/* ** ** 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, 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 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& event); sp 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 mThread; /*const*/ sp mClient; // see explanation at ~TrackBase() why not const sp mCblkMemory; audio_track_cblk_t* mCblk; sp 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 >mSyncEvents; const bool mIsOut; sp 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 mServerLatencyFromTrack{}; // latency from track or server timestamp. std::atomic mServerLatencyMs{}; // last latency pushed from server thread. std::atomic 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; PatchTrackBase(sp proxy, const ThreadBase& thread, const Timeout& timeout); void setPeerTimeout(std::chrono::nanoseconds timeout); template void setPeerProxy(const sp &proxy, bool holdReference) { mPeerReferenceHold = holdReference ? proxy : nullptr; mPeerProxy = proxy.get(); } void clearPeerProxy() { mPeerReferenceHold.clear(); mPeerProxy = nullptr; } protected: const sp mProxy; sp mPeerReferenceHold; // keeps mPeerProxy alive during access. PatchProxyBufferProvider* mPeerProxy = nullptr; struct timespec mPeerTimeout{}; };