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android / platform / frameworks / av / 42a25c46b844518ff0d0b920c20c519e1417be69 / . / media / libmedia / AudioTrack.cpp
blob: ff5fe1d05ebdfcb404e07e33ae48452d1361d8a8 [file] [log] [blame]
/*
**
** Copyright 2007, The Android Open Source Project
**
** Licensed under the Apache License, Version 2.0 (the "License");
** you may not use this file except in compliance with the License.
** You may obtain a copy of the License at
**
** http://www.apache.org/licenses/LICENSE-2.0
**
** Unless required by applicable law or agreed to in writing, software
** distributed under the License is distributed on an "AS IS" BASIS,
** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
** See the License for the specific language governing permissions and
** limitations under the License.
*/
//#define LOG_NDEBUG 0
#define LOG_TAG "AudioTrack"
#include <inttypes.h>
#include <math.h>
#include <sys/resource.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/AudioPolicyHelper.h>
#include <media/AudioResamplerPublic.h>
#define WAIT_PERIOD_MS 10
#define WAIT_STREAM_END_TIMEOUT_SEC 120
static const int kMaxLoopCountNotifications = 32;
namespace android {
// ---------------------------------------------------------------------------
// 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;
}
// 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;
}
// Must match similar computation in createTrack_l in Threads.cpp.
// TODO: Move to a common library
static size_t calculateMinFrameCount(
uint32_t afLatencyMs, uint32_t afFrameCount, uint32_t afSampleRate,
uint32_t sampleRate, float speed)
{
// Ensure that buffer depth covers at least audio hardware latency
uint32_t minBufCount = afLatencyMs / ((1000 * afFrameCount) / afSampleRate);
if (minBufCount < 2) {
minBufCount = 2;
}
ALOGV("calculateMinFrameCount afLatency %u afFrameCount %u afSampleRate %u "
"sampleRate %u speed %f minBufCount: %u",
afLatencyMs, afFrameCount, afSampleRate, sampleRate, speed, minBufCount);
return minBufCount * sourceFramesNeededWithTimestretch(
sampleRate, afFrameCount, afSampleRate, speed);
}
// 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 = calculateMinFrameCount(afLatency, afFrameCount, afSampleRate, sampleRate, 1.0f);
// 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;
}
// ---------------------------------------------------------------------------
AudioTrack::AudioTrack()
: mStatus(NO_INIT),
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT),
mPausedPosition(0),
mSelectedDeviceId(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,
uint32_t notificationFrames,
int sessionId,
transfer_type transferType,
const audio_offload_info_t *offloadInfo,
int uid,
pid_t pid,
const audio_attributes_t* pAttributes,
bool doNotReconnect)
: mStatus(NO_INIT),
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT),
mPausedPosition(0),
mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE)
{
mStatus = set(streamType, sampleRate, format, channelMask,
frameCount, flags, cbf, user, notificationFrames,
0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType,
offloadInfo, uid, pid, pAttributes, doNotReconnect);
}
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,
uint32_t notificationFrames,
int sessionId,
transfer_type transferType,
const audio_offload_info_t *offloadInfo,
int uid,
pid_t pid,
const audio_attributes_t* pAttributes,
bool doNotReconnect)
: mStatus(NO_INIT),
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT),
mPausedPosition(0),
mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE)
{
mStatus = set(streamType, sampleRate, format, channelMask,
0 /*frameCount*/, flags, cbf, user, notificationFrames,
sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo,
uid, pid, pAttributes, doNotReconnect);
}
AudioTrack::~AudioTrack()
{
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(mDeviceCallback, 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,
uint32_t notificationFrames,
const sp<IMemory>& sharedBuffer,
bool threadCanCallJava,
int sessionId,
transfer_type transferType,
const audio_offload_info_t *offloadInfo,
int uid,
pid_t pid,
const audio_attributes_t* pAttributes,
bool doNotReconnect)
{
ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
"flags #%x, notificationFrames %u, sessionId %d, transferType %d, uid %d, pid %d",
streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames,
sessionId, transferType, uid, pid);
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");
return BAD_VALUE;
}
break;
case TRANSFER_OBTAIN:
case TRANSFER_SYNC:
if (sharedBuffer != 0) {
ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0");
return BAD_VALUE;
}
break;
case TRANSFER_SHARED:
if (sharedBuffer == 0) {
ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0");
return BAD_VALUE;
}
break;
default:
ALOGE("Invalid transfer type %d", transferType);
return BAD_VALUE;
}
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");
return INVALID_OPERATION;
}
// 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);
return BAD_VALUE;
}
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);
}
}
// these below should probably come from the audioFlinger too...
if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
}
// validate parameters
if (!audio_is_valid_format(format)) {
ALOGE("Invalid format %#x", format);
return BAD_VALUE;
}
mFormat = format;
if (!audio_is_output_channel(channelMask)) {
ALOGE("Invalid channel mask %#x", channelMask);
return BAD_VALUE;
}
mChannelMask = channelMask;
uint32_t 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_is_linear_pcm(format)) {
mFrameSize = channelCount * audio_bytes_per_sample(format);
} else {
mFrameSize = sizeof(uint8_t);
}
} else {
ALOG_ASSERT(audio_is_linear_pcm(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) {
return BAD_VALUE;
}
mSampleRate = sampleRate;
mOriginalSampleRate = sampleRate;
mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
// 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;
}
mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f;
mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f;
mSendLevel = 0.0f;
// mFrameCount is initialized in createTrack_l
mReqFrameCount = frameCount;
mNotificationFramesReq = notificationFrames;
mNotificationFramesAct = 0;
if (sessionId == AUDIO_SESSION_ALLOCATE) {
mSessionId = AudioSystem::newAudioUniqueId();
} else {
mSessionId = sessionId;
}
int callingpid = IPCThreadState::self()->getCallingPid();
int mypid = getpid();
if (uid == -1 || (callingpid != mypid)) {
mClientUid = IPCThreadState::self()->getCallingUid();
} else {
mClientUid = uid;
}
if (pid == -1 || (callingpid != mypid)) {
mClientPid = callingpid;
} else {
mClientPid = pid;
}
mAuxEffectId = 0;
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_t status = createTrack_l();
if (status != NO_ERROR) {
if (mAudioTrackThread != 0) {
mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
mAudioTrackThread->requestExitAndWait();
mAudioTrackThread.clear();
}
return status;
}
mStatus = NO_ERROR;
mState = STATE_STOPPED;
mUserData = user;
mLoopCount = 0;
mLoopStart = 0;
mLoopEnd = 0;
mLoopCountNotified = 0;
mMarkerPosition = 0;
mMarkerReached = false;
mNewPosition = 0;
mUpdatePeriod = 0;
mPosition = 0;
mReleased = 0;
mStartUs = 0;
AudioSystem::acquireAudioSessionId(mSessionId, mClientPid);
mSequence = 1;
mObservedSequence = mSequence;
mInUnderrun = false;
mPreviousTimestampValid = false;
mTimestampStartupGlitchReported = false;
mRetrogradeMotionReported = false;
return NO_ERROR;
}
// -------------------------------------------------------------------------
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();
if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) {
// reset current position as seen by client to 0
mPosition = 0;
mPreviousTimestampValid = false;
mTimestampStartupGlitchReported = false;
mRetrogradeMotionReported = false;
// If previousState == STATE_STOPPED, we reactivate markers (mMarkerPosition != 0)
// as the position is reset to 0. This is legacy behavior. This is not done
// in stop() to avoid a race condition where the last marker event is issued twice.
// Note: the if is technically unnecessary because previousState == STATE_FLUSHED
// is only for streaming tracks, and mMarkerReached is already set to false.
if (previousState == STATE_STOPPED) {
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).
mStartUs = getNowUs();
// 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_DISABLED, &mCblk->mFlags);
sp<AudioTrackThread> t = mAudioTrackThread;
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);
}
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");
}
if (status != NO_ERROR) {
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;
mReleased = 0;
}
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 || mState == STATE_FLUSHED) {
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 (mIsTimed || 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
{
if (mIsTimed) {
return 0;
}
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
{
if (mIsTimed) {
return 0;
}
return mOriginalSampleRate;
}
status_t AudioTrack::setPlaybackRate(const AudioPlaybackRate &playbackRate)
{
AutoMutex lock(mLock);
if (isAudioPlaybackRateEqual(playbackRate, mPlaybackRate)) {
return NO_ERROR;
}
if (mIsTimed || isOffloadedOrDirect_l()) {
return INVALID_OPERATION;
}
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
return INVALID_OPERATION;
}
// 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;
if (!isAudioPlaybackRateValid(playbackRateTemp)) {
return BAD_VALUE;
}
// Check if the buffer size is compatible.
if (!isSampleRateSpeedAllowed_l(effectiveRate, effectiveSpeed)) {
ALOGV("setPlaybackRate(%f, %f) failed", playbackRate.mSpeed, playbackRate.mPitch);
return BAD_VALUE;
}
// Check resampler ratios are within bounds
if (effectiveRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
ALOGV("setPlaybackRate(%f, %f) failed. Resample rate exceeds max accepted value",
playbackRate.mSpeed, playbackRate.mPitch);
return BAD_VALUE;
}
if (effectiveRate * AUDIO_RESAMPLER_UP_RATIO_MAX < mSampleRate) {
ALOGV("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;
}
status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
{
if (mSharedBuffer == 0 || mIsTimed || 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);
*marker = mMarkerPosition;
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 || mIsTimed || 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);
if (isOffloadedOrDirect_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();
}
return NO_ERROR;
}
status_t AudioTrack::getBufferPosition(uint32_t *position)
{
if (mSharedBuffer == 0 || mIsTimed) {
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 || mIsTimed || 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;
android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
}
return NO_ERROR;
}
audio_port_handle_t AudioTrack::getOutputDevice() {
AutoMutex lock(mLock);
return mSelectedDeviceId;
}
audio_port_handle_t AudioTrack::getRoutedDeviceId() {
AutoMutex lock(mLock);
if (mOutput == AUDIO_IO_HANDLE_NONE) {
return AUDIO_PORT_HANDLE_NONE;
}
return AudioSystem::getDeviceIdForIo(mOutput);
}
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;
}
// -------------------------------------------------------------------------
// must be called with mLock held
status_t AudioTrack::createTrack_l()
{
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
if (audioFlinger == 0) {
ALOGE("Could not get audioflinger");
return NO_INIT;
}
if (mDeviceCallback != 0 && mOutput != AUDIO_IO_HANDLE_NONE) {
AudioSystem::removeAudioDeviceCallback(mDeviceCallback, mOutput);
}
audio_io_handle_t output;
audio_stream_type_t streamType = mStreamType;
audio_attributes_t *attr = (mStreamType == AUDIO_STREAM_DEFAULT) ? &mAttributes : NULL;
status_t status;
status = AudioSystem::getOutputForAttr(attr, &output,
(audio_session_t)mSessionId, &streamType, mClientUid,
mSampleRate, mFormat, mChannelMask,
mFlags, mSelectedDeviceId, mOffloadInfo);
if (status != NO_ERROR || output == AUDIO_IO_HANDLE_NONE) {
ALOGE("Could not get audio output for session %d, stream type %d, usage %d, sample rate %u, format %#x,"
" channel mask %#x, flags %#x",
mSessionId, streamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask, mFlags);
return BAD_VALUE;
}
{
// Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger,
// we must release it ourselves if anything goes wrong.
// Not all of these values are needed under all conditions, but it is easier to get them all
status = AudioSystem::getLatency(output, &mAfLatency);
if (status != NO_ERROR) {
ALOGE("getLatency(%d) failed status %d", output, status);
goto release;
}
ALOGV("createTrack_l() output %d afLatency %u", output, mAfLatency);
status = AudioSystem::getFrameCount(output, &mAfFrameCount);
if (status != NO_ERROR) {
ALOGE("getFrameCount(output=%d) status %d", output, status);
goto release;
}
status = AudioSystem::getSamplingRate(output, &mAfSampleRate);
if (status != NO_ERROR) {
ALOGE("getSamplingRate(output=%d) status %d", output, status);
goto release;
}
if (mSampleRate == 0) {
mSampleRate = mAfSampleRate;
mOriginalSampleRate = mAfSampleRate;
}
// Client decides whether the track is TIMED (see below), but 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
(mSharedBuffer != 0) ||
// use case 2: callback transfer mode
(mTransfer == TRANSFER_CALLBACK) ||
// use case 3: obtain/release mode
(mTransfer == TRANSFER_OBTAIN)) &&
// matching sample rate
(mSampleRate == mAfSampleRate))) {
ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client; transfer %d, track %u Hz, output %u Hz",
mTransfer, mSampleRate, mAfSampleRate);
// once denied, do not request again if IAudioTrack is re-created
mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
}
// The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where
// n = 1 fast track with single buffering; nBuffering is ignored
// n = 2 fast track with double buffering
// n = 2 normal track, (including those with sample rate conversion)
// n >= 3 very high latency or very small notification interval (unused).
const uint32_t nBuffering = 2;
mNotificationFramesAct = mNotificationFramesReq;
size_t frameCount = mReqFrameCount;
if (!audio_is_linear_pcm(mFormat)) {
if (mSharedBuffer != 0) {
// Same comment as below about ignoring frameCount parameter for set()
frameCount = mSharedBuffer->size();
} else if (frameCount == 0) {
frameCount = mAfFrameCount;
}
if (mNotificationFramesAct != frameCount) {
mNotificationFramesAct = frameCount;
}
} else if (mSharedBuffer != 0) {
// FIXME: Ensure client side memory buffers need
// not have additional alignment beyond sample
// (e.g. 16 bit stereo accessed as 32 bit frame).
size_t alignment = audio_bytes_per_sample(mFormat);
if (alignment & 1) {
// for AUDIO_FORMAT_PCM_24_BIT_PACKED (not exposed through Java).
alignment = 1;
}
if (mChannelCount > 1) {
// More than 2 channels does not require stronger alignment than stereo
alignment <<= 1;
}
if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) {
ALOGE("Invalid buffer alignment: address %p, channel count %u",
mSharedBuffer->pointer(), mChannelCount);
status = BAD_VALUE;
goto release;
}
// When initializing a shared buffer AudioTrack via constructors,
// there's no frameCount parameter.
// But when initializing a shared buffer AudioTrack via set(),
// there _is_ a frameCount parameter. We silently ignore it.
frameCount = mSharedBuffer->size() / mFrameSize;
} else {
// For fast tracks the frame count calculations and checks are done by server
if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
// for normal tracks precompute the frame count based on speed.
const size_t minFrameCount = calculateMinFrameCount(
mAfLatency, mAfFrameCount, mAfSampleRate, mSampleRate,
mPlaybackRate.mSpeed);
if (frameCount < minFrameCount) {
frameCount = minFrameCount;
}
}
}
IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT;
if (mIsTimed) {
trackFlags |= IAudioFlinger::TRACK_TIMED;
}
pid_t tid = -1;
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
trackFlags |= IAudioFlinger::TRACK_FAST;
if (mAudioTrackThread != 0) {
tid = mAudioTrackThread->getTid();
}
}
if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
trackFlags |= IAudioFlinger::TRACK_OFFLOAD;
}
if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
trackFlags |= IAudioFlinger::TRACK_DIRECT;
}
size_t temp = frameCount; // temp may be replaced by a revised value of frameCount,
// but we will still need the original value also
int originalSessionId = mSessionId;
sp<IAudioTrack> track = audioFlinger->createTrack(streamType,
mSampleRate,
mFormat,
mChannelMask,
&temp,
&trackFlags,
mSharedBuffer,
output,
tid,
&mSessionId,
mClientUid,
&status);
ALOGE_IF(originalSessionId != AUDIO_SESSION_ALLOCATE && mSessionId != originalSessionId,
"session ID changed from %d to %d", originalSessionId, mSessionId);
if (status != NO_ERROR) {
ALOGE("AudioFlinger could not create track, status: %d", status);
goto release;
}
ALOG_ASSERT(track != 0);
// AudioFlinger now owns the reference to the I/O handle,
// so we are no longer responsible for releasing it.
sp<IMemory> iMem = track->getCblk();
if (iMem == 0) {
ALOGE("Could not get control block");
return NO_INIT;
}
void *iMemPointer = iMem->pointer();
if (iMemPointer == NULL) {
ALOGE("Could not get control block pointer");
return NO_INIT;
}
// 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;
// note that temp is the (possibly revised) value of frameCount
if (temp < frameCount || (frameCount == 0 && temp == 0)) {
// In current design, AudioTrack client checks and ensures frame count validity before
// passing it to AudioFlinger so AudioFlinger should not return a different value except
// for fast track as it uses a special method of assigning frame count.
ALOGW("Requested frameCount %zu but received frameCount %zu", frameCount, temp);
}
frameCount = temp;
mAwaitBoost = false;
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
if (trackFlags & IAudioFlinger::TRACK_FAST) {
ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu", frameCount);
mAwaitBoost = true;
} else {
ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu", frameCount);
// once denied, do not request again if IAudioTrack is re-created
mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
}
}
if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) {
ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful");
} else {
ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server");
mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
// FIXME This is a warning, not an error, so don't return error status
//return NO_INIT;
}
}
if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
if (trackFlags & IAudioFlinger::TRACK_DIRECT) {
ALOGV("AUDIO_OUTPUT_FLAG_DIRECT successful");
} else {
ALOGW("AUDIO_OUTPUT_FLAG_DIRECT denied by server");
mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_DIRECT);
// FIXME This is a warning, not an error, so don't return error status
//return NO_INIT;
}
}
// Make sure that application is notified with sufficient margin before underrun
if (mSharedBuffer == 0 && audio_is_linear_pcm(mFormat)) {
// Theoretically double-buffering is not required for fast tracks,
// due to tighter scheduling. But in practice, to accommodate kernels with
// scheduling jitter, and apps with computation jitter, we use double-buffering
// for fast tracks just like normal streaming tracks.
if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount / nBuffering) {
mNotificationFramesAct = frameCount / nBuffering;
}
}
// We retain a copy of the I/O handle, but don't own the reference
mOutput = output;
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");
return NO_INIT;
}
}
mAudioTrack->attachAuxEffect(mAuxEffectId);
// FIXME doesn't take into account speed or future sample rate changes (until restoreTrack)
// FIXME don't believe this lie
mLatency = mAfLatency + (1000*frameCount) / mSampleRate;
mFrameCount = frameCount;
// If IAudioTrack is re-created, don't let the requested frameCount
// decrease. This can confuse clients that cache frameCount().
if (frameCount > mReqFrameCount) {
mReqFrameCount = frameCount;
}
// reset server position to 0 as we have new cblk.
mServer = 0;
// update proxy
if (mSharedBuffer == 0) {
mStaticProxy.clear();
mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize);
} else {
mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, 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);
if (mDeviceCallback != 0) {
AudioSystem::addAudioDeviceCallback(mDeviceCallback, mOutput);
}
return NO_ERROR;
}
release:
AudioSystem::releaseOutput(output, streamType, (audio_session_t)mSessionId);
if (status == NO_ERROR) {
status = NO_INIT;
}
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;
// 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) && (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
if (mState == STATE_ACTIVE) {
audio_track_cblk_t* cblk = mCblk;
if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & 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 || mIsTimed) {
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;
}
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);
}
return written;
}
// -------------------------------------------------------------------------
TimedAudioTrack::TimedAudioTrack() {
mIsTimed = true;
}
status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer)
{
AutoMutex lock(mLock);
status_t result = UNKNOWN_ERROR;
#if 1
// acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
// while we are accessing the cblk
sp<IAudioTrack> audioTrack = mAudioTrack;
sp<IMemory> iMem = mCblkMemory;
#endif
// If the track is not invalid already, try to allocate a buffer. alloc
// fails indicating that the server is dead, flag the track as invalid so
// we can attempt to restore in just a bit.
audio_track_cblk_t* cblk = mCblk;
if (!(cblk->mFlags & CBLK_INVALID)) {
result = mAudioTrack->allocateTimedBuffer(size, buffer);
if (result == DEAD_OBJECT) {
android_atomic_or(CBLK_INVALID, &cblk->mFlags);
}
}
// If the track is invalid at this point, attempt to restore it. and try the
// allocation one more time.
if (cblk->mFlags & CBLK_INVALID) {
result = restoreTrack_l("allocateTimedBuffer");
if (result == NO_ERROR) {
result = mAudioTrack->allocateTimedBuffer(size, buffer);
}
}
return result;
}
status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer,
int64_t pts)
{
status_t status = mAudioTrack->queueTimedBuffer(buffer, pts);
{
AutoMutex lock(mLock);
audio_track_cblk_t* cblk = mCblk;
// restart track if it was disabled by audioflinger due to previous underrun
if (buffer->size() != 0 && status == NO_ERROR &&
(mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) {
android_atomic_and(~CBLK_DISABLED, &cblk->mFlags);
ALOGW("queueTimedBuffer() track %p disabled, restarting", this);
// FIXME ignoring status
mAudioTrack->start();
}
}
return status;
}
status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform,
TargetTimeline target)
{
return mAudioTrack->setMediaTimeTransform(xform, target);
}
// -------------------------------------------------------------------------
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);
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
size_t position = updateAndGetPosition_l();
// Manage marker callback
bool markerReached = false;
size_t markerPosition = mMarkerPosition;
// FIXME fails for wraparound, need 64 bits
if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) {
mMarkerReached = markerReached = true;
}
// Determine number of new position callback(s) that will be needed, while locked
size_t newPosCount = 0;
size_t newPosition = mNewPosition;
size_t updatePeriod = mUpdatePeriod;
// FIXME fails for wraparound, need 64 bits
if (updatePeriod > 0 && position >= newPosition) {
newPosCount = ((position - newPosition) / 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;
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;
}
if (loopPeriod > 0 && loopPeriod < minFrames) {
// loopPeriod is already adjusted for actual position.
minFrames = loopPeriod;
}
if (updatePeriod > 0) {
minFrames = min(minFrames, uint32_t(newPosition - position));
}
// 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) {
ns = framesToNanoseconds(minFrames, sampleRate, speed) + 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;
}
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_is_linear_pcm(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_is_linear_pcm(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)
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);
// 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
}
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 the old static buffer position
size_t bufferPosition = 0;
int loopCount = 0;
if (mStaticProxy != 0) {
mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
}
// 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) {
// 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");
}
}
}
if (mState == STATE_ACTIVE) {
result = mAudioTrack->start();
}
}
if (result != NO_ERROR) {
ALOGW("restoreTrack_l() failed status %d", result);
mState = STATE_STOPPED;
mReleased = 0;
}
return result;
}
uint32_t AudioTrack::updateAndGetPosition_l()
{
// This is the sole place to read server consumed frames
uint32_t newServer = mProxy->getPosition();
int32_t delta = newServer - mServer;
mServer = newServer;
// 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.
if (delta < 0) {
ALOGE("detected illegal retrograde motion by the server: mServer advanced by %d", delta);
delta = 0;
}
return mPosition += (uint32_t) delta;
}
bool AudioTrack::isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed) const
{
// 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 =
calculateMinFrameCount(mAfLatency, mAfFrameCount, mAfSampleRate, sampleRate, speed);
ALOGV("isSampleRateSpeedAllowed_l mFrameCount %zu minFrameCount %zu",
mFrameCount, minFrameCount);
return mFrameCount >= minFrameCount;
}
status_t AudioTrack::setParameters(const String8& keyValuePairs)
{
AutoMutex lock(mLock);
return mAudioTrack->setParameters(keyValuePairs);
}
status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp)
{
AutoMutex lock(mLock);
bool previousTimestampValid = mPreviousTimestampValid;
// Set false here to cover all the error return cases.
mPreviousTimestampValid = false;
// FIXME not implemented for fast tracks; should use proxy and SSQ
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
return INVALID_OPERATION;
}
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 = mAudioTrack->getTimestamp(timestamp);
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, &timestamp.mTime);
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 (mStartUs != 0 && mSampleRate != 0) {
static const int kTimeJitterUs = 100000; // 100 ms
static const int k1SecUs = 1000000;
const int64_t timeNow = getNowUs();
if (timeNow < mStartUs + k1SecUs) { // within first second of starting
const int64_t timestampTimeUs = convertTimespecToUs(timestamp.mTime);
if (timestampTimeUs < mStartUs) {
return WOULD_BLOCK; // stale timestamp time, occurs before start.
}
const int64_t deltaTimeUs = timestampTimeUs - mStartUs;
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) {
mStartUs = 0; // don't check again, we got valid nonzero position.
}
} else {
mStartUs = 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.
if ((uint32_t) (mServer - timestamp.mPosition) > mPosition) {
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.
// If (mPosition - mServer) can be negative then should use:
// (int32_t)(mPosition - mServer)
timestamp.mPosition += mPosition - mServer;
// 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) {
if (previousTimestampValid) {
#define TIME_TO_NANOS(time) ((uint64_t)time.tv_sec * 1000000000 + time.tv_nsec)
const uint64_t previousTimeNanos = TIME_TO_NANOS(mPreviousTimestamp.mTime);
const uint64_t currentTimeNanos = TIME_TO_NANOS(timestamp.mTime);
#undef TIME_TO_NANOS
if (currentTimeNanos < previousTimeNanos) {
ALOGW("retrograde timestamp time");
// FIXME Consider blocking this from propagating upwards.
}
// Looking at signed delta will work even when the timestamps
// are wrapping around.
int32_t deltaPosition = static_cast<int32_t>(timestamp.mPosition
- mPreviousTimestamp.mPosition);
// position can bobble slightly as an artifact; this hides the bobble
static const int32_t MINIMUM_POSITION_DELTA = 8;
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 < MINIMUM_POSITION_DELTA) {
timestamp = mPreviousTimestamp; // Use last valid timestamp.
}
}
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
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
result.append(" AudioTrack::dump\n");
snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType,
mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]);
result.append(buffer);
snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%zu)\n", mFormat,
mChannelCount, mFrameCount);
result.append(buffer);
snprintf(buffer, 255, " sample rate(%u), speed(%f), status(%d)\n",
mSampleRate, mPlaybackRate.mSpeed, mStatus);
result.append(buffer);
snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency);
result.append(buffer);
::write(fd, result.string(), result.size());
return NO_ERROR;
}
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 == callback) {
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(mDeviceCallback, mOutput);
}
status = AudioSystem::addAudioDeviceCallback(callback, 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 != callback) {
ALOGW("%s removing different callback!", __FUNCTION__);
return INVALID_OPERATION;
}
if (mOutput != AUDIO_IO_HANDLE_NONE) {
AudioSystem::removeAudioDeviceCallback(mDeviceCallback, mOutput);
}
mDeviceCallback = 0;
return NO_ERROR;
}
// =========================================================================
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) {
mMyCond.wait(mMyLock);
// caller will check for exitPending()
return true;
}
if (mIgnoreNextPausedInt) {
mIgnoreNextPausedInt = false;
mPausedInt = false;
}
if (mPausedInt) {
if (mPausedNs > 0) {
(void) mMyCond.waitRelative(mMyLock, mPausedNs);
} else {
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