You can not select more than 25 topics
Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
3032 lines
111 KiB
3032 lines
111 KiB
/*
|
|
**
|
|
** 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 const int32_t NANOS_PER_SECOND = 1000000000;
|
|
|
|
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;
|
|
}
|
|
|
|
static inline nsecs_t convertTimespecToNs(const struct timespec &tv)
|
|
{
|
|
return tv.tv_sec * (long long)NANOS_PER_SECOND + tv.tv_nsec;
|
|
}
|
|
|
|
// 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 /*, uint32_t notificationsPerBufferReq*/)
|
|
{
|
|
// Ensure that buffer depth covers at least audio hardware latency
|
|
uint32_t minBufCount = afLatencyMs / ((1000 * afFrameCount) / afSampleRate);
|
|
if (minBufCount < 2) {
|
|
minBufCount = 2;
|
|
}
|
|
#if 0
|
|
// The notificationsPerBufferReq parameter is not yet used for non-fast tracks,
|
|
// but keeping the code here to make it easier to add later.
|
|
if (minBufCount < notificationsPerBufferReq) {
|
|
minBufCount = notificationsPerBufferReq;
|
|
}
|
|
#endif
|
|
ALOGV("calculateMinFrameCount afLatency %u afFrameCount %u afSampleRate %u "
|
|
"sampleRate %u speed %f minBufCount: %u" /*" notificationsPerBufferReq %u"*/,
|
|
afLatencyMs, afFrameCount, afSampleRate, sampleRate, speed, minBufCount
|
|
/*, notificationsPerBufferReq*/);
|
|
return minBufCount * sourceFramesNeededWithTimestretch(
|
|
sampleRate, afFrameCount, afSampleRate, speed);
|
|
}
|
|
|
|
// static
|
|
status_t AudioTrack::getMinFrameCount(
|
|
size_t* frameCount,
|
|
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
|
|
/*, 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;
|
|
}
|
|
|
|
// ---------------------------------------------------------------------------
|
|
|
|
AudioTrack::AudioTrack()
|
|
: mStatus(NO_INIT),
|
|
mState(STATE_STOPPED),
|
|
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
|
|
mPreviousSchedulingGroup(SP_DEFAULT),
|
|
mPausedPosition(0),
|
|
mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
|
|
mPortId(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)
|
|
: mStatus(NO_INIT),
|
|
mState(STATE_STOPPED),
|
|
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
|
|
mPreviousSchedulingGroup(SP_DEFAULT),
|
|
mPausedPosition(0),
|
|
mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
|
|
mPortId(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, maxRequiredSpeed);
|
|
}
|
|
|
|
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),
|
|
mPortId(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, maxRequiredSpeed);
|
|
}
|
|
|
|
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,
|
|
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)
|
|
{
|
|
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;
|
|
|
|
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);
|
|
}
|
|
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);
|
|
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_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) {
|
|
return BAD_VALUE;
|
|
}
|
|
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);
|
|
return BAD_VALUE;
|
|
}
|
|
if (frameCount > 0) {
|
|
ALOGE("notificationFrames=%d not permitted with non-zero frameCount=%zu",
|
|
notificationFrames, frameCount);
|
|
return BAD_VALUE;
|
|
}
|
|
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;
|
|
if (sessionId == AUDIO_SESSION_ALLOCATE) {
|
|
mSessionId = (audio_session_t) AudioSystem::newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
|
|
} else {
|
|
mSessionId = sessionId;
|
|
}
|
|
int callingpid = IPCThreadState::self()->getCallingPid();
|
|
int mypid = getpid();
|
|
if (uid == AUDIO_UID_INVALID || (callingpid != mypid)) {
|
|
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_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;
|
|
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;
|
|
mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID;
|
|
mStartTs.mPosition = 0;
|
|
mUnderrunCountOffset = 0;
|
|
mFramesWritten = 0;
|
|
mFramesWrittenServerOffset = 0;
|
|
mFramesWrittenAtRestore = -1; // -1 is a unique initializer.
|
|
mVolumeHandler = new VolumeHandler();
|
|
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();
|
|
|
|
// save start timestamp
|
|
if (isOffloadedOrDirect_l()) {
|
|
if (getTimestamp_l(mStartTs) != OK) {
|
|
mStartTs.mPosition = 0;
|
|
}
|
|
} else {
|
|
if (getTimestamp_l(&mStartEts) != OK) {
|
|
mStartEts.clear();
|
|
}
|
|
}
|
|
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);
|
|
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).
|
|
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_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->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 (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)) {
|
|
ALOGV("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)) {
|
|
ALOGV("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) {
|
|
ALOGV("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) {
|
|
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;
|
|
}
|
|
|
|
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);
|
|
}
|
|
}
|
|
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;
|
|
|
|
// mFlags (not mOrigFlags) is modified depending on whether fast request is accepted.
|
|
// After fast request is denied, we will request again if IAudioTrack is re-created.
|
|
|
|
status_t status;
|
|
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
|
|
config.sample_rate = mSampleRate;
|
|
config.channel_mask = mChannelMask;
|
|
config.format = mFormat;
|
|
config.offload_info = mOffloadInfoCopy;
|
|
status = AudioSystem::getOutputForAttr(attr, &output,
|
|
mSessionId, &streamType, mClientUid,
|
|
&config,
|
|
mFlags, mSelectedDeviceId, &mPortId);
|
|
|
|
if (status != NO_ERROR || output == AUDIO_IO_HANDLE_NONE) {
|
|
ALOGE("Could not get audio output for session %d, stream type %d, usage %d, sample rate %u,"
|
|
" format %#x, channel mask %#x, flags %#x",
|
|
mSessionId, streamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask,
|
|
mFlags);
|
|
return BAD_VALUE;
|
|
}
|
|
{
|
|
// Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger,
|
|
// we must release it ourselves if anything goes wrong.
|
|
|
|
// Not all of these values are needed under all conditions, but it is easier to get them all
|
|
status = AudioSystem::getLatency(output, &mAfLatency);
|
|
if (status != NO_ERROR) {
|
|
ALOGE("getLatency(%d) failed status %d", output, status);
|
|
goto release;
|
|
}
|
|
ALOGV("createTrack_l() output %d afLatency %u", output, mAfLatency);
|
|
|
|
status = AudioSystem::getFrameCount(output, &mAfFrameCount);
|
|
if (status != NO_ERROR) {
|
|
ALOGE("getFrameCount(output=%d) status %d", output, status);
|
|
goto release;
|
|
}
|
|
|
|
// TODO consider making this a member variable if there are other uses for it later
|
|
size_t afFrameCountHAL;
|
|
status = AudioSystem::getFrameCountHAL(output, &afFrameCountHAL);
|
|
if (status != NO_ERROR) {
|
|
ALOGE("getFrameCountHAL(output=%d) status %d", output, status);
|
|
goto release;
|
|
}
|
|
ALOG_ASSERT(afFrameCountHAL > 0);
|
|
|
|
status = AudioSystem::getSamplingRate(output, &mAfSampleRate);
|
|
if (status != NO_ERROR) {
|
|
ALOGE("getSamplingRate(output=%d) status %d", output, status);
|
|
goto release;
|
|
}
|
|
if (mSampleRate == 0) {
|
|
mSampleRate = mAfSampleRate;
|
|
mOriginalSampleRate = mAfSampleRate;
|
|
}
|
|
|
|
// Client can only express a preference for FAST. Server will perform additional tests.
|
|
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
|
|
bool useCaseAllowed =
|
|
// 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) ||
|
|
// use case 4: synchronous write
|
|
((mTransfer == TRANSFER_SYNC) && mThreadCanCallJava);
|
|
// sample rates must also match
|
|
bool fastAllowed = useCaseAllowed && (mSampleRate == mAfSampleRate);
|
|
if (!fastAllowed) {
|
|
ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client; transfer %d, "
|
|
"track %u Hz, output %u Hz",
|
|
mTransfer, mSampleRate, mAfSampleRate);
|
|
mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
|
|
}
|
|
}
|
|
|
|
mNotificationFramesAct = mNotificationFramesReq;
|
|
|
|
size_t frameCount = mReqFrameCount;
|
|
if (!audio_has_proportional_frames(mFormat)) {
|
|
|
|
if (mSharedBuffer != 0) {
|
|
// Same comment as below about ignoring frameCount parameter for set()
|
|
frameCount = mSharedBuffer->size();
|
|
} else if (frameCount == 0) {
|
|
frameCount = mAfFrameCount;
|
|
}
|
|
if (mNotificationFramesAct != frameCount) {
|
|
mNotificationFramesAct = frameCount;
|
|
}
|
|
} else if (mSharedBuffer != 0) {
|
|
// FIXME: Ensure client side memory buffers need
|
|
// not have additional alignment beyond sample
|
|
// (e.g. 16 bit stereo accessed as 32 bit frame).
|
|
size_t alignment = audio_bytes_per_sample(mFormat);
|
|
if (alignment & 1) {
|
|
// for AUDIO_FORMAT_PCM_24_BIT_PACKED (not exposed through Java).
|
|
alignment = 1;
|
|
}
|
|
if (mChannelCount > 1) {
|
|
// More than 2 channels does not require stronger alignment than stereo
|
|
alignment <<= 1;
|
|
}
|
|
if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) {
|
|
ALOGE("Invalid buffer alignment: address %p, channel count %u",
|
|
mSharedBuffer->pointer(), mChannelCount);
|
|
status = BAD_VALUE;
|
|
goto release;
|
|
}
|
|
|
|
// When initializing a shared buffer AudioTrack via constructors,
|
|
// there's no frameCount parameter.
|
|
// But when initializing a shared buffer AudioTrack via set(),
|
|
// there _is_ a frameCount parameter. We silently ignore it.
|
|
frameCount = mSharedBuffer->size() / mFrameSize;
|
|
} else {
|
|
size_t minFrameCount = 0;
|
|
// For fast tracks the frame count calculations and checks are mostly done by server,
|
|
// but we try to respect the application's request for notifications per buffer.
|
|
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
|
|
if (mNotificationsPerBufferReq > 0) {
|
|
// Avoid possible arithmetic overflow during multiplication.
|
|
// mNotificationsPerBuffer is clamped to a small integer earlier, so it is unlikely.
|
|
if (mNotificationsPerBufferReq > SIZE_MAX / afFrameCountHAL) {
|
|
ALOGE("Requested notificationPerBuffer=%u ignored for HAL frameCount=%zu",
|
|
mNotificationsPerBufferReq, afFrameCountHAL);
|
|
} else {
|
|
minFrameCount = afFrameCountHAL * mNotificationsPerBufferReq;
|
|
}
|
|
}
|
|
} else {
|
|
// for normal tracks precompute the frame count based on speed.
|
|
const float speed = !isPurePcmData_l() || isOffloadedOrDirect_l() ? 1.0f :
|
|
max(mMaxRequiredSpeed, mPlaybackRate.mSpeed);
|
|
minFrameCount = calculateMinFrameCount(
|
|
mAfLatency, mAfFrameCount, mAfSampleRate, mSampleRate,
|
|
speed /*, 0 mNotificationsPerBufferReq*/);
|
|
}
|
|
if (frameCount < minFrameCount) {
|
|
frameCount = minFrameCount;
|
|
}
|
|
}
|
|
|
|
audio_output_flags_t flags = mFlags;
|
|
|
|
pid_t tid = -1;
|
|
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
|
|
if (mAudioTrackThread != 0 && !mThreadCanCallJava) {
|
|
tid = mAudioTrackThread->getTid();
|
|
}
|
|
}
|
|
|
|
size_t temp = frameCount; // temp may be replaced by a revised value of frameCount,
|
|
// but we will still need the original value also
|
|
audio_session_t originalSessionId = mSessionId;
|
|
sp<IAudioTrack> track = audioFlinger->createTrack(streamType,
|
|
mSampleRate,
|
|
mFormat,
|
|
mChannelMask,
|
|
&temp,
|
|
&flags,
|
|
mSharedBuffer,
|
|
output,
|
|
mClientPid,
|
|
tid,
|
|
&mSessionId,
|
|
mClientUid,
|
|
&status,
|
|
mPortId);
|
|
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.
|
|
|
|
// FIXME compare to AudioRecord
|
|
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 (flags & AUDIO_OUTPUT_FLAG_FAST) {
|
|
ALOGI("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu -> %zu", frameCount, temp);
|
|
if (!mThreadCanCallJava) {
|
|
mAwaitBoost = true;
|
|
}
|
|
} else {
|
|
ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu -> %zu", frameCount,
|
|
temp);
|
|
}
|
|
}
|
|
mFlags = flags;
|
|
|
|
// Make sure that application is notified with sufficient margin before underrun.
|
|
// The client can divide the AudioTrack buffer into sub-buffers,
|
|
// and expresses its desire to server as the notification frame count.
|
|
if (mSharedBuffer == 0 && audio_is_linear_pcm(mFormat)) {
|
|
size_t maxNotificationFrames;
|
|
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
|
|
// notify every HAL buffer, regardless of the size of the track buffer
|
|
maxNotificationFrames = afFrameCountHAL;
|
|
} else {
|
|
// For normal tracks, use at least double-buffering if no sample rate conversion,
|
|
// or at least triple-buffering if there is sample rate conversion
|
|
const int nBuffering = mOriginalSampleRate == mAfSampleRate ? 2 : 3;
|
|
maxNotificationFrames = frameCount / nBuffering;
|
|
}
|
|
if (mNotificationFramesAct == 0 || mNotificationFramesAct > maxNotificationFrames) {
|
|
if (mNotificationFramesAct == 0) {
|
|
ALOGD("Client defaulted notificationFrames to %zu for frameCount %zu",
|
|
maxNotificationFrames, frameCount);
|
|
} else {
|
|
ALOGW("Client adjusted notificationFrames from %u to %zu for frameCount %zu",
|
|
mNotificationFramesAct, maxNotificationFrames, frameCount);
|
|
}
|
|
mNotificationFramesAct = (uint32_t) maxNotificationFrames;
|
|
}
|
|
}
|
|
|
|
// 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, 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;
|
|
|
|
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) {
|
|
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;
|
|
}
|
|
|
|
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();
|
|
}
|
|
|
|
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) {
|
|
// 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("restoreTrack_l() failed status %d", result);
|
|
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) 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
|
|
/*, 0 mNotificationsPerBufferReq*/);
|
|
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);
|
|
}
|
|
|
|
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) {
|
|
// 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 = convertTimespecToNs(timestamp.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.tv_sec = limit / NANOS_PER_SECOND;
|
|
timestamp.mTime.tv_nsec = limit % NANOS_PER_SECOND; // compiler opt.
|
|
}
|
|
}
|
|
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 (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.
|
|
// 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) {
|
|
if (previousTimestampValid) {
|
|
const int64_t previousTimeNanos = convertTimespecToNs(mPreviousTimestamp.mTime);
|
|
const int64_t currentTimeNanos = convertTimespecToNs(timestamp.mTime);
|
|
if (currentTimeNanos < previousTimeNanos) {
|
|
ALOGW("retrograde timestamp time corrected, %lld < %lld",
|
|
(long long)currentTimeNanos, (long long)previousTimeNanos);
|
|
timestamp.mTime = mPreviousTimestamp.mTime;
|
|
}
|
|
|
|
// 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 =
|
|
convertTimespecToNs(timestamp.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
|
|
{
|
|
|
|
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::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 == 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;
|
|
}
|
|
|
|
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 mStartUs > 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) {
|
|
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
|