/*
**
** Copyright 2012, 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.
*/
#ifndef INCLUDING_FROM_AUDIOFLINGER_H
#error This header file should only be included from AudioFlinger.h
#endif
// base for record and playback
class TrackBase : public ExtendedAudioBufferProvider, public RefBase {
public:
enum track_state {
IDLE,
FLUSHED, // for PlaybackTracks only
STOPPED,
// next 2 states are currently used for fast tracks
// and offloaded tracks only
STOPPING_1, // waiting for first underrun
STOPPING_2, // waiting for presentation complete
RESUMING, // for PlaybackTracks only
ACTIVE,
PAUSING,
PAUSED,
STARTING_1, // for RecordTrack only
STARTING_2, // for RecordTrack only
};
// where to allocate the data buffer
enum alloc_type {
ALLOC_CBLK, // allocate immediately after control block
ALLOC_READONLY, // allocate from a separate read-only heap per thread
ALLOC_PIPE, // do not allocate; use the pipe buffer
ALLOC_LOCAL, // allocate a local buffer
ALLOC_NONE, // do not allocate:use the buffer passed to TrackBase constructor
};
enum track_type {
TYPE_DEFAULT,
TYPE_OUTPUT,
TYPE_PATCH,
};
TrackBase(ThreadBase *thread,
const sp<Client>& client,
const audio_attributes_t& mAttr,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
void *buffer,
size_t bufferSize,
audio_session_t sessionId,
pid_t creatorPid,
uid_t uid,
bool isOut,
alloc_type alloc = ALLOC_CBLK,
track_type type = TYPE_DEFAULT,
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
virtual ~TrackBase();
virtual status_t initCheck() const;
virtual status_t start(AudioSystem::sync_event_t event,
audio_session_t triggerSession) = 0;
virtual void stop() = 0;
sp<IMemory> getCblk() const { return mCblkMemory; }
audio_track_cblk_t* cblk() const { return mCblk; }
audio_session_t sessionId() const { return mSessionId; }
uid_t uid() const { return mUid; }
pid_t creatorPid() const { return mCreatorPid; }
audio_port_handle_t portId() const { return mPortId; }
virtual status_t setSyncEvent(const sp<SyncEvent>& event);
sp<IMemory> getBuffers() const { return mBufferMemory; }
void* buffer() const { return mBuffer; }
size_t bufferSize() const { return mBufferSize; }
virtual bool isFastTrack() const = 0;
virtual bool isDirect() const = 0;
bool isOutputTrack() const { return (mType == TYPE_OUTPUT); }
bool isPatchTrack() const { return (mType == TYPE_PATCH); }
bool isExternalTrack() const { return !isOutputTrack() && !isPatchTrack(); }
virtual void invalidate() { mIsInvalid = true; }
bool isInvalid() const { return mIsInvalid; }
void terminate() { mTerminated = true; }
bool isTerminated() const { return mTerminated; }
audio_attributes_t attributes() const { return mAttr; }
#ifdef TEE_SINK
void dumpTee(int fd, const std::string &reason) const {
mTee.dump(fd, reason);
}
#endif
/** returns the buffer contents size converted to time in milliseconds
* for PCM Playback or Record streaming tracks. The return value is zero for
* PCM static tracks and not defined for non-PCM tracks.
*
* This may be called without the thread lock.
*/
virtual double bufferLatencyMs() const {
return mServerProxy->framesReadySafe() * 1000 / sampleRate();
}
/** returns whether the track supports server latency computation.
* This is set in the constructor and constant throughout the track lifetime.
*/
bool isServerLatencySupported() const { return mServerLatencySupported; }
/** computes the server latency for PCM Playback or Record track
* to the device sink/source. This is the time for the next frame in the track buffer
* written or read from the server thread to the device source or sink.
*
* This may be called without the thread lock, but latencyMs and fromTrack
* may be not be synchronized. For example PatchPanel may not obtain the
* thread lock before calling.
*
* \param latencyMs on success is set to the latency in milliseconds of the
* next frame written/read by the server thread to/from the track buffer
* from the device source/sink.
* \param fromTrack on success is set to true if latency was computed directly
* from the track timestamp; otherwise set to false if latency was
* estimated from the server timestamp.
* fromTrack may be nullptr or omitted if not required.
*
* \returns OK or INVALID_OPERATION on failure.
*/
status_t getServerLatencyMs(double *latencyMs, bool *fromTrack = nullptr) const {
if (!isServerLatencySupported()) {
return INVALID_OPERATION;
}
// if no thread lock is acquired, these atomics are not
// synchronized with each other, considered a benign race.
const double serverLatencyMs = mServerLatencyMs.load();
if (serverLatencyMs == 0.) {
return INVALID_OPERATION;
}
if (fromTrack != nullptr) {
*fromTrack = mServerLatencyFromTrack.load();
}
*latencyMs = serverLatencyMs;
return OK;
}
/** computes the total client latency for PCM Playback or Record tracks
* for the next client app access to the device sink/source; i.e. the
* server latency plus the buffer latency.
*
* This may be called without the thread lock, but latencyMs and fromTrack
* may be not be synchronized. For example PatchPanel may not obtain the
* thread lock before calling.
*
* \param latencyMs on success is set to the latency in milliseconds of the
* next frame written/read by the client app to/from the track buffer
* from the device sink/source.
* \param fromTrack on success is set to true if latency was computed directly
* from the track timestamp; otherwise set to false if latency was
* estimated from the server timestamp.
* fromTrack may be nullptr or omitted if not required.
*
* \returns OK or INVALID_OPERATION on failure.
*/
status_t getTrackLatencyMs(double *latencyMs, bool *fromTrack = nullptr) const {
double serverLatencyMs;
status_t status = getServerLatencyMs(&serverLatencyMs, fromTrack);
if (status == OK) {
*latencyMs = serverLatencyMs + bufferLatencyMs();
}
return status;
}
// TODO: Consider making this external.
struct FrameTime {
int64_t frames;
int64_t timeNs;
};
// KernelFrameTime is updated per "mix" period even for non-pcm tracks.
void getKernelFrameTime(FrameTime *ft) const {
*ft = mKernelFrameTime.load();
}
audio_format_t format() const { return mFormat; }
int id() const { return mId; }
protected:
DISALLOW_COPY_AND_ASSIGN(TrackBase);
// AudioBufferProvider interface
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) = 0;
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
// ExtendedAudioBufferProvider interface is only needed for Track,
// but putting it in TrackBase avoids the complexity of virtual inheritance
virtual size_t framesReady() const { return SIZE_MAX; }
uint32_t channelCount() const { return mChannelCount; }
audio_channel_mask_t channelMask() const { return mChannelMask; }
virtual uint32_t sampleRate() const { return mSampleRate; }
bool isStopped() const {
return (mState == STOPPED || mState == FLUSHED);
}
// for fast tracks and offloaded tracks only
bool isStopping() const {
return mState == STOPPING_1 || mState == STOPPING_2;
}
bool isStopping_1() const {
return mState == STOPPING_1;
}
bool isStopping_2() const {
return mState == STOPPING_2;
}
// Upper case characters are final states.
// Lower case characters are transitory.
const char *getTrackStateString() const {
if (isTerminated()) {
return "T ";
}
switch (mState) {
case IDLE:
return "I ";
case STOPPING_1: // for Fast and Offload
return "s1";
case STOPPING_2: // for Fast and Offload
return "s2";
case STOPPED:
return "S ";
case RESUMING:
return "r ";
case ACTIVE:
return "A ";
case PAUSING:
return "p ";
case PAUSED:
return "P ";
case FLUSHED:
return "F ";
case STARTING_1: // for RecordTrack
return "r1";
case STARTING_2: // for RecordTrack
return "r2";
default:
return "? ";
}
}
bool isOut() const { return mIsOut; }
// true for Track, false for RecordTrack,
// this could be a track type if needed later
const wp<ThreadBase> mThread;
/*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const
sp<IMemory> mCblkMemory;
audio_track_cblk_t* mCblk;
sp<IMemory> mBufferMemory; // currently non-0 for fast RecordTrack only
void* mBuffer; // start of track buffer, typically in shared memory
// except for OutputTrack when it is in local memory
size_t mBufferSize; // size of mBuffer in bytes
// we don't really need a lock for these
track_state mState;
const audio_attributes_t mAttr;
const uint32_t mSampleRate; // initial sample rate only; for tracks which
// support dynamic rates, the current value is in control block
const audio_format_t mFormat;
const audio_channel_mask_t mChannelMask;
const uint32_t mChannelCount;
const size_t mFrameSize; // AudioFlinger's view of frame size in shared memory,
// where for AudioTrack (but not AudioRecord),
// 8-bit PCM samples are stored as 16-bit
const size_t mFrameCount;// size of track buffer given at createTrack() or
// createRecord(), and then adjusted as needed
const audio_session_t mSessionId;
uid_t mUid;
Vector < sp<SyncEvent> >mSyncEvents;
const bool mIsOut;
sp<ServerProxy> mServerProxy;
const int mId;
#ifdef TEE_SINK
NBAIO_Tee mTee;
#endif
bool mTerminated;
track_type mType; // must be one of TYPE_DEFAULT, TYPE_OUTPUT, TYPE_PATCH ...
audio_io_handle_t mThreadIoHandle; // I/O handle of the thread the track is attached to
audio_port_handle_t mPortId; // unique ID for this track used by audio policy
bool mIsInvalid; // non-resettable latch, set by invalidate()
bool mServerLatencySupported = false;
std::atomic<bool> mServerLatencyFromTrack{}; // latency from track or server timestamp.
std::atomic<double> mServerLatencyMs{}; // last latency pushed from server thread.
std::atomic<FrameTime> mKernelFrameTime{}; // last frame time on kernel side.
const pid_t mCreatorPid; // can be different from mclient->pid() for instance
// when created by NuPlayer on behalf of a client
};
// PatchProxyBufferProvider interface is implemented by PatchTrack and PatchRecord.
// it provides buffer access methods that map those of a ClientProxy (see AudioTrackShared.h)
class PatchProxyBufferProvider
{
public:
virtual ~PatchProxyBufferProvider() {}
virtual status_t obtainBuffer(Proxy::Buffer* buffer,
const struct timespec *requested = NULL) = 0;
virtual void releaseBuffer(Proxy::Buffer* buffer) = 0;
};
class PatchTrackBase : public PatchProxyBufferProvider
{
public:
using Timeout = std::optional<std::chrono::nanoseconds>;
PatchTrackBase(sp<ClientProxy> proxy, const ThreadBase& thread,
const Timeout& timeout);
void setPeerTimeout(std::chrono::nanoseconds timeout);
template <typename T>
void setPeerProxy(const sp<T> &proxy, bool holdReference) {
mPeerReferenceHold = holdReference ? proxy : nullptr;
mPeerProxy = proxy.get();
}
void clearPeerProxy() {
mPeerReferenceHold.clear();
mPeerProxy = nullptr;
}
protected:
const sp<ClientProxy> mProxy;
sp<RefBase> mPeerReferenceHold; // keeps mPeerProxy alive during access.
PatchProxyBufferProvider* mPeerProxy = nullptr;
struct timespec mPeerTimeout{};
};