = 0x0;
strcpy(mAttributes.tags, "");
}
AudioTrack::AudioTrack(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
audio_output_flags_t flags,
callback_t cbf,
void* user,
int32_t notificationFrames,
audio_session_t sessionId,
transfer_type transferType,
const audio_offload_info_t *offloadInfo,
uid_t uid,
pid_t pid,
const audio_attributes_t* pAttributes,
bool doNotReconnect,
float maxRequiredSpeed,
audio_port_handle_t selectedDeviceId)
: mStatus(NO_INIT),
mState(STATE_STOPPED),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT),
mPausedPosition(0)
{
(void)set(streamType, sampleRate, format, channelMask,
frameCount, flags, cbf, user, notificationFrames,
0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType,
offloadInfo, uid, pid, pAttributes, doNotReconnect, maxRequiredSpeed, selectedDeviceId);
}
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,
int32_t notificationFrames,
audio_session_t sessionId,
transfer_type transferType,
const audio_offload_info_t *offloadInfo,
uid_t uid,
pid_t pid,
const audio_attributes_t* pAttributes,
bool doNotReconnect,
float maxRequiredSpeed)
: mStatus(NO_INIT),
mState(STATE_STOPPED),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT),
mPausedPosition(0),
mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE)
{
(void)set(streamType, sampleRate, format, channelMask,
0 /*frameCount*/, flags, cbf, user, notificationFrames,
sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo,
uid, pid, pAttributes, doNotReconnect, maxRequiredSpeed);
}
AudioTrack::~AudioTrack()
{
// pull together the numbers, before we clean up our structures
mMediaMetrics.gather(this);
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) {
mProxy->interrupt();
mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
mAudioTrackThread->requestExitAndWait();
mAudioTrackThread.clear();
}
// No lock here: worst case we remove a NULL callback which will be a nop
if (mDeviceCallback != 0 && mOutput != AUDIO_IO_HANDLE_NONE) {
AudioSystem::removeAudioDeviceCallback(this, mOutput);
}
IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
mAudioTrack.clear();
mCblkMemory.clear();
mSharedBuffer.clear();
IPCThreadState::self()->flushCommands();
ALOGV("~AudioTrack, releasing session id %d from %d on behalf of %d",
mSessionId, IPCThreadState::self()->getCallingPid(), mClientPid);
AudioSystem::releaseAudioSessionId(mSessionId, mClientPid);
}
}
status_t AudioTrack::set(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
audio_output_flags_t flags,
callback_t cbf,
void* user,
int32_t notificationFrames,
const sp<IMemory>& sharedBuffer,
bool threadCanCallJava,
audio_session_t sessionId,
transfer_type transferType,
const audio_offload_info_t *offloadInfo,
uid_t uid,
pid_t pid,
const audio_attributes_t* pAttributes,
bool doNotReconnect,
float maxRequiredSpeed,
audio_port_handle_t selectedDeviceId)
{
status_t status;
uint32_t channelCount;
pid_t callingPid;
pid_t myPid;
ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
"flags #%x, notificationFrames %d, sessionId %d, transferType %d, uid %d, pid %d",
streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames,
sessionId, transferType, uid, pid);
mThreadCanCallJava = threadCanCallJava;
mSelectedDeviceId = selectedDeviceId;
mSessionId = sessionId;
switch (transferType) {
case TRANSFER_DEFAULT:
if (sharedBuffer != 0) {
transferType = TRANSFER_SHARED;
} else if (cbf == NULL || threadCanCallJava) {
transferType = TRANSFER_SYNC;
} else {
transferType = TRANSFER_CALLBACK;
}
break;
case TRANSFER_CALLBACK:
if (cbf == NULL || sharedBuffer != 0) {
ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0");
status = BAD_VALUE;
goto exit;
}
break;
case TRANSFER_OBTAIN:
case TRANSFER_SYNC:
if (sharedBuffer != 0) {
ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0");
status = BAD_VALUE;
goto exit;
}
break;
case TRANSFER_SHARED:
if (sharedBuffer == 0) {
ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0");
status = BAD_VALUE;
goto exit;
}
break;
default:
ALOGE("Invalid transfer type %d", transferType);
status = BAD_VALUE;
goto exit;
}
mSharedBuffer = sharedBuffer;
mTransfer = transferType;
mDoNotReconnect = doNotReconnect;
ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %zu", sharedBuffer->pointer(),
sharedBuffer->size());
ALOGV("set() streamType %d frameCount %zu flags %04x", streamType, frameCount, flags);
// invariant that mAudioTrack != 0 is true only after set() returns successfully
if (mAudioTrack != 0) {
ALOGE("Track already in use");
status = INVALID_OPERATION;
goto exit;
}
// handle default values first.
if (streamType == AUDIO_STREAM_DEFAULT) {
streamType = AUDIO_STREAM_MUSIC;
}
if (pAttributes == NULL) {
if (uint32_t(streamType) >= AUDIO_STREAM_PUBLIC_CNT) {
ALOGE("Invalid stream type %d", streamType);
status = BAD_VALUE;
goto exit;
}
mStreamType = streamType;
} else {
// stream type shouldn't be looked at, this track has audio attributes
memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t));
ALOGV("Building AudioTrack with attributes: usage=%d content=%d flags=0x%x tags=[%s]",
mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags);
mStreamType = AUDIO_STREAM_DEFAULT;
if ((mAttributes.flags & AUDIO_FLAG_HW_AV_SYNC) != 0) {
flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_HW_AV_SYNC);
}
if ((mAttributes.flags & AUDIO_FLAG_LOW_LATENCY) != 0) {
flags = (audio_output_flags_t) (flags | AUDIO_OUTPUT_FLAG_FAST);
}
// check deep buffer after flags have been modified above
if (flags == AUDIO_OUTPUT_FLAG_NONE && (mAttributes.flags & AUDIO_FLAG_DEEP_BUFFER) != 0) {
flags = AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
}
}
// these below should probably come from the audioFlinger too...
if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
} else if (format == AUDIO_FORMAT_IEC61937) { // HDMI pass-through?
mAttributes.flags |= AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO;
}
// validate parameters
if (!audio_is_valid_format(format)) {
ALOGE("Invalid format %#x", format);
status = BAD_VALUE;
goto exit;
}
mFormat = format;
if (!audio_is_output_channel(channelMask)) {
ALOGE("Invalid channel mask %#x", channelMask);
status = BAD_VALUE;
goto exit;
}
mChannelMask = channelMask;
channelCount = audio_channel_count_from_out_mask(channelMask);
mChannelCount = channelCount;
// force direct flag if format is not linear PCM
// or offload was requested
if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
|| !audio_is_linear_pcm(format)) {
ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
? "Offload request, forcing to Direct Output"
: "Not linear PCM, forcing to Direct Output");
flags = (audio_output_flags_t)
// FIXME why can't we allow direct AND fast?
((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST);
}
// force direct flag if HW A/V sync requested
if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
}
if (flags & AUDIO_OUTPUT_FLAG_DIRECT) {
if (audio_has_proportional_frames(format)) {
mFrameSize = channelCount * audio_bytes_per_sample(format);
} else {
mFrameSize = sizeof(uint8_t);
}
} else {
ALOG_ASSERT(audio_has_proportional_frames(format));
mFrameSize = channelCount * audio_bytes_per_sample(format);
// createTrack will return an error if PCM format is not supported by server,
// so no need to check for specific PCM formats here
}
// sampling rate must be specified for direct outputs
if (sampleRate == 0 && (flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
status = BAD_VALUE;
goto exit;
}
mSampleRate = sampleRate;
mOriginalSampleRate = sampleRate;
mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
// 1.0 <= mMaxRequiredSpeed <= AUDIO_TIMESTRETCH_SPEED_MAX
mMaxRequiredSpeed = min(max(maxRequiredSpeed, 1.0f), AUDIO_TIMESTRETCH_SPEED_MAX);
// Make copy of input parameter offloadInfo so that in the future:
// (a) createTrack_l doesn't need it as an input parameter
// (b) we can support re-creation of offloaded tracks
if (offloadInfo != NULL) {
mOffloadInfoCopy = *offloadInfo;
mOffloadInfo = &mOffloadInfoCopy;
} else {
mOffloadInfo = NULL;
memset(&mOffloadInfoCopy, 0, sizeof(audio_offload_info_t));
}
mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f;
mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f;
mSendLevel = 0.0f;
// mFrameCount is initialized in createTrack_l
mReqFrameCount = frameCount;
if (notificationFrames >= 0) {
mNotificationFramesReq = notificationFrames;
mNotificationsPerBufferReq = 0;
} else {
if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
ALOGE("notificationFrames=%d not permitted for non-fast track",
notificationFrames);
status = BAD_VALUE;
goto exit;
}
if (frameCount > 0) {
ALOGE("notificationFrames=%d not permitted with non-zero frameCount=%zu",
notificationFrames, frameCount);
status = BAD_VALUE;
goto exit;
}
mNotificationFramesReq = 0;
const uint32_t minNotificationsPerBuffer = 1;
const uint32_t maxNotificationsPerBuffer = 8;
mNotificationsPerBufferReq = min(maxNotificationsPerBuffer,
max((uint32_t) -notificationFrames, minNotificationsPerBuffer));
ALOGW_IF(mNotificationsPerBufferReq != (uint32_t) -notificationFrames,
"notificationFrames=%d clamped to the range -%u to -%u",
notificationFrames, minNotificationsPerBuffer, maxNotificationsPerBuffer);
}
mNotificationFramesAct = 0;
callingPid = IPCThreadState::self()->getCallingPid();
myPid = getpid();
if (uid == AUDIO_UID_INVALID || (callingPid != myPid)) {
mClientUid = IPCThreadState::self()->getCallingUid();
} else {
mClientUid = uid;
}
if (pid == -1 || (callingPid != myPid)) {
mClientPid = callingPid;
} else {
mClientPid = pid;
}
mAuxEffectId = 0;
mOrigFlags = mFlags = flags;
mCbf = cbf;
if (cbf != NULL) {
mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
// thread begins in paused state, and will not reference us until start()
}
// create the IAudioTrack
status = createTrack_l();
if (status != NO_ERROR) {
if (mAudioTrackThread != 0) {
mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
mAudioTrackThread->requestExitAndWait();
mAudioTrackThread.clear();
}
goto exit;
}
mUserData = user;
mLoopCount = 0;
mLoopStart = 0;
mLoopEnd = 0;
mLoopCountNotified = 0;
mMarkerPosition = 0;
mMarkerReached = false;
mNewPosition = 0;
mUpdatePeriod = 0;
mPosition = 0;
mReleased = 0;
mStartNs = 0;
mStartFromZeroUs = 0;
AudioSystem::acquireAudioSessionId(mSessionId, mClientPid);
mSequence = 1;
mObservedSequence = mSequence;
mInUnderrun = false;
mPreviousTimestampValid = false;
mTimestampStartupGlitchReported = false;
mRetrogradeMotionReported = false;
mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID;
mStartTs.mPosition = 0;
mUnderrunCountOffset = 0;
mFramesWritten = 0;
mFramesWrittenServerOffset = 0;
mFramesWrittenAtRestore = -1; // -1 is a unique initializer.
mVolumeHandler = new media::VolumeHandler();
exit:
mStatus = status;
return status;
}
// -------------------------------------------------------------------------
status_t AudioTrack::start()
{
AutoMutex lock(mLock);
if (mState == STATE_ACTIVE) {
return INVALID_OPERATION;
}
mInUnderrun = true;
State previousState = mState;
if (previousState == STATE_PAUSED_STOPPING) {
mState = STATE_STOPPING;
} else {
mState = STATE_ACTIVE;
}
(void) updateAndGetPosition_l();
// save start timestamp
if (isOffloadedOrDirect_l()) {
if (getTimestamp_l(mStartTs) != OK) {
mStartTs.mPosition = 0;
}
} else {
if (getTimestamp_l(&mStartEts) != OK) {
mStartEts.clear();
}
}
mStartNs = systemTime(); // save this for timestamp adjustment after starting.
if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) {
// reset current position as seen by client to 0
mPosition = 0;
mPreviousTimestampValid = false;
mTimestampStartupGlitchReported = false;
mRetrogradeMotionReported = false;
mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID;
if (!isOffloadedOrDirect_l()
&& mStartEts.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] > 0) {
// Server side has consumed something, but is it finished consuming?
// It is possible since flush and stop are asynchronous that the server
// is still active at this point.
ALOGV("start: server read:%lld cumulative flushed:%lld client written:%lld",
(long long)(mFramesWrittenServerOffset
+ mStartEts.mPosition[ExtendedTimestamp::LOCATION_SERVER]),
(long long)mStartEts.mFlushed,
(long long)mFramesWritten);
// mStartEts is already adjusted by mFramesWrittenServerOffset, so we delta adjust.
mFramesWrittenServerOffset -= mStartEts.mPosition[ExtendedTimestamp::LOCATION_SERVER];
}
mFramesWritten = 0;
mProxy->clearTimestamp(); // need new server push for valid timestamp
mMarkerReached = false;
// For offloaded tracks, we don't know if the hardware counters are really zero here,
// since the flush is asynchronous and stop may not fully drain.
// We save the time when the track is started to later verify whether
// the counters are realistic (i.e. start from zero after this time).
mStartFromZeroUs = mStartNs / 1000;
// force refresh of remaining frames by processAudioBuffer() as last
// write before stop could be partial.
mRefreshRemaining = true;
}
mNewPosition = mPosition + mUpdatePeriod;
int32_t flags = android_atomic_and(~(CBLK_STREAM_END_DONE | CBLK_DISABLED), &mCblk->mFlags);
status_t status = NO_ERROR;
if (!(flags & CBLK_INVALID)) {
status = mAudioTrack->start();
if (status == DEAD_OBJECT) {
flags |= CBLK_INVALID;
}
}
if (flags & CBLK_INVALID) {
status = restoreTrack_l("start");
}
// resume or pause the callback thread as needed.
sp<AudioTrackThread> t = mAudioTrackThread;
if (status == NO_ERROR) {
if (t != 0) {
if (previousState == STATE_STOPPING) {
mProxy->interrupt();
} else {
t->resume();
}
} else {
mPreviousPriority = getpriority(PRIO_PROCESS, 0);
get_sched_policy(0, &mPreviousSchedulingGroup);
androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
}
// Start our local VolumeHandler for restoration purposes.
mVolumeHandler->setStarted();
} else {
ALOGE("start() status %d", status);
mState = previousState;
if (t != 0) {
if (previousState != STATE_STOPPING) {
t->pause();
}
} else {
setpriority(PRIO_PROCESS, 0, mPreviousPriority);
set_sched_policy(0, mPreviousSchedulingGroup);
}
}
return status;
}
void AudioTrack::stop()
{
AutoMutex lock(mLock);
if (mState != STATE_ACTIVE && mState != STATE_PAUSED) {
return;
}
if (isOffloaded_l()) {
mState = STATE_STOPPING;
} else {
mState = STATE_STOPPED;
ALOGD_IF(mSharedBuffer == nullptr,
"stop() called with %u frames delivered", mReleased.value());
mReleased = 0;
}
mProxy->stop(); // notify server not to read beyond current client position until start().
mProxy->interrupt();
mAudioTrack->stop();
// Note: legacy handling - stop does not clear playback marker
// and periodic update counter, but flush does for streaming tracks.
if (mSharedBuffer != 0) {
// clear buffer position and loop count.
mStaticProxy->setBufferPositionAndLoop(0 /* position */,
0 /* loopStart */, 0 /* loopEnd */, 0 /* loopCount */);
}
sp<AudioTrackThread> t = mAudioTrackThread;
if (t != 0) {
if (!isOffloaded_l()) {
t->pause();
}
} else {
setpriority(PRIO_PROCESS, 0, mPreviousPriority);
set_sched_policy(0, mPreviousSchedulingGroup);
}
}
bool AudioTrack::stopped() const
{
AutoMutex lock(mLock);
return mState != STATE_ACTIVE;
}
void AudioTrack::flush()
{
if (mSharedBuffer != 0) {
return;
}
AutoMutex lock(mLock);
if (mState == STATE_ACTIVE) {
return;
}
flush_l();
}
void AudioTrack::flush_l()
{
ALOG_ASSERT(mState != STATE_ACTIVE);
// clear playback marker and periodic update counter
mMarkerPosition = 0;
mMarkerReached = false;
mUpdatePeriod = 0;
mRefreshRemaining = true;
mState = STATE_FLUSHED;
mReleased = 0;
if (isOffloaded_l()) {
mProxy->interrupt();
}
mProxy->flush();
mAudioTrack->flush();
}
void AudioTrack::pause()
{
AutoMutex lock(mLock);
if (mState == STATE_ACTIVE) {
mState = STATE_PAUSED;
} else if (mState == STATE_STOPPING) {
mState = STATE_PAUSED_STOPPING;
} else {
return;
}
mProxy->interrupt();
mAudioTrack->pause();
if (isOffloaded_l()) {
if (mOutput != AUDIO_IO_HANDLE_NONE) {
// An offload output can be re-used between two audio tracks having
// the same configuration. A timestamp query for a paused track
// while the other is running would return an incorrect time.
// To fix this, cache the playback position on a pause() and return
// this time when requested until the track is resumed.
// OffloadThread sends HAL pause in its threadLoop. Time saved
// here can be slightly off.
// TODO: check return code for getRenderPosition.
uint32_t halFrames;
AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition);
ALOGV("AudioTrack::pause for offload, cache current position %u", mPausedPosition);
}
}
}
status_t AudioTrack::setVolume(float left, float right)
{
// This duplicates a test by AudioTrack JNI, but that is not the only caller
if (isnanf(left) || left < GAIN_FLOAT_ZERO || left > GAIN_FLOAT_UNITY ||
isnanf(right) || right < GAIN_FLOAT_ZERO || right > GAIN_FLOAT_UNITY) {
return BAD_VALUE;
}
AutoMutex lock(mLock);
mVolume[AUDIO_INTERLEAVE_LEFT] = left;
mVolume[AUDIO_INTERLEAVE_RIGHT] = right;
mProxy->setVolumeLR(gain_minifloat_pack(gain_from_float(left), gain_from_float(right)));
if (isOffloaded_l()) {
mAudioTrack->signal();
}
return NO_ERROR;
}
status_t AudioTrack::setVolume(float volume)
{
return setVolume(volume, volume);
}
status_t AudioTrack::setAuxEffectSendLevel(float level)
{
// This duplicates a test by AudioTrack JNI, but that is not the only caller
if (isnanf(level) || level < GAIN_FLOAT_ZERO || level > GAIN_FLOAT_UNITY) {
return BAD_VALUE;
}
AutoMutex lock(mLock);
mSendLevel = level;
mProxy->setSendLevel(level);
return NO_ERROR;
}
void AudioTrack::getAuxEffectSendLevel(float* level) const
{
if (level != NULL) {
*level = mSendLevel;
}
}
status_t AudioTrack::setSampleRate(uint32_t rate)
{
AutoMutex lock(mLock);
if (rate == mSampleRate) {
return NO_ERROR;
}
if (isOffloadedOrDirect_l() || (mFlags & AUDIO_OUTPUT_FLAG_FAST)) {
return INVALID_OPERATION;
}
if (mOutput == AUDIO_IO_HANDLE_NONE) {
return NO_INIT;
}
// NOTE: it is theoretically possible, but highly unlikely, that a device change
// could mean a previously allowed sampling rate is no longer allowed.
uint32_t afSamplingRate;
if (AudioSystem::getSamplingRate(mOutput, &afSamplingRate) != NO_ERROR) {
return NO_INIT;
}
// pitch is emulated by adjusting speed and sampleRate
const uint32_t effectiveSampleRate = adjustSampleRate(rate, mPlaybackRate.mPitch);
if (rate == 0 || effectiveSampleRate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
return BAD_VALUE;
}
// TODO: Should we also check if the buffer size is compatible?
mSampleRate = rate;
mProxy->setSampleRate(effectiveSampleRate);
return NO_ERROR;
}
uint32_t AudioTrack::getSampleRate() const
{
AutoMutex lock(mLock);
// sample rate can be updated during playback by the offloaded decoder so we need to
// query the HAL and update if needed.
// FIXME use Proxy return channel to update the rate from server and avoid polling here
if (isOffloadedOrDirect_l()) {
if (mOutput != AUDIO_IO_HANDLE_NONE) {
uint32_t sampleRate = 0;
status_t status = AudioSystem::getSamplingRate(mOutput, &sampleRate);
if (status == NO_ERROR) {
mSampleRate = sampleRate;
}
}
}
return mSampleRate;
}
uint32_t AudioTrack::getOriginalSampleRate() const
{
return mOriginalSampleRate;
}
status_t AudioTrack::setPlaybackRate(const AudioPlaybackRate &playbackRate)
{
AutoMutex lock(mLock);
if (isAudioPlaybackRateEqual(playbackRate, mPlaybackRate)) {
return NO_ERROR;
}
if (isOffloadedOrDirect_l()) {
return INVALID_OPERATION;
}
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
return INVALID_OPERATION;
}
ALOGV("setPlaybackRate (input): mSampleRate:%u mSpeed:%f mPitch:%f",
mSampleRate, playbackRate.mSpeed, playbackRate.mPitch);
// pitch is emulated by adjusting speed and sampleRate
const uint32_t effectiveRate = adjustSampleRate(mSampleRate, playbackRate.mPitch);
const float effectiveSpeed = adjustSpeed(playbackRate.mSpeed, playbackRate.mPitch);
const float effectivePitch = adjustPitch(playbackRate.mPitch);
AudioPlaybackRate playbackRateTemp = playbackRate;
playbackRateTemp.mSpeed = effectiveSpeed;
playbackRateTemp.mPitch = effectivePitch;
ALOGV("setPlaybackRate (effective): mSampleRate:%u mSpeed:%f mPitch:%f",
effectiveRate, effectiveSpeed, effectivePitch);
if (!isAudioPlaybackRateValid(playbackRateTemp)) {
ALOGW("setPlaybackRate(%f, %f) failed (effective rate out of bounds)",
playbackRate.mSpeed, playbackRate.mPitch);
return BAD_VALUE;
}
// Check if the buffer size is compatible.
if (!isSampleRateSpeedAllowed_l(effectiveRate, effectiveSpeed)) {
ALOGW("setPlaybackRate(%f, %f) failed (buffer size)",
playbackRate.mSpeed, playbackRate.mPitch);
return BAD_VALUE;
}
// Check resampler ratios are within bounds
if ((uint64_t)effectiveRate > (uint64_t)mSampleRate *
(uint64_t)AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
ALOGW("setPlaybackRate(%f, %f) failed. Resample rate exceeds max accepted value",
playbackRate.mSpeed, playbackRate.mPitch);
return BAD_VALUE;
}
if ((uint64_t)effectiveRate * (uint64_t)AUDIO_RESAMPLER_UP_RATIO_MAX < (uint64_t)mSampleRate) {
ALOGW("setPlaybackRate(%f, %f) failed. Resample rate below min accepted value",
playbackRate.mSpeed, playbackRate.mPitch);
return BAD_VALUE;
}
mPlaybackRate = playbackRate;
//set effective rates
mProxy->setPlaybackRate(playbackRateTemp);
mProxy->setSampleRate(effectiveRate); // FIXME: not quite "atomic" with setPlaybackRate
return NO_ERROR;
}
const AudioPlaybackRate& AudioTrack::getPlaybackRate() const
{
AutoMutex lock(mLock);
return mPlaybackRate;
}
ssize_t AudioTrack::getBufferSizeInFrames()
{
AutoMutex lock(mLock);
if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
return NO_INIT;
}
return (ssize_t) mProxy->getBufferSizeInFrames();
}
status_t AudioTrack::getBufferDurationInUs(int64_t *duration)
{
if (duration == nullptr) {
return BAD_VALUE;
}
AutoMutex lock(mLock);
if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
return NO_INIT;
}
ssize_t bufferSizeInFrames = (ssize_t) mProxy->getBufferSizeInFrames();
if (bufferSizeInFrames < 0) {
return (status_t)bufferSizeInFrames;
}
*duration = (int64_t)((double)bufferSizeInFrames * 1000000
/ ((double)mSampleRate * mPlaybackRate.mSpeed));
return NO_ERROR;
}
ssize_t AudioTrack::setBufferSizeInFrames(size_t bufferSizeInFrames)
{
AutoMutex lock(mLock);
if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
return NO_INIT;
}
// Reject if timed track or compressed audio.
if (!audio_is_linear_pcm(mFormat)) {
return INVALID_OPERATION;
}
return (ssize_t) mProxy->setBufferSizeInFrames((uint32_t) bufferSizeInFrames);
}
status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
{
if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
return INVALID_OPERATION;
}
if (loopCount == 0) {
;
} else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount &&
loopEnd - loopStart >= MIN_LOOP) {
;
} else {
return BAD_VALUE;
}
AutoMutex lock(mLock);
// See setPosition() regarding setting parameters such as loop points or position while active
if (mState == STATE_ACTIVE) {
return INVALID_OPERATION;
}
setLoop_l(loopStart, loopEnd, loopCount);
return NO_ERROR;
}
void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
{
// We do not update the periodic notification point.
// mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
mLoopCount = loopCount;
mLoopEnd = loopEnd;
mLoopStart = loopStart;
mLoopCountNotified = loopCount;
mStaticProxy->setLoop(loopStart, loopEnd, loopCount);
// Waking the AudioTrackThread is not needed as this cannot be called when active.
}
status_t AudioTrack::setMarkerPosition(uint32_t marker)
{
// The only purpose of setting marker position is to get a callback
if (mCbf == NULL || isOffloadedOrDirect()) {
return INVALID_OPERATION;
}
AutoMutex lock(mLock);
mMarkerPosition = marker;
mMarkerReached = false;
sp<AudioTrackThread> t = mAudioTrackThread;
if (t != 0) {
t->wake();
}
return NO_ERROR;
}
status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
{
if (isOffloadedOrDirect()) {
return INVALID_OPERATION;
}
if (marker == NULL) {
return BAD_VALUE;
}
AutoMutex lock(mLock);
mMarkerPosition.getValue(marker);
return NO_ERROR;
}
status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
{
// The only purpose of setting position update period is to get a callback
if (mCbf == NULL || isOffloadedOrDirect()) {
return INVALID_OPERATION;
}
AutoMutex lock(mLock);
mNewPosition = updateAndGetPosition_l() + updatePeriod;
mUpdatePeriod = updatePeriod;
sp<AudioTrackThread> t = mAudioTrackThread;
if (t != 0) {
t->wake();
}
return NO_ERROR;
}
status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const
{
if (isOffloadedOrDirect()) {
return INVALID_OPERATION;
}
if (updatePeriod == NULL) {
return BAD_VALUE;
}
AutoMutex lock(mLock);
*updatePeriod = mUpdatePeriod;
return NO_ERROR;
}
status_t AudioTrack::setPosition(uint32_t position)
{
if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
return INVALID_OPERATION;
}
if (position > mFrameCount) {
return BAD_VALUE;
}
AutoMutex lock(mLock);
// Currently we require that the player is inactive before setting parameters such as position
// or loop points. Otherwise, there could be a race condition: the application could read the
// current position, compute a new position or loop parameters, and then set that position or
// loop parameters but it would do the "wrong" thing since the position has continued to advance
// in the mean time. If we ever provide a sequencer in server, we could allow a way for the app
// to specify how it wants to handle such scenarios.
if (mState == STATE_ACTIVE) {
return INVALID_OPERATION;
}
// After setting the position, use full update period before notification.
mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
mStaticProxy->setBufferPosition(position);
// Waking the AudioTrackThread is not needed as this cannot be called when active.
return NO_ERROR;
}
status_t AudioTrack::getPosition(uint32_t *position)
{
if (position == NULL) {
return BAD_VALUE;
}
AutoMutex lock(mLock);
// FIXME: offloaded and direct tracks call into the HAL for render positions
// for compressed/synced data; however, we use proxy position for pure linear pcm data
// as we do not know the capability of the HAL for pcm position support and standby.
// There may be some latency differences between the HAL position and the proxy position.
if (isOffloadedOrDirect_l() && !isPurePcmData_l()) {
uint32_t dspFrames = 0;
if (isOffloaded_l() && ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING))) {
ALOGV("getPosition called in paused state, return cached position %u", mPausedPosition);
*position = mPausedPosition;
return NO_ERROR;
}
if (mOutput != AUDIO_IO_HANDLE_NONE) {
uint32_t halFrames; // actually unused
(void) AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames);
// FIXME: on getRenderPosition() error, we return OK with frame position 0.
}
// FIXME: dspFrames may not be zero in (mState == STATE_STOPPED || mState == STATE_FLUSHED)
// due to hardware latency. We leave this behavior for now.
*position = dspFrames;
} else {
if (mCblk->mFlags & CBLK_INVALID) {
(void) restoreTrack_l("getPosition");
// FIXME: for compatibility with the Java API we ignore the restoreTrack_l()
// error here (e.g. DEAD_OBJECT) and return OK with the last recorded server position.
}
// IAudioTrack::stop() isn't synchronous; we don't know when presentation completes
*position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ?
0 : updateAndGetPosition_l().value();
}
return NO_ERROR;
}
status_t AudioTrack::getBufferPosition(uint32_t *position)
{
if (mSharedBuffer == 0) {
return INVALID_OPERATION;
}
if (position == NULL) {
return BAD_VALUE;
}
AutoMutex lock(mLock);
*position = mStaticProxy->getBufferPosition();
return NO_ERROR;
}
status_t AudioTrack::reload()
{
if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
return INVALID_OPERATION;
}
AutoMutex lock(mLock);
// See setPosition() regarding setting parameters such as loop points or position while active
if (mState == STATE_ACTIVE) {
return INVALID_OPERATION;
}
mNewPosition = mUpdatePeriod;
(void) updateAndGetPosition_l();
mPosition = 0;
mPreviousTimestampValid = false;
#if 0
// The documentation is not clear on the behavior of reload() and the restoration
// of loop count. Historically we have not restored loop count, start, end,
// but it makes sense if one desires to repeat playing a particular sound.
if (mLoopCount != 0) {
mLoopCountNotified = mLoopCount;
mStaticProxy->setLoop(mLoopStart, mLoopEnd, mLoopCount);
}
#endif
mStaticProxy->setBufferPosition(0);
return NO_ERROR;
}
audio_io_handle_t AudioTrack::getOutput() const
{
AutoMutex lock(mLock);
return mOutput;
}
status_t AudioTrack::setOutputDevice(audio_port_handle_t deviceId) {
AutoMutex lock(mLock);
if (mSelectedDeviceId != deviceId) {
mSelectedDeviceId = deviceId;
if (mStatus == NO_ERROR) {
android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
mProxy->interrupt();
}
}
return NO_ERROR;
}
audio_port_handle_t AudioTrack::getOutputDevice() {
AutoMutex lock(mLock);
return mSelectedDeviceId;
}
// must be called with mLock held
void AudioTrack::updateRoutedDeviceId_l()
{
// if the track is inactive, do not update actual device as the output stream maybe routed
// to a device not relevant to this client because of other active use cases.
if (mState != STATE_ACTIVE) {
return;
}
if (mOutput != AUDIO_IO_HANDLE_NONE) {
audio_port_handle_t deviceId = AudioSystem::getDeviceIdForIo(mOutput);
if (deviceId != AUDIO_PORT_HANDLE_NONE) {
mRoutedDeviceId = deviceId;
}
}
}
audio_port_handle_t AudioTrack::getRoutedDeviceId() {
AutoMutex lock(mLock);
updateRoutedDeviceId_l();
return mRoutedDeviceId;
}
status_t AudioTrack::attachAuxEffect(int effectId)
{
AutoMutex lock(mLock);
status_t status = mAudioTrack->attachAuxEffect(effectId);
if (status == NO_ERROR) {
mAuxEffectId = effectId;
}
return status;
}
audio_stream_type_t AudioTrack::streamType() const
{
if (mStreamType == AUDIO_STREAM_DEFAULT) {
return audio_attributes_to_stream_type(&mAttributes);
}
return mStreamType;
}
uint32_t AudioTrack::latency()
{
AutoMutex lock(mLock);
updateLatency_l();
return mLatency;
}
// -------------------------------------------------------------------------
// must be called with mLock held
void AudioTrack::updateLatency_l()
{
status_t status = AudioSystem::getLatency(mOutput, &mAfLatency);
if (status != NO_ERROR) {
ALOGW("getLatency(%d) failed status %d", mOutput, status);
} else {
// FIXME don't believe this lie
mLatency = mAfLatency + (1000LL * mFrameCount) / mSampleRate;
}
}
// TODO Move this macro to a common header file for enum to string conversion in audio framework.
#define MEDIA_CASE_ENUM(name) case name: return #name
const char * AudioTrack::convertTransferToText(transfer_type transferType) {
switch (transferType) {
MEDIA_CASE_ENUM(TRANSFER_DEFAULT);
MEDIA_CASE_ENUM(TRANSFER_CALLBACK);
MEDIA_CASE_ENUM(TRANSFER_OBTAIN);
MEDIA_CASE_ENUM(TRANSFER_SYNC);
MEDIA_CASE_ENUM(TRANSFER_SHARED);
default:
return "UNRECOGNIZED";
}
}
status_t AudioTrack::createTrack_l()
{
status_t status;
bool callbackAdded = false;
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
if (audioFlinger == 0) {
ALOGE("Could not get audioflinger");
status = NO_INIT;
goto exit;
}
{
// 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.
// Client can only express a preference for FAST. Server will perform additional tests.
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
// either of these use cases:
// use case 1: shared buffer
bool sharedBuffer = mSharedBuffer != 0;
bool transferAllowed =
// use case 2: callback transfer mode
(mTransfer == TRANSFER_CALLBACK) ||
// use case 3: obtain/release mode
(mTransfer == TRANSFER_OBTAIN) ||
// use case 4: synchronous write
((mTransfer == TRANSFER_SYNC) && mThreadCanCallJava);
bool fastAllowed = sharedBuffer || transferAllowed;
if (!fastAllowed) {
ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client, not shared buffer and transfer = %s",
convertTransferToText(mTransfer));
mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
}
}
IAudioFlinger::CreateTrackInput input;
if (mStreamType != AUDIO_STREAM_DEFAULT) {
stream_type_to_audio_attributes(mStreamType, &input.attr);
} else {
input.attr = mAttributes;
}
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) {
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;
IAudioFlinger::CreateTrackOutput output;
sp<IAudioTrack> track = audioFlinger->createTrack(input,
output,
&status);
if (status != NO_ERROR || output.outputId == AUDIO_IO_HANDLE_NONE) {
ALOGE("AudioFlinger could not create track, status: %d output %d", status, output.outputId);
if (status == NO_ERROR) {
status = NO_INIT;
}
goto exit;
}
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.
// FIXME compare to AudioRecord
sp<IMemory> iMem = track->getCblk();
if (iMem == 0) {
ALOGE("Could not get control block");
status = NO_INIT;
goto exit;
}
void *iMemPointer = iMem->pointer();
if (iMemPointer == NULL) {
ALOGE("Could not get control block pointer");
status = NO_INIT;
goto exit;
}
// invariant that mAudioTrack != 0 is true only after set() returns successfully
if (mAudioTrack != 0) {
IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
mDeathNotifier.clear();
}
mAudioTrack = track;
mCblkMemory = iMem;
IPCThreadState::self()->flushCommands();
audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
mCblk = cblk;
mAwaitBoost = false;
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
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", mReqFrameCount,
mFrameCount);
}
}
mFlags = output.flags;
//mOutput != output includes the case where mOutput == AUDIO_IO_HANDLE_NONE for first creation
if (mDeviceCallback != 0 && mOutput != output.outputId) {
if (mOutput != AUDIO_IO_HANDLE_NONE) {
AudioSystem::removeAudioDeviceCallback(this, mOutput);
}
AudioSystem::addAudioDeviceCallback(this, output.outputId);
callbackAdded = true;
}
// We retain a copy of the I/O handle, but don't own the reference
mOutput = output.outputId;
mRefreshRemaining = true;
// Starting address of buffers in shared memory. If there is a shared buffer, buffers
// is the value of pointer() for the shared buffer, otherwise buffers points
// immediately after the control block. This address is for the mapping within client
// address space. AudioFlinger::TrackBase::mBuffer is for the server address space.
void* buffers;
if (mSharedBuffer == 0) {
buffers = cblk + 1;
} else {
buffers = mSharedBuffer->pointer();
if (buffers == NULL) {
ALOGE("Could not get buffer pointer");
status = NO_INIT;
goto exit;
}
}
mAudioTrack->attachAuxEffect(mAuxEffectId);
// If IAudioTrack is re-created, don't let the requested frameCount
// decrease. This can confuse clients that cache frameCount().
if (mFrameCount > mReqFrameCount) {
mReqFrameCount = mFrameCount;
}
// reset server position to 0 as we have new cblk.
mServer = 0;
// update proxy
if (mSharedBuffer == 0) {
mStaticProxy.clear();
mProxy = new AudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
} else {
mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
mProxy = mStaticProxy;
}
mProxy->setVolumeLR(gain_minifloat_pack(
gain_from_float(mVolume[AUDIO_INTERLEAVE_LEFT]),
gain_from_float(mVolume[AUDIO_INTERLEAVE_RIGHT])));
mProxy->setSendLevel(mSendLevel);
const uint32_t effectiveSampleRate = adjustSampleRate(mSampleRate, mPlaybackRate.mPitch);
const float effectiveSpeed = adjustSpeed(mPlaybackRate.mSpeed, mPlaybackRate.mPitch);
const float effectivePitch = adjustPitch(mPlaybackRate.mPitch);
mProxy->setSampleRate(effectiveSampleRate);
AudioPlaybackRate playbackRateTemp = mPlaybackRate;
playbackRateTemp.mSpeed = effectiveSpeed;
playbackRateTemp.mPitch = effectivePitch;
mProxy->setPlaybackRate(playbackRateTemp);
mProxy->setMinimum(mNotificationFramesAct);
mDeathNotifier = new DeathNotifier(this);
IInterface::asBinder(mAudioTrack)->linkToDeath(mDeathNotifier, this);
}
exit:
if (status != NO_ERROR && callbackAdded) {
// note: mOutput is always valid is callbackAdded is true
AudioSystem::removeAudioDeviceCallback(this, mOutput);
}
mStatus = status;
// sp<IAudioTrack> track destructor will cause releaseOutput() to be called by AudioFlinger
return status;
}
status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_t *nonContig)
{
if (audioBuffer == NULL) {
if (nonContig != NULL) {
*nonContig = 0;
}
return BAD_VALUE;
}
if (mTransfer != TRANSFER_OBTAIN) {
audioBuffer->frameCount = 0;
audioBuffer->size = 0;
audioBuffer->raw = NULL;
if (nonContig != NULL) {
*nonContig = 0;
}
return INVALID_OPERATION;
}
const struct timespec *requested;
struct timespec timeout;
if (waitCount == -1) {
requested = &ClientProxy::kForever;
} else if (waitCount == 0) {
requested = &ClientProxy::kNonBlocking;
} else if (waitCount > 0) {
long long ms = WAIT_PERIOD_MS * (long long) waitCount;
timeout.tv_sec = ms / 1000;
timeout.tv_nsec = (int) (ms % 1000) * 1000000;
requested = &timeout;
} else {
ALOGE("%s invalid waitCount %d", __func__, waitCount);
requested = NULL;
}
return obtainBuffer(audioBuffer, requested, NULL /*elapsed*/, nonContig);
}
status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
struct timespec *elapsed, size_t *nonContig)
{
// previous and new IAudioTrack sequence numbers are used to detect track re-creation
uint32_t oldSequence = 0;
uint32_t newSequence;
Proxy::Buffer buffer;
status_t status = NO_ERROR;
static const int32_t kMaxTries = 5;
int32_t tryCounter = kMaxTries;
do {
// obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to
// keep them from going away if another thread re-creates the track during obtainBuffer()
sp<AudioTrackClientProxy> proxy;
sp<IMemory> iMem;
{ // start of lock scope
AutoMutex lock(mLock);
newSequence = mSequence;
// did previous obtainBuffer() fail due to media server death or voluntary invalidation?
if (status == DEAD_OBJECT) {
// re-create track, unless someone else has already done so
if (newSequence == oldSequence) {
status = restoreTrack_l("obtainBuffer");
if (status != NO_ERROR) {
buffer.mFrameCount = 0;
buffer.mRaw = NULL;
buffer.mNonContig = 0;
break;
}
}
}
oldSequence = newSequence;
if (status == NOT_ENOUGH_DATA) {
restartIfDisabled();
}
// Keep the extra references
proxy = mProxy;
iMem = mCblkMemory;
if (mState == STATE_STOPPING) {
status = -EINTR;
buffer.mFrameCount = 0;
buffer.mRaw = NULL;
buffer.mNonContig = 0;
break;
}
// Non-blocking if track is stopped or paused
if (mState != STATE_ACTIVE) {
requested = &ClientProxy::kNonBlocking;
}
} // end of lock scope
buffer.mFrameCount = audioBuffer->frameCount;
// FIXME starts the requested timeout and elapsed over from scratch
status = proxy->obtainBuffer(&buffer, requested, elapsed);
} while (((status == DEAD_OBJECT) || (status == NOT_ENOUGH_DATA)) && (tryCounter-- > 0));
audioBuffer->frameCount = buffer.mFrameCount;
audioBuffer->size = buffer.mFrameCount * mFrameSize;
audioBuffer->raw = buffer.mRaw;
if (nonContig != NULL) {
*nonContig = buffer.mNonContig;
}
return status;
}
void AudioTrack::releaseBuffer(const Buffer* audioBuffer)
{
// FIXME add error checking on mode, by adding an internal version
if (mTransfer == TRANSFER_SHARED) {
return;
}
size_t stepCount = audioBuffer->size / mFrameSize;
if (stepCount == 0) {
return;
}
Proxy::Buffer buffer;
buffer.mFrameCount = stepCount;
buffer.mRaw = audioBuffer->raw;
AutoMutex lock(mLock);
mReleased += stepCount;
mInUnderrun = false;
mProxy->releaseBuffer(&buffer);
// restart track if it was disabled by audioflinger due to previous underrun
restartIfDisabled();
}
void AudioTrack::restartIfDisabled()
{
int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags);
if ((mState == STATE_ACTIVE) && (flags & CBLK_DISABLED)) {
ALOGW("releaseBuffer() track %p disabled due to previous underrun, restarting", this);
// FIXME ignoring status
mAudioTrack->start();
}
}
// -------------------------------------------------------------------------
ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking)
{
if (mTransfer != TRANSFER_SYNC) {
return INVALID_OPERATION;
}
if (isDirect()) {
AutoMutex lock(mLock);
int32_t flags = android_atomic_and(
~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
&mCblk->mFlags);
if (flags & CBLK_INVALID) {
return DEAD_OBJECT;
}
}
if (ssize_t(userSize) < 0 || (buffer == NULL && 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=%zu (%zd)", buffer, userSize, userSize);
return BAD_VALUE;
}
size_t written = 0;
Buffer audioBuffer;
while (userSize >= mFrameSize) {
audioBuffer.frameCount = userSize / mFrameSize;
status_t err = obtainBuffer(&audioBuffer,
blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking);
if (err < 0) {
if (written > 0) {
break;
}
if (err == TIMED_OUT || err == -EINTR) {
err = WOULD_BLOCK;
}
return ssize_t(err);
}
size_t toWrite = audioBuffer.size;
memcpy(audioBuffer.i8, buffer, toWrite);
buffer = ((const char *) buffer) + toWrite;
userSize -= toWrite;
written += toWrite;
releaseBuffer(&audioBuffer);
}
if (written > 0) {
mFramesWritten += written / mFrameSize;
}
return written;
}
// -------------------------------------------------------------------------
nsecs_t AudioTrack::processAudioBuffer()
{
// Currently the AudioTrack thread is not created if there are no callbacks.
// Would it ever make sense to run the thread, even without callbacks?
// If so, then replace this by checks at each use for mCbf != NULL.
LOG_ALWAYS_FATAL_IF(mCblk == NULL);
mLock.lock();
if (mAwaitBoost) {
mAwaitBoost = false;
mLock.unlock();
static const int32_t kMaxTries = 5;
int32_t tryCounter = kMaxTries;
uint32_t pollUs = 10000;
do {
int policy = sched_getscheduler(0) & ~SCHED_RESET_ON_FORK;
if (policy == SCHED_FIFO || policy == SCHED_RR) {
break;
}
usleep(pollUs);
pollUs <<= 1;
} while (tryCounter-- > 0);
if (tryCounter < 0) {
ALOGE("did not receive expected priority boost on time");
}
// Run again immediately
return 0;
}
// Can only reference mCblk while locked
int32_t flags = android_atomic_and(
~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags);
// Check for track invalidation
if (flags & CBLK_INVALID) {
// for offloaded tracks restoreTrack_l() will just update the sequence and clear
// AudioSystem cache. We should not exit here but after calling the callback so
// that the upper layers can recreate the track
if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) {
status_t status __unused = restoreTrack_l("processAudioBuffer");
// FIXME unused status
// after restoration, continue below to make sure that the loop and buffer events
// are notified because they have been cleared from mCblk->mFlags above.
}
}
bool waitStreamEnd = mState == STATE_STOPPING;
bool active = mState == STATE_ACTIVE;
// Manage underrun callback, must be done under lock to avoid race with releaseBuffer()
bool newUnderrun = false;
if (flags & CBLK_UNDERRUN) {
#if 0
// Currently in shared buffer mode, when the server reaches the end of buffer,
// the track stays active in continuous underrun state. It's up to the application
// to pause or stop the track, or set the position to a new offset within buffer.
// This was some experimental code to auto-pause on underrun. Keeping it here
// in "if 0" so we can re-visit this if we add a real sequencer for shared memory content.
if (mTransfer == TRANSFER_SHARED) {
mState = STATE_PAUSED;
active = false;
}
#endif
if (!mInUnderrun) {
mInUnderrun = true;
newUnderrun = true;
}
}
// Get current position of server
Modulo<uint32_t> position(updateAndGetPosition_l());
// Manage marker callback
bool markerReached = false;
Modulo<uint32_t> markerPosition(mMarkerPosition);
// uses 32 bit wraparound for comparison with position.
if (!mMarkerReached && markerPosition.value() > 0 && position >= markerPosition) {
mMarkerReached = markerReached = true;
}
// Determine number of new position callback(s) that will be needed, while locked
size_t newPosCount = 0;
Modulo<uint32_t> newPosition(mNewPosition);
uint32_t updatePeriod = mUpdatePeriod;
// FIXME fails for wraparound, need 64 bits
if (updatePeriod > 0 && position >= newPosition) {
newPosCount = ((position - newPosition).value() / updatePeriod) + 1;
mNewPosition += updatePeriod * newPosCount;
}
// Cache other fields that will be needed soon
uint32_t sampleRate = mSampleRate;
float speed = mPlaybackRate.mSpeed;
const uint32_t notificationFrames = mNotificationFramesAct;
if (mRefreshRemaining) {
mRefreshRemaining = false;
mRemainingFrames = notificationFrames;
mRetryOnPartialBuffer = false;
}
size_t misalignment = mProxy->getMisalignment();
uint32_t sequence = mSequence;
sp<AudioTrackClientProxy> proxy = mProxy;
// Determine the number of new loop callback(s) that will be needed, while locked.
int loopCountNotifications = 0;
uint32_t loopPeriod = 0; // time in frames for next EVENT_LOOP_END or EVENT_BUFFER_END
if (mLoopCount > 0) {
int loopCount;
size_t bufferPosition;
mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
loopPeriod = ((loopCount > 0) ? mLoopEnd : mFrameCount) - bufferPosition;
loopCountNotifications = min(mLoopCountNotified - loopCount, kMaxLoopCountNotifications);
mLoopCountNotified = loopCount; // discard any excess notifications
} else if (mLoopCount < 0) {
// FIXME: We're not accurate with notification count and position with infinite looping
// since loopCount from server side will always return -1 (we could decrement it).
size_t bufferPosition = mStaticProxy->getBufferPosition();
loopCountNotifications = int((flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) != 0);
loopPeriod = mLoopEnd - bufferPosition;
} else if (/* mLoopCount == 0 && */ mSharedBuffer != 0) {
size_t bufferPosition = mStaticProxy->getBufferPosition();
loopPeriod = mFrameCount - bufferPosition;
}
// These fields don't need to be cached, because they are assigned only by set():
// mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFlags
// mFlags is also assigned by createTrack_l(), but not the bit we care about.
mLock.unlock();
// get anchor time to account for callbacks.
const nsecs_t timeBeforeCallbacks = systemTime();
if (waitStreamEnd) {
// FIXME: Instead of blocking in proxy->waitStreamEndDone(), Callback thread
// should wait on proxy futex and handle CBLK_STREAM_END_DONE within this function
// (and make sure we don't callback for more data while we're stopping).
// This helps with position, marker notifications, and track invalidation.
struct timespec timeout;
timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC;
timeout.tv_nsec = 0;
status_t status = proxy->waitStreamEndDone(&timeout);
switch (status) {
case NO_ERROR:
case DEAD_OBJECT:
case TIMED_OUT:
if (status != DEAD_OBJECT) {
// for DEAD_OBJECT, we do not send a EVENT_STREAM_END after stop();
// instead, the application should handle the EVENT_NEW_IAUDIOTRACK.
mCbf(EVENT_STREAM_END, mUserData, NULL);
}
{
AutoMutex lock(mLock);
// The previously assigned value of waitStreamEnd is no longer valid,
// since the mutex has been unlocked and either the callback handler
// or another thread could have re-started the AudioTrack during that time.
waitStreamEnd = mState == STATE_STOPPING;
if (waitStreamEnd) {
mState = STATE_STOPPED;
mReleased = 0;
}
}
if (waitStreamEnd && status != DEAD_OBJECT) {
return NS_INACTIVE;
}
break;
}
return 0;
}
// perform callbacks while unlocked
if (newUnderrun) {
mCbf(EVENT_UNDERRUN, mUserData, NULL);
}
while (loopCountNotifications > 0) {
mCbf(EVENT_LOOP_END, mUserData, NULL);
--loopCountNotifications;
}
if (flags & CBLK_BUFFER_END) {
mCbf(EVENT_BUFFER_END, mUserData, NULL);
}
if (markerReached) {
mCbf(EVENT_MARKER, mUserData, &markerPosition);
}
while (newPosCount > 0) {
size_t temp = newPosition.value(); // FIXME size_t != uint32_t
mCbf(EVENT_NEW_POS, mUserData, &temp);
newPosition += updatePeriod;
newPosCount--;
}
if (mObservedSequence != sequence) {
mObservedSequence = sequence;
mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL);
// for offloaded tracks, just wait for the upper layers to recreate the track
if (isOffloadedOrDirect()) {
return NS_INACTIVE;
}
}
// if inactive, then don't run me again until re-started
if (!active) {
return NS_INACTIVE;
}
// Compute the estimated time until the next timed event (position, markers, loops)
// FIXME only for non-compressed audio
uint32_t minFrames = ~0;
if (!markerReached && position < markerPosition) {
minFrames = (markerPosition - position).value();
}
if (loopPeriod > 0 && loopPeriod < minFrames) {
// loopPeriod is already adjusted for actual position.
minFrames = loopPeriod;
}
if (updatePeriod > 0) {
minFrames = min(minFrames, (newPosition - position).value());
}
// If > 0, poll periodically to recover from a stuck server. A good value is 2.
static const uint32_t kPoll = 0;
if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) {
minFrames = kPoll * notificationFrames;
}
// This "fudge factor" avoids soaking CPU, and compensates for late progress by server
static const nsecs_t kWaitPeriodNs = WAIT_PERIOD_MS * 1000000LL;
const nsecs_t timeAfterCallbacks = systemTime();
// Convert frame units to time units
nsecs_t ns = NS_WHENEVER;
if (minFrames != (uint32_t) ~0) {
// AudioFlinger consumption of client data may be irregular when coming out of device
// standby since the kernel buffers require filling. This is throttled to no more than 2x
// the expected rate in the MixerThread. Hence, we reduce the estimated time to wait by one
// half (but no more than half a second) to improve callback accuracy during these temporary
// data surges.
const nsecs_t estimatedNs = framesToNanoseconds(minFrames, sampleRate, speed);
constexpr nsecs_t maxThrottleCompensationNs = 500000000LL;
ns = estimatedNs - min(estimatedNs / 2, maxThrottleCompensationNs) + kWaitPeriodNs;
ns -= (timeAfterCallbacks - timeBeforeCallbacks); // account for callback time
// TODO: Should we warn if the callback time is too long?
if (ns < 0) ns = 0;
}
// If not supplying data by EVENT_MORE_DATA, then we're done
if (mTransfer != TRANSFER_CALLBACK) {
return ns;
}
// EVENT_MORE_DATA callback handling.
// Timing for linear pcm audio data formats can be derived directly from the
// buffer fill level.
// Timing for compressed data is not directly available from the buffer fill level,
// rather indirectly from waiting for blocking mode callbacks or waiting for obtain()
// to return a certain fill level.
struct timespec timeout;
const struct timespec *requested = &ClientProxy::kForever;
if (ns != NS_WHENEVER) {
timeout.tv_sec = ns / 1000000000LL;
timeout.tv_nsec = ns % 1000000000LL;
ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000);
requested = &timeout;
}
size_t writtenFrames = 0;
while (mRemainingFrames > 0) {
Buffer audioBuffer;
audioBuffer.frameCount = mRemainingFrames;
size_t nonContig;
status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig);
LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0),
"obtainBuffer() err=%d frameCount=%zu", err, audioBuffer.frameCount);
requested = &ClientProxy::kNonBlocking;
size_t avail = audioBuffer.frameCount + nonContig;
ALOGV("obtainBuffer(%u) returned %zu = %zu + %zu err %d",
mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
if (err != NO_ERROR) {
if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR ||
(isOffloaded() && (err == DEAD_OBJECT))) {
// FIXME bug 25195759
return 1000000;
}
ALOGE("Error %d obtaining an audio buffer, giving up.", err);
return NS_NEVER;
}
if (mRetryOnPartialBuffer && audio_has_proportional_frames(mFormat)) {
mRetryOnPartialBuffer = false;
if (avail < mRemainingFrames) {
if (ns > 0) { // account for obtain time
const nsecs_t timeNow = systemTime();
ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks));
}
nsecs_t myns = framesToNanoseconds(mRemainingFrames - avail, sampleRate, speed);
if (ns < 0 /* NS_WHENEVER */ || myns < ns) {
ns = myns;
}
return ns;
}
}
size_t reqSize = audioBuffer.size;
mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
size_t writtenSize = audioBuffer.size;
// Sanity check on returned size
if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) {
ALOGE("EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes",
reqSize, ssize_t(writtenSize));
return NS_NEVER;
}
if (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
// mCbf(EVENT_MORE_DATA, ...) might either
// (1) Block until it can fill the buffer, returning 0 size on EOS.
// (2) Block until it can fill the buffer, returning 0 data (silence) on EOS.
// (3) Return 0 size when no data is available, does not wait for more data.
//
// (1) and (2) occurs with AudioPlayer/AwesomePlayer; (3) occurs with NuPlayer.
// We try to compute the wait time to avoid a tight sleep-wait cycle,
// especially for case (3).
//
// The decision to support (1) and (2) affect the sizing of mRemainingFrames
// and this loop; whereas for case (3) we could simply check once with the full
// buffer size and skip the loop entirely.
nsecs_t myns;
if (audio_has_proportional_frames(mFormat)) {
// time to wait based on buffer occupancy
const nsecs_t datans = mRemainingFrames <= avail ? 0 :
framesToNanoseconds(mRemainingFrames - avail, sampleRate, speed);
// audio flinger thread buffer size (TODO: adjust for fast tracks)
// FIXME: use mAfFrameCountHAL instead of mAfFrameCount below for fast tracks.
const nsecs_t afns = framesToNanoseconds(mAfFrameCount, mAfSampleRate, speed);
// add a half the AudioFlinger buffer time to avoid soaking CPU if datans is 0.
myns = datans + (afns / 2);
} else {
// FIXME: This could ping quite a bit if the buffer isn't full.
// Note that when mState is stopping we waitStreamEnd, so it never gets here.
myns = kWaitPeriodNs;
}
if (ns > 0) { // account for obtain and callback time
const nsecs_t timeNow = systemTime();
ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks));
}
if (ns < 0 /* NS_WHENEVER */ || myns < ns) {
ns = myns;
}
return ns;
}
size_t releasedFrames = writtenSize / mFrameSize;
audioBuffer.frameCount = releasedFrames;
mRemainingFrames -= releasedFrames;
if (misalignment >= releasedFrames) {
misalignment -= releasedFrames;
} else {
misalignment = 0;
}
releaseBuffer(&audioBuffer);
writtenFrames += releasedFrames;
// FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer
// if callback doesn't like to accept the full chunk
if (writtenSize < reqSize) {
continue;
}
// There could be enough non-contiguous frames available to satisfy the remaining request
if (mRemainingFrames <= nonContig) {
continue;
}
#if 0
// This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a
// sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA
// that total to a sum == notificationFrames.
if (0 < misalignment && misalignment <= mRemainingFrames) {
mRemainingFrames = misalignment;
return ((double)mRemainingFrames * 1100000000) / ((double)sampleRate * speed);
}
#endif
}
if (writtenFrames > 0) {
AutoMutex lock(mLock);
mFramesWritten += writtenFrames;
}
mRemainingFrames = notificationFrames;
mRetryOnPartialBuffer = true;
// A lot has transpired since ns was calculated, so run again immediately and re-calculate
return 0;
}
status_t AudioTrack::restoreTrack_l(const char *from)
{
ALOGW("dead IAudioTrack, %s, creating a new one from %s()",
isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from);
++mSequence;
// refresh the audio configuration cache in this process to make sure we get new
// output parameters and new IAudioFlinger in createTrack_l()
AudioSystem::clearAudioConfigCache();
if (isOffloadedOrDirect_l() || mDoNotReconnect) {
// FIXME re-creation of offloaded and direct tracks is not yet implemented;
// reconsider enabling for linear PCM encodings when position can be preserved.
return DEAD_OBJECT;
}
// Save so we can return count since creation.
mUnderrunCountOffset = getUnderrunCount_l();
// save the old static buffer position
uint32_t staticPosition = 0;
size_t bufferPosition = 0;
int loopCount = 0;
if (mStaticProxy != 0) {
mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
staticPosition = mStaticProxy->getPosition().unsignedValue();
}
// See b/74409267. Connecting to a BT A2DP device supporting multiple codecs
// causes a lot of churn on the service side, and it can reject starting
// playback of a previously created track. May also apply to other cases.
const int INITIAL_RETRIES = 3;
int retries = INITIAL_RETRIES;
retry:
if (retries < INITIAL_RETRIES) {
// See the comment for clearAudioConfigCache at the start of the function.
AudioSystem::clearAudioConfigCache();
}
mFlags = mOrigFlags;
// If a new IAudioTrack is successfully 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.
// If a new IAudioTrack cannot be created, the previous (dead) instance will be left intact.
status_t result = createTrack_l();
if (result != NO_ERROR) {
ALOGW("%s(): createTrack_l failed, do not retry", __func__);
retries = 0;
} else {
// take the frames that will be lost by track recreation into account in saved position
// For streaming tracks, this is the amount we obtained from the user/client
// (not the number actually consumed at the server - those are already lost).
if (mStaticProxy == 0) {
mPosition = mReleased;
}
// Continue playback from last known position and restore loop.
if (mStaticProxy != 0) {
if (loopCount != 0) {
mStaticProxy->setBufferPositionAndLoop(bufferPosition,
mLoopStart, mLoopEnd, loopCount);
} else {
mStaticProxy->setBufferPosition(bufferPosition);
if (bufferPosition == mFrameCount) {
ALOGD("restoring track at end of static buffer");
}
}
}
// restore volume handler
mVolumeHandler->forall([this](const VolumeShaper &shaper) -> VolumeShaper::Status {
sp<VolumeShaper::Operation> operationToEnd =
new VolumeShaper::Operation(shaper.mOperation);
// TODO: Ideally we would restore to the exact xOffset position
// as returned by getVolumeShaperState(), but we don't have that
// information when restoring at the client unless we periodically poll
// the server or create shared memory state.
//
// For now, we simply advance to the end of the VolumeShaper effect
// if it has been started.
if (shaper.isStarted()) {
operationToEnd->setNormalizedTime(1.f);
}
return mAudioTrack->applyVolumeShaper(shaper.mConfiguration, operationToEnd);
});
if (mState == STATE_ACTIVE) {
result = mAudioTrack->start();
}
// server resets to zero so we offset
mFramesWrittenServerOffset =
mStaticProxy.get() != nullptr ? staticPosition : mFramesWritten;
mFramesWrittenAtRestore = mFramesWrittenServerOffset;
}
if (result != NO_ERROR) {
ALOGW("%s() failed status %d, retries %d", __func__, result, retries);
if (--retries > 0) {
goto retry;
}
mState = STATE_STOPPED;
mReleased = 0;
}
return result;
}
Modulo<uint32_t> AudioTrack::updateAndGetPosition_l()
{
// This is the sole place to read server consumed frames
Modulo<uint32_t> newServer(mProxy->getPosition());
const int32_t delta = (newServer - mServer).signedValue();
// TODO There is controversy about whether there can be "negative jitter" in server position.
// This should be investigated further, and if possible, it should be addressed.
// A more definite failure mode is infrequent polling by client.
// One could call (void)getPosition_l() in releaseBuffer(),
// so mReleased and mPosition are always lock-step as best possible.
// That should ensure delta never goes negative for infrequent polling
// unless the server has more than 2^31 frames in its buffer,
// in which case the use of uint32_t for these counters has bigger issues.
ALOGE_IF(delta < 0,
"detected illegal retrograde motion by the server: mServer advanced by %d",
delta);
mServer = newServer;
if (delta > 0) { // avoid retrograde
mPosition += delta;
}
return mPosition;
}
bool AudioTrack::isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed)
{
updateLatency_l();
// applicable for mixing tracks only (not offloaded or direct)
if (mStaticProxy != 0) {
return true; // static tracks do not have issues with buffer sizing.
}
const size_t minFrameCount =
AudioSystem::calculateMinFrameCount(mAfLatency, mAfFrameCount, mAfSampleRate,
sampleRate, speed /*, 0 mNotificationsPerBufferReq*/);
const bool allowed = mFrameCount >= minFrameCount;
ALOGD_IF(!allowed,
"isSampleRateSpeedAllowed_l denied "
"mAfLatency:%u mAfFrameCount:%zu mAfSampleRate:%u sampleRate:%u speed:%f "
"mFrameCount:%zu < minFrameCount:%zu",
mAfLatency, mAfFrameCount, mAfSampleRate, sampleRate, speed,
mFrameCount, minFrameCount);
return allowed;
}
status_t AudioTrack::setParameters(const String8& keyValuePairs)
{
AutoMutex lock(mLock);
return mAudioTrack->setParameters(keyValuePairs);
}
status_t AudioTrack::selectPresentation(int presentationId, int programId)
{
AutoMutex lock(mLock);
AudioParameter param = AudioParameter();
param.addInt(String8(AudioParameter::keyPresentationId), presentationId);
param.addInt(String8(AudioParameter::keyProgramId), programId);
ALOGV("PresentationId/ProgramId[%s]",param.toString().string());
return mAudioTrack->setParameters(param.toString());
}
VolumeShaper::Status AudioTrack::applyVolumeShaper(
const sp<VolumeShaper::Configuration>& configuration,
const sp<VolumeShaper::Operation>& operation)
{
AutoMutex lock(mLock);
mVolumeHandler->setIdIfNecessary(configuration);
VolumeShaper::Status status = mAudioTrack->applyVolumeShaper(configuration, operation);
if (status == DEAD_OBJECT) {
if (restoreTrack_l("applyVolumeShaper") == OK) {
status = mAudioTrack->applyVolumeShaper(configuration, operation);
}
}
if (status >= 0) {
// save VolumeShaper for restore
mVolumeHandler->applyVolumeShaper(configuration, operation);
if (mState == STATE_ACTIVE || mState == STATE_STOPPING) {
mVolumeHandler->setStarted();
}
} else {
// warn only if not an expected restore failure.
ALOGW_IF(!((isOffloadedOrDirect_l() || mDoNotReconnect) && status == DEAD_OBJECT),
"applyVolumeShaper failed: %d", status);
}
return status;
}
sp<VolumeShaper::State> AudioTrack::getVolumeShaperState(int id)
{
AutoMutex lock(mLock);
sp<VolumeShaper::State> state = mAudioTrack->getVolumeShaperState(id);
if (state.get() == nullptr && (mCblk->mFlags & CBLK_INVALID) != 0) {
if (restoreTrack_l("getVolumeShaperState") == OK) {
state = mAudioTrack->getVolumeShaperState(id);
}
}
return state;
}
status_t AudioTrack::getTimestamp(ExtendedTimestamp *timestamp)
{
if (timestamp == nullptr) {
return BAD_VALUE;
}
AutoMutex lock(mLock);
return getTimestamp_l(timestamp);
}
status_t AudioTrack::getTimestamp_l(ExtendedTimestamp *timestamp)
{
if (mCblk->mFlags & CBLK_INVALID) {
const status_t status = restoreTrack_l("getTimestampExtended");
if (status != OK) {
// per getTimestamp() API doc in header, we return DEAD_OBJECT here,
// recommending that the track be recreated.
return DEAD_OBJECT;
}
}
// check for offloaded/direct here in case restoring somehow changed those flags.
if (isOffloadedOrDirect_l()) {
return INVALID_OPERATION; // not supported
}
status_t status = mProxy->getTimestamp(timestamp);
LOG_ALWAYS_FATAL_IF(status != OK, "status %d not allowed from proxy getTimestamp", status);
bool found = false;
timestamp->mPosition[ExtendedTimestamp::LOCATION_CLIENT] = mFramesWritten;
timestamp->mTimeNs[ExtendedTimestamp::LOCATION_CLIENT] = 0;
// server side frame offset in case AudioTrack has been restored.
for (int i = ExtendedTimestamp::LOCATION_SERVER;
i < ExtendedTimestamp::LOCATION_MAX; ++i) {
if (timestamp->mTimeNs[i] >= 0) {
// apply server offset (frames flushed is ignored
// so we don't report the jump when the flush occurs).
timestamp->mPosition[i] += mFramesWrittenServerOffset;
found = true;
}
}
return found ? OK : WOULD_BLOCK;
}
status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp)
{
AutoMutex lock(mLock);
return getTimestamp_l(timestamp);
}
status_t AudioTrack::getTimestamp_l(AudioTimestamp& timestamp)
{
bool previousTimestampValid = mPreviousTimestampValid;
// Set false here to cover all the error return cases.
mPreviousTimestampValid = false;
switch (mState) {
case STATE_ACTIVE:
case STATE_PAUSED:
break; // handle below
case STATE_FLUSHED:
case STATE_STOPPED:
return WOULD_BLOCK;
case STATE_STOPPING:
case STATE_PAUSED_STOPPING:
if (!isOffloaded_l()) {
return INVALID_OPERATION;
}
break; // offloaded tracks handled below
default:
LOG_ALWAYS_FATAL("Invalid mState in getTimestamp(): %d", mState);
break;
}
if (mCblk->mFlags & CBLK_INVALID) {
const status_t status = restoreTrack_l("getTimestamp");
if (status != OK) {
// per getTimestamp() API doc in header, we return DEAD_OBJECT here,
// recommending that the track be recreated.
return DEAD_OBJECT;
}
}
// The presented frame count must always lag behind the consumed frame count.
// To avoid a race, read the presented frames first. This ensures that presented <= consumed.
status_t status;
if (isOffloadedOrDirect_l()) {
// use Binder to get timestamp
status = mAudioTrack->getTimestamp(timestamp);
} else {
// read timestamp from shared memory
ExtendedTimestamp ets;
status = mProxy->getTimestamp(&ets);
if (status == OK) {
ExtendedTimestamp::Location location;
status = ets.getBestTimestamp(×tamp, &location);
if (status == OK) {
updateLatency_l();
// It is possible that the best location has moved from the kernel to the server.
// In this case we adjust the position from the previous computed latency.
if (location == ExtendedTimestamp::LOCATION_SERVER) {
ALOGW_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_KERNEL,
"getTimestamp() location moved from kernel to server");
// check that the last kernel OK time info exists and the positions
// are valid (if they predate the current track, the positions may
// be zero or negative).
const int64_t frames =
(ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 ||
ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0 ||
ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] <= 0 ||
ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] <= 0)
?
int64_t((double)mAfLatency * mSampleRate * mPlaybackRate.mSpeed
/ 1000)
:
(ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK]
- ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK]);
ALOGV("frame adjustment:%lld timestamp:%s",
(long long)frames, ets.toString().c_str());
if (frames >= ets.mPosition[location]) {
timestamp.mPosition = 0;
} else {
timestamp.mPosition = (uint32_t)(ets.mPosition[location] - frames);
}
} else if (location == ExtendedTimestamp::LOCATION_KERNEL) {
ALOGV_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_SERVER,
"getTimestamp() location moved from server to kernel");
}
// We update the timestamp time even when paused.
if (mState == STATE_PAUSED /* not needed: STATE_PAUSED_STOPPING */) {
const int64_t now = systemTime();
const int64_t at = audio_utils_ns_from_timespec(×tamp.mTime);
const int64_t lag =
(ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 ||
ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0)
? int64_t(mAfLatency * 1000000LL)
: (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK]
- ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK])
* NANOS_PER_SECOND / mSampleRate;
const int64_t limit = now - lag; // no earlier than this limit
if (at < limit) {
ALOGV("timestamp pause lag:%lld adjusting from %lld to %lld",
(long long)lag, (long long)at, (long long)limit);
timestamp.mTime = convertNsToTimespec(limit);
}
}
mPreviousLocation = location;
} else {
// right after AudioTrack is started, one may not find a timestamp
ALOGV("getBestTimestamp did not find timestamp");
}
}
if (status == INVALID_OPERATION) {
// INVALID_OPERATION occurs when no timestamp has been issued by the server;
// other failures are signaled by a negative time.
// If we come out of FLUSHED or STOPPED where the position is known
// to be zero we convert this to WOULD_BLOCK (with the implicit meaning of
// "zero" for NuPlayer). We don't convert for track restoration as position
// does not reset.
ALOGV("timestamp server offset:%lld restore frames:%lld",
(long long)mFramesWrittenServerOffset, (long long)mFramesWrittenAtRestore);
if (mFramesWrittenServerOffset != mFramesWrittenAtRestore) {
status = WOULD_BLOCK;
}
}
}
if (status != NO_ERROR) {
ALOGV_IF(status != WOULD_BLOCK, "getTimestamp error:%#x", status);
return status;
}
if (isOffloadedOrDirect_l()) {
if (isOffloaded_l() && (mState == STATE_PAUSED || mState == STATE_PAUSED_STOPPING)) {
// use cached paused position in case another offloaded track is running.
timestamp.mPosition = mPausedPosition;
clock_gettime(CLOCK_MONOTONIC, ×tamp.mTime);
// TODO: adjust for delay
return NO_ERROR;
}
// Check whether a pending flush or stop has completed, as those commands may
// be asynchronous or return near finish or exhibit glitchy behavior.
//
// Originally this showed up as the first timestamp being a continuation of
// the previous song under gapless playback.
// However, we sometimes see zero timestamps, then a glitch of
// the previous song's position, and then correct timestamps afterwards.
if (mStartFromZeroUs != 0 && mSampleRate != 0) {
static const int kTimeJitterUs = 100000; // 100 ms
static const int k1SecUs = 1000000;
const int64_t timeNow = getNowUs();
if (timeNow < mStartFromZeroUs + k1SecUs) { // within first second of starting
const int64_t timestampTimeUs = convertTimespecToUs(timestamp.mTime);
if (timestampTimeUs < mStartFromZeroUs) {
return WOULD_BLOCK; // stale timestamp time, occurs before start.
}
const int64_t deltaTimeUs = timestampTimeUs - mStartFromZeroUs;
const int64_t deltaPositionByUs = (double)timestamp.mPosition * 1000000
/ ((double)mSampleRate * mPlaybackRate.mSpeed);
if (deltaPositionByUs > deltaTimeUs + kTimeJitterUs) {
// Verify that the counter can't count faster than the sample rate
// since the start time. If greater, then that means we may have failed
// to completely flush or stop the previous playing track.
ALOGW_IF(!mTimestampStartupGlitchReported,
"getTimestamp startup glitch detected"
" deltaTimeUs(%lld) deltaPositionUs(%lld) tsmPosition(%u)",
(long long)deltaTimeUs, (long long)deltaPositionByUs,
timestamp.mPosition);
mTimestampStartupGlitchReported = true;
if (previousTimestampValid
&& mPreviousTimestamp.mPosition == 0 /* should be true if valid */) {
timestamp = mPreviousTimestamp;
mPreviousTimestampValid = true;
return NO_ERROR;
}
return WOULD_BLOCK;
}
if (deltaPositionByUs != 0) {
mStartFromZeroUs = 0; // don't check again, we got valid nonzero position.
}
} else {
mStartFromZeroUs = 0; // don't check again, start time expired.
}
mTimestampStartupGlitchReported = false;
}
} else {
// Update the mapping between local consumed (mPosition) and server consumed (mServer)
(void) updateAndGetPosition_l();
// Server consumed (mServer) and presented both use the same server time base,
// and server consumed is always >= presented.
// The delta between these represents the number of frames in the buffer pipeline.
// If this delta between these is greater than the client position, it means that
// actually presented is still stuck at the starting line (figuratively speaking),
// waiting for the first frame to go by. So we can't report a valid timestamp yet.
// Note: We explicitly use non-Modulo comparison here - potential wrap issue when
// mPosition exceeds 32 bits.
// TODO Remove when timestamp is updated to contain pipeline status info.
const int32_t pipelineDepthInFrames = (mServer - timestamp.mPosition).signedValue();
if (pipelineDepthInFrames > 0 /* should be true, but we check anyways */
&& (uint32_t)pipelineDepthInFrames > mPosition.value()) {
return INVALID_OPERATION;
}
// Convert timestamp position from server time base to client time base.
// TODO The following code should work OK now because timestamp.mPosition is 32-bit.
// But if we change it to 64-bit then this could fail.
// Use Modulo computation here.
timestamp.mPosition = (mPosition - mServer + timestamp.mPosition).value();
// Immediately after a call to getPosition_l(), mPosition and
// mServer both represent the same frame position. mPosition is
// in client's point of view, and mServer is in server's point of
// view. So the difference between them is the "fudge factor"
// between client and server views due to stop() and/or new
// IAudioTrack. And timestamp.mPosition is initially in server's
// point of view, so we need to apply the same fudge factor to it.
}
// Prevent retrograde motion in timestamp.
// This is sometimes caused by erratic reports of the available space in the ALSA drivers.
if (status == NO_ERROR) {
// previousTimestampValid is set to false when starting after a stop or flush.
if (previousTimestampValid) {
const int64_t previousTimeNanos =
audio_utils_ns_from_timespec(&mPreviousTimestamp.mTime);
int64_t currentTimeNanos = audio_utils_ns_from_timespec(×tamp.mTime);
// Fix stale time when checking timestamp right after start().
//
// For offload compatibility, use a default lag value here.
// Any time discrepancy between this update and the pause timestamp is handled
// by the retrograde check afterwards.
const int64_t lagNs = int64_t(mAfLatency * 1000000LL);
const int64_t limitNs = mStartNs - lagNs;
if (currentTimeNanos < limitNs) {
ALOGD("correcting timestamp time for pause, "
"currentTimeNanos: %lld < limitNs: %lld < mStartNs: %lld",
(long long)currentTimeNanos, (long long)limitNs, (long long)mStartNs);
timestamp.mTime = convertNsToTimespec(limitNs);
currentTimeNanos = limitNs;
}
// retrograde check
if (currentTimeNanos < previousTimeNanos) {
ALOGW("retrograde timestamp time corrected, %lld < %lld",
(long long)currentTimeNanos, (long long)previousTimeNanos);
timestamp.mTime = mPreviousTimestamp.mTime;
// currentTimeNanos not used below.
}
// Looking at signed delta will work even when the timestamps
// are wrapping around.
int32_t deltaPosition = (Modulo<uint32_t>(timestamp.mPosition)
- mPreviousTimestamp.mPosition).signedValue();
if (deltaPosition < 0) {
// Only report once per position instead of spamming the log.
if (!mRetrogradeMotionReported) {
ALOGW("retrograde timestamp position corrected, %d = %u - %u",
deltaPosition,
timestamp.mPosition,
mPreviousTimestamp.mPosition);
mRetrogradeMotionReported = true;
}
} else {
mRetrogradeMotionReported = false;
}
if (deltaPosition < 0) {
timestamp.mPosition = mPreviousTimestamp.mPosition;
deltaPosition = 0;
}
#if 0
// Uncomment this to verify audio timestamp rate.
const int64_t deltaTime =
audio_utils_ns_from_timespec(×tamp.mTime) - previousTimeNanos;
if (deltaTime != 0) {
const int64_t computedSampleRate =
deltaPosition * (long long)NANOS_PER_SECOND / deltaTime;
ALOGD("computedSampleRate:%u sampleRate:%u",
(unsigned)computedSampleRate, mSampleRate);
}
#endif
}
mPreviousTimestamp = timestamp;
mPreviousTimestampValid = true;
}
return status;
}
String8 AudioTrack::getParameters(const String8& keys)
{
audio_io_handle_t output = getOutput();
if (output != AUDIO_IO_HANDLE_NONE) {
return AudioSystem::getParameters(output, keys);
} else {
return String8::empty();
}
}
bool AudioTrack::isOffloaded() const
{
AutoMutex lock(mLock);
return isOffloaded_l();
}
bool AudioTrack::isDirect() const
{
AutoMutex lock(mLock);
return isDirect_l();
}
bool AudioTrack::isOffloadedOrDirect() const
{
AutoMutex lock(mLock);
return isOffloadedOrDirect_l();
}
status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const
{
String8 result;
result.append(" AudioTrack::dump\n");
result.appendFormat(" status(%d), state(%d), session Id(%d), flags(%#x)\n",
mStatus, mState, mSessionId, mFlags);
result.appendFormat(" stream type(%d), left - right volume(%f, %f)\n",
(mStreamType == AUDIO_STREAM_DEFAULT) ?
audio_attributes_to_stream_type(&mAttributes) : mStreamType,
mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]);
result.appendFormat(" format(%#x), channel mask(%#x), channel count(%u)\n",
mFormat, mChannelMask, mChannelCount);
result.appendFormat(" sample rate(%u), original sample rate(%u), speed(%f)\n",
mSampleRate, mOriginalSampleRate, mPlaybackRate.mSpeed);
result.appendFormat(" frame count(%zu), req. frame count(%zu)\n",
mFrameCount, mReqFrameCount);
result.appendFormat(" notif. frame count(%u), req. notif. frame count(%u),"
" req. notif. per buff(%u)\n",
mNotificationFramesAct, mNotificationFramesReq, mNotificationsPerBufferReq);
result.appendFormat(" latency (%d), selected device Id(%d), routed device Id(%d)\n",
mLatency, mSelectedDeviceId, mRoutedDeviceId);
result.appendFormat(" output(%d) AF latency (%u) AF frame count(%zu) AF SampleRate(%u)\n",
mOutput, mAfLatency, mAfFrameCount, mAfSampleRate);
::write(fd, result.string(), result.size());
return NO_ERROR;
}
uint32_t AudioTrack::getUnderrunCount() const
{
AutoMutex lock(mLock);
return getUnderrunCount_l();
}
uint32_t AudioTrack::getUnderrunCount_l() const
{
return mProxy->getUnderrunCount() + mUnderrunCountOffset;
}
uint32_t AudioTrack::getUnderrunFrames() const
{
AutoMutex lock(mLock);
return mProxy->getUnderrunFrames();
}
status_t AudioTrack::addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback)
{
if (callback == 0) {
ALOGW("%s adding NULL callback!", __FUNCTION__);
return BAD_VALUE;
}
AutoMutex lock(mLock);
if (mDeviceCallback.unsafe_get() == callback.get()) {
ALOGW("%s adding same callback!", __FUNCTION__);
return INVALID_OPERATION;
}
status_t status = NO_ERROR;
if (mOutput != AUDIO_IO_HANDLE_NONE) {
if (mDeviceCallback != 0) {
ALOGW("%s callback already present!", __FUNCTION__);
AudioSystem::removeAudioDeviceCallback(this, mOutput);
}
status = AudioSystem::addAudioDeviceCallback(this, mOutput);
}
mDeviceCallback = callback;
return status;
}
status_t AudioTrack::removeAudioDeviceCallback(
const sp<AudioSystem::AudioDeviceCallback>& callback)
{
if (callback == 0) {
ALOGW("%s removing NULL callback!", __FUNCTION__);
return BAD_VALUE;
}
AutoMutex lock(mLock);
if (mDeviceCallback.unsafe_get() != callback.get()) {
ALOGW("%s removing different callback!", __FUNCTION__);
return INVALID_OPERATION;
}
mDeviceCallback.clear();
if (mOutput != AUDIO_IO_HANDLE_NONE) {
AudioSystem::removeAudioDeviceCallback(this, mOutput);
}
return NO_ERROR;
}
void AudioTrack::onAudioDeviceUpdate(audio_io_handle_t audioIo,
audio_port_handle_t deviceId)
{
sp<AudioSystem::AudioDeviceCallback> callback;
{
AutoMutex lock(mLock);
if (audioIo != mOutput) {
return;
}
callback = mDeviceCallback.promote();
// only update device if the track is active as route changes due to other use cases are
// irrelevant for this client
if (mState == STATE_ACTIVE) {
mRoutedDeviceId = deviceId;
}
}
if (callback.get() != nullptr) {
callback->onAudioDeviceUpdate(mOutput, mRoutedDeviceId);
}
}
status_t AudioTrack::pendingDuration(int32_t *msec, ExtendedTimestamp::Location location)
{
if (msec == nullptr ||
(location != ExtendedTimestamp::LOCATION_SERVER
&& location != ExtendedTimestamp::LOCATION_KERNEL)) {
return BAD_VALUE;
}
AutoMutex lock(mLock);
// inclusive of offloaded and direct tracks.
//
// It is possible, but not enabled, to allow duration computation for non-pcm
// audio_has_proportional_frames() formats because currently they have
// the drain rate equivalent to the pcm sample rate * framesize.
if (!isPurePcmData_l()) {
return INVALID_OPERATION;
}
ExtendedTimestamp ets;
if (getTimestamp_l(&ets) == OK
&& ets.mTimeNs[location] > 0) {
int64_t diff = ets.mPosition[ExtendedTimestamp::LOCATION_CLIENT]
- ets.mPosition[location];
if (diff < 0) {
*msec = 0;
} else {
// ms is the playback time by frames
int64_t ms = (int64_t)((double)diff * 1000 /
((double)mSampleRate * mPlaybackRate.mSpeed));
// clockdiff is the timestamp age (negative)
int64_t clockdiff = (mState != STATE_ACTIVE) ? 0 :
ets.mTimeNs[location]
+ ets.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_MONOTONIC]
- systemTime(SYSTEM_TIME_MONOTONIC);
//ALOGV("ms: %lld clockdiff: %lld", (long long)ms, (long long)clockdiff);
static const int NANOS_PER_MILLIS = 1000000;
*msec = (int32_t)(ms + clockdiff / NANOS_PER_MILLIS);
}
return NO_ERROR;
}
if (location != ExtendedTimestamp::LOCATION_SERVER) {
return INVALID_OPERATION; // LOCATION_KERNEL is not available
}
// use server position directly (offloaded and direct arrive here)
updateAndGetPosition_l();
int32_t diff = (Modulo<uint32_t>(mFramesWritten) - mPosition).signedValue();
*msec = (diff <= 0) ? 0
: (int32_t)((double)diff * 1000 / ((double)mSampleRate * mPlaybackRate.mSpeed));
return NO_ERROR;
}
bool AudioTrack::hasStarted()
{
AutoMutex lock(mLock);
switch (mState) {
case STATE_STOPPED:
if (isOffloadedOrDirect_l()) {
// check if we have started in the past to return true.
return mStartFromZeroUs > 0;
}
// A normal audio track may still be draining, so
// check if stream has ended. This covers fasttrack position
// instability and start/stop without any data written.
if (mProxy->getStreamEndDone()) {
return true;
}
// fall through
case STATE_ACTIVE:
case STATE_STOPPING:
break;
case STATE_PAUSED:
case STATE_PAUSED_STOPPING:
case STATE_FLUSHED:
return false; // we're not active
default:
LOG_ALWAYS_FATAL("Invalid mState in hasStarted(): %d", mState);
break;
}
// wait indicates whether we need to wait for a timestamp.
// This is conservatively figured - if we encounter an unexpected error
// then we will not wait.
bool wait = false;
if (isOffloadedOrDirect_l()) {
AudioTimestamp ts;
status_t status = getTimestamp_l(ts);
if (status == WOULD_BLOCK) {
wait = true;
} else if (status == OK) {
wait = (ts.mPosition == 0 || ts.mPosition == mStartTs.mPosition);
}
ALOGV("hasStarted wait:%d ts:%u start position:%lld",
(int)wait,
ts.mPosition,
(long long)mStartTs.mPosition);
} else {
int location = ExtendedTimestamp::LOCATION_SERVER; // for ALOG
ExtendedTimestamp ets;
status_t status = getTimestamp_l(&ets);
if (status == WOULD_BLOCK) { // no SERVER or KERNEL frame info in ets
wait = true;
} else if (status == OK) {
for (location = ExtendedTimestamp::LOCATION_KERNEL;
location >= ExtendedTimestamp::LOCATION_SERVER; --location) {
if (ets.mTimeNs[location] < 0 || mStartEts.mTimeNs[location] < 0) {
continue;
}
wait = ets.mPosition[location] == 0
|| ets.mPosition[location] == mStartEts.mPosition[location];
break;
}
}
ALOGV("hasStarted wait:%d ets:%lld start position:%lld",
(int)wait,
(long long)ets.mPosition[location],
(long long)mStartEts.mPosition[location]);
}
return !wait;
}
// =========================================================================
void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused)
{
sp<AudioTrack> audioTrack = mAudioTrack.promote();
if (audioTrack != 0) {
AutoMutex lock(audioTrack->mLock);
audioTrack->mProxy->binderDied();
}
}
// =========================================================================
AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava)
: Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL),
mIgnoreNextPausedInt(false)
{
}
AudioTrack::AudioTrackThread::~AudioTrackThread()
{
}
bool AudioTrack::AudioTrackThread::threadLoop()
{
{
AutoMutex _l(mMyLock);
if (mPaused) {
// TODO check return value and handle or log
mMyCond.wait(mMyLock);
// caller will check for exitPending()
return true;
}
if (mIgnoreNextPausedInt) {
mIgnoreNextPausedInt = false;
mPausedInt = false;
}
if (mPausedInt) {
// TODO use futex instead of condition, for event flag "or"
if (mPausedNs > 0) {
// TODO check return value and handle or log
(void) mMyCond.waitRelative(mMyLock, mPausedNs);
} else {
// TODO check return value and handle or log
mMyCond.wait(mMyLock);
}
mPausedInt = false;
return true;
}
}
if (exitPending()) {
return false;
}
nsecs_t ns = mReceiver.processAudioBuffer();
switch (ns) {
case 0:
return true;
case NS_INACTIVE:
pauseInternal();
return true;
case NS_NEVER:
return false;
case NS_WHENEVER:
// Event driven: call wake() when callback notifications conditions change.
ns = INT64_MAX;
// fall through
default:
LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns);
pauseInternal(ns);
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);
mIgnoreNextPausedInt = true;
if (mPaused || mPausedInt) {
mPaused = false;
mPausedInt = false;
mMyCond.signal();
}
}
void AudioTrack::AudioTrackThread::wake()
{
AutoMutex _l(mMyLock);
if (!mPaused) {
// wake() might be called while servicing a callback - ignore the next
// pause time and call processAudioBuffer.
mIgnoreNextPausedInt = true;
if (mPausedInt && mPausedNs > 0) {
// audio track is active and internally paused with timeout.
mPausedInt = false;
mMyCond.signal();
}
}
}
void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns)
{
AutoMutex _l(mMyLock);
mPausedInt = true;
mPausedNs = ns;
}
} // namespace android