From 9f80dd223d83d9bb9077fb6baee056cee4eaf7e5 Mon Sep 17 00:00:00 2001 From: Glenn Kasten Date: Tue, 18 Dec 2012 15:57:32 -0800 Subject: New control block for AudioTrack and AudioRecord Main differences between old and new control block: - removes the mutex, which was a potential source of priority inversion - circular indices into shared buffer, which is now always a power-of-2 size Change-Id: I4e9b7fa99858b488ac98a441fa70e31dbba1b865 --- include/media/AudioBufferProvider.h | 15 + include/media/AudioRecord.h | 251 ++++-- include/media/AudioTrack.h | 274 ++++--- include/private/media/AudioTrackShared.h | 391 +++++++--- media/libmedia/AudioRecord.cpp | 832 +++++++++++--------- media/libmedia/AudioTrack.cpp | 1219 ++++++++++++++++-------------- media/libmedia/AudioTrackShared.cpp | 716 +++++++++++++++--- media/libmedia/ToneGenerator.cpp | 4 +- services/audioflinger/AudioFlinger.h | 1 + services/audioflinger/PlaybackTracks.h | 7 +- services/audioflinger/RecordTracks.h | 1 + services/audioflinger/Threads.cpp | 59 +- services/audioflinger/TrackBase.h | 2 +- services/audioflinger/Tracks.cpp | 274 +++---- 14 files changed, 2496 insertions(+), 1550 deletions(-) diff --git a/include/media/AudioBufferProvider.h b/include/media/AudioBufferProvider.h index 43e4de7..ef392f0 100644 --- a/include/media/AudioBufferProvider.h +++ b/include/media/AudioBufferProvider.h @@ -26,6 +26,8 @@ class AudioBufferProvider { public: + // FIXME merge with AudioTrackShared::Buffer, AudioTrack::Buffer, and AudioRecord::Buffer + // and rename getNextBuffer() to obtainBuffer() struct Buffer { Buffer() : raw(NULL), frameCount(0) { } union { @@ -44,6 +46,19 @@ public: // pts is the local time when the next sample yielded by getNextBuffer // will be rendered. // Pass kInvalidPTS if the PTS is unknown or not applicable. + // On entry: + // buffer != NULL + // buffer->raw unused + // buffer->frameCount maximum number of desired frames + // On successful return: + // status NO_ERROR + // buffer->raw non-NULL pointer to buffer->frameCount contiguous available frames + // buffer->frameCount number of contiguous available frames at buffer->raw, + // 0 < buffer->frameCount <= entry value + // On error return: + // status != NO_ERROR + // buffer->raw NULL + // buffer->frameCount 0 virtual status_t getNextBuffer(Buffer* buffer, int64_t pts = kInvalidPTS) = 0; virtual void releaseBuffer(Buffer* buffer) = 0; diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h index 38c6548..81be803 100644 --- a/include/media/AudioRecord.h +++ b/include/media/AudioRecord.h @@ -14,26 +14,24 @@ * limitations under the License. */ -#ifndef AUDIORECORD_H_ -#define AUDIORECORD_H_ +#ifndef ANDROID_AUDIORECORD_H +#define ANDROID_AUDIORECORD_H -#include #include #include #include -#include -#include -#include #include namespace android { +// ---------------------------------------------------------------------------- + class audio_track_cblk_t; class AudioRecordClientProxy; // ---------------------------------------------------------------------------- -class AudioRecord : virtual public RefBase +class AudioRecord : public RefBase { public: @@ -49,6 +47,8 @@ public: // (See setMarkerPosition()). EVENT_NEW_POS = 3, // Record head is at a new position // (See setPositionUpdatePeriod()). + EVENT_NEW_IAUDIORECORD = 4, // IAudioRecord was re-created, either due to re-routing and + // voluntary invalidation by mediaserver, or mediaserver crash. }; /* Client should declare Buffer on the stack and pass address to obtainBuffer() @@ -58,11 +58,16 @@ public: class Buffer { public: + // FIXME use m prefix size_t frameCount; // number of sample frames corresponding to size; // on input it is the number of frames available, // on output is the number of frames actually drained - size_t size; // total size in bytes == frameCount * frameSize + size_t size; // input/output in bytes == frameCount * frameSize + // FIXME this is redundant with respect to frameCount, + // and TRANSFER_OBTAIN mode is broken for 8-bit data + // since we don't define the frame format + union { void* raw; short* i16; // signed 16-bit @@ -84,6 +89,7 @@ public: * - EVENT_OVERRUN: unused. * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames. * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames. + * - EVENT_NEW_IAUDIORECORD: unused. */ typedef void (*callback_t)(int event, void* user, void *info); @@ -101,20 +107,28 @@ public: audio_format_t format, audio_channel_mask_t channelMask); + /* How data is transferred from AudioRecord + */ + enum transfer_type { + TRANSFER_DEFAULT, // not specified explicitly; determine from other parameters + TRANSFER_CALLBACK, // callback EVENT_MORE_DATA + TRANSFER_OBTAIN, // FIXME deprecated: call obtainBuffer() and releaseBuffer() + TRANSFER_SYNC, // synchronous read() + }; + /* Constructs an uninitialized AudioRecord. No connection with - * AudioFlinger takes place. + * AudioFlinger takes place. Use set() after this. */ AudioRecord(); /* Creates an AudioRecord object and registers it with AudioFlinger. * Once created, the track needs to be started before it can be used. - * Unspecified values are set to the audio hardware's current - * values. + * Unspecified values are set to appropriate default values. * * Parameters: * - * inputSource: Select the audio input to record to (e.g. AUDIO_SOURCE_DEFAULT). - * sampleRate: Track sampling rate in Hz. + * inputSource: Select the audio input to record from (e.g. AUDIO_SOURCE_DEFAULT). + * sampleRate: Data sink sampling rate in Hz. * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed * 16 bits per sample). * channelMask: Channel mask. @@ -124,11 +138,13 @@ public: * be larger if the requested size is not compatible with current audio HAL * latency. Zero means to use a default value. * cbf: Callback function. If not null, this function is called periodically - * to consume new PCM data. + * to consume new PCM data and inform of marker, position updates, etc. * user: Context for use by the callback receiver. * notificationFrames: The callback function is called each time notificationFrames PCM * frames are ready in record track output buffer. * sessionId: Not yet supported. + * transferType: How data is transferred from AudioRecord. + * threadCanCallJava: Not present in parameter list, and so is fixed at false. */ AudioRecord(audio_source_t inputSource, @@ -139,22 +155,26 @@ public: callback_t cbf = NULL, void* user = NULL, int notificationFrames = 0, - int sessionId = 0); - + int sessionId = 0, + transfer_type transferType = TRANSFER_DEFAULT); /* Terminates the AudioRecord and unregisters it from AudioFlinger. * Also destroys all resources associated with the AudioRecord. */ ~AudioRecord(); - - /* Initialize an uninitialized AudioRecord. + /* Initialize an AudioRecord that was created using the AudioRecord() constructor. + * Don't call set() more than once, or after an AudioRecord() constructor that takes parameters. * Returned status (from utils/Errors.h) can be: * - NO_ERROR: successful intialization - * - INVALID_OPERATION: AudioRecord is already intitialized or record device is already in use + * - INVALID_OPERATION: AudioRecord is already initialized or record device is already in use * - BAD_VALUE: invalid parameter (channels, format, sampleRate...) * - NO_INIT: audio server or audio hardware not initialized * - PERMISSION_DENIED: recording is not allowed for the requesting process + * + * Parameters not listed in the AudioRecord constructors above: + * + * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. */ status_t set(audio_source_t inputSource = AUDIO_SOURCE_DEFAULT, uint32_t sampleRate = 0, @@ -165,30 +185,29 @@ public: void* user = NULL, int notificationFrames = 0, bool threadCanCallJava = false, - int sessionId = 0); - + int sessionId = 0, + transfer_type transferType = TRANSFER_DEFAULT); /* Result of constructing the AudioRecord. This must be checked * before using any AudioRecord API (except for set()), because using * an uninitialized AudioRecord produces undefined results. * See set() method above for possible return codes. */ - status_t initCheck() const; + status_t initCheck() const { return mStatus; } /* Returns this track's estimated latency in milliseconds. * This includes the latency due to AudioRecord buffer size, * and audio hardware driver. */ - uint32_t latency() const; + uint32_t latency() const { return mLatency; } /* getters, see constructor and set() */ - audio_format_t format() const; - uint32_t channelCount() const; - size_t frameCount() const; - size_t frameSize() const { return mFrameSize; } - audio_source_t inputSource() const; - + audio_format_t format() const { return mFormat; } + uint32_t channelCount() const { return mChannelCount; } + size_t frameCount() const { return mFrameCount; } + size_t frameSize() const { return mFrameSize; } + audio_source_t inputSource() const { return mInputSource; } /* After it's created the track is not active. Call start() to * make it active. If set, the callback will start being called. @@ -198,26 +217,29 @@ public: status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, int triggerSession = 0); - /* Stop a track. If set, the callback will cease being called and - * obtainBuffer returns STOPPED. Note that obtainBuffer() still works - * and will drain buffers until the pool is exhausted. + /* Stop a track. If set, the callback will cease being called. Note that obtainBuffer() still + * works and will drain buffers until the pool is exhausted, and then will return WOULD_BLOCK. */ void stop(); bool stopped() const; - /* Get sample rate for this record track in Hz. + /* Return the sink sample rate for this record track in Hz. + * Unlike AudioTrack, the sample rate is const after initialization, so doesn't need a lock. */ - uint32_t getSampleRate() const; + uint32_t getSampleRate() const { return mSampleRate; } /* Sets marker position. When record reaches the number of frames specified, * a callback with event type EVENT_MARKER is called. Calling setMarkerPosition * with marker == 0 cancels marker notification callback. + * To set a marker at a position which would compute as 0, + * a workaround is to the set the marker at a nearby position such as ~0 or 1. * If the AudioRecord has been opened with no callback function associated, * the operation will fail. * * Parameters: * - * marker: marker position expressed in frames. + * marker: marker position expressed in wrapping (overflow) frame units, + * like the return value of getPosition(). * * Returned status (from utils/Errors.h) can be: * - NO_ERROR: successful operation @@ -226,13 +248,13 @@ public: status_t setMarkerPosition(uint32_t marker); status_t getMarkerPosition(uint32_t *marker) const; - /* Sets position update period. Every time the number of frames specified has been recorded, * a callback with event type EVENT_NEW_POS is called. * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification * callback. * If the AudioRecord has been opened with no callback function associated, * the operation will fail. + * Extremely small values may be rounded up to a value the implementation can support. * * Parameters: * @@ -245,13 +267,13 @@ public: status_t setPositionUpdatePeriod(uint32_t updatePeriod); status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; - - /* Gets record head position. The position is the total number of frames - * recorded since record start. + /* Return the total number of frames recorded since recording started. + * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. + * It is reset to zero by stop(). * * Parameters: * - * position: Address where to return record head position within AudioRecord buffer. + * position: Address where to return record head position. * * Returned status (from utils/Errors.h) can be: * - NO_ERROR: successful operation @@ -276,38 +298,70 @@ public: * * Returned value: * AudioRecord session ID. + * + * No lock needed because session ID doesn't change after first set(). */ - int getSessionId() const; - - /* Obtains a buffer of "frameCount" frames. The buffer must be - * drained entirely, and then released with releaseBuffer(). - * If the track is stopped, obtainBuffer() returns - * STOPPED instead of NO_ERROR as long as there are buffers available, - * at which point NO_MORE_BUFFERS is returned. + int getSessionId() const { return mSessionId; } + + /* Obtains a buffer of up to "audioBuffer->frameCount" full frames. + * After draining these frames of data, the caller should release them with releaseBuffer(). + * If the track buffer is not empty, obtainBuffer() returns as many contiguous + * full frames as are available immediately. + * If the track buffer is empty and track is stopped, obtainBuffer() returns WOULD_BLOCK + * regardless of the value of waitCount. + * If the track buffer is empty and track is not stopped, obtainBuffer() blocks with a + * maximum timeout based on waitCount; see chart below. * Buffers will be returned until the pool * is exhausted, at which point obtainBuffer() will either block - * or return WOULD_BLOCK depending on the value of the "blocking" + * or return WOULD_BLOCK depending on the value of the "waitCount" * parameter. * + * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications, + * which should use read() or callback EVENT_MORE_DATA instead. + * * Interpretation of waitCount: * +n limits wait time to n * WAIT_PERIOD_MS, * -1 causes an (almost) infinite wait time, * 0 non-blocking. + * + * Buffer fields + * On entry: + * frameCount number of frames requested + * After error return: + * frameCount 0 + * size 0 + * raw undefined + * After successful return: + * frameCount actual number of frames available, <= number requested + * size actual number of bytes available + * raw pointer to the buffer */ - enum { - NO_MORE_BUFFERS = 0x80000001, // same name in AudioFlinger.h, ok to be different value - STOPPED = 1 - }; + /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */ + status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount) + __attribute__((__deprecated__)); - status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount); +private: + /* New internal API. + * If nonContig is non-NULL, it is an output parameter that will be set to the number of + * additional non-contiguous frames that are available immediately. + * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), + * in case the requested amount of frames is in two or more non-contiguous regions. + * FIXME requested and elapsed are both relative times. Consider changing to absolute time. + */ + status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, + struct timespec *elapsed = NULL, size_t *nonContig = NULL); +public: - /* Release an emptied buffer of "frameCount" frames for AudioFlinger to re-fill. */ + /* Release an emptied buffer of "audioBuffer->frameCount" frames for AudioFlinger to re-fill. */ + // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed void releaseBuffer(Buffer* audioBuffer); - /* As a convenience we provide a read() interface to the audio buffer. - * This is implemented on top of obtainBuffer/releaseBuffer. + * Input parameter 'size' is in byte units. + * This is implemented on top of obtainBuffer/releaseBuffer. For best + * performance use callbacks. Returns actual number of bytes read >= 0, + * or a negative status code. */ ssize_t read(void* buffer, size_t size); @@ -336,68 +390,113 @@ private: void pause(); // suspend thread from execution at next loop boundary void resume(); // allow thread to execute, if not requested to exit + void pauseConditional(); + // like pause(), but only if prior resume() wasn't latched private: friend class AudioRecord; virtual bool threadLoop(); - AudioRecord& mReceiver; + AudioRecord& mReceiver; virtual ~AudioRecordThread(); Mutex mMyLock; // Thread::mLock is private Condition mMyCond; // Thread::mThreadExitedCondition is private bool mPaused; // whether thread is currently paused + bool mResumeLatch; // whether next pauseConditional() will be a nop }; // body of AudioRecordThread::threadLoop() - bool processAudioBuffer(const sp& thread); - + // returns the maximum amount of time before we would like to run again, where: + // 0 immediately + // > 0 no later than this many nanoseconds from now + // NS_WHENEVER still active but no particular deadline + // NS_INACTIVE inactive so don't run again until re-started + // NS_NEVER never again + static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; + nsecs_t processAudioBuffer(const sp& thread); + + // caller must hold lock on mLock for all _l methods status_t openRecord_l(uint32_t sampleRate, audio_format_t format, size_t frameCount, - audio_io_handle_t input); + audio_io_handle_t input, + size_t epoch); + audio_io_handle_t getInput_l(); - status_t restoreRecord_l(audio_track_cblk_t*& cblk); + + // FIXME enum is faster than strcmp() for parameter 'from' + status_t restoreRecord_l(const char *from); sp mAudioRecordThread; mutable Mutex mLock; - bool mActive; // protected by mLock + // Current client state: false = stopped, true = active. Protected by mLock. If more states + // are added, consider changing this to enum State { ... } mState as in AudioTrack. + bool mActive; // for client callback handler callback_t mCbf; // callback handler for events, or NULL - void* mUserData; + void* mUserData; // for client callback handler // for notification APIs - uint32_t mNotificationFrames; - uint32_t mRemainingFrames; - uint32_t mMarkerPosition; // in frames + uint32_t mNotificationFrames; // frames between each notification callback + bool mRefreshRemaining; // processAudioBuffer() should refresh next 2 + + // These are private to processAudioBuffer(), and are not protected by a lock + uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() + bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() + int mObservedSequence; // last observed value of mSequence + + uint32_t mMarkerPosition; // in wrapping (overflow) frame units bool mMarkerReached; uint32_t mNewPosition; // in frames - uint32_t mUpdatePeriod; // in ms + uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS + + status_t mStatus; // constant after constructor or set() uint32_t mSampleRate; size_t mFrameCount; audio_format_t mFormat; - uint8_t mChannelCount; + uint32_t mChannelCount; size_t mFrameSize; // app-level frame size == AudioFlinger frame size audio_source_t mInputSource; - status_t mStatus; - uint32_t mLatency; + uint32_t mLatency; // in ms audio_channel_mask_t mChannelMask; - audio_io_handle_t mInput; // returned by AudioSystem::getInput() int mSessionId; + transfer_type mTransfer; + + audio_io_handle_t mInput; // returned by AudioSystem::getInput() // may be changed if IAudioRecord object is re-created sp mAudioRecord; sp mCblkMemory; - audio_track_cblk_t* mCblk; - void* mBuffers; // starting address of buffers in shared memory + audio_track_cblk_t* mCblk; // re-load after mLock.unlock() - int mPreviousPriority; // before start() + int mPreviousPriority; // before start() SchedPolicy mPreviousSchedulingGroup; - AudioRecordClientProxy* mProxy; + + // The proxy should only be referenced while a lock is held because the proxy isn't + // multi-thread safe. + // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, + // provided that the caller also holds an extra reference to the proxy and shared memory to keep + sp mProxy; + + bool mInOverrun; // whether recorder is currently in overrun state + +private: + class DeathNotifier : public IBinder::DeathRecipient { + public: + DeathNotifier(AudioRecord* audioRecord) : mAudioRecord(audioRecord) { } + protected: + virtual void binderDied(const wp& who); + private: + const wp mAudioRecord; + }; + + sp mDeathNotifier; + uint32_t mSequence; // incremented for each new IAudioRecord attempt }; }; // namespace android -#endif /*AUDIORECORD_H_*/ +#endif // ANDROID_AUDIORECORD_H diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h index 8dbc9ee..e9bb76a 100644 --- a/include/media/AudioTrack.h +++ b/include/media/AudioTrack.h @@ -17,18 +17,9 @@ #ifndef ANDROID_AUDIOTRACK_H #define ANDROID_AUDIOTRACK_H -#include -#include - -#include -#include -#include - -#include -#include -#include -#include #include +#include +#include #include namespace android { @@ -37,10 +28,11 @@ namespace android { class audio_track_cblk_t; class AudioTrackClientProxy; +class StaticAudioTrackClientProxy; // ---------------------------------------------------------------------------- -class AudioTrack : virtual public RefBase +class AudioTrack : public RefBase { public: enum channel_index { @@ -49,7 +41,7 @@ public: RIGHT = 1 }; - /* Events used by AudioTrack callback function (audio_track_cblk_t). + /* Events used by AudioTrack callback function (callback_t). * Keep in sync with frameworks/base/media/java/android/media/AudioTrack.java NATIVE_EVENT_*. */ enum event_type { @@ -64,7 +56,10 @@ public: // (See setMarkerPosition()). EVENT_NEW_POS = 4, // Playback head is at a new position // (See setPositionUpdatePeriod()). - EVENT_BUFFER_END = 5 // Playback head is at the end of the buffer. + EVENT_BUFFER_END = 5, // Playback head is at the end of the buffer. + // Not currently used by android.media.AudioTrack. + EVENT_NEW_IAUDIOTRACK = 6, // IAudioTrack was re-created, either due to re-routing and + // voluntary invalidation by mediaserver, or mediaserver crash. }; /* Client should declare Buffer on the stack and pass address to obtainBuffer() @@ -74,19 +69,23 @@ public: class Buffer { public: + // FIXME use m prefix size_t frameCount; // number of sample frames corresponding to size; // on input it is the number of frames desired, // on output is the number of frames actually filled - size_t size; // input/output in byte units + size_t size; // input/output in bytes == frameCount * frameSize + // FIXME this is redundant with respect to frameCount, + // and TRANSFER_OBTAIN mode is broken for 8-bit data + // since we don't define the frame format + union { void* raw; - short* i16; // signed 16-bit - int8_t* i8; // unsigned 8-bit, offset by 0x80 + short* i16; // signed 16-bit + int8_t* i8; // unsigned 8-bit, offset by 0x80 }; }; - /* As a convenience, if a callback is supplied, a handler thread * is automatically created with the appropriate priority. This thread * invokes the callback when a new buffer becomes available or various conditions occur. @@ -100,9 +99,10 @@ public: * written. * - EVENT_UNDERRUN: unused. * - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining. - * - EVENT_MARKER: pointer to an uint32_t containing the marker position in frames. - * - EVENT_NEW_POS: pointer to an uint32_t containing the new position in frames. + * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames. + * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames. * - EVENT_BUFFER_END: unused. + * - EVENT_NEW_IAUDIOTRACK: unused. */ typedef void (*callback_t)(int event, void* user, void *info); @@ -114,9 +114,19 @@ public: * - NO_INIT: audio server or audio hardware not initialized */ - static status_t getMinFrameCount(size_t* frameCount, - audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT, - uint32_t sampleRate = 0); + static status_t getMinFrameCount(size_t* frameCount, + audio_stream_type_t streamType, + uint32_t sampleRate); + + /* How data is transferred to AudioTrack + */ + enum transfer_type { + TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters + TRANSFER_CALLBACK, // callback EVENT_MORE_DATA + TRANSFER_OBTAIN, // FIXME deprecated: call obtainBuffer() and releaseBuffer() + TRANSFER_SYNC, // synchronous write() + TRANSFER_SHARED, // shared memory + }; /* Constructs an uninitialized AudioTrack. No connection with * AudioFlinger takes place. Use set() after this. @@ -128,13 +138,13 @@ public: * Unspecified values are set to appropriate default values. * With this constructor, the track is configured for streaming mode. * Data to be rendered is supplied by write() or by the callback EVENT_MORE_DATA. - * Intermixing a combination of write() and non-ignored EVENT_MORE_DATA is deprecated. + * Intermixing a combination of write() and non-ignored EVENT_MORE_DATA is not allowed. * * Parameters: * * streamType: Select the type of audio stream this track is attached to * (e.g. AUDIO_STREAM_MUSIC). - * sampleRate: Track sampling rate in Hz. + * sampleRate: Data source sampling rate in Hz. * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed * 16 bits per sample). * channelMask: Channel mask. @@ -149,9 +159,10 @@ public: * user: Context for use by the callback receiver. * notificationFrames: The callback function is called each time notificationFrames PCM * frames have been consumed from track input buffer. + * This is expressed in units of frames at the initial source sample rate. * sessionId: Specific session ID, or zero to use default. - * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. - * If not present in parameter list, then fixed at false. + * transferType: How data is transferred to AudioTrack. + * threadCanCallJava: Not present in parameter list, and so is fixed at false. */ AudioTrack( audio_stream_type_t streamType, @@ -163,7 +174,8 @@ public: callback_t cbf = NULL, void* user = NULL, int notificationFrames = 0, - int sessionId = 0); + int sessionId = 0, + transfer_type transferType = TRANSFER_DEFAULT); /* Creates an audio track and registers it with AudioFlinger. * With this constructor, the track is configured for static buffer mode. @@ -174,7 +186,6 @@ public: * The write() method is not supported in this case. * It is recommended to pass a callback function to be notified of playback end by an * EVENT_UNDERRUN event. - * FIXME EVENT_MORE_DATA still occurs; it must be ignored. */ AudioTrack( audio_stream_type_t streamType, @@ -186,7 +197,8 @@ public: callback_t cbf = NULL, void* user = NULL, int notificationFrames = 0, - int sessionId = 0); + int sessionId = 0, + transfer_type transferType = TRANSFER_DEFAULT); /* Terminates the AudioTrack and unregisters it from AudioFlinger. * Also destroys all resources associated with the AudioTrack. @@ -195,7 +207,8 @@ protected: virtual ~AudioTrack(); public: - /* Initialize an uninitialized AudioTrack. + /* Initialize an AudioTrack that was created using the AudioTrack() constructor. + * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. * Returned status (from utils/Errors.h) can be: * - NO_ERROR: successful initialization * - INVALID_OPERATION: AudioTrack is already initialized @@ -203,6 +216,10 @@ public: * - NO_INIT: audio server or audio hardware not initialized * If sharedBuffer is non-0, the frameCount parameter is ignored and * replaced by the shared buffer's total allocated size in frame units. + * + * Parameters not listed in the AudioTrack constructors above: + * + * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. */ status_t set(audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT, uint32_t sampleRate = 0, @@ -215,7 +232,8 @@ public: int notificationFrames = 0, const sp& sharedBuffer = 0, bool threadCanCallJava = false, - int sessionId = 0); + int sessionId = 0, + transfer_type transferType = TRANSFER_DEFAULT); /* Result of constructing the AudioTrack. This must be checked * before using any AudioTrack API (except for set()), because using @@ -235,14 +253,15 @@ public: audio_stream_type_t streamType() const { return mStreamType; } audio_format_t format() const { return mFormat; } - /* Return frame size in bytes, which for linear PCM is channelCount * (bit depth per channel / 8). + /* Return frame size in bytes, which for linear PCM is + * channelCount * (bit depth per channel / 8). * channelCount is determined from channelMask, and bit depth comes from format. * For non-linear formats, the frame size is typically 1 byte. */ - uint32_t channelCount() const { return mChannelCount; } + size_t frameSize() const { return mFrameSize; } + uint32_t channelCount() const { return mChannelCount; } uint32_t frameCount() const { return mFrameCount; } - size_t frameSize() const { return mFrameSize; } /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ sp sharedBuffer() const { return mSharedBuffer; } @@ -255,10 +274,9 @@ public: /* Stop a track. * In static buffer mode, the track is stopped immediately. - * In streaming mode, the callback will cease being called and - * obtainBuffer returns STOPPED. Note that obtainBuffer() still works - * and will fill up buffers until the pool is exhausted. - * The stop does not occur immediately: any data remaining in the buffer + * In streaming mode, the callback will cease being called. Note that obtainBuffer() still + * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK. + * In streaming mode the stop does not occur immediately: any data remaining in the buffer * is first drained, mixed, and output, and only then is the track marked as stopped. */ void stop(); @@ -272,7 +290,7 @@ public: void flush(); /* Pause a track. After pause, the callback will cease being called and - * obtainBuffer returns STOPPED. Note that obtainBuffer() still works + * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works * and will fill up buffers until the pool is exhausted. * Volume is ramped down over the next mix buffer following the pause request, * and then the track is marked as paused. It can be resumed with ramp up by start(). @@ -296,11 +314,11 @@ public: status_t setAuxEffectSendLevel(float level); void getAuxEffectSendLevel(float* level) const; - /* Set sample rate for this track in Hz, mostly used for games' sound effects + /* Set source sample rate for this track in Hz, mostly used for games' sound effects */ status_t setSampleRate(uint32_t sampleRate); - /* Return current sample rate in Hz, or 0 if unknown */ + /* Return current source sample rate in Hz, or 0 if unknown */ uint32_t getSampleRate() const; /* Enables looping and sets the start and end points of looping. @@ -322,7 +340,7 @@ public: * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). * * If the loop period (loopEnd - loopStart) is too small for the implementation to support, - * setLoop() will return BAD_VALUE. + * setLoop() will return BAD_VALUE. loopCount must be >= -1. * */ status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); @@ -330,7 +348,7 @@ public: /* Sets marker position. When playback reaches the number of frames specified, a callback with * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker * notification callback. To set a marker at a position which would compute as 0, - * a workaround is to the set the marker at a nearby position such as -1 or 1. + * a workaround is to the set the marker at a nearby position such as ~0 or 1. * If the AudioTrack has been opened with no callback function associated, the operation will * fail. * @@ -390,16 +408,22 @@ public: /* Return the total number of frames played since playback start. * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. * It is reset to zero by flush(), reload(), and stop(). + * + * Parameters: + * + * position: Address where to return play head position. + * + * Returned status (from utils/Errors.h) can be: + * - NO_ERROR: successful operation + * - BAD_VALUE: position is NULL */ - status_t getPosition(uint32_t *position); + status_t getPosition(uint32_t *position) const; -#if 0 /* For static buffer mode only, this returns the current playback position in frames * relative to start of buffer. It is analogous to the new API for * setLoop() and setPosition(). After underrun, the position will be at end of buffer. */ status_t getBufferPosition(uint32_t *position); -#endif /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids * rewriting the buffer before restarting playback after a stop. @@ -446,15 +470,19 @@ public: */ status_t attachAuxEffect(int effectId); - /* Obtains a buffer of "frameCount" frames. The buffer must be - * filled entirely, and then released with releaseBuffer(). - * If the track is stopped, obtainBuffer() returns - * STOPPED instead of NO_ERROR as long as there are buffers available, - * at which point NO_MORE_BUFFERS is returned. + /* Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. + * After filling these slots with data, the caller should release them with releaseBuffer(). + * If the track buffer is not full, obtainBuffer() returns as many contiguous + * [empty slots for] frames as are available immediately. + * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK + * regardless of the value of waitCount. + * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a + * maximum timeout based on waitCount; see chart below. * Buffers will be returned until the pool * is exhausted, at which point obtainBuffer() will either block - * or return WOULD_BLOCK depending on the value of the "blocking" + * or return WOULD_BLOCK depending on the value of the "waitCount" * parameter. + * Each sample is 16-bit signed PCM. * * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications, * which should use write() or callback EVENT_MORE_DATA instead. @@ -477,24 +505,35 @@ public: * raw pointer to the buffer */ - enum { - NO_MORE_BUFFERS = 0x80000001, // same name in AudioFlinger.h, ok to be different value - STOPPED = 1 - }; + /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */ + status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount) + __attribute__((__deprecated__)); - status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount); +private: + /* New internal API + * If nonContig is non-NULL, it is an output parameter that will be set to the number of + * additional non-contiguous frames that are available immediately. + * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), + * in case the requested amount of frames is in two or more non-contiguous regions. + * FIXME requested and elapsed are both relative times. Consider changing to absolute time. + */ + status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, + struct timespec *elapsed = NULL, size_t *nonContig = NULL); +public: - /* Release a filled buffer of "frameCount" frames for AudioFlinger to process. */ + /* Release a filled buffer of "audioBuffer->frameCount" frames for AudioFlinger to process. */ + // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed void releaseBuffer(Buffer* audioBuffer); /* As a convenience we provide a write() interface to the audio buffer. + * Input parameter 'size' is in byte units. * This is implemented on top of obtainBuffer/releaseBuffer. For best * performance use callbacks. Returns actual number of bytes written >= 0, * or one of the following negative status codes: * INVALID_OPERATION AudioTrack is configured for shared buffer mode * BAD_VALUE size is invalid - * STOPPED AudioTrack was stopped during the write - * NO_MORE_BUFFERS when obtainBuffer() returns same + * WOULD_BLOCK when obtainBuffer() returns same, or + * AudioTrack was stopped during the write * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). * Not supported for static buffer mode. */ @@ -503,7 +542,13 @@ public: /* * Dumps the state of an audio track. */ - status_t dump(int fd, const Vector& args) const; + status_t dump(int fd, const Vector& args) const; + + /* + * Return the total number of frames which AudioFlinger desired but were unavailable, + * and thus which resulted in an underrun. Reset to zero by stop(). + */ + uint32_t getUnderrunFrames() const; protected: /* copying audio tracks is not allowed */ @@ -522,19 +567,29 @@ protected: void pause(); // suspend thread from execution at next loop boundary void resume(); // allow thread to execute, if not requested to exit + void pauseConditional(); + // like pause(), but only if prior resume() wasn't latched private: friend class AudioTrack; virtual bool threadLoop(); - AudioTrack& mReceiver; - ~AudioTrackThread(); + AudioTrack& mReceiver; + virtual ~AudioTrackThread(); Mutex mMyLock; // Thread::mLock is private Condition mMyCond; // Thread::mThreadExitedCondition is private bool mPaused; // whether thread is currently paused + bool mResumeLatch; // whether next pauseConditional() will be a nop }; // body of AudioTrackThread::threadLoop() - bool processAudioBuffer(const sp& thread); + // returns the maximum amount of time before we would like to run again, where: + // 0 immediately + // > 0 no later than this many nanoseconds from now + // NS_WHENEVER still active but no particular deadline + // NS_INACTIVE inactive so don't run again until re-started + // NS_NEVER never again + static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; + nsecs_t processAudioBuffer(const sp& thread); // caller must hold lock on mLock for all _l methods status_t createTrack_l(audio_stream_type_t streamType, @@ -543,20 +598,24 @@ protected: size_t frameCount, audio_output_flags_t flags, const sp& sharedBuffer, - audio_io_handle_t output); + audio_io_handle_t output, + size_t epoch); - // can only be called when !mActive + // can only be called when mState != STATE_ACTIVE void flush_l(); - status_t setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); + void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); audio_io_handle_t getOutput_l(); - status_t restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart); - bool stopped_l() const { return !mActive; } + // FIXME enum is faster than strcmp() for parameter 'from' + status_t restoreTrack_l(const char *from); + + // may be changed if IAudioTrack is re-created sp mAudioTrack; sp mCblkMemory; - sp mAudioTrackThread; + audio_track_cblk_t* mCblk; // re-load after mLock.unlock() + sp mAudioTrackThread; float mVolume[2]; float mSendLevel; uint32_t mSampleRate; @@ -564,62 +623,89 @@ protected: size_t mReqFrameCount; // frame count to request the next time a new // IAudioTrack is needed - audio_track_cblk_t* mCblk; // re-load after mLock.unlock() - - // Starting address of buffers in shared memory. If there is a shared buffer, mBuffers - // is the value of pointer() for the shared buffer, otherwise mBuffers 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* mBuffers; + // constant after constructor or set() audio_format_t mFormat; // as requested by client, not forced to 16-bit audio_stream_type_t mStreamType; uint32_t mChannelCount; audio_channel_mask_t mChannelMask; + transfer_type mTransfer; - // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. - // For 8-bit PCM data, mFrameSizeAF is - // twice as large because data is expanded to 16-bit before being stored in buffer. + // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. For 8-bit PCM data, it's + // twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer. size_t mFrameSize; // app-level frame size size_t mFrameSizeAF; // AudioFlinger frame size status_t mStatus; - uint32_t mLatency; - bool mActive; // protected by mLock + // can change dynamically when IAudioTrack invalidated + uint32_t mLatency; // in ms + + // Indicates the current track state. Protected by mLock. + enum State { + STATE_ACTIVE, + STATE_STOPPED, + STATE_PAUSED, + STATE_FLUSHED, + } mState; callback_t mCbf; // callback handler for events, or NULL void* mUserData; // for client callback handler // for notification APIs uint32_t mNotificationFramesReq; // requested number of frames between each - // notification callback + // notification callback, + // at initial source sample rate uint32_t mNotificationFramesAct; // actual number of frames between each - // notification callback + // notification callback, + // at initial source sample rate + bool mRefreshRemaining; // processAudioBuffer() should refresh next 2 + + // These are private to processAudioBuffer(), and are not protected by a lock + uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() + bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() + int mObservedSequence; // last observed value of mSequence + sp mSharedBuffer; - int mLoopCount; - uint32_t mRemainingFrames; + uint32_t mLoopPeriod; // in frames, zero means looping is disabled uint32_t mMarkerPosition; // in wrapping (overflow) frame units bool mMarkerReached; uint32_t mNewPosition; // in frames - uint32_t mUpdatePeriod; // in frames + uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS - bool mFlushed; // FIXME will be made obsolete by making flush() synchronous audio_output_flags_t mFlags; int mSessionId; int mAuxEffectId; - // When locking both mLock and mCblk->lock, must lock in this order to avoid deadlock: - // 1. mLock - // 2. mCblk->lock - // It is OK to lock only mCblk->lock. mutable Mutex mLock; bool mIsTimed; int mPreviousPriority; // before start() SchedPolicy mPreviousSchedulingGroup; - AudioTrackClientProxy* mProxy; bool mAwaitBoost; // thread should wait for priority boost before running + + // The proxy should only be referenced while a lock is held because the proxy isn't + // multi-thread safe, especially the SingleStateQueue part of the proxy. + // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, + // provided that the caller also holds an extra reference to the proxy and shared memory to keep + // them around in case they are replaced during the obtainBuffer(). + sp mStaticProxy; // for type safety only + sp mProxy; // primary owner of the memory + + bool mInUnderrun; // whether track is currently in underrun state + +private: + class DeathNotifier : public IBinder::DeathRecipient { + public: + DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } + protected: + virtual void binderDied(const wp& who); + private: + const wp mAudioTrack; + }; + + sp mDeathNotifier; + uint32_t mSequence; // incremented for each new IAudioTrack attempt }; class TimedAudioTrack : public AudioTrack diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h index 41e20f8..681f557 100644 --- a/include/private/media/AudioTrackShared.h +++ b/include/private/media/AudioTrackShared.h @@ -22,32 +22,46 @@ #include #include +#include +#include +#include +#include namespace android { // ---------------------------------------------------------------------------- -// Maximum cumulated timeout milliseconds before restarting audioflinger thread -#define MAX_STARTUP_TIMEOUT_MS 3000 // Longer timeout period at startup to cope with A2DP - // init time -#define MAX_RUN_TIMEOUT_MS 1000 -#define WAIT_PERIOD_MS 10 - -#define CBLK_UNDERRUN 0x01 // set: underrun (out) or overrrun (in), clear: no underrun or overrun +#define CBLK_UNDERRUN 0x01 // set by server immediately on output underrun, cleared by client #define CBLK_FORCEREADY 0x02 // set: track is considered ready immediately by AudioFlinger, // clear: track is ready when buffer full #define CBLK_INVALID 0x04 // track buffer invalidated by AudioFlinger, need to re-create -#define CBLK_DISABLED 0x08 // track disabled by AudioFlinger due to underrun, need to re-start +#define CBLK_DISABLED 0x08 // output track disabled by AudioFlinger due to underrun, + // need to re-start. Unlike CBLK_UNDERRUN, this is not set + // immediately, but only after a long string of underruns. +// 0x10 unused +#define CBLK_LOOP_CYCLE 0x20 // set by server each time a loop cycle other than final one completes +#define CBLK_LOOP_FINAL 0x40 // set by server when the final loop cycle completes +#define CBLK_BUFFER_END 0x80 // set by server when the position reaches end of buffer if not looping +#define CBLK_OVERRUN 0x100 // set by server immediately on input overrun, cleared by client +#define CBLK_INTERRUPT 0x200 // set by client on interrupt(), cleared by client in obtainBuffer() struct AudioTrackSharedStreaming { // similar to NBAIO MonoPipe - volatile int32_t mFront; - volatile int32_t mRear; + // in continuously incrementing frame units, take modulo buffer size, which must be a power of 2 + volatile int32_t mFront; // read by server + volatile int32_t mRear; // write by client + volatile int32_t mFlush; // incremented by client to indicate a request to flush; + // server notices and discards all data between mFront and mRear + volatile uint32_t mUnderrunFrames; // server increments for each unavailable but desired frame }; -// future +typedef SingleStateQueue StaticAudioTrackSingleStateQueue; + struct AudioTrackSharedStatic { - int mReserved; + StaticAudioTrackSingleStateQueue::Shared + mSingleStateQueue; + size_t mBufferPosition; // updated asynchronously by server, + // "for entertainment purposes only" }; // ---------------------------------------------------------------------------- @@ -55,65 +69,61 @@ struct AudioTrackSharedStatic { // Important: do not add any virtual methods, including ~ struct audio_track_cblk_t { + // Since the control block is always located in shared memory, this constructor + // is only used for placement new(). It is never used for regular new() or stack. + audio_track_cblk_t(); + /*virtual*/ ~audio_track_cblk_t() { } + friend class Proxy; + friend class ClientProxy; friend class AudioTrackClientProxy; friend class AudioRecordClientProxy; friend class ServerProxy; + friend class AudioTrackServerProxy; + friend class AudioRecordServerProxy; // The data members are grouped so that members accessed frequently and in the same context // are in the same line of data cache. - Mutex lock; // sizeof(int) - Condition cv; // sizeof(int) - - // next 4 are offsets within "buffers" - volatile uint32_t user; - volatile uint32_t server; - uint32_t userBase; - uint32_t serverBase; - int mPad1; // unused, but preserves cache line alignment + volatile uint32_t server; // updated asynchronously by server, + // "for entertainment purposes only" size_t frameCount_; // used during creation to pass actual track buffer size // from AudioFlinger to client, and not referenced again - // FIXME remove here and replace by createTrack() in/out parameter + // FIXME remove here and replace by createTrack() in/out + // parameter // renamed to "_" to detect incorrect use - // Cache line boundary (32 bytes) + volatile int32_t mFutex; // semaphore: down (P) by client, + // up (V) by server or binderDied() or interrupt() + +private: - uint32_t loopStart; - uint32_t loopEnd; // read-only for server, read/write for client - int loopCount; // read/write for client + size_t mMinimum; // server wakes up client if available >= mMinimum // Channel volumes are fixed point U4.12, so 0x1000 means 1.0. // Left channel is in [0:15], right channel is in [16:31]. // Always read and write the combined pair atomically. // For AudioTrack only, not used by AudioRecord. -private: uint32_t mVolumeLR; uint32_t mSampleRate; // AudioTrack only: client's requested sample rate in Hz // or 0 == default. Write-only client, read-only server. + // client write-only, server read-only + uint16_t mSendLevel; // Fixed point U4.12 so 0x1000 means 1.0 + uint8_t mPad2; // unused public: // read-only for client, server writes once at initialization and is then read-only uint8_t mName; // normal tracks: track name, fast tracks: track index - // used by client only - uint16_t bufferTimeoutMs; // Maximum cumulated timeout before restarting - // audioflinger - - uint16_t waitTimeMs; // Cumulated wait time, used by client only -private: - // client write-only, server read-only - uint16_t mSendLevel; // Fixed point U4.12 so 0x1000 means 1.0 -public: volatile int32_t flags; // Cache line boundary (32 bytes) -#if 0 +public: union { AudioTrackSharedStreaming mStreaming; AudioTrackSharedStatic mStatic; @@ -121,25 +131,6 @@ public: } u; // Cache line boundary (32 bytes) -#endif - - // Since the control block is always located in shared memory, this constructor - // is only used for placement new(). It is never used for regular new() or stack. - audio_track_cblk_t(); - -private: - // if there is a shared buffer, "buffers" is the value of pointer() for the shared - // buffer, otherwise "buffers" points immediately after the control block - void* buffer(void *buffers, uint32_t frameSize, size_t offset) const; - - bool tryLock(); - - // isOut == true means AudioTrack, isOut == false means AudioRecord - bool stepServer(size_t stepCount, size_t frameCount, bool isOut); - uint32_t stepUser(size_t stepCount, size_t frameCount, bool isOut); - uint32_t framesAvailable(size_t frameCount, bool isOut); - uint32_t framesAvailable_l(size_t frameCount, bool isOut); - uint32_t framesReady(bool isOut); }; // ---------------------------------------------------------------------------- @@ -147,29 +138,31 @@ private: // Proxy for shared memory control block, to isolate callers from needing to know the details. // There is exactly one ClientProxy and one ServerProxy per shared memory control block. // The proxies are located in normal memory, and are not multi-thread safe within a given side. -class Proxy { +class Proxy : public RefBase { protected: - Proxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize) - : mCblk(cblk), mBuffers(buffers), mFrameCount(frameCount), mFrameSize(frameSize) { } + Proxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, bool isOut, + bool clientInServer); virtual ~Proxy() { } public: - void* buffer(size_t offset) const { - return mCblk->buffer(mBuffers, mFrameSize, offset); - } + struct Buffer { + size_t mFrameCount; // number of frames available in this buffer + void* mRaw; // pointer to first frame + size_t mNonContig; // number of additional non-contiguous frames available + }; protected: // These refer to shared memory, and are virtual addresses with respect to the current process. // They may have different virtual addresses within the other process. - audio_track_cblk_t* const mCblk; // the control block - void* const mBuffers; // starting address of buffers - - const size_t mFrameCount; // not necessarily a power of 2 - const size_t mFrameSize; // in bytes -#if 0 - const size_t mFrameCountP2; // mFrameCount rounded to power of 2, streaming mode -#endif - + audio_track_cblk_t* const mCblk; // the control block + void* const mBuffers; // starting address of buffers + + const size_t mFrameCount; // not necessarily a power of 2 + const size_t mFrameSize; // in bytes + const size_t mFrameCountP2; // mFrameCount rounded to power of 2, streaming mode + const bool mIsOut; // true for AudioTrack, false for AudioRecord + const bool mClientInServer; // true for OutputTrack, false for AudioTrack & AudioRecord + bool mIsShutdown; // latch set to true when shared memory corruption detected }; // ---------------------------------------------------------------------------- @@ -177,9 +170,86 @@ protected: // Proxy seen by AudioTrack client and AudioRecord client class ClientProxy : public Proxy { protected: - ClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize) - : Proxy(cblk, buffers, frameCount, frameSize) { } + ClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, + bool isOut, bool clientInServer); virtual ~ClientProxy() { } + +public: + static const struct timespec kForever; + static const struct timespec kNonBlocking; + + // Obtain a buffer with filled frames (reading) or empty frames (writing). + // It is permitted to call obtainBuffer() multiple times in succession, without any intervening + // calls to releaseBuffer(). In that case, the final obtainBuffer() is the one that effectively + // sets or extends the unreleased frame count. + // On entry: + // buffer->mFrameCount should be initialized to maximum number of desired frames, + // which must be > 0. + // buffer->mNonContig is unused. + // buffer->mRaw is unused. + // requested is the requested timeout in local monotonic delta time units: + // NULL or &kNonBlocking means non-blocking (zero timeout). + // &kForever means block forever (infinite timeout). + // Other values mean a specific timeout in local monotonic delta time units. + // elapsed is a pointer to a location that will hold the total local monotonic time that + // elapsed while blocked, or NULL if not needed. + // On exit: + // buffer->mFrameCount has the actual number of contiguous available frames, + // which is always 0 when the return status != NO_ERROR. + // buffer->mNonContig is the number of additional non-contiguous available frames. + // buffer->mRaw is a pointer to the first available frame, + // or NULL when buffer->mFrameCount == 0. + // The return status is one of: + // NO_ERROR Success, buffer->mFrameCount > 0. + // WOULD_BLOCK Non-blocking mode and no frames are available. + // TIMED_OUT Timeout occurred before any frames became available. + // This can happen even for infinite timeout, due to a spurious wakeup. + // In this case, the caller should investigate and then re-try as appropriate. + // DEAD_OBJECT Server has died or invalidated, caller should destroy this proxy and re-create. + // -EINTR Call has been interrupted. Look around to see why, and then perhaps try again. + // NO_INIT Shared memory is corrupt. + // BAD_VALUE On entry buffer == NULL or buffer->mFrameCount == 0. + status_t obtainBuffer(Buffer* buffer, const struct timespec *requested = NULL, + struct timespec *elapsed = NULL); + + // Release (some of) the frames last obtained. + // On entry, buffer->mFrameCount should have the number of frames to release, + // which must (cumulatively) be <= the number of frames last obtained but not yet released. + // buffer->mRaw is ignored, but is normally same pointer returned by last obtainBuffer(). + // It is permitted to call releaseBuffer() multiple times to release the frames in chunks. + // On exit: + // buffer->mFrameCount is zero. + // buffer->mRaw is NULL. + void releaseBuffer(Buffer* buffer); + + // Call after detecting server's death + void binderDied(); + + // Call to force an obtainBuffer() to return quickly with -EINTR + void interrupt(); + + size_t getPosition() { + return mEpoch + mCblk->server; + } + + void setEpoch(size_t epoch) { + mEpoch = epoch; + } + + void setMinimum(size_t minimum) { + mCblk->mMinimum = minimum; + } + + // Return the number of frames that would need to be obtained and released + // in order for the client to be aligned at start of buffer + virtual size_t getMisalignment(); + + size_t getEpoch() const { + return mEpoch; + } + +private: + size_t mEpoch; }; // ---------------------------------------------------------------------------- @@ -187,8 +257,10 @@ protected: // Proxy used by AudioTrack client, which also includes AudioFlinger::PlaybackThread::OutputTrack class AudioTrackClientProxy : public ClientProxy { public: - AudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize) - : ClientProxy(cblk, buffers, frameCount, frameSize) { } + AudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, + size_t frameSize, bool clientInServer = false) + : ClientProxy(cblk, buffers, frameCount, frameSize, true /*isOut*/, + clientInServer) { } virtual ~AudioTrackClientProxy() { } // No barriers on the following operations, so the ordering of loads/stores @@ -208,27 +280,36 @@ public: mCblk->mSampleRate = sampleRate; } - // called by: - // PlaybackThread::OutputTrack::write - // AudioTrack::createTrack_l - // AudioTrack::releaseBuffer - // AudioTrack::reload - // AudioTrack::restoreTrack_l (2 places) - size_t stepUser(size_t stepCount) { - return mCblk->stepUser(stepCount, mFrameCount, true /*isOut*/); + virtual void flush(); + + virtual uint32_t getUnderrunFrames() const { + return mCblk->u.mStreaming.mUnderrunFrames; } +}; + +class StaticAudioTrackClientProxy : public AudioTrackClientProxy { +public: + StaticAudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, + size_t frameSize); + virtual ~StaticAudioTrackClientProxy() { } + + virtual void flush(); + +#define MIN_LOOP 16 // minimum length of each loop iteration in frames + void setLoop(size_t loopStart, size_t loopEnd, int loopCount); + size_t getBufferPosition(); - // called by AudioTrack::obtainBuffer and AudioTrack::processBuffer - size_t framesAvailable() { - return mCblk->framesAvailable(mFrameCount, true /*isOut*/); + virtual size_t getMisalignment() { + return 0; } - // called by AudioTrack::obtainBuffer and PlaybackThread::OutputTrack::obtainBuffer - // FIXME remove this API since it assumes a lock that should be invisible to caller - size_t framesAvailable_l() { - return mCblk->framesAvailable_l(mFrameCount, true /*isOut*/); + virtual uint32_t getUnderrunFrames() const { + return 0; } +private: + StaticAudioTrackSingleStateQueue::Mutator mMutator; + size_t mBufferPosition; // so that getBufferPosition() appears to be synchronous }; // ---------------------------------------------------------------------------- @@ -236,60 +317,122 @@ public: // Proxy used by AudioRecord client class AudioRecordClientProxy : public ClientProxy { public: - AudioRecordClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize) - : ClientProxy(cblk, buffers, frameCount, frameSize) { } + AudioRecordClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, + size_t frameSize) + : ClientProxy(cblk, buffers, frameCount, frameSize, + false /*isOut*/, false /*clientInServer*/) { } ~AudioRecordClientProxy() { } - - // called by AudioRecord::releaseBuffer - size_t stepUser(size_t stepCount) { - return mCblk->stepUser(stepCount, mFrameCount, false /*isOut*/); - } - - // called by AudioRecord::processBuffer - size_t framesAvailable() { - return mCblk->framesAvailable(mFrameCount, false /*isOut*/); - } - - // called by AudioRecord::obtainBuffer - size_t framesReady() { - return mCblk->framesReady(false /*isOut*/); - } - }; // ---------------------------------------------------------------------------- // Proxy used by AudioFlinger server class ServerProxy : public Proxy { +protected: + ServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, + bool isOut, bool clientInServer); public: - ServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, bool isOut) - : Proxy(cblk, buffers, frameCount, frameSize), mIsOut(isOut) { } virtual ~ServerProxy() { } - // for AudioTrack and AudioRecord - bool step(size_t stepCount) { return mCblk->stepServer(stepCount, mFrameCount, mIsOut); } + // Obtain a buffer with filled frames (writing) or empty frames (reading). + // It is permitted to call obtainBuffer() multiple times in succession, without any intervening + // calls to releaseBuffer(). In that case, the final obtainBuffer() is the one that effectively + // sets or extends the unreleased frame count. + // Always non-blocking. + // On entry: + // buffer->mFrameCount should be initialized to maximum number of desired frames, + // which must be > 0. + // buffer->mNonContig is unused. + // buffer->mRaw is unused. + // On exit: + // buffer->mFrameCount has the actual number of contiguous available frames, + // which is always 0 when the return status != NO_ERROR. + // buffer->mNonContig is the number of additional non-contiguous available frames. + // buffer->mRaw is a pointer to the first available frame, + // or NULL when buffer->mFrameCount == 0. + // The return status is one of: + // NO_ERROR Success, buffer->mFrameCount > 0. + // WOULD_BLOCK No frames are available. + // NO_INIT Shared memory is corrupt. + virtual status_t obtainBuffer(Buffer* buffer); + + // Release (some of) the frames last obtained. + // On entry, buffer->mFrameCount should have the number of frames to release, + // which must (cumulatively) be <= the number of frames last obtained but not yet released. + // It is permitted to call releaseBuffer() multiple times to release the frames in chunks. + // buffer->mRaw is ignored, but is normally same pointer returned by last obtainBuffer(). + // On exit: + // buffer->mFrameCount is zero. + // buffer->mRaw is NULL. + virtual void releaseBuffer(Buffer* buffer); +protected: + size_t mUnreleased; // unreleased frames remaining from most recent obtainBuffer() + size_t mAvailToClient; // estimated frames available to client prior to releaseBuffer() +private: + int32_t mFlush; // our copy of cblk->u.mStreaming.mFlush, for streaming output only + bool mDeferWake; // whether another releaseBuffer() is expected soon +}; + +// Proxy used by AudioFlinger for servicing AudioTrack +class AudioTrackServerProxy : public ServerProxy { +public: + AudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, + size_t frameSize, bool clientInServer = false) + : ServerProxy(cblk, buffers, frameCount, frameSize, true /*isOut*/, clientInServer) { } +protected: + virtual ~AudioTrackServerProxy() { } + +public: // return value of these methods must be validated by the caller uint32_t getSampleRate() const { return mCblk->mSampleRate; } uint16_t getSendLevel_U4_12() const { return mCblk->mSendLevel; } uint32_t getVolumeLR() const { return mCblk->mVolumeLR; } - // for AudioTrack only - size_t framesReady() { - ALOG_ASSERT(mIsOut); - return mCblk->framesReady(true); - } + // estimated total number of filled frames available to server to read, + // which may include non-contiguous frames + virtual size_t framesReady(); + + // Currently AudioFlinger will call framesReady() for a fast track from two threads: + // FastMixer thread, and normal mixer thread. This is dangerous, as the proxy is intended + // to be called from at most one thread of server, and one thread of client. + // As a temporary workaround, this method informs the proxy implementation that it + // should avoid doing a state queue poll from within framesReady(). + // FIXME Change AudioFlinger to not call framesReady() from normal mixer thread. + virtual void framesReadyIsCalledByMultipleThreads() { } +}; - // for AudioRecord only, called by RecordThread::RecordTrack::getNextBuffer - // FIXME remove this API since it assumes a lock that should be invisible to caller - size_t framesAvailableIn_l() { - ALOG_ASSERT(!mIsOut); - return mCblk->framesAvailable_l(mFrameCount, false); - } +class StaticAudioTrackServerProxy : public AudioTrackServerProxy { +public: + StaticAudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, + size_t frameSize); +protected: + virtual ~StaticAudioTrackServerProxy() { } + +public: + virtual size_t framesReady(); + virtual void framesReadyIsCalledByMultipleThreads(); + virtual status_t obtainBuffer(Buffer* buffer); + virtual void releaseBuffer(Buffer* buffer); private: - const bool mIsOut; // true for AudioTrack, false for AudioRecord + ssize_t pollPosition(); // poll for state queue update, and return current position + StaticAudioTrackSingleStateQueue::Observer mObserver; + size_t mPosition; // server's current play position in frames, relative to 0 + size_t mEnd; // cached value computed from mState, safe for asynchronous read + bool mFramesReadyIsCalledByMultipleThreads; + StaticAudioTrackState mState; +}; +// Proxy used by AudioFlinger for servicing AudioRecord +class AudioRecordServerProxy : public ServerProxy { +public: + AudioRecordServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, + size_t frameSize) + : ServerProxy(cblk, buffers, frameCount, frameSize, false /*isOut*/, + false /*clientInServer*/) { } +protected: + virtual ~AudioRecordServerProxy() { } }; // ---------------------------------------------------------------------------- diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp index a2b8ae2..9faa497 100644 --- a/media/libmedia/AudioRecord.cpp +++ b/media/libmedia/AudioRecord.cpp @@ -19,18 +19,13 @@ #define LOG_TAG "AudioRecord" #include -#include - #include -#include -#include #include -#include -#include #include - #include +#define WAIT_PERIOD_MS 10 + namespace android { // --------------------------------------------------------------------------- @@ -41,7 +36,9 @@ status_t AudioRecord::getMinFrameCount( audio_format_t format, audio_channel_mask_t channelMask) { - if (frameCount == NULL) return BAD_VALUE; + if (frameCount == NULL) { + return BAD_VALUE; + } // default to 0 in case of error *frameCount = 0; @@ -75,8 +72,7 @@ status_t AudioRecord::getMinFrameCount( AudioRecord::AudioRecord() : mStatus(NO_INIT), mSessionId(0), - mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT), - mProxy(NULL) + mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT) { } @@ -89,14 +85,15 @@ AudioRecord::AudioRecord( callback_t cbf, void* user, int notificationFrames, - int sessionId) + int sessionId, + transfer_type transferType) : mStatus(NO_INIT), mSessionId(0), mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT), mProxy(NULL) { - mStatus = set(inputSource, sampleRate, format, channelMask, - frameCount, cbf, user, notificationFrames, false /*threadCanCallJava*/, sessionId); + mStatus = set(inputSource, sampleRate, format, channelMask, frameCount, cbf, user, + notificationFrames, false /*threadCanCallJava*/, sessionId, transferType); } AudioRecord::~AudioRecord() @@ -111,11 +108,13 @@ AudioRecord::~AudioRecord() mAudioRecordThread->requestExitAndWait(); mAudioRecordThread.clear(); } - mAudioRecord.clear(); + if (mAudioRecord != 0) { + mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this); + mAudioRecord.clear(); + } IPCThreadState::self()->flushCommands(); AudioSystem::releaseAudioSessionId(mSessionId); } - delete mProxy; } status_t AudioRecord::set( @@ -128,8 +127,32 @@ status_t AudioRecord::set( void* user, int notificationFrames, bool threadCanCallJava, - int sessionId) + int sessionId, + transfer_type transferType) { + switch (transferType) { + case TRANSFER_DEFAULT: + if (cbf == NULL || threadCanCallJava) { + transferType = TRANSFER_SYNC; + } else { + transferType = TRANSFER_CALLBACK; + } + break; + case TRANSFER_CALLBACK: + if (cbf == NULL) { + ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL"); + return BAD_VALUE; + } + break; + case TRANSFER_OBTAIN: + case TRANSFER_SYNC: + break; + default: + ALOGE("Invalid transfer type %d", transferType); + return BAD_VALUE; + } + mTransfer = transferType; + // FIXME "int" here is legacy and will be replaced by size_t later if (frameCountInt < 0) { ALOGE("Invalid frame count %d", frameCountInt); @@ -143,6 +166,7 @@ status_t AudioRecord::set( AutoMutex lock(mLock); if (mAudioRecord != 0) { + ALOGE("Track already in use"); return INVALID_OPERATION; } @@ -159,14 +183,16 @@ status_t AudioRecord::set( if (format == AUDIO_FORMAT_DEFAULT) { format = AUDIO_FORMAT_PCM_16_BIT; } + // validate parameters if (!audio_is_valid_format(format)) { - ALOGE("Invalid format"); + ALOGE("Invalid format %d", format); return BAD_VALUE; } mFormat = format; if (!audio_is_input_channel(channelMask)) { + ALOGE("Invalid channel mask %#x", channelMask); return BAD_VALUE; } mChannelMask = channelMask; @@ -200,6 +226,7 @@ status_t AudioRecord::set( size_t minFrameCount = 0; status_t status = getMinFrameCount(&minFrameCount, sampleRate, format, channelMask); if (status != NO_ERROR) { + ALOGE("getMinFrameCount() failed; status %d", status); return status; } ALOGV("AudioRecord::set() minFrameCount = %d", minFrameCount); @@ -207,6 +234,7 @@ status_t AudioRecord::set( if (frameCount == 0) { frameCount = minFrameCount; } else if (frameCount < minFrameCount) { + ALOGE("frameCount %u < minFrameCount %u", frameCount, minFrameCount); return BAD_VALUE; } @@ -215,7 +243,7 @@ status_t AudioRecord::set( } // create the IAudioRecord - status = openRecord_l(sampleRate, format, frameCount, input); + status = openRecord_l(sampleRate, format, frameCount, input, 0 /*epoch*/); if (status != NO_ERROR) { return status; } @@ -233,7 +261,7 @@ status_t AudioRecord::set( mActive = false; mCbf = cbf; mNotificationFrames = notificationFrames; - mRemainingFrames = notificationFrames; + mRefreshRemaining = true; mUserData = user; // TODO: add audio hardware input latency here mLatency = (1000*mFrameCount) / sampleRate; @@ -244,117 +272,78 @@ status_t AudioRecord::set( mInputSource = inputSource; mInput = input; AudioSystem::acquireAudioSessionId(mSessionId); + mSequence = 1; + mObservedSequence = mSequence; + mInOverrun = false; return NO_ERROR; } -status_t AudioRecord::initCheck() const -{ - return mStatus; -} - -// ------------------------------------------------------------------------- - -uint32_t AudioRecord::latency() const -{ - return mLatency; -} - -audio_format_t AudioRecord::format() const -{ - return mFormat; -} - -uint32_t AudioRecord::channelCount() const -{ - return mChannelCount; -} - -size_t AudioRecord::frameCount() const -{ - return mFrameCount; -} - -audio_source_t AudioRecord::inputSource() const -{ - return mInputSource; -} - // ------------------------------------------------------------------------- status_t AudioRecord::start(AudioSystem::sync_event_t event, int triggerSession) { - status_t ret = NO_ERROR; - sp t = mAudioRecordThread; - ALOGV("start, sync event %d trigger session %d", event, triggerSession); AutoMutex lock(mLock); - // acquire a strong reference on the IAudioRecord and IMemory so that they cannot be destroyed - // while we are accessing the cblk - sp audioRecord = mAudioRecord; - sp iMem = mCblkMemory; - audio_track_cblk_t* cblk = mCblk; + if (mActive) { + return NO_ERROR; + } - if (!mActive) { - mActive = true; + // reset current position as seen by client to 0 + mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); - cblk->lock.lock(); - if (!(cblk->flags & CBLK_INVALID)) { - cblk->lock.unlock(); - ALOGV("mAudioRecord->start()"); - ret = mAudioRecord->start(event, triggerSession); - cblk->lock.lock(); - if (ret == DEAD_OBJECT) { - android_atomic_or(CBLK_INVALID, &cblk->flags); - } - } - if (cblk->flags & CBLK_INVALID) { - audio_track_cblk_t* temp = cblk; - ret = restoreRecord_l(temp); - cblk = temp; + mNewPosition = mProxy->getPosition() + mUpdatePeriod; + int32_t flags = android_atomic_acquire_load(&mCblk->flags); + + status_t status = NO_ERROR; + if (!(flags & CBLK_INVALID)) { + ALOGV("mAudioRecord->start()"); + status = mAudioRecord->start(event, triggerSession); + if (status == DEAD_OBJECT) { + flags |= CBLK_INVALID; } - cblk->lock.unlock(); - if (ret == NO_ERROR) { - mNewPosition = cblk->user + mUpdatePeriod; - cblk->bufferTimeoutMs = (event == AudioSystem::SYNC_EVENT_NONE) ? MAX_RUN_TIMEOUT_MS : - AudioSystem::kSyncRecordStartTimeOutMs; - cblk->waitTimeMs = 0; - if (t != 0) { - t->resume(); - } else { - mPreviousPriority = getpriority(PRIO_PROCESS, 0); - get_sched_policy(0, &mPreviousSchedulingGroup); - androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); - } + } + if (flags & CBLK_INVALID) { + status = restoreRecord_l("start"); + } + + if (status != NO_ERROR) { + ALOGE("start() status %d", status); + } else { + mActive = true; + sp t = mAudioRecordThread; + if (t != 0) { + t->resume(); } else { - mActive = false; + mPreviousPriority = getpriority(PRIO_PROCESS, 0); + get_sched_policy(0, &mPreviousSchedulingGroup); + androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); } } - return ret; + return status; } void AudioRecord::stop() { - sp t = mAudioRecordThread; - - ALOGV("stop"); - AutoMutex lock(mLock); - if (mActive) { - mActive = false; - mCblk->cv.signal(); - mAudioRecord->stop(); - // the record head position will reset to 0, so if a marker is set, we need - // to activate it again - mMarkerReached = false; - if (t != 0) { - t->pause(); - } else { - setpriority(PRIO_PROCESS, 0, mPreviousPriority); - set_sched_policy(0, mPreviousSchedulingGroup); - } + if (!mActive) { + return; + } + + mActive = false; + mProxy->interrupt(); + mAudioRecord->stop(); + // the record head position will reset to 0, so if a marker is set, we need + // to activate it again + mMarkerReached = false; + sp t = mAudioRecordThread; + if (t != 0) { + t->pause(); + } else { + setpriority(PRIO_PROCESS, 0, mPreviousPriority); + set_sched_policy(0, mPreviousSchedulingGroup); } } @@ -364,14 +353,11 @@ bool AudioRecord::stopped() const return !mActive; } -uint32_t AudioRecord::getSampleRate() const -{ - return mSampleRate; -} - status_t AudioRecord::setMarkerPosition(uint32_t marker) { - if (mCbf == NULL) return INVALID_OPERATION; + if (mCbf == NULL) { + return INVALID_OPERATION; + } AutoMutex lock(mLock); mMarkerPosition = marker; @@ -382,7 +368,9 @@ status_t AudioRecord::setMarkerPosition(uint32_t marker) status_t AudioRecord::getMarkerPosition(uint32_t *marker) const { - if (marker == NULL) return BAD_VALUE; + if (marker == NULL) { + return BAD_VALUE; + } AutoMutex lock(mLock); *marker = mMarkerPosition; @@ -392,13 +380,12 @@ status_t AudioRecord::getMarkerPosition(uint32_t *marker) const status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod) { - if (mCbf == NULL) return INVALID_OPERATION; - - uint32_t curPosition; - getPosition(&curPosition); + if (mCbf == NULL) { + return INVALID_OPERATION; + } AutoMutex lock(mLock); - mNewPosition = curPosition + updatePeriod; + mNewPosition = mProxy->getPosition() + updatePeriod; mUpdatePeriod = updatePeriod; return NO_ERROR; @@ -406,7 +393,9 @@ status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod) status_t AudioRecord::getPositionUpdatePeriod(uint32_t *updatePeriod) const { - if (updatePeriod == NULL) return BAD_VALUE; + if (updatePeriod == NULL) { + return BAD_VALUE; + } AutoMutex lock(mLock); *updatePeriod = mUpdatePeriod; @@ -416,10 +405,12 @@ status_t AudioRecord::getPositionUpdatePeriod(uint32_t *updatePeriod) const status_t AudioRecord::getPosition(uint32_t *position) const { - if (position == NULL) return BAD_VALUE; + if (position == NULL) { + return BAD_VALUE; + } AutoMutex lock(mLock); - *position = mCblk->user; + *position = mProxy->getPosition(); return NO_ERROR; } @@ -427,7 +418,7 @@ status_t AudioRecord::getPosition(uint32_t *position) const unsigned int AudioRecord::getInputFramesLost() const { // no need to check mActive, because if inactive this will return 0, which is what we want - return AudioSystem::getInputFramesLost(mInput); + return AudioSystem::getInputFramesLost(getInput()); } // ------------------------------------------------------------------------- @@ -437,7 +428,8 @@ status_t AudioRecord::openRecord_l( uint32_t sampleRate, audio_format_t format, size_t frameCount, - audio_io_handle_t input) + audio_io_handle_t input, + size_t epoch) { status_t status; const sp& audioFlinger = AudioSystem::get_audio_flinger(); @@ -447,7 +439,7 @@ status_t AudioRecord::openRecord_l( } pid_t tid = -1; - // FIXME see similar logic at AudioTrack + // FIXME see similar logic at AudioTrack for tid int originalSessionId = mSessionId; sp record = audioFlinger->openRecord(input, @@ -470,133 +462,138 @@ status_t AudioRecord::openRecord_l( ALOGE("Could not get control block"); return NO_INIT; } - mAudioRecord.clear(); + if (mAudioRecord != 0) { + mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this); + mDeathNotifier.clear(); + } mAudioRecord = record; - mCblkMemory.clear(); mCblkMemory = iMem; audio_track_cblk_t* cblk = static_cast(iMem->pointer()); mCblk = cblk; - mBuffers = (char*)cblk + sizeof(audio_track_cblk_t); - cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; - cblk->waitTimeMs = 0; + + // starting address of buffers in shared memory + void *buffers = (char*)cblk + sizeof(audio_track_cblk_t); // update proxy - delete mProxy; - mProxy = new AudioRecordClientProxy(cblk, mBuffers, frameCount, mFrameSize); + mProxy = new AudioRecordClientProxy(cblk, buffers, frameCount, mFrameSize); + mProxy->setEpoch(epoch); + mProxy->setMinimum(mNotificationFrames); + + mDeathNotifier = new DeathNotifier(this); + mAudioRecord->asBinder()->linkToDeath(mDeathNotifier, this); return NO_ERROR; } status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) { - ALOG_ASSERT(mStatus == NO_ERROR && mProxy != NULL); + if (audioBuffer == NULL) { + return BAD_VALUE; + } + if (mTransfer != TRANSFER_OBTAIN) { + audioBuffer->frameCount = 0; + audioBuffer->size = 0; + audioBuffer->raw = NULL; + return INVALID_OPERATION; + } - AutoMutex lock(mLock); - bool active; - status_t result = NO_ERROR; - audio_track_cblk_t* cblk = mCblk; - uint32_t framesReq = audioBuffer->frameCount; - uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS; - - audioBuffer->frameCount = 0; - audioBuffer->size = 0; - - size_t framesReady = mProxy->framesReady(); - - if (framesReady == 0) { - cblk->lock.lock(); - goto start_loop_here; - while (framesReady == 0) { - active = mActive; - if (CC_UNLIKELY(!active)) { - cblk->lock.unlock(); - return NO_MORE_BUFFERS; - } - if (CC_UNLIKELY(!waitCount)) { - cblk->lock.unlock(); - return WOULD_BLOCK; - } - if (!(cblk->flags & CBLK_INVALID)) { - mLock.unlock(); - // this condition is in shared memory, so if IAudioRecord and control block - // are replaced due to mediaserver death or IAudioRecord invalidation then - // cv won't be signalled, but fortunately the timeout will limit the wait - result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); - cblk->lock.unlock(); - mLock.lock(); - if (!mActive) { - return status_t(STOPPED); - } - // IAudioRecord may have been re-created while mLock was unlocked - cblk = mCblk; - cblk->lock.lock(); - } - if (cblk->flags & CBLK_INVALID) { - goto create_new_record; - } - if (CC_UNLIKELY(result != NO_ERROR)) { - cblk->waitTimeMs += waitTimeMs; - if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) { - ALOGW( "obtainBuffer timed out (is the CPU pegged?) " - "user=%08x, server=%08x", cblk->user, cblk->server); - cblk->lock.unlock(); - // callback thread or sync event hasn't changed - result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0); - cblk->lock.lock(); - if (result == DEAD_OBJECT) { - android_atomic_or(CBLK_INVALID, &cblk->flags); -create_new_record: - audio_track_cblk_t* temp = cblk; - result = AudioRecord::restoreRecord_l(temp); - cblk = temp; - } - if (result != NO_ERROR) { - ALOGW("obtainBuffer create Track error %d", result); - cblk->lock.unlock(); - return result; + const struct timespec *requested; + 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; + struct timespec timeout; + 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); +} + +status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, + struct timespec *elapsed, size_t *nonContig) +{ + // previous and new IAudioRecord 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 proxy; + sp 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 = restoreRecord_l("obtainBuffer"); + if (status != NO_ERROR) { + break; } - cblk->waitTimeMs = 0; - } - if (--waitCount == 0) { - cblk->lock.unlock(); - return TIMED_OUT; } } - // read the server count again -start_loop_here: - framesReady = mProxy->framesReady(); - } - cblk->lock.unlock(); - } + oldSequence = newSequence; - cblk->waitTimeMs = 0; - // reset time out to running value after obtaining a buffer - cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; + // Keep the extra references + proxy = mProxy; + iMem = mCblkMemory; - if (framesReq > framesReady) { - framesReq = framesReady; - } + // Non-blocking if track is stopped + if (!mActive) { + requested = &ClientProxy::kNonBlocking; + } - uint32_t u = cblk->user; - uint32_t bufferEnd = cblk->userBase + mFrameCount; + } // end of lock scope - if (framesReq > bufferEnd - u) { - framesReq = bufferEnd - u; - } + buffer.mFrameCount = audioBuffer->frameCount; + // FIXME starts the requested timeout and elapsed over from scratch + status = proxy->obtainBuffer(&buffer, requested, elapsed); + + } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); - audioBuffer->frameCount = framesReq; - audioBuffer->size = framesReq * mFrameSize; - audioBuffer->raw = mProxy->buffer(u); - active = mActive; - return active ? status_t(NO_ERROR) : status_t(STOPPED); + audioBuffer->frameCount = buffer.mFrameCount; + audioBuffer->size = buffer.mFrameCount * mFrameSize; + audioBuffer->raw = buffer.mRaw; + if (nonContig != NULL) { + *nonContig = buffer.mNonContig; + } + return status; } void AudioRecord::releaseBuffer(Buffer* audioBuffer) { - ALOG_ASSERT(mStatus == NO_ERROR && mProxy != NULL); + // all TRANSFER_* are valid + + size_t stepCount = audioBuffer->size / mFrameSize; + if (stepCount == 0) { + return; + } + + Proxy::Buffer buffer; + buffer.mFrameCount = stepCount; + buffer.mRaw = audioBuffer->raw; AutoMutex lock(mLock); - (void) mProxy->stepUser(audioBuffer->frameCount); + mInOverrun = false; + mProxy->releaseBuffer(&buffer); + + // the server does not automatically disable recorder on overrun, so no need to restart } audio_io_handle_t AudioRecord::getInput() const @@ -616,215 +613,304 @@ audio_io_handle_t AudioRecord::getInput_l() return mInput; } -int AudioRecord::getSessionId() const -{ - // no lock needed because session ID doesn't change after first set() - return mSessionId; -} - // ------------------------------------------------------------------------- ssize_t AudioRecord::read(void* buffer, size_t userSize) { - ssize_t read = 0; - Buffer audioBuffer; - int8_t *dst = static_cast(buffer); + if (mTransfer != TRANSFER_SYNC) { + return INVALID_OPERATION; + } - if (ssize_t(userSize) < 0) { - // sanity-check. user is most-likely passing an error code. - ALOGE("AudioRecord::read(buffer=%p, size=%u (%d)", - buffer, userSize, userSize); + 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("AudioRecord::read(buffer=%p, size=%u (%d)", buffer, userSize, userSize); return BAD_VALUE; } - mLock.lock(); - // acquire a strong reference on the IAudioRecord and IMemory so that they cannot be destroyed - // while we are accessing the cblk - sp audioRecord = mAudioRecord; - sp iMem = mCblkMemory; - mLock.unlock(); - - do { + ssize_t read = 0; + Buffer audioBuffer; - audioBuffer.frameCount = userSize/frameSize(); + while (userSize >= mFrameSize) { + audioBuffer.frameCount = userSize / mFrameSize; - // By using a wait count corresponding to twice the timeout period in - // obtainBuffer() we give a chance to recover once for a read timeout - // (if media_server crashed for instance) before returning a length of - // 0 bytes read to the client - status_t err = obtainBuffer(&audioBuffer, ((2 * MAX_RUN_TIMEOUT_MS) / WAIT_PERIOD_MS)); + status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); if (err < 0) { - // out of buffers, return #bytes written - if (err == status_t(NO_MORE_BUFFERS)) { + if (read > 0) { break; } - if (err == status_t(TIMED_OUT)) { - // return partial transfer count - return read; - } return ssize_t(err); } size_t bytesRead = audioBuffer.size; - memcpy(dst, audioBuffer.i8, bytesRead); - - dst += bytesRead; + memcpy(buffer, audioBuffer.i8, bytesRead); + buffer = ((char *) buffer) + bytesRead; userSize -= bytesRead; read += bytesRead; releaseBuffer(&audioBuffer); - } while (userSize); + } return read; } // ------------------------------------------------------------------------- -bool AudioRecord::processAudioBuffer(const sp& thread) +nsecs_t AudioRecord::processAudioBuffer(const sp& thread) { - Buffer audioBuffer; - uint32_t frames = mRemainingFrames; - size_t readSize; - mLock.lock(); - // acquire a strong reference on the IAudioRecord and IMemory so that they cannot be destroyed - // while we are accessing the cblk - sp audioRecord = mAudioRecord; - sp iMem = mCblkMemory; - audio_track_cblk_t* cblk = mCblk; + + // Can only reference mCblk while locked + int32_t flags = android_atomic_and(~CBLK_OVERRUN, &mCblk->flags); + + // Check for track invalidation + if (flags & CBLK_INVALID) { + (void) restoreRecord_l("processAudioBuffer"); + mLock.unlock(); + // Run again immediately, but with a new IAudioRecord + return 0; + } + bool active = mActive; - uint32_t markerPosition = mMarkerPosition; - uint32_t newPosition = mNewPosition; - uint32_t user = cblk->user; - // determine whether a marker callback will be needed, while locked - bool needMarker = !mMarkerReached && (mMarkerPosition > 0) && (user >= mMarkerPosition); - if (needMarker) { - mMarkerReached = true; - } - // determine the number of new position callback(s) that will be needed, while locked + + // Manage overrun callback, must be done under lock to avoid race with releaseBuffer() + bool newOverrun = false; + if (flags & CBLK_OVERRUN) { + if (!mInOverrun) { + mInOverrun = true; + newOverrun = true; + } + } + + // Get current position of server + size_t position = mProxy->getPosition(); + + // Manage marker callback + bool markerReached = false; + size_t markerPosition = mMarkerPosition; + // FIXME fails for wraparound, need 64 bits + if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { + mMarkerReached = markerReached = true; + } + + // Determine the number of new position callback(s) that will be needed, while locked + size_t newPosCount = 0; + size_t newPosition = mNewPosition; uint32_t updatePeriod = mUpdatePeriod; - uint32_t needNewPos = updatePeriod > 0 && user >= newPosition ? - ((user - newPosition) / updatePeriod) + 1 : 0; - mNewPosition = newPosition + updatePeriod * needNewPos; + // FIXME fails for wraparound, need 64 bits + if (updatePeriod > 0 && position >= newPosition) { + newPosCount = ((position - newPosition) / updatePeriod) + 1; + mNewPosition += updatePeriod * newPosCount; + } + + // Cache other fields that will be needed soon + size_t notificationFrames = mNotificationFrames; + if (mRefreshRemaining) { + mRefreshRemaining = false; + mRemainingFrames = notificationFrames; + mRetryOnPartialBuffer = false; + } + size_t misalignment = mProxy->getMisalignment(); + int32_t sequence = mSequence; + + // These fields don't need to be cached, because they are assigned only by set(): + // mTransfer, mCbf, mUserData, mSampleRate + mLock.unlock(); - // perform marker callback, while unlocked - if (needMarker) { + // perform callbacks while unlocked + if (newOverrun) { + mCbf(EVENT_OVERRUN, mUserData, NULL); + } + if (markerReached) { mCbf(EVENT_MARKER, mUserData, &markerPosition); } - - // perform new position callback(s), while unlocked - for (; needNewPos > 0; --needNewPos) { - uint32_t temp = newPosition; + while (newPosCount > 0) { + size_t temp = newPosition; mCbf(EVENT_NEW_POS, mUserData, &temp); newPosition += updatePeriod; + newPosCount--; + } + if (mObservedSequence != sequence) { + mObservedSequence = sequence; + mCbf(EVENT_NEW_IAUDIORECORD, mUserData, NULL); } - do { - audioBuffer.frameCount = frames; - // Calling obtainBuffer() with a wait count of 1 - // limits wait time to WAIT_PERIOD_MS. This prevents from being - // stuck here not being able to handle timed events (position, markers). - status_t err = obtainBuffer(&audioBuffer, 1); - if (err < NO_ERROR) { - if (err != TIMED_OUT) { - ALOGE_IF(err != status_t(NO_MORE_BUFFERS), - "Error obtaining an audio buffer, giving up."); - return false; + // 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) + uint32_t minFrames = ~0; + if (!markerReached && position < markerPosition) { + minFrames = markerPosition - position; + } + if (updatePeriod > 0 && updatePeriod < minFrames) { + minFrames = updatePeriod; + } + + // 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; + } + + // Convert frame units to time units + nsecs_t ns = NS_WHENEVER; + if (minFrames != (uint32_t) ~0) { + // This "fudge factor" avoids soaking CPU, and compensates for late progress by server + static const nsecs_t kFudgeNs = 10000000LL; // 10 ms + ns = ((minFrames * 1000000000LL) / mSampleRate) + kFudgeNs; + } + + // If not supplying data by EVENT_MORE_DATA, then we're done + if (mTransfer != TRANSFER_CALLBACK) { + return ns; + } + + struct timespec timeout; + const struct timespec *requested = &ClientProxy::kForever; + if (ns != NS_WHENEVER) { + timeout.tv_sec = ns / 1000000000LL; + timeout.tv_nsec = ns % 1000000000LL; + ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); + requested = &timeout; + } + + while (mRemainingFrames > 0) { + + Buffer audioBuffer; + audioBuffer.frameCount = mRemainingFrames; + size_t nonContig; + status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); + LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), + "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); + requested = &ClientProxy::kNonBlocking; + size_t avail = audioBuffer.frameCount + nonContig; + ALOGV("obtainBuffer(%u) returned %u = %u + %u", + mRemainingFrames, avail, audioBuffer.frameCount, nonContig); + if (err != NO_ERROR) { + if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) { + break; + } + ALOGE("Error %d obtaining an audio buffer, giving up.", err); + return NS_NEVER; + } + + if (mRetryOnPartialBuffer) { + mRetryOnPartialBuffer = false; + if (avail < mRemainingFrames) { + int64_t myns = ((mRemainingFrames - avail) * + 1100000000LL) / mSampleRate; + if (ns < 0 || myns < ns) { + ns = myns; + } + return ns; } - break; } - if (err == status_t(STOPPED)) return false; size_t reqSize = audioBuffer.size; mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); - readSize = audioBuffer.size; + size_t readSize = audioBuffer.size; // Sanity check on returned size - if (ssize_t(readSize) <= 0) { - // The callback is done filling buffers + if (ssize_t(readSize) < 0 || readSize > reqSize) { + ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", + reqSize, (int) readSize); + return NS_NEVER; + } + + if (readSize == 0) { + // The callback is done consuming buffers // Keep this thread going to handle timed events and - // still try to get more data in intervals of WAIT_PERIOD_MS + // still try to provide more data in intervals of WAIT_PERIOD_MS // but don't just loop and block the CPU, so wait - usleep(WAIT_PERIOD_MS*1000); - break; + return WAIT_PERIOD_MS * 1000000LL; } - if (readSize > reqSize) readSize = reqSize; - audioBuffer.size = readSize; - audioBuffer.frameCount = readSize/frameSize(); - frames -= audioBuffer.frameCount; + size_t releasedFrames = readSize / mFrameSize; + audioBuffer.frameCount = releasedFrames; + mRemainingFrames -= releasedFrames; + if (misalignment >= releasedFrames) { + misalignment -= releasedFrames; + } else { + misalignment = 0; + } releaseBuffer(&audioBuffer); - } while (frames); + // 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 (readSize < reqSize) { + continue; + } + // There could be enough non-contiguous frames available to satisfy the remaining request + if (mRemainingFrames <= nonContig) { + continue; + } - // Manage overrun callback - if (active && (mProxy->framesAvailable() == 0)) { - // The value of active is stale, but we are almost sure to be active here because - // otherwise we would have exited when obtainBuffer returned STOPPED earlier. - ALOGV("Overrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags); - if (!(android_atomic_or(CBLK_UNDERRUN, &cblk->flags) & CBLK_UNDERRUN)) { - mCbf(EVENT_OVERRUN, mUserData, NULL); +#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 (mRemainingFrames * 1100000000LL) / mSampleRate; } - } +#endif - if (frames == 0) { - mRemainingFrames = mNotificationFrames; - } else { - mRemainingFrames = frames; } - return true; + mRemainingFrames = notificationFrames; + mRetryOnPartialBuffer = true; + + // A lot has transpired since ns was calculated, so run again immediately and re-calculate + return 0; } -// must be called with mLock and cblk.lock held. Callers must also hold strong references on -// the IAudioRecord and IMemory in case they are recreated here. -// If the IAudioRecord is successfully restored, the cblk pointer is updated -status_t AudioRecord::restoreRecord_l(audio_track_cblk_t*& refCblk) +status_t AudioRecord::restoreRecord_l(const char *from) { + ALOGW("dead IAudioRecord, creating a new one from %s()", from); + ++mSequence; status_t result; - audio_track_cblk_t* cblk = refCblk; - audio_track_cblk_t* newCblk = cblk; - ALOGW("dead IAudioRecord, creating a new one"); - - // signal old cblk condition so that other threads waiting for available buffers stop - // waiting now - cblk->cv.broadcast(); - cblk->lock.unlock(); - // if the new IAudioRecord is created, openRecord_l() will modify the // following member variables: mAudioRecord, mCblkMemory and mCblk. // It will also delete the strong references on previous IAudioRecord and IMemory - result = openRecord_l(mSampleRate, mFormat, mFrameCount, getInput_l()); + size_t position = mProxy->getPosition(); + mNewPosition = position + mUpdatePeriod; + result = openRecord_l(mSampleRate, mFormat, mFrameCount, getInput_l(), position); if (result == NO_ERROR) { - newCblk = mCblk; - // callback thread or sync event hasn't changed - result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0); + if (mActive) { + // callback thread or sync event hasn't changed + // FIXME this fails if we have a new AudioFlinger instance + result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0); + } } if (result != NO_ERROR) { + ALOGW("restoreRecord_l() failed status %d", result); mActive = false; } - ALOGV("restoreRecord_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x", - result, mActive, newCblk, cblk, newCblk->flags, cblk->flags); - - if (result == NO_ERROR) { - // from now on we switch to the newly created cblk - refCblk = newCblk; - } - newCblk->lock.lock(); + return result; +} - ALOGW_IF(result != NO_ERROR, "restoreRecord_l() error %d", result); +// ========================================================================= - return result; +void AudioRecord::DeathNotifier::binderDied(const wp& who) +{ + sp audioRecord = mAudioRecord.promote(); + if (audioRecord != 0) { + AutoMutex lock(audioRecord->mLock); + audioRecord->mProxy->binderDied(); + } } // ========================================================================= AudioRecord::AudioRecordThread::AudioRecordThread(AudioRecord& receiver, bool bCanCallJava) - : Thread(bCanCallJava), mReceiver(receiver), mPaused(true) + : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mResumeLatch(false) { } @@ -842,10 +928,26 @@ bool AudioRecord::AudioRecordThread::threadLoop() return true; } } - if (!mReceiver.processAudioBuffer(this)) { - pause(); + nsecs_t ns = mReceiver.processAudioBuffer(this); + switch (ns) { + case 0: + return true; + case NS_WHENEVER: + sleep(1); + return true; + case NS_INACTIVE: + pauseConditional(); + return true; + case NS_NEVER: + return false; + default: + LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); + struct timespec req; + req.tv_sec = ns / 1000000000LL; + req.tv_nsec = ns % 1000000000LL; + nanosleep(&req, NULL /*rem*/); + return true; } - return true; } void AudioRecord::AudioRecordThread::requestExit() @@ -859,6 +961,17 @@ void AudioRecord::AudioRecordThread::pause() { AutoMutex _l(mMyLock); mPaused = true; + mResumeLatch = false; +} + +void AudioRecord::AudioRecordThread::pauseConditional() +{ + AutoMutex _l(mMyLock); + if (mResumeLatch) { + mResumeLatch = false; + } else { + mPaused = true; + } } void AudioRecord::AudioRecordThread::resume() @@ -866,7 +979,10 @@ void AudioRecord::AudioRecordThread::resume() AutoMutex _l(mMyLock); if (mPaused) { mPaused = false; + mResumeLatch = false; mMyCond.signal(); + } else { + mResumeLatch = true; } } diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp index 77fc6f6..faca054 100644 --- a/media/libmedia/AudioTrack.cpp +++ b/media/libmedia/AudioTrack.cpp @@ -19,31 +19,14 @@ //#define LOG_NDEBUG 0 #define LOG_TAG "AudioTrack" -#include -#include -#include - -#include #include - -#include - -#include +#include +#include #include - #include -#include -#include -#include -#include - -#include -#include +#include -#include -#include - -#include +#define WAIT_PERIOD_MS 10 namespace android { // --------------------------------------------------------------------------- @@ -82,7 +65,9 @@ status_t AudioTrack::getMinFrameCount( // Ensure that buffer depth covers at least audio hardware latency uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); - if (minBufCount < 2) minBufCount = 2; + if (minBufCount < 2) { + minBufCount = 2; + } *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : afFrameCount * minBufCount * sampleRate / afSampleRate; @@ -97,8 +82,7 @@ AudioTrack::AudioTrack() : mStatus(NO_INIT), mIsTimed(false), mPreviousPriority(ANDROID_PRIORITY_NORMAL), - mPreviousSchedulingGroup(SP_DEFAULT), - mProxy(NULL) + mPreviousSchedulingGroup(SP_DEFAULT) { } @@ -112,16 +96,16 @@ AudioTrack::AudioTrack( callback_t cbf, void* user, int notificationFrames, - int sessionId) + int sessionId, + transfer_type transferType) : mStatus(NO_INIT), mIsTimed(false), mPreviousPriority(ANDROID_PRIORITY_NORMAL), - mPreviousSchedulingGroup(SP_DEFAULT), - mProxy(NULL) + mPreviousSchedulingGroup(SP_DEFAULT) { mStatus = set(streamType, sampleRate, format, channelMask, frameCount, flags, cbf, user, notificationFrames, - 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId); + 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType); } AudioTrack::AudioTrack( @@ -134,27 +118,20 @@ AudioTrack::AudioTrack( callback_t cbf, void* user, int notificationFrames, - int sessionId) + int sessionId, + transfer_type transferType) : mStatus(NO_INIT), mIsTimed(false), mPreviousPriority(ANDROID_PRIORITY_NORMAL), - mPreviousSchedulingGroup(SP_DEFAULT), - mProxy(NULL) + mPreviousSchedulingGroup(SP_DEFAULT) { - if (sharedBuffer == 0) { - ALOGE("sharedBuffer must be non-0"); - mStatus = BAD_VALUE; - return; - } mStatus = set(streamType, sampleRate, format, channelMask, 0 /*frameCount*/, flags, cbf, user, notificationFrames, - sharedBuffer, false /*threadCanCallJava*/, sessionId); + sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType); } AudioTrack::~AudioTrack() { - ALOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer()); - if (mStatus == NO_ERROR) { // Make sure that callback function exits in the case where // it is looping on buffer full condition in obtainBuffer(). @@ -165,11 +142,13 @@ AudioTrack::~AudioTrack() mAudioTrackThread->requestExitAndWait(); mAudioTrackThread.clear(); } - mAudioTrack.clear(); + if (mAudioTrack != 0) { + mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); + mAudioTrack.clear(); + } IPCThreadState::self()->flushCommands(); AudioSystem::releaseAudioSessionId(mSessionId); } - delete mProxy; } status_t AudioTrack::set( @@ -184,8 +163,44 @@ status_t AudioTrack::set( int notificationFrames, const sp& sharedBuffer, bool threadCanCallJava, - int sessionId) + int sessionId, + transfer_type transferType) { + 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; + } + mTransfer = transferType; + // FIXME "int" here is legacy and will be replaced by size_t later if (frameCountInt < 0) { ALOGE("Invalid frame count %d", frameCountInt); @@ -199,6 +214,7 @@ status_t AudioTrack::set( ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags); AutoMutex lock(mLock); + if (mAudioTrack != 0) { ALOGE("Track already in use"); return INVALID_OPERATION; @@ -228,7 +244,7 @@ status_t AudioTrack::set( // validate parameters if (!audio_is_valid_format(format)) { - ALOGE("Invalid format"); + ALOGE("Invalid format %d", format); return BAD_VALUE; } @@ -281,6 +297,7 @@ status_t AudioTrack::set( mFrameCount = frameCount; mReqFrameCount = frameCount; mNotificationFramesReq = notificationFrames; + mNotificationFramesAct = 0; mSessionId = sessionId; mAuxEffectId = 0; mFlags = flags; @@ -298,7 +315,8 @@ status_t AudioTrack::set( frameCount, flags, sharedBuffer, - output); + output, + 0 /*epoch*/); if (status != NO_ERROR) { if (mAudioTrackThread != 0) { @@ -309,20 +327,21 @@ status_t AudioTrack::set( } mStatus = NO_ERROR; - mStreamType = streamType; mFormat = format; - mSharedBuffer = sharedBuffer; - mActive = false; + mState = STATE_STOPPED; mUserData = user; - mLoopCount = 0; + mLoopPeriod = 0; mMarkerPosition = 0; mMarkerReached = false; mNewPosition = 0; mUpdatePeriod = 0; - mFlushed = false; AudioSystem::acquireAudioSessionId(mSessionId); + mSequence = 1; + mObservedSequence = mSequence; + mInUnderrun = false; + return NO_ERROR; } @@ -330,87 +349,45 @@ status_t AudioTrack::set( void AudioTrack::start() { - sp t = mAudioTrackThread; - - ALOGV("start %p", this); - AutoMutex lock(mLock); - // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed - // while we are accessing the cblk - sp audioTrack = mAudioTrack; - sp iMem = mCblkMemory; - audio_track_cblk_t* cblk = mCblk; + if (mState == STATE_ACTIVE) { + return; + } - if (!mActive) { - mFlushed = false; - mActive = true; - mNewPosition = cblk->server + mUpdatePeriod; - cblk->lock.lock(); - cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; - cblk->waitTimeMs = 0; - android_atomic_and(~CBLK_DISABLED, &cblk->flags); - if (t != 0) { - t->resume(); - } else { - mPreviousPriority = getpriority(PRIO_PROCESS, 0); - get_sched_policy(0, &mPreviousSchedulingGroup); - androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); - } + mInUnderrun = true; - ALOGV("start %p before lock cblk %p", this, cblk); - status_t status = NO_ERROR; - if (!(cblk->flags & CBLK_INVALID)) { - cblk->lock.unlock(); - ALOGV("mAudioTrack->start()"); - status = mAudioTrack->start(); - cblk->lock.lock(); - if (status == DEAD_OBJECT) { - android_atomic_or(CBLK_INVALID, &cblk->flags); - } - } - if (cblk->flags & CBLK_INVALID) { - audio_track_cblk_t* temp = cblk; - status = restoreTrack_l(temp, true /*fromStart*/); - cblk = temp; - } - cblk->lock.unlock(); - if (status != NO_ERROR) { - ALOGV("start() failed"); - mActive = false; - if (t != 0) { - t->pause(); - } else { - setpriority(PRIO_PROCESS, 0, mPreviousPriority); - set_sched_policy(0, mPreviousSchedulingGroup); - } - } + State previousState = mState; + mState = STATE_ACTIVE; + if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { + // reset current position as seen by client to 0 + mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); } + mNewPosition = mProxy->getPosition() + mUpdatePeriod; + int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->flags); -} - -void AudioTrack::stop() -{ sp t = mAudioTrackThread; + if (t != 0) { + t->resume(); + } else { + mPreviousPriority = getpriority(PRIO_PROCESS, 0); + get_sched_policy(0, &mPreviousSchedulingGroup); + androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); + } - ALOGV("stop %p", this); - - AutoMutex lock(mLock); - if (mActive) { - mActive = false; - mCblk->cv.signal(); - mAudioTrack->stop(); - // Cancel loops (If we are in the middle of a loop, playback - // would not stop until loopCount reaches 0). - setLoop_l(0, 0, 0); - // the playback head position will reset to 0, so if a marker is set, we need - // to activate it again - mMarkerReached = false; - // Force flush if a shared buffer is used otherwise audioflinger - // will not stop before end of buffer is reached. - // It may be needed to make sure that we stop playback, likely in case looping is on. - if (mSharedBuffer != 0) { - flush_l(); + status_t status = NO_ERROR; + if (!(flags & CBLK_INVALID)) { + status = mAudioTrack->start(); + if (status == DEAD_OBJECT) { + flags |= CBLK_INVALID; } + } + if (flags & CBLK_INVALID) { + status = restoreTrack_l("start"); + } + + if (status != NO_ERROR) { + ALOGE("start() status %d", status); + mState = previousState; if (t != 0) { t->pause(); } else { @@ -419,57 +396,85 @@ void AudioTrack::stop() } } + // FIXME discarding status +} + +void AudioTrack::stop() +{ + AutoMutex lock(mLock); + // FIXME pause then stop should not be a nop + if (mState != STATE_ACTIVE) { + return; + } + + mState = STATE_STOPPED; + mProxy->interrupt(); + mAudioTrack->stop(); + // the playback head position will reset to 0, so if a marker is set, we need + // to activate it again + mMarkerReached = false; +#if 0 + // Force flush if a shared buffer is used otherwise audioflinger + // will not stop before end of buffer is reached. + // It may be needed to make sure that we stop playback, likely in case looping is on. + if (mSharedBuffer != 0) { + flush_l(); + } +#endif + sp t = mAudioTrackThread; + if (t != 0) { + t->pause(); + } else { + setpriority(PRIO_PROCESS, 0, mPreviousPriority); + set_sched_policy(0, mPreviousSchedulingGroup); + } } bool AudioTrack::stopped() const { AutoMutex lock(mLock); - return stopped_l(); + return mState != STATE_ACTIVE; } void AudioTrack::flush() { + if (mSharedBuffer != 0) { + return; + } AutoMutex lock(mLock); - if (!mActive && mSharedBuffer == 0) { - flush_l(); + if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { + return; } + flush_l(); } void AudioTrack::flush_l() { - ALOGV("flush"); - ALOG_ASSERT(!mActive); + ALOG_ASSERT(mState != STATE_ACTIVE); // clear playback marker and periodic update counter mMarkerPosition = 0; mMarkerReached = false; mUpdatePeriod = 0; - mFlushed = true; + mState = STATE_FLUSHED; + mProxy->flush(); mAudioTrack->flush(); - // Release AudioTrack callback thread in case it was waiting for new buffers - // in AudioTrack::obtainBuffer() - mCblk->cv.signal(); } void AudioTrack::pause() { - ALOGV("pause"); AutoMutex lock(mLock); - if (mActive) { - mActive = false; - mCblk->cv.signal(); - mAudioTrack->pause(); + if (mState != STATE_ACTIVE) { + return; } + mState = STATE_PAUSED; + mProxy->interrupt(); + mAudioTrack->pause(); } status_t AudioTrack::setVolume(float left, float right) { - if (mStatus != NO_ERROR) { - return mStatus; - } - ALOG_ASSERT(mProxy != NULL); - if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { return BAD_VALUE; } @@ -490,18 +495,11 @@ status_t AudioTrack::setVolume(float volume) status_t AudioTrack::setAuxEffectSendLevel(float level) { - ALOGV("setAuxEffectSendLevel(%f)", level); - - if (mStatus != NO_ERROR) { - return mStatus; - } - ALOG_ASSERT(mProxy != NULL); - if (level < 0.0f || level > 1.0f) { return BAD_VALUE; } - AutoMutex lock(mLock); + AutoMutex lock(mLock); mSendLevel = level; mProxy->setSendLevel(level); @@ -511,18 +509,17 @@ status_t AudioTrack::setAuxEffectSendLevel(float level) void AudioTrack::getAuxEffectSendLevel(float* level) const { if (level != NULL) { - *level = mSendLevel; + *level = mSendLevel; } } status_t AudioTrack::setSampleRate(uint32_t rate) { - uint32_t afSamplingRate; - if (mIsTimed) { return INVALID_OPERATION; } + uint32_t afSamplingRate; if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { return NO_INIT; } @@ -550,78 +547,44 @@ uint32_t AudioTrack::getSampleRate() const status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) { - AutoMutex lock(mLock); - return setLoop_l(loopStart, loopEnd, loopCount); -} - -// must be called with mLock held -status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) -{ if (mSharedBuffer == 0 || mIsTimed) { return INVALID_OPERATION; } - if (loopCount < 0 && loopCount != -1) { - return BAD_VALUE; - } - -#if 0 - // This will be for the new interpretation of loopStart and loopEnd - - if (loopCount != 0) { - if (loopStart >= mFrameCount || loopEnd >= mFrameCount || loopStart >= loopEnd) { - return BAD_VALUE; - } - uint32_t periodFrames = loopEnd - loopStart; - if (periodFrames < PERIOD_FRAMES_MIN) { - return BAD_VALUE; - } - } - - // The remainder of this code still uses the old interpretation -#endif - - audio_track_cblk_t* cblk = mCblk; - - Mutex::Autolock _l(cblk->lock); - if (loopCount == 0) { - cblk->loopStart = UINT_MAX; - cblk->loopEnd = UINT_MAX; - cblk->loopCount = 0; - mLoopCount = 0; - return NO_ERROR; - } - - if (loopStart >= loopEnd || - loopEnd - loopStart > mFrameCount || - cblk->server > loopStart) { - ALOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, " - "user %d", loopStart, loopEnd, loopCount, mFrameCount, cblk->user); + ; + } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && + loopEnd - loopStart >= MIN_LOOP) { + ; + } else { return BAD_VALUE; } - if ((mSharedBuffer != 0) && (loopEnd > mFrameCount)) { - ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, " - "framecount %d", - loopStart, loopEnd, mFrameCount); - 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; } - - cblk->loopStart = loopStart; - cblk->loopEnd = loopEnd; - cblk->loopCount = loopCount; - mLoopCount = loopCount; - + setLoop_l(loopStart, loopEnd, loopCount); return NO_ERROR; } +void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) +{ + // FIXME If setting a loop also sets position to start of loop, then + // this is correct. Otherwise it should be removed. + mNewPosition = mProxy->getPosition() + mUpdatePeriod; + mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; + mStaticProxy->setLoop(loopStart, loopEnd, loopCount); +} + status_t AudioTrack::setMarkerPosition(uint32_t marker) { if (mCbf == NULL) { return INVALID_OPERATION; } + AutoMutex lock(mLock); mMarkerPosition = marker; mMarkerReached = false; @@ -634,6 +597,7 @@ status_t AudioTrack::getMarkerPosition(uint32_t *marker) const return BAD_VALUE; } + AutoMutex lock(mLock); *marker = mMarkerPosition; return NO_ERROR; @@ -645,9 +609,8 @@ status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) return INVALID_OPERATION; } - uint32_t curPosition; - getPosition(&curPosition); - mNewPosition = curPosition + updatePeriod; + AutoMutex lock(mLock); + mNewPosition = mProxy->getPosition() + updatePeriod; mUpdatePeriod = updatePeriod; return NO_ERROR; @@ -659,6 +622,7 @@ status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const return BAD_VALUE; } + AutoMutex lock(mLock); *updatePeriod = mUpdatePeriod; return NO_ERROR; @@ -669,49 +633,44 @@ status_t AudioTrack::setPosition(uint32_t position) if (mSharedBuffer == 0 || mIsTimed) { return INVALID_OPERATION; } - - AutoMutex lock(mLock); - - if (!stopped_l()) { - return INVALID_OPERATION; - } - -#if 0 - // This will be for the new interpretation of position - - if (position >= mFrameCount) { + if (position > mFrameCount) { return BAD_VALUE; } - // The remainder of this code still uses the old interpretation -#endif - - audio_track_cblk_t* cblk = mCblk; - Mutex::Autolock _l(cblk->lock); - - if (position > cblk->user) { - 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; } - - cblk->server = position; - android_atomic_or(CBLK_FORCEREADY, &cblk->flags); + mNewPosition = mProxy->getPosition() + mUpdatePeriod; + mLoopPeriod = 0; + // FIXME Check whether loops and setting position are incompatible in old code. + // If we use setLoop for both purposes we lose the capability to set the position while looping. + mStaticProxy->setLoop(position, mFrameCount, 0); return NO_ERROR; } -status_t AudioTrack::getPosition(uint32_t *position) +status_t AudioTrack::getPosition(uint32_t *position) const { if (position == NULL) { return BAD_VALUE; } + AutoMutex lock(mLock); - *position = mFlushed ? 0 : mCblk->server; + // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes + *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 0 : + mProxy->getPosition(); return NO_ERROR; } -#if 0 -status_t AudioTrack::getBufferPosition(uint32_t *position) +status_t AudioTrack::getBufferPosition(size_t *position) { if (mSharedBuffer == 0 || mIsTimed) { return INVALID_OPERATION; @@ -719,33 +678,28 @@ status_t AudioTrack::getBufferPosition(uint32_t *position) if (position == NULL) { return BAD_VALUE; } - *position = 0; + AutoMutex lock(mLock); + *position = mStaticProxy->getBufferPosition(); return NO_ERROR; } -#endif status_t AudioTrack::reload() { - if (mStatus != NO_ERROR) { - return mStatus; - } - ALOG_ASSERT(mProxy != NULL); - if (mSharedBuffer == 0 || mIsTimed) { return INVALID_OPERATION; } AutoMutex lock(mLock); - - if (!stopped_l()) { + // See setPosition() regarding setting parameters such as loop points or position while active + if (mState == STATE_ACTIVE) { return INVALID_OPERATION; } - - flush_l(); - - (void) mProxy->stepUser(mFrameCount); - + mNewPosition = mUpdatePeriod; + mLoopPeriod = 0; + // FIXME The new code cannot reload while keeping a loop specified. + // Need to check how the old code handled this, and whether it's a significant change. + mStaticProxy->setLoop(0, mFrameCount, 0); return NO_ERROR; } @@ -764,7 +718,7 @@ audio_io_handle_t AudioTrack::getOutput_l() status_t AudioTrack::attachAuxEffect(int effectId) { - ALOGV("attachAuxEffect(%d)", effectId); + AutoMutex lock(mLock); status_t status = mAudioTrack->attachAuxEffect(effectId); if (status == NO_ERROR) { mAuxEffectId = effectId; @@ -782,7 +736,8 @@ status_t AudioTrack::createTrack_l( size_t frameCount, audio_output_flags_t flags, const sp& sharedBuffer, - audio_io_handle_t output) + audio_io_handle_t output, + size_t epoch) { status_t status; const sp& audioFlinger = AudioSystem::get_audio_flinger(); @@ -792,7 +747,8 @@ status_t AudioTrack::createTrack_l( } uint32_t afLatency; - if (AudioSystem::getLatency(output, streamType, &afLatency) != NO_ERROR) { + if ((status = AudioSystem::getLatency(output, streamType, &afLatency)) != NO_ERROR) { + ALOGE("getLatency(%d) failed status %d", output, status); return NO_INIT; } @@ -820,7 +776,10 @@ status_t AudioTrack::createTrack_l( frameCount = sharedBuffer->size(); } else if (frameCount == 0) { size_t afFrameCount; - if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) { + status = AudioSystem::getFrameCount(output, streamType, &afFrameCount); + if (status != NO_ERROR) { + ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, + status); return NO_INIT; } frameCount = afFrameCount; @@ -851,11 +810,16 @@ status_t AudioTrack::createTrack_l( // FIXME move these calculations and associated checks to server uint32_t afSampleRate; - if (AudioSystem::getSamplingRate(output, streamType, &afSampleRate) != NO_ERROR) { + status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate); + if (status != NO_ERROR) { + ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType, + status); return NO_INIT; } size_t afFrameCount; - if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) { + status = AudioSystem::getFrameCount(output, streamType, &afFrameCount); + if (status != NO_ERROR) { + ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status); return NO_INIT; } @@ -875,12 +839,9 @@ status_t AudioTrack::createTrack_l( if (frameCount == 0) { frameCount = minFrameCount; } - if (mNotificationFramesAct == 0) { - mNotificationFramesAct = frameCount/2; - } // Make sure that application is notified with sufficient margin // before underrun - if (mNotificationFramesAct > frameCount/2) { + if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/2) { mNotificationFramesAct = frameCount/2; } if (frameCount < minFrameCount) { @@ -930,6 +891,10 @@ status_t AudioTrack::createTrack_l( ALOGE("Could not get control block"); return NO_INIT; } + if (mAudioTrack != 0) { + mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); + mDeathNotifier.clear(); + } mAudioTrack = track; mCblkMemory = iMem; audio_track_cblk_t* cblk = static_cast(iMem->pointer()); @@ -947,26 +912,38 @@ status_t AudioTrack::createTrack_l( if (trackFlags & IAudioFlinger::TRACK_FAST) { ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); mAwaitBoost = true; + if (sharedBuffer == 0) { + // double-buffering is not required for fast tracks, due to tighter scheduling + if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount) { + mNotificationFramesAct = frameCount; + } + } } else { ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount); // once denied, do not request again if IAudioTrack is re-created flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); mFlags = flags; - } - if (sharedBuffer == 0) { - mNotificationFramesAct = frameCount/2; + if (sharedBuffer == 0) { + if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/2) { + mNotificationFramesAct = frameCount/2; + } + } } } + 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 (sharedBuffer == 0) { - mBuffers = (char*)cblk + sizeof(audio_track_cblk_t); + buffers = (char*)cblk + sizeof(audio_track_cblk_t); } else { - mBuffers = sharedBuffer->pointer(); + buffers = sharedBuffer->pointer(); } mAudioTrack->attachAuxEffect(mAuxEffectId); - cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; - cblk->waitTimeMs = 0; - mRemainingFrames = mNotificationFramesAct; // FIXME don't believe this lie mLatency = afLatency + (1000*frameCount) / sampleRate; mFrameCount = frameCount; @@ -977,147 +954,143 @@ status_t AudioTrack::createTrack_l( } // update proxy - delete mProxy; - mProxy = new AudioTrackClientProxy(cblk, mBuffers, frameCount, mFrameSizeAF); + if (sharedBuffer == 0) { + mStaticProxy.clear(); + mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); + } else { + mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); + mProxy = mStaticProxy; + } mProxy->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000)); mProxy->setSendLevel(mSendLevel); mProxy->setSampleRate(mSampleRate); - if (sharedBuffer != 0) { - // Force buffer full condition as data is already present in shared memory - mProxy->stepUser(frameCount); - } + mProxy->setEpoch(epoch); + mProxy->setMinimum(mNotificationFramesAct); + + mDeathNotifier = new DeathNotifier(this); + mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); return NO_ERROR; } status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) { - ALOG_ASSERT(mStatus == NO_ERROR && mProxy != NULL); + if (audioBuffer == NULL) { + return BAD_VALUE; + } + if (mTransfer != TRANSFER_OBTAIN) { + audioBuffer->frameCount = 0; + audioBuffer->size = 0; + audioBuffer->raw = NULL; + return INVALID_OPERATION; + } + + const struct timespec *requested; + 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; + struct timespec timeout; + 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); +} + +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; - AutoMutex lock(mLock); - bool active; - status_t result = NO_ERROR; - audio_track_cblk_t* cblk = mCblk; - uint32_t framesReq = audioBuffer->frameCount; - uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS; + Proxy::Buffer buffer; + status_t status = NO_ERROR; - audioBuffer->frameCount = 0; - audioBuffer->size = 0; + static const int32_t kMaxTries = 5; + int32_t tryCounter = kMaxTries; - size_t framesAvail = mProxy->framesAvailable(); + 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 proxy; + sp iMem; - cblk->lock.lock(); - if (cblk->flags & CBLK_INVALID) { - goto create_new_track; - } - cblk->lock.unlock(); - - if (framesAvail == 0) { - cblk->lock.lock(); - goto start_loop_here; - while (framesAvail == 0) { - active = mActive; - if (CC_UNLIKELY(!active)) { - ALOGV("Not active and NO_MORE_BUFFERS"); - cblk->lock.unlock(); - return NO_MORE_BUFFERS; - } - if (CC_UNLIKELY(!waitCount)) { - cblk->lock.unlock(); - return WOULD_BLOCK; - } - if (!(cblk->flags & CBLK_INVALID)) { - mLock.unlock(); - // this condition is in shared memory, so if IAudioTrack and control block - // are replaced due to mediaserver death or IAudioTrack invalidation then - // cv won't be signalled, but fortunately the timeout will limit the wait - result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); - cblk->lock.unlock(); - mLock.lock(); - if (!mActive) { - return status_t(STOPPED); - } - // IAudioTrack may have been re-created while mLock was unlocked - cblk = mCblk; - cblk->lock.lock(); - } + { // start of lock scope + AutoMutex lock(mLock); - if (cblk->flags & CBLK_INVALID) { - goto create_new_track; - } - if (CC_UNLIKELY(result != NO_ERROR)) { - cblk->waitTimeMs += waitTimeMs; - if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) { - // timing out when a loop has been set and we have already written upto loop end - // is a normal condition: no need to wake AudioFlinger up. - if (cblk->user < cblk->loopEnd) { - ALOGW("obtainBuffer timed out (is the CPU pegged?) %p name=%#x user=%08x, " - "server=%08x", this, cblk->mName, cblk->user, cblk->server); - //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140) - cblk->lock.unlock(); - result = mAudioTrack->start(); - cblk->lock.lock(); - if (result == DEAD_OBJECT) { - android_atomic_or(CBLK_INVALID, &cblk->flags); -create_new_track: - audio_track_cblk_t* temp = cblk; - result = restoreTrack_l(temp, false /*fromStart*/); - cblk = temp; - } - if (result != NO_ERROR) { - ALOGW("obtainBuffer create Track error %d", result); - cblk->lock.unlock(); - return result; - } + 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) { + break; } - cblk->waitTimeMs = 0; } + } + oldSequence = newSequence; - if (--waitCount == 0) { - cblk->lock.unlock(); - return TIMED_OUT; - } + // Keep the extra references + proxy = mProxy; + iMem = mCblkMemory; + + // Non-blocking if track is stopped or paused + if (mState != STATE_ACTIVE) { + requested = &ClientProxy::kNonBlocking; } - // read the server count again - start_loop_here: - framesAvail = mProxy->framesAvailable_l(); - } - cblk->lock.unlock(); - } - cblk->waitTimeMs = 0; + } // end of lock scope - if (framesReq > framesAvail) { - framesReq = framesAvail; - } + buffer.mFrameCount = audioBuffer->frameCount; + // FIXME starts the requested timeout and elapsed over from scratch + status = proxy->obtainBuffer(&buffer, requested, elapsed); - uint32_t u = cblk->user; - uint32_t bufferEnd = cblk->userBase + mFrameCount; + } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); - if (framesReq > bufferEnd - u) { - framesReq = bufferEnd - u; + audioBuffer->frameCount = buffer.mFrameCount; + audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; + audioBuffer->raw = buffer.mRaw; + if (nonContig != NULL) { + *nonContig = buffer.mNonContig; } - - audioBuffer->frameCount = framesReq; - audioBuffer->size = framesReq * mFrameSizeAF; - audioBuffer->raw = mProxy->buffer(u); - active = mActive; - return active ? status_t(NO_ERROR) : status_t(STOPPED); + return status; } void AudioTrack::releaseBuffer(Buffer* audioBuffer) { - ALOG_ASSERT(mStatus == NO_ERROR && mProxy != NULL); + if (mTransfer == TRANSFER_SHARED) { + return; + } + + size_t stepCount = audioBuffer->size / mFrameSizeAF; + if (stepCount == 0) { + return; + } + + Proxy::Buffer buffer; + buffer.mFrameCount = stepCount; + buffer.mRaw = audioBuffer->raw; AutoMutex lock(mLock); - audio_track_cblk_t* cblk = mCblk; - (void) mProxy->stepUser(audioBuffer->frameCount); - if (audioBuffer->frameCount > 0) { - // restart track if it was disabled by audioflinger due to previous underrun - if (mActive && (cblk->flags & CBLK_DISABLED)) { - android_atomic_and(~CBLK_DISABLED, &cblk->flags); - ALOGW("releaseBuffer() track %p name=%#x disabled, restarting", this, cblk->mName); + mInUnderrun = false; + mProxy->releaseBuffer(&buffer); + + // restart track if it was disabled by audioflinger due to previous underrun + if (mState == STATE_ACTIVE) { + audio_track_cblk_t* cblk = mCblk; + if (android_atomic_and(~CBLK_DISABLED, &cblk->flags) & CBLK_DISABLED) { + ALOGW("releaseBuffer() track %p name=%#x disabled due to previous underrun, restarting", + this, cblk->mName); + // FIXME ignoring status mAudioTrack->start(); } } @@ -1127,68 +1100,46 @@ void AudioTrack::releaseBuffer(Buffer* audioBuffer) ssize_t AudioTrack::write(const void* buffer, size_t userSize) { - - if (mSharedBuffer != 0 || mIsTimed) { + if (mTransfer != TRANSFER_SYNC || mIsTimed) { return INVALID_OPERATION; } - if (ssize_t(userSize) < 0) { + 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=%u (%d)", - buffer, userSize, userSize); + ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", buffer, userSize, userSize); return BAD_VALUE; } - ALOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive); - - if (userSize == 0) { - return 0; - } - - // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed - // while we are accessing the cblk - mLock.lock(); - sp audioTrack = mAudioTrack; - sp iMem = mCblkMemory; - mLock.unlock(); - - // since mLock is unlocked the IAudioTrack and shared memory may be re-created, - // so all cblk references might still refer to old shared memory, but that should be benign - - ssize_t written = 0; - const int8_t *src = (const int8_t *)buffer; + size_t written = 0; Buffer audioBuffer; - size_t frameSz = frameSize(); - do { - audioBuffer.frameCount = userSize/frameSz; + while (userSize >= mFrameSize) { + audioBuffer.frameCount = userSize / mFrameSize; - status_t err = obtainBuffer(&audioBuffer, -1); + status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); if (err < 0) { - // out of buffers, return #bytes written - if (err == status_t(NO_MORE_BUFFERS)) { + if (written > 0) { break; } return ssize_t(err); } size_t toWrite; - if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { // Divide capacity by 2 to take expansion into account - toWrite = audioBuffer.size>>1; - memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) src, toWrite); + toWrite = audioBuffer.size >> 1; + memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); } else { toWrite = audioBuffer.size; - memcpy(audioBuffer.i8, src, toWrite); + memcpy(audioBuffer.i8, buffer, toWrite); } - src += toWrite; + buffer = ((const char *) buffer) + toWrite; userSize -= toWrite; written += toWrite; releaseBuffer(&audioBuffer); - } while (userSize >= frameSz); + } return written; } @@ -1204,10 +1155,12 @@ status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp* buffer) AutoMutex lock(mLock); status_t result = UNKNOWN_ERROR; +#if 1 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed // while we are accessing the cblk sp audioTrack = mAudioTrack; sp iMem = mCblkMemory; +#endif // If the track is not invalid already, try to allocate a buffer. alloc // fails indicating that the server is dead, flag the track as invalid so @@ -1223,13 +1176,9 @@ status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp* buffer) // If the track is invalid at this point, attempt to restore it. and try the // allocation one more time. if (cblk->flags & CBLK_INVALID) { - cblk->lock.lock(); - audio_track_cblk_t* temp = cblk; - result = restoreTrack_l(temp, false /*fromStart*/); - cblk = temp; - cblk->lock.unlock(); + result = restoreTrack_l("allocateTimedBuffer"); - if (result == OK) { + if (result == NO_ERROR) { result = mAudioTrack->allocateTimedBuffer(size, buffer); } } @@ -1246,9 +1195,10 @@ status_t TimedAudioTrack::queueTimedBuffer(const sp& buffer, audio_track_cblk_t* cblk = mCblk; // restart track if it was disabled by audioflinger due to previous underrun if (buffer->size() != 0 && status == NO_ERROR && - mActive && (cblk->flags & CBLK_DISABLED)) { + (mState == STATE_ACTIVE) && (cblk->flags & CBLK_DISABLED)) { android_atomic_and(~CBLK_DISABLED, &cblk->flags); ALOGW("queueTimedBuffer() track %p disabled, restarting", this); + // FIXME ignoring status mAudioTrack->start(); } } @@ -1263,12 +1213,8 @@ status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, // ------------------------------------------------------------------------- -bool AudioTrack::processAudioBuffer(const sp& thread) +nsecs_t AudioTrack::processAudioBuffer(const sp& thread) { - Buffer audioBuffer; - uint32_t frames; - size_t writtenSize; - mLock.lock(); if (mAwaitBoost) { mAwaitBoost = false; @@ -1289,86 +1235,181 @@ bool AudioTrack::processAudioBuffer(const sp& thread) } return true; } - // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed - // while we are accessing the cblk - sp audioTrack = mAudioTrack; - sp iMem = mCblkMemory; - audio_track_cblk_t* cblk = mCblk; - bool active = mActive; - mLock.unlock(); - // since mLock is unlocked the IAudioTrack and shared memory may be re-created, - // so all cblk references might still refer to old shared memory, but that should be benign + // Can only reference mCblk while locked + int32_t flags = android_atomic_and( + ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->flags); - // Manage underrun callback - if (active && (mProxy->framesAvailable() == mFrameCount)) { - ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags); - if (!(android_atomic_or(CBLK_UNDERRUN, &cblk->flags) & CBLK_UNDERRUN)) { - mCbf(EVENT_UNDERRUN, mUserData, 0); - if (cblk->server == mFrameCount) { - mCbf(EVENT_BUFFER_END, mUserData, 0); - } - if (mSharedBuffer != 0) { - return false; - } - } + // Check for track invalidation + if (flags & CBLK_INVALID) { + (void) restoreTrack_l("processAudioBuffer"); + mLock.unlock(); + // Run again immediately, but with a new IAudioTrack + return 0; } - // Manage loop end callback - while (mLoopCount > cblk->loopCount) { - int loopCount = -1; - mLoopCount--; - if (mLoopCount >= 0) loopCount = mLoopCount; + bool active = mState == STATE_ACTIVE; - mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount); + // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() + bool newUnderrun = false; + if (flags & CBLK_UNDERRUN) { +#if 0 + // Currently in shared buffer mode, when the server reaches the end of buffer, + // the track stays active in continuous underrun state. It's up to the application + // to pause or stop the track, or set the position to a new offset within buffer. + // This was some experimental code to auto-pause on underrun. Keeping it here + // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. + if (mTransfer == TRANSFER_SHARED) { + mState = STATE_PAUSED; + active = false; + } +#endif + if (!mInUnderrun) { + mInUnderrun = true; + newUnderrun = true; + } } + // Get current position of server + size_t position = mProxy->getPosition(); + // Manage marker callback - if (!mMarkerReached && (mMarkerPosition > 0)) { - if (cblk->server >= mMarkerPosition) { - mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition); - mMarkerReached = true; - } + bool markerReached = false; + size_t markerPosition = mMarkerPosition; + // FIXME fails for wraparound, need 64 bits + if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { + mMarkerReached = markerReached = true; + } + + // Determine number of new position callback(s) that will be needed, while locked + size_t newPosCount = 0; + size_t newPosition = mNewPosition; + size_t updatePeriod = mUpdatePeriod; + // FIXME fails for wraparound, need 64 bits + if (updatePeriod > 0 && position >= newPosition) { + newPosCount = ((position - newPosition) / updatePeriod) + 1; + mNewPosition += updatePeriod * newPosCount; + } + + // Cache other fields that will be needed soon + uint32_t loopPeriod = mLoopPeriod; + uint32_t sampleRate = mSampleRate; + size_t notificationFrames = mNotificationFramesAct; + if (mRefreshRemaining) { + mRefreshRemaining = false; + mRemainingFrames = notificationFrames; + mRetryOnPartialBuffer = false; + } + size_t misalignment = mProxy->getMisalignment(); + int32_t sequence = mSequence; + + // These fields don't need to be cached, because they are assigned only by set(): + // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags + // mFlags is also assigned by createTrack_l(), but not the bit we care about. + + mLock.unlock(); + + // perform callbacks while unlocked + if (newUnderrun) { + mCbf(EVENT_UNDERRUN, mUserData, NULL); + } + // FIXME we will miss loops if loop cycle was signaled several times since last call + // to processAudioBuffer() + if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { + mCbf(EVENT_LOOP_END, mUserData, NULL); + } + if (flags & CBLK_BUFFER_END) { + mCbf(EVENT_BUFFER_END, mUserData, NULL); + } + if (markerReached) { + mCbf(EVENT_MARKER, mUserData, &markerPosition); + } + while (newPosCount > 0) { + size_t temp = newPosition; + mCbf(EVENT_NEW_POS, mUserData, &temp); + newPosition += updatePeriod; + newPosCount--; + } + if (mObservedSequence != sequence) { + mObservedSequence = sequence; + mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); } - // Manage new position callback - if (mUpdatePeriod > 0) { - while (cblk->server >= mNewPosition) { - mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition); - mNewPosition += mUpdatePeriod; - } + // if inactive, then don't run me again until re-started + if (!active) { + return NS_INACTIVE; } - // If Shared buffer is used, no data is requested from client. - if (mSharedBuffer != 0) { - frames = 0; - } else { - frames = mRemainingFrames; + // Compute the estimated time until the next timed event (position, markers, loops) + // FIXME only for non-compressed audio + uint32_t minFrames = ~0; + if (!markerReached && position < markerPosition) { + minFrames = markerPosition - position; + } + if (loopPeriod > 0 && loopPeriod < minFrames) { + minFrames = loopPeriod; + } + if (updatePeriod > 0 && updatePeriod < minFrames) { + minFrames = updatePeriod; } - // See description of waitCount parameter at declaration of obtainBuffer(). - // The logic below prevents us from being stuck below at obtainBuffer() - // not being able to handle timed events (position, markers, loops). - int32_t waitCount = -1; - if (mUpdatePeriod || (!mMarkerReached && mMarkerPosition) || mLoopCount) { - waitCount = 1; + // 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; } - do { + // Convert frame units to time units + nsecs_t ns = NS_WHENEVER; + if (minFrames != (uint32_t) ~0) { + // This "fudge factor" avoids soaking CPU, and compensates for late progress by server + static const nsecs_t kFudgeNs = 10000000LL; // 10 ms + ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; + } + + // If not supplying data by EVENT_MORE_DATA, then we're done + if (mTransfer != TRANSFER_CALLBACK) { + return ns; + } - audioBuffer.frameCount = frames; + struct timespec timeout; + const struct timespec *requested = &ClientProxy::kForever; + if (ns != NS_WHENEVER) { + timeout.tv_sec = ns / 1000000000LL; + timeout.tv_nsec = ns % 1000000000LL; + ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); + requested = &timeout; + } + + while (mRemainingFrames > 0) { - status_t err = obtainBuffer(&audioBuffer, waitCount); - if (err < NO_ERROR) { - if (err != TIMED_OUT) { - ALOGE_IF(err != status_t(NO_MORE_BUFFERS), - "Error obtaining an audio buffer, giving up."); - return false; + 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=%u", err, audioBuffer.frameCount); + requested = &ClientProxy::kNonBlocking; + size_t avail = audioBuffer.frameCount + nonContig; + ALOGV("obtainBuffer(%u) returned %u = %u + %u", + mRemainingFrames, avail, audioBuffer.frameCount, nonContig); + if (err != NO_ERROR) { + if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) { + return 0; } - break; + ALOGE("Error %d obtaining an audio buffer, giving up.", err); + return NS_NEVER; } - if (err == status_t(STOPPED)) { - return false; + + if (mRetryOnPartialBuffer) { + mRetryOnPartialBuffer = false; + if (avail < mRemainingFrames) { + int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; + if (ns < 0 || myns < ns) { + ns = myns; + } + return ns; + } } // Divide buffer size by 2 to take into account the expansion @@ -1380,66 +1421,76 @@ bool AudioTrack::processAudioBuffer(const sp& thread) size_t reqSize = audioBuffer.size; mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); - writtenSize = audioBuffer.size; + size_t writtenSize = audioBuffer.size; + size_t writtenFrames = writtenSize / mFrameSize; // Sanity check on returned size - if (ssize_t(writtenSize) <= 0) { + if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { + ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", + reqSize, (int) 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 - usleep(WAIT_PERIOD_MS*1000); - break; - } - - if (writtenSize > reqSize) { - writtenSize = reqSize; + return WAIT_PERIOD_MS * 1000000LL; } if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { // 8 to 16 bit conversion, note that source and destination are the same address memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); - writtenSize <<= 1; + audioBuffer.size <<= 1; } - audioBuffer.size = writtenSize; - // NOTE: cblk->frameSize is not equal to AudioTrack::frameSize() for - // 8 bit PCM data: in this case, cblk->frameSize is based on a sample size of - // 16 bit. - audioBuffer.frameCount = writtenSize / mFrameSizeAF; - - frames -= audioBuffer.frameCount; + size_t releasedFrames = audioBuffer.size / mFrameSizeAF; + audioBuffer.frameCount = releasedFrames; + mRemainingFrames -= releasedFrames; + if (misalignment >= releasedFrames) { + misalignment -= releasedFrames; + } else { + misalignment = 0; + } releaseBuffer(&audioBuffer); - } - while (frames); - if (frames == 0) { - mRemainingFrames = mNotificationFramesAct; - } else { - mRemainingFrames = frames; + // 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 (mRemainingFrames * 1100000000LL) / sampleRate; + } +#endif + } - return true; + mRemainingFrames = notificationFrames; + mRetryOnPartialBuffer = true; + + // A lot has transpired since ns was calculated, so run again immediately and re-calculate + return 0; } -// must be called with mLock and refCblk.lock held. Callers must also hold strong references on -// the IAudioTrack and IMemory in case they are recreated here. -// If the IAudioTrack is successfully restored, the refCblk pointer is updated -// FIXME Don't depend on caller to hold strong references. -status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& refCblk, bool fromStart) +status_t AudioTrack::restoreTrack_l(const char *from) { + ALOGW("dead IAudioTrack, creating a new one from %s()", from); + ++mSequence; status_t result; - audio_track_cblk_t* cblk = refCblk; - audio_track_cblk_t* newCblk = cblk; - ALOGW("dead IAudioTrack, creating a new one from %s", - fromStart ? "start()" : "obtainBuffer()"); - - // signal old cblk condition so that other threads waiting for available buffers stop - // waiting now - cblk->cv.broadcast(); - cblk->lock.unlock(); - // refresh the audio configuration cache in this process to make sure we get new // output parameters in getOutput_l() and createTrack_l() AudioSystem::clearAudioConfigCache(); @@ -1447,68 +1498,47 @@ status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& refCblk, bool fromStart // if the new IAudioTrack is 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 + size_t position = mProxy->getPosition(); + mNewPosition = position + mUpdatePeriod; + size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; result = createTrack_l(mStreamType, mSampleRate, mFormat, mReqFrameCount, // so that frame count never goes down mFlags, mSharedBuffer, - getOutput_l()); + getOutput_l(), + position /*epoch*/); if (result == NO_ERROR) { - uint32_t user = cblk->user; - uint32_t server = cblk->server; + // continue playback from last known position, but + // don't attempt to restore loop after invalidation; it's difficult and not worthwhile + if (mStaticProxy != NULL) { + mLoopPeriod = 0; + mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); + } + // FIXME How do we simulate the fact that all frames present in the buffer at the time of + // track destruction have been played? This is critical for SoundPool implementation + // This must be broken, and needs to be tested/debugged. +#if 0 // restore write index and set other indexes to reflect empty buffer status - newCblk = mCblk; - newCblk->user = user; - newCblk->server = user; - newCblk->userBase = user; - newCblk->serverBase = user; - // restore loop: this is not guaranteed to succeed if new frame count is not - // compatible with loop length - setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount); - size_t frames = 0; - if (!fromStart) { - newCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; + if (!strcmp(from, "start")) { // Make sure that a client relying on callback events indicating underrun or // the actual amount of audio frames played (e.g SoundPool) receives them. if (mSharedBuffer == 0) { - if (user > server) { - frames = ((user - server) > mFrameCount) ? - mFrameCount : (user - server); - memset(mBuffers, 0, frames * mFrameSizeAF); - } // restart playback even if buffer is not completely filled. - android_atomic_or(CBLK_FORCEREADY, &newCblk->flags); + android_atomic_or(CBLK_FORCEREADY, &mCblk->flags); } } - if (mSharedBuffer != 0) { - frames = mFrameCount; - } - if (frames > 0) { - // stepUser() clears CBLK_UNDERRUN flag enabling underrun callbacks to - // the client - mProxy->stepUser(frames); - } - if (mActive) { +#endif + if (mState == STATE_ACTIVE) { result = mAudioTrack->start(); - ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result); - } - if (fromStart && result == NO_ERROR) { - mNewPosition = newCblk->server + mUpdatePeriod; } } - ALOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result); - ALOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x", - result, mActive, newCblk, cblk, newCblk->flags, cblk->flags); - - if (result == NO_ERROR) { - // from now on we switch to the newly created cblk - refCblk = newCblk; + if (result != NO_ERROR) { + ALOGW("restoreTrack_l() failed status %d", result); + mState = STATE_STOPPED; } - newCblk->lock.lock(); - - ALOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d", result); return result; } @@ -1529,16 +1559,33 @@ status_t AudioTrack::dump(int fd, const Vector& args) const result.append(buffer); snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); result.append(buffer); - snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency); + snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); result.append(buffer); ::write(fd, result.string(), result.size()); return NO_ERROR; } +uint32_t AudioTrack::getUnderrunFrames() const +{ + AutoMutex lock(mLock); + return mProxy->getUnderrunFrames(); +} + +// ========================================================================= + +void AudioTrack::DeathNotifier::binderDied(const wp& who) +{ + sp 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) + : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mResumeLatch(false) { } @@ -1556,10 +1603,26 @@ bool AudioTrack::AudioTrackThread::threadLoop() return true; } } - if (!mReceiver.processAudioBuffer(this)) { - pause(); + nsecs_t ns = mReceiver.processAudioBuffer(this); + switch (ns) { + case 0: + return true; + case NS_WHENEVER: + sleep(1); + return true; + case NS_INACTIVE: + pauseConditional(); + return true; + case NS_NEVER: + return false; + default: + LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); + struct timespec req; + req.tv_sec = ns / 1000000000LL; + req.tv_nsec = ns % 1000000000LL; + nanosleep(&req, NULL /*rem*/); + return true; } - return true; } void AudioTrack::AudioTrackThread::requestExit() @@ -1573,6 +1636,17 @@ void AudioTrack::AudioTrackThread::pause() { AutoMutex _l(mMyLock); mPaused = true; + mResumeLatch = false; +} + +void AudioTrack::AudioTrackThread::pauseConditional() +{ + AutoMutex _l(mMyLock); + if (mResumeLatch) { + mResumeLatch = false; + } else { + mPaused = true; + } } void AudioTrack::AudioTrackThread::resume() @@ -1580,7 +1654,10 @@ void AudioTrack::AudioTrackThread::resume() AutoMutex _l(mMyLock); if (mPaused) { mPaused = false; + mResumeLatch = false; mMyCond.signal(); + } else { + mResumeLatch = true; } } diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp index 13d47c9..f034164 100644 --- a/media/libmedia/AudioTrackShared.cpp +++ b/media/libmedia/AudioTrackShared.cpp @@ -19,178 +19,664 @@ #include #include +extern "C" { +#include "../private/bionic_futex.h" +} namespace android { audio_track_cblk_t::audio_track_cblk_t() - : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0), - userBase(0), serverBase(0), frameCount_(0), - loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), mVolumeLR(0x10001000), - mSampleRate(0), mSendLevel(0), flags(0) + : server(0), frameCount_(0), mFutex(0), mMinimum(0), + mVolumeLR(0x10001000), mSampleRate(0), mSendLevel(0), mName(0), flags(0) { + memset(&u, 0, sizeof(u)); } -uint32_t audio_track_cblk_t::stepUser(size_t stepCount, size_t frameCount, bool isOut) +// --------------------------------------------------------------------------- + +Proxy::Proxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, + bool isOut, bool clientInServer) + : mCblk(cblk), mBuffers(buffers), mFrameCount(frameCount), mFrameSize(frameSize), + mFrameCountP2(roundup(frameCount)), mIsOut(isOut), mClientInServer(clientInServer), + mIsShutdown(false) { - ALOGV("stepuser %08x %08x %d", user, server, stepCount); +} - uint32_t u = user; - u += stepCount; - // Ensure that user is never ahead of server for AudioRecord - if (isOut) { - // If stepServer() has been called once, switch to normal obtainBuffer() timeout period - if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) { - bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; - } - } else if (u > server) { - ALOGW("stepUser occurred after track reset"); - u = server; +// --------------------------------------------------------------------------- + +ClientProxy::ClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, + size_t frameSize, bool isOut, bool clientInServer) + : Proxy(cblk, buffers, frameCount, frameSize, isOut, clientInServer), mEpoch(0) +{ +} + +const struct timespec ClientProxy::kForever = {INT_MAX /*tv_sec*/, 0 /*tv_nsec*/}; +const struct timespec ClientProxy::kNonBlocking = {0 /*tv_sec*/, 0 /*tv_nsec*/}; + +#define MEASURE_NS 10000000 // attempt to provide accurate timeouts if requested >= MEASURE_NS + +// To facilitate quicker recovery from server failure, this value limits the timeout per each futex +// wait. However it does not protect infinite timeouts. If defined to be zero, there is no limit. +// FIXME May not be compatible with audio tunneling requirements where timeout should be in the +// order of minutes. +#define MAX_SEC 5 + +status_t ClientProxy::obtainBuffer(Buffer* buffer, const struct timespec *requested, + struct timespec *elapsed) +{ + if (buffer == NULL || buffer->mFrameCount == 0) { + ALOGE("%s BAD_VALUE", __func__); + return BAD_VALUE; } + struct timespec total; // total elapsed time spent waiting + total.tv_sec = 0; + total.tv_nsec = 0; + bool measure = elapsed != NULL; // whether to measure total elapsed time spent waiting - if (u >= frameCount) { - // common case, user didn't just wrap - if (u - frameCount >= userBase ) { - userBase += frameCount; + status_t status; + enum { + TIMEOUT_ZERO, // requested == NULL || *requested == 0 + TIMEOUT_INFINITE, // *requested == infinity + TIMEOUT_FINITE, // 0 < *requested < infinity + TIMEOUT_CONTINUE, // additional chances after TIMEOUT_FINITE + } timeout; + if (requested == NULL) { + timeout = TIMEOUT_ZERO; + } else if (requested->tv_sec == 0 && requested->tv_nsec == 0) { + timeout = TIMEOUT_ZERO; + } else if (requested->tv_sec == INT_MAX) { + timeout = TIMEOUT_INFINITE; + } else { + timeout = TIMEOUT_FINITE; + if (requested->tv_sec > 0 || requested->tv_nsec >= MEASURE_NS) { + measure = true; + } + } + struct timespec before; + bool beforeIsValid = false; + audio_track_cblk_t* cblk = mCblk; + bool ignoreInitialPendingInterrupt = true; + // check for shared memory corruption + if (mIsShutdown) { + status = NO_INIT; + goto end; + } + for (;;) { + int32_t flags = android_atomic_and(~CBLK_INTERRUPT, &cblk->flags); + // check for track invalidation by server, or server death detection + if (flags & CBLK_INVALID) { + ALOGV("Track invalidated"); + status = DEAD_OBJECT; + goto end; + } + // check for obtainBuffer interrupted by client + if (!ignoreInitialPendingInterrupt && (flags & CBLK_INTERRUPT)) { + ALOGV("obtainBuffer() interrupted by client"); + status = -EINTR; + goto end; + } + ignoreInitialPendingInterrupt = false; + // compute number of frames available to write (AudioTrack) or read (AudioRecord) + int32_t front; + int32_t rear; + if (mIsOut) { + // The barrier following the read of mFront is probably redundant. + // We're about to perform a conditional branch based on 'filled', + // which will force the processor to observe the read of mFront + // prior to allowing data writes starting at mRaw. + // However, the processor may support speculative execution, + // and be unable to undo speculative writes into shared memory. + // The barrier will prevent such speculative execution. + front = android_atomic_acquire_load(&cblk->u.mStreaming.mFront); + rear = cblk->u.mStreaming.mRear; + } else { + // On the other hand, this barrier is required. + rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear); + front = cblk->u.mStreaming.mFront; + } + ssize_t filled = rear - front; + // pipe should not be overfull + if (!(0 <= filled && (size_t) filled <= mFrameCount)) { + ALOGE("Shared memory control block is corrupt (filled=%d); shutting down", filled); + mIsShutdown = true; + status = NO_INIT; + goto end; + } + // don't allow filling pipe beyond the nominal size + size_t avail = mIsOut ? mFrameCount - filled : filled; + if (avail > 0) { + // 'avail' may be non-contiguous, so return only the first contiguous chunk + size_t part1; + if (mIsOut) { + rear &= mFrameCountP2 - 1; + part1 = mFrameCountP2 - rear; + } else { + front &= mFrameCountP2 - 1; + part1 = mFrameCountP2 - front; + } + if (part1 > avail) { + part1 = avail; + } + if (part1 > buffer->mFrameCount) { + part1 = buffer->mFrameCount; + } + buffer->mFrameCount = part1; + buffer->mRaw = part1 > 0 ? + &((char *) mBuffers)[(mIsOut ? rear : front) * mFrameSize] : NULL; + buffer->mNonContig = avail - part1; + // mUnreleased = part1; + status = NO_ERROR; + break; + } + struct timespec remaining; + const struct timespec *ts; + switch (timeout) { + case TIMEOUT_ZERO: + status = WOULD_BLOCK; + goto end; + case TIMEOUT_INFINITE: + ts = NULL; + break; + case TIMEOUT_FINITE: + timeout = TIMEOUT_CONTINUE; + if (MAX_SEC == 0) { + ts = requested; + break; + } + // fall through + case TIMEOUT_CONTINUE: + // FIXME we do not retry if requested < 10ms? needs documentation on this state machine + if (!measure || requested->tv_sec < total.tv_sec || + (requested->tv_sec == total.tv_sec && requested->tv_nsec <= total.tv_nsec)) { + status = TIMED_OUT; + goto end; + } + remaining.tv_sec = requested->tv_sec - total.tv_sec; + if ((remaining.tv_nsec = requested->tv_nsec - total.tv_nsec) < 0) { + remaining.tv_nsec += 1000000000; + remaining.tv_sec++; + } + if (0 < MAX_SEC && MAX_SEC < remaining.tv_sec) { + remaining.tv_sec = MAX_SEC; + remaining.tv_nsec = 0; + } + ts = &remaining; + break; + default: + LOG_FATAL("%s timeout=%d", timeout); + ts = NULL; + break; + } + int32_t old = android_atomic_dec(&cblk->mFutex); + if (old <= 0) { + int rc; + if (measure && !beforeIsValid) { + clock_gettime(CLOCK_MONOTONIC, &before); + beforeIsValid = true; + } + int ret = __futex_syscall4(&cblk->mFutex, + mClientInServer ? FUTEX_WAIT_PRIVATE : FUTEX_WAIT, old - 1, ts); + // update total elapsed time spent waiting + if (measure) { + struct timespec after; + clock_gettime(CLOCK_MONOTONIC, &after); + total.tv_sec += after.tv_sec - before.tv_sec; + long deltaNs = after.tv_nsec - before.tv_nsec; + if (deltaNs < 0) { + deltaNs += 1000000000; + total.tv_sec--; + } + if ((total.tv_nsec += deltaNs) >= 1000000000) { + total.tv_nsec -= 1000000000; + total.tv_sec++; + } + before = after; + beforeIsValid = true; + } + switch (ret) { + case 0: // normal wakeup by server, or by binderDied() + case -EWOULDBLOCK: // benign race condition with server + case -EINTR: // wait was interrupted by signal or other spurious wakeup + case -ETIMEDOUT: // time-out expired + break; + default: + ALOGE("%s unexpected error %d", __func__, ret); + status = -ret; + goto end; + } } - } else if (u >= userBase + frameCount) { - // user just wrapped - userBase += frameCount; } - user = u; - - // Clear flow control error condition as new data has been written/read to/from buffer. - if (flags & CBLK_UNDERRUN) { - android_atomic_and(~CBLK_UNDERRUN, &flags); +end: + if (status != NO_ERROR) { + buffer->mFrameCount = 0; + buffer->mRaw = NULL; + buffer->mNonContig = 0; + } + if (elapsed != NULL) { + *elapsed = total; } + if (requested == NULL) { + requested = &kNonBlocking; + } + if (measure) { + ALOGV("requested %d.%03d elapsed %d.%03d", requested->tv_sec, requested->tv_nsec / 1000000, + total.tv_sec, total.tv_nsec / 1000000); + } + return status; +} - return u; +void ClientProxy::releaseBuffer(Buffer* buffer) +{ + size_t stepCount = buffer->mFrameCount; + // FIXME + // check mUnreleased + // verify that stepCount <= frameCount returned by the last obtainBuffer() + // verify stepCount not > total frame count of pipe + if (stepCount == 0) { + return; + } + audio_track_cblk_t* cblk = mCblk; + // Both of these barriers are required + if (mIsOut) { + int32_t rear = cblk->u.mStreaming.mRear; + android_atomic_release_store(stepCount + rear, &cblk->u.mStreaming.mRear); + } else { + int32_t front = cblk->u.mStreaming.mFront; + android_atomic_release_store(stepCount + front, &cblk->u.mStreaming.mFront); + } } -bool audio_track_cblk_t::stepServer(size_t stepCount, size_t frameCount, bool isOut) +void ClientProxy::binderDied() { - ALOGV("stepserver %08x %08x %d", user, server, stepCount); + audio_track_cblk_t* cblk = mCblk; + if (!(android_atomic_or(CBLK_INVALID, &cblk->flags) & CBLK_INVALID)) { + // it seems that a FUTEX_WAKE_PRIVATE will not wake a FUTEX_WAIT, even within same process + (void) __futex_syscall3(&cblk->mFutex, mClientInServer ? FUTEX_WAKE_PRIVATE : FUTEX_WAKE, + 1); + } +} - if (!tryLock()) { - ALOGW("stepServer() could not lock cblk"); - return false; +void ClientProxy::interrupt() +{ + audio_track_cblk_t* cblk = mCblk; + if (!(android_atomic_or(CBLK_INTERRUPT, &cblk->flags) & CBLK_INTERRUPT)) { + (void) __futex_syscall3(&cblk->mFutex, mClientInServer ? FUTEX_WAKE_PRIVATE : FUTEX_WAKE, + 1); } +} - uint32_t s = server; - bool flushed = (s == user); +size_t ClientProxy::getMisalignment() +{ + audio_track_cblk_t* cblk = mCblk; + return (mFrameCountP2 - (mIsOut ? cblk->u.mStreaming.mRear : cblk->u.mStreaming.mFront)) & + (mFrameCountP2 - 1); +} - s += stepCount; - if (isOut) { - // Mark that we have read the first buffer so that next time stepUser() is called - // we switch to normal obtainBuffer() timeout period - if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) { - bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1; - } - // It is possible that we receive a flush() - // while the mixer is processing a block: in this case, - // stepServer() is called After the flush() has reset u & s and - // we have s > u - if (flushed) { - ALOGW("stepServer occurred after track reset"); - s = user; +// --------------------------------------------------------------------------- + +void AudioTrackClientProxy::flush() +{ + mCblk->u.mStreaming.mFlush++; +} + +// --------------------------------------------------------------------------- + +StaticAudioTrackClientProxy::StaticAudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, + size_t frameCount, size_t frameSize) + : AudioTrackClientProxy(cblk, buffers, frameCount, frameSize), + mMutator(&cblk->u.mStatic.mSingleStateQueue), mBufferPosition(0) +{ +} + +void StaticAudioTrackClientProxy::flush() +{ + LOG_FATAL("static flush"); +} + +void StaticAudioTrackClientProxy::setLoop(size_t loopStart, size_t loopEnd, int loopCount) +{ + StaticAudioTrackState newState; + newState.mLoopStart = loopStart; + newState.mLoopEnd = loopEnd; + newState.mLoopCount = loopCount; + mBufferPosition = loopStart; + (void) mMutator.push(newState); +} + +size_t StaticAudioTrackClientProxy::getBufferPosition() +{ + size_t bufferPosition; + if (mMutator.ack()) { + bufferPosition = mCblk->u.mStatic.mBufferPosition; + if (bufferPosition > mFrameCount) { + bufferPosition = mFrameCount; } + } else { + bufferPosition = mBufferPosition; } + return bufferPosition; +} + +// --------------------------------------------------------------------------- - if (s >= loopEnd) { - ALOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd); - s = loopStart; - if (--loopCount == 0) { - loopEnd = UINT_MAX; - loopStart = UINT_MAX; +ServerProxy::ServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, + size_t frameSize, bool isOut, bool clientInServer) + : Proxy(cblk, buffers, frameCount, frameSize, isOut, clientInServer), mUnreleased(0), + mAvailToClient(0), mFlush(0), mDeferWake(false) +{ +} + +status_t ServerProxy::obtainBuffer(Buffer* buffer) +{ + if (mIsShutdown) { + buffer->mFrameCount = 0; + buffer->mRaw = NULL; + buffer->mNonContig = 0; + mUnreleased = 0; + return NO_INIT; + } + audio_track_cblk_t* cblk = mCblk; + // compute number of frames available to write (AudioTrack) or read (AudioRecord), + // or use previous cached value from framesReady(), with added barrier if it omits. + int32_t front; + int32_t rear; + // See notes on barriers at ClientProxy::obtainBuffer() + if (mIsOut) { + int32_t flush = cblk->u.mStreaming.mFlush; + rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear); + if (flush != mFlush) { + front = rear; + mFlush = flush; + } else { + front = cblk->u.mStreaming.mFront; } + } else { + front = android_atomic_acquire_load(&cblk->u.mStreaming.mFront); + rear = cblk->u.mStreaming.mRear; + } + ssize_t filled = rear - front; + // pipe should not already be overfull + if (!(0 <= filled && (size_t) filled <= mFrameCount)) { + ALOGE("Shared memory control block is corrupt (filled=%d); shutting down", filled); + mIsShutdown = true; + } + if (mIsShutdown) { + buffer->mFrameCount = 0; + buffer->mRaw = NULL; + buffer->mNonContig = 0; + mUnreleased = 0; + return NO_INIT; } + // don't allow filling pipe beyond the nominal size + size_t availToServer; + if (mIsOut) { + availToServer = filled; + mAvailToClient = mFrameCount - filled; + } else { + availToServer = mFrameCount - filled; + mAvailToClient = filled; + } + // 'availToServer' may be non-contiguous, so return only the first contiguous chunk + size_t part1; + if (mIsOut) { + front &= mFrameCountP2 - 1; + part1 = mFrameCountP2 - front; + } else { + rear &= mFrameCountP2 - 1; + part1 = mFrameCountP2 - rear; + } + if (part1 > availToServer) { + part1 = availToServer; + } + size_t ask = buffer->mFrameCount; + if (part1 > ask) { + part1 = ask; + } + // is assignment redundant in some cases? + buffer->mFrameCount = part1; + buffer->mRaw = part1 > 0 ? + &((char *) mBuffers)[(mIsOut ? front : rear) * mFrameSize] : NULL; + buffer->mNonContig = availToServer - part1; + mUnreleased = part1; + // optimization to avoid waking up the client too early + // FIXME need to test for recording + mDeferWake = part1 < ask && availToServer >= ask; + return part1 > 0 ? NO_ERROR : WOULD_BLOCK; +} - if (s >= frameCount) { - // common case, server didn't just wrap - if (s - frameCount >= serverBase ) { - serverBase += frameCount; - } - } else if (s >= serverBase + frameCount) { - // server just wrapped - serverBase += frameCount; +void ServerProxy::releaseBuffer(Buffer* buffer) +{ + if (mIsShutdown) { + buffer->mFrameCount = 0; + buffer->mRaw = NULL; + buffer->mNonContig = 0; + return; + } + size_t stepCount = buffer->mFrameCount; + LOG_ALWAYS_FATAL_IF(stepCount > mUnreleased); + if (stepCount == 0) { + buffer->mRaw = NULL; + buffer->mNonContig = 0; + return; + } + mUnreleased -= stepCount; + audio_track_cblk_t* cblk = mCblk; + if (mIsOut) { + int32_t front = cblk->u.mStreaming.mFront; + android_atomic_release_store(stepCount + front, &cblk->u.mStreaming.mFront); + } else { + int32_t rear = cblk->u.mStreaming.mRear; + android_atomic_release_store(stepCount + rear, &cblk->u.mStreaming.mRear); } - server = s; + mCblk->server += stepCount; - if (!(flags & CBLK_INVALID)) { - cv.signal(); + size_t half = mFrameCount / 2; + if (half == 0) { + half = 1; + } + size_t minimum = cblk->mMinimum; + if (minimum == 0) { + minimum = mIsOut ? half : 1; + } else if (minimum > half) { + minimum = half; + } + if (!mDeferWake && mAvailToClient + stepCount >= minimum) { + ALOGV("mAvailToClient=%u stepCount=%u minimum=%u", mAvailToClient, stepCount, minimum); + // could client be sleeping, or not need this increment and counter overflows? + int32_t old = android_atomic_inc(&cblk->mFutex); + if (old == -1) { + (void) __futex_syscall3(&cblk->mFutex, + mClientInServer ? FUTEX_WAKE_PRIVATE : FUTEX_WAKE, 1); + } } - lock.unlock(); - return true; + + buffer->mFrameCount = 0; + buffer->mRaw = NULL; + buffer->mNonContig = 0; } -void* audio_track_cblk_t::buffer(void *buffers, size_t frameSize, uint32_t offset) const +// --------------------------------------------------------------------------- + +size_t AudioTrackServerProxy::framesReady() { - return (int8_t *)buffers + (offset - userBase) * frameSize; + LOG_ALWAYS_FATAL_IF(!mIsOut); + + if (mIsShutdown) { + return 0; + } + audio_track_cblk_t* cblk = mCblk; + // the acquire might not be necessary since not doing a subsequent read + int32_t rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear); + ssize_t filled = rear - cblk->u.mStreaming.mFront; + // pipe should not already be overfull + if (!(0 <= filled && (size_t) filled <= mFrameCount)) { + ALOGE("Shared memory control block is corrupt (filled=%d); shutting down", filled); + mIsShutdown = true; + return 0; + } + // cache this value for later use by obtainBuffer(), with added barrier + // and racy if called by normal mixer thread + // ignores flush(), so framesReady() may report a larger mFrameCount than obtainBuffer() + return filled; } -uint32_t audio_track_cblk_t::framesAvailable(size_t frameCount, bool isOut) +// --------------------------------------------------------------------------- + +StaticAudioTrackServerProxy::StaticAudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers, + size_t frameCount, size_t frameSize) + : AudioTrackServerProxy(cblk, buffers, frameCount, frameSize), + mObserver(&cblk->u.mStatic.mSingleStateQueue), mPosition(0), + mEnd(frameCount), mFramesReadyIsCalledByMultipleThreads(false) { - Mutex::Autolock _l(lock); - return framesAvailable_l(frameCount, isOut); + mState.mLoopStart = 0; + mState.mLoopEnd = 0; + mState.mLoopCount = 0; } -uint32_t audio_track_cblk_t::framesAvailable_l(size_t frameCount, bool isOut) +void StaticAudioTrackServerProxy::framesReadyIsCalledByMultipleThreads() { - uint32_t u = user; - uint32_t s = server; + mFramesReadyIsCalledByMultipleThreads = true; +} - if (isOut) { - uint32_t limit = (s < loopStart) ? s : loopStart; - return limit + frameCount - u; - } else { - return frameCount + u - s; +size_t StaticAudioTrackServerProxy::framesReady() +{ + // FIXME + // This is racy if called by normal mixer thread, + // as we're reading 2 independent variables without a lock. + // Can't call mObserver.poll(), as we might be called from wrong thread. + // If looping is enabled, should return a higher number (since includes non-contiguous). + size_t position = mPosition; + if (!mFramesReadyIsCalledByMultipleThreads) { + ssize_t positionOrStatus = pollPosition(); + if (positionOrStatus >= 0) { + position = (size_t) positionOrStatus; + } } + size_t end = mEnd; + return position < end ? end - position : 0; } -uint32_t audio_track_cblk_t::framesReady(bool isOut) +ssize_t StaticAudioTrackServerProxy::pollPosition() { - uint32_t u = user; - uint32_t s = server; - - if (isOut) { - if (u < loopEnd) { - return u - s; - } else { - // do not block on mutex shared with client on AudioFlinger side - if (!tryLock()) { - ALOGW("framesReady() could not lock cblk"); - return 0; + size_t position = mPosition; + StaticAudioTrackState state; + if (mObserver.poll(state)) { + bool valid = false; + size_t loopStart = state.mLoopStart; + size_t loopEnd = state.mLoopEnd; + if (state.mLoopCount == 0) { + if (loopStart > mFrameCount) { + loopStart = mFrameCount; } - uint32_t frames = UINT_MAX; - if (loopCount >= 0) { - frames = (loopEnd - loopStart)*loopCount + u - s; + // ignore loopEnd + mPosition = position = loopStart; + mEnd = mFrameCount; + mState.mLoopCount = 0; + valid = true; + } else { + if (loopStart < loopEnd && loopEnd <= mFrameCount && + loopEnd - loopStart >= MIN_LOOP) { + if (!(loopStart <= position && position < loopEnd)) { + mPosition = position = loopStart; + } + mEnd = loopEnd; + mState = state; + valid = true; } - lock.unlock(); - return frames; } - } else { - return s - u; + if (!valid) { + ALOGE("%s client pushed an invalid state, shutting down", __func__); + mIsShutdown = true; + return (ssize_t) NO_INIT; + } + mCblk->u.mStatic.mBufferPosition = position; } + return (ssize_t) position; } -bool audio_track_cblk_t::tryLock() +status_t StaticAudioTrackServerProxy::obtainBuffer(Buffer* buffer) { - // the code below simulates lock-with-timeout - // we MUST do this to protect the AudioFlinger server - // as this lock is shared with the client. - status_t err; + if (mIsShutdown) { + buffer->mFrameCount = 0; + buffer->mRaw = NULL; + buffer->mNonContig = 0; + mUnreleased = 0; + return NO_INIT; + } + ssize_t positionOrStatus = pollPosition(); + if (positionOrStatus < 0) { + buffer->mFrameCount = 0; + buffer->mRaw = NULL; + buffer->mNonContig = 0; + mUnreleased = 0; + return (status_t) positionOrStatus; + } + size_t position = (size_t) positionOrStatus; + size_t avail; + if (position < mEnd) { + avail = mEnd - position; + size_t wanted = buffer->mFrameCount; + if (avail < wanted) { + buffer->mFrameCount = avail; + } else { + avail = wanted; + } + buffer->mRaw = &((char *) mBuffers)[position * mFrameSize]; + } else { + avail = 0; + buffer->mFrameCount = 0; + buffer->mRaw = NULL; + } + buffer->mNonContig = 0; // FIXME should be > 0 for looping + mUnreleased = avail; + return NO_ERROR; +} - err = lock.tryLock(); - if (err == -EBUSY) { // just wait a bit - usleep(1000); - err = lock.tryLock(); +void StaticAudioTrackServerProxy::releaseBuffer(Buffer* buffer) +{ + size_t stepCount = buffer->mFrameCount; + LOG_ALWAYS_FATAL_IF(stepCount > mUnreleased); + if (stepCount == 0) { + buffer->mRaw = NULL; + buffer->mNonContig = 0; + return; } - if (err != NO_ERROR) { - // probably, the client just died. - return false; + mUnreleased -= stepCount; + audio_track_cblk_t* cblk = mCblk; + size_t position = mPosition; + size_t newPosition = position + stepCount; + int32_t setFlags = 0; + if (!(position <= newPosition && newPosition <= mFrameCount)) { + ALOGW("%s newPosition %u outside [%u, %u]", __func__, newPosition, position, mFrameCount); + newPosition = mFrameCount; + } else if (mState.mLoopCount != 0 && newPosition == mState.mLoopEnd) { + if (mState.mLoopCount == -1 || --mState.mLoopCount != 0) { + newPosition = mState.mLoopStart; + setFlags = CBLK_LOOP_CYCLE; + } else { + mEnd = mFrameCount; // this is what allows playback to continue after the loop + setFlags = CBLK_LOOP_FINAL; + } } - return true; + if (newPosition == mFrameCount) { + setFlags |= CBLK_BUFFER_END; + } + mPosition = newPosition; + + cblk->server += stepCount; + cblk->u.mStatic.mBufferPosition = newPosition; + if (setFlags != 0) { + (void) android_atomic_or(setFlags, &cblk->flags); + // this would be a good place to wake a futex + } + + buffer->mFrameCount = 0; + buffer->mRaw = NULL; + buffer->mNonContig = 0; } +// --------------------------------------------------------------------------- + } // namespace android diff --git a/media/libmedia/ToneGenerator.cpp b/media/libmedia/ToneGenerator.cpp index ebe1ba1..f9ad31d 100644 --- a/media/libmedia/ToneGenerator.cpp +++ b/media/libmedia/ToneGenerator.cpp @@ -1060,7 +1060,9 @@ bool ToneGenerator::initAudioTrack() { this, // user 0, // notificationFrames 0, // sharedBuffer - mThreadCanCallJava); + mThreadCanCallJava, + 0, // sessionId + AudioTrack::TRANSFER_CALLBACK); if (mpAudioTrack->initCheck() != NO_ERROR) { ALOGE("AudioTrack->initCheck failed"); diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h index cf68848..05dbab1 100644 --- a/services/audioflinger/AudioFlinger.h +++ b/services/audioflinger/AudioFlinger.h @@ -56,6 +56,7 @@ #include #include +#include namespace android { diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h index a749d7a..b1286d3 100644 --- a/services/audioflinger/PlaybackTracks.h +++ b/services/audioflinger/PlaybackTracks.h @@ -46,6 +46,8 @@ public: void destroy(); int name() const { return mName; } + virtual uint32_t sampleRate() const; + audio_stream_type_t streamType() const { return mStreamType; } @@ -139,6 +141,7 @@ private: // 'volatile' means accessed without lock or // barrier, but is read/written atomically bool mIsInvalid; // non-resettable latch, set by invalidate() + AudioTrackServerProxy* mAudioTrackServerProxy; }; // end of Track class TimedTrack : public Track { @@ -255,10 +258,6 @@ public: private: - enum { - NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value - }; - status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs); void clearBufferQueue(); diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h index 6c0d1d3..ffe3e9f 100644 --- a/services/audioflinger/RecordTracks.h +++ b/services/audioflinger/RecordTracks.h @@ -57,4 +57,5 @@ private: // releaseBuffer() not overridden bool mOverflow; // overflow on most recent attempt to fill client buffer + AudioRecordServerProxy* mAudioRecordServerProxy; }; diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp index ee52fcb..0773534 100644 --- a/services/audioflinger/Threads.cpp +++ b/services/audioflinger/Threads.cpp @@ -139,7 +139,7 @@ static const int kPriorityFastMixer = 3; // FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or // N-buffering, so AudioFlinger could allocate the right amount of memory. // See the client's minBufCount and mNotificationFramesAct calculations for details. -static const int kFastTrackMultiplier = 2; +static const int kFastTrackMultiplier = 1; // ---------------------------------------------------------------------------- @@ -1327,7 +1327,7 @@ status_t AudioFlinger::PlaybackThread::addTrack_l(const sp& track) // the track is newly added, make sure it fills up all its // buffers before playing. This is to ensure the client will // effectively get the latency it requested. - track->mFillingUpStatus = Track::FS_FILLING; + track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; track->mResetDone = false; track->mPresentationCompleteFrames = 0; mActiveTracks.add(track); @@ -2596,24 +2596,35 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac // app does not call stop() and relies on underrun to stop: // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed // during last round + size_t desiredFrames; + if (t->sampleRate() == mSampleRate) { + desiredFrames = mNormalFrameCount; + } else { + // +1 for rounding and +1 for additional sample needed for interpolation + desiredFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; + // add frames already consumed but not yet released by the resampler + // because cblk->framesReady() will include these frames + desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); + // the minimum track buffer size is normally twice the number of frames necessary + // to fill one buffer and the resampler should not leave more than one buffer worth + // of unreleased frames after each pass, but just in case... + ALOG_ASSERT(desiredFrames <= cblk->frameCount_); + } uint32_t minFrames = 1; if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { - if (t->sampleRate() == mSampleRate) { - minFrames = mNormalFrameCount; - } else { - // +1 for rounding and +1 for additional sample needed for interpolation - minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; - // add frames already consumed but not yet released by the resampler - // because cblk->framesReady() will include these frames - minFrames += mAudioMixer->getUnreleasedFrames(track->name()); - // the minimum track buffer size is normally twice the number of frames necessary - // to fill one buffer and the resampler should not leave more than one buffer worth - // of unreleased frames after each pass, but just in case... - ALOG_ASSERT(minFrames <= cblk->frameCount_); - } + minFrames = desiredFrames; } - if ((track->framesReady() >= minFrames) && track->isReady() && + // It's not safe to call framesReady() for a static buffer track, so assume it's ready + size_t framesReady; + if (track->sharedBuffer() == 0) { + framesReady = track->framesReady(); + } else if (track->isStopped()) { + framesReady = 0; + } else { + framesReady = 1; + } + if ((framesReady >= minFrames) && track->isReady() && !track->isPaused() && !track->isTerminated()) { ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, @@ -2664,7 +2675,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac // read original volumes with volume control float typeVolume = mStreamTypes[track->streamType()].volume; float v = masterVolume * typeVolume; - ServerProxy *proxy = track->mServerProxy; + AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; uint32_t vlr = proxy->getVolumeLR(); vl = vlr & 0xFFFF; vr = vlr >> 16; @@ -2737,7 +2748,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); // limit track sample rate to 2 x output sample rate, which changes at re-configuration uint32_t maxSampleRate = mSampleRate * 2; - uint32_t reqSampleRate = track->mServerProxy->getSampleRate(); + uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); if (reqSampleRate == 0) { reqSampleRate = mSampleRate; } else if (reqSampleRate > maxSampleRate) { @@ -2768,6 +2779,13 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac mixerStatus = MIXER_TRACKS_READY; } } else { + // only implemented for normal tracks, not fast tracks + if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { + // we missed desiredFrames whatever the actual number of frames missing was + cblk->u.mStreaming.mUnderrunFrames += desiredFrames; + // FIXME also wake futex so that underrun is noticed more quickly + (void) android_atomic_or(CBLK_UNDERRUN, &cblk->flags); + } // clear effect chain input buffer if an active track underruns to avoid sending // previous audio buffer again to effects chain = getEffectChain_l(track->sessionId()); @@ -3170,7 +3188,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep } else { float typeVolume = mStreamTypes[track->streamType()].volume; float v = mMasterVolume * typeVolume; - uint32_t vlr = track->mServerProxy->getVolumeLR(); + uint32_t vlr = track->mAudioTrackServerProxy->getVolumeLR(); float v_clamped = v * (vlr & 0xFFFF); if (v_clamped > MAX_GAIN) { v_clamped = MAX_GAIN; @@ -3696,7 +3714,8 @@ bool AudioFlinger::RecordThread::threadLoop() } buffer.frameCount = mFrameCount; - if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { + status_t status = mActiveTrack->getNextBuffer(&buffer); + if (CC_LIKELY(status == NO_ERROR)) { readOnce = true; size_t framesOut = buffer.frameCount; if (mResampler == NULL) { diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h index fac7071..55d96fa 100644 --- a/services/audioflinger/TrackBase.h +++ b/services/audioflinger/TrackBase.h @@ -74,7 +74,7 @@ protected: audio_channel_mask_t channelMask() const { return mChannelMask; } - uint32_t sampleRate() const; // FIXME inline after cblk sr moved + virtual uint32_t sampleRate() const { return mSampleRate; } // Return a pointer to the start of a contiguous slice of the track buffer. // Parameter 'offset' is the requested start position, expressed in diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp index 41a763d..bfc197c 100644 --- a/services/audioflinger/Tracks.cpp +++ b/services/audioflinger/Tracks.cpp @@ -98,7 +98,7 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase( // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); size_t size = sizeof(audio_track_cblk_t); - size_t bufferSize = frameCount * mFrameSize; + size_t bufferSize = (sharedBuffer == 0 ? roundup(frameCount) : frameCount) * mFrameSize; if (sharedBuffer == 0) { size += bufferSize; } @@ -124,22 +124,16 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase( new(mCblk) audio_track_cblk_t(); // clear all buffers mCblk->frameCount_ = frameCount; -// uncomment the following lines to quickly test 32-bit wraparound -// mCblk->user = 0xffff0000; -// mCblk->server = 0xffff0000; -// mCblk->userBase = 0xffff0000; -// mCblk->serverBase = 0xffff0000; if (sharedBuffer == 0) { mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); memset(mBuffer, 0, bufferSize); - // Force underrun condition to avoid false underrun callback until first data is - // written to buffer (other flags are cleared) - mCblk->flags = CBLK_UNDERRUN; } else { mBuffer = sharedBuffer->pointer(); +#if 0 + mCblk->flags = CBLK_FORCEREADY; // FIXME hack, need to fix the track ready logic +#endif } mBufferEnd = (uint8_t *)mBuffer + bufferSize; - mServerProxy = new ServerProxy(mCblk, mBuffer, frameCount, mFrameSize, isOut); #ifdef TEE_SINK if (mTeeSinkTrackEnabled) { @@ -199,51 +193,17 @@ void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buf } #endif - buffer->raw = NULL; - mStepCount = buffer->frameCount; - // FIXME See note at getNextBuffer() - (void) step(); // ignore return value of step() + ServerProxy::Buffer buf; + buf.mFrameCount = buffer->frameCount; + buf.mRaw = buffer->raw; buffer->frameCount = 0; -} - -bool AudioFlinger::ThreadBase::TrackBase::step() { - bool result = mServerProxy->step(mStepCount); - if (!result) { - ALOGV("stepServer failed acquiring cblk mutex"); - mStepServerFailed = true; - } - return result; + buffer->raw = NULL; + mServerProxy->releaseBuffer(&buf); } void AudioFlinger::ThreadBase::TrackBase::reset() { - audio_track_cblk_t* cblk = this->cblk(); - - cblk->user = 0; - cblk->server = 0; - cblk->userBase = 0; - cblk->serverBase = 0; - mStepServerFailed = false; ALOGV("TrackBase::reset"); -} - -uint32_t AudioFlinger::ThreadBase::TrackBase::sampleRate() const { - return mServerProxy->getSampleRate(); -} - -void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { - audio_track_cblk_t* cblk = this->cblk(); - int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase) * mFrameSize; - int8_t *bufferEnd = bufferStart + frames * mFrameSize; - - // Check validity of returned pointer in case the track control block would have been corrupted. - ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), - "TrackBase::getBuffer buffer out of range:\n" - " start: %p, end %p , mBuffer %p mBufferEnd %p\n" - " server %u, serverBase %u, user %u, userBase %u, frameSize %u", - bufferStart, bufferEnd, mBuffer, mBufferEnd, - cblk->server, cblk->serverBase, cblk->user, cblk->userBase, mFrameSize); - - return bufferStart; + // FIXME still needed? } status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp& event) @@ -362,9 +322,18 @@ AudioFlinger::PlaybackThread::Track::Track( mFastIndex(-1), mUnderrunCount(0), mCachedVolume(1.0), - mIsInvalid(false) + mIsInvalid(false), + mAudioTrackServerProxy(NULL) { if (mCblk != NULL) { + if (sharedBuffer == 0) { + mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount, + mFrameSize); + } else { + mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount, + mFrameSize); + } + mServerProxy = mAudioTrackServerProxy; // to avoid leaking a track name, do not allocate one unless there is an mCblk mName = thread->getTrackName_l(channelMask, sessionId); mCblk->mName = mName; @@ -374,6 +343,7 @@ AudioFlinger::PlaybackThread::Track::Track( } // only allocate a fast track index if we were able to allocate a normal track name if (flags & IAudioFlinger::TRACK_FAST) { + mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads(); ALOG_ASSERT(thread->mFastTrackAvailMask != 0); int i = __builtin_ctz(thread->mFastTrackAvailMask); ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); @@ -432,12 +402,12 @@ void AudioFlinger::PlaybackThread::Track::destroy() /*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) { result.append(" Name Client Type Fmt Chn mask Session StpCnt fCount S F SRate " - "L dB R dB Server User Main buf Aux Buf Flags Underruns\n"); + "L dB R dB Server Main buf Aux Buf Flags Underruns\n"); } void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) { - uint32_t vlr = mServerProxy->getVolumeLR(); + uint32_t vlr = mAudioTrackServerProxy->getVolumeLR(); if (isFastTrack()) { sprintf(buffer, " F %2d", mFastIndex); } else { @@ -496,7 +466,7 @@ void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) break; } snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %5u %5.2g %5.2g " - "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", + "0x%08x 0x%08x 0x%08x %#5x %9u%c\n", (mClient == 0) ? getpid_cached : mClient->pid(), mStreamType, mFormat, @@ -506,11 +476,10 @@ void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) mFrameCount, stateChar, mFillingUpStatus, - mServerProxy->getSampleRate(), + mAudioTrackServerProxy->getSampleRate(), 20.0 * log10((vlr & 0xFFFF) / 4096.0), 20.0 * log10((vlr >> 16) / 4096.0), mCblk->server, - mCblk->user, (int)mMainBuffer, (int)mAuxBuffer, mCblk->flags, @@ -518,53 +487,27 @@ void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) nowInUnderrun); } +uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const { + return mAudioTrackServerProxy->getSampleRate(); +} + // AudioBufferProvider interface status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( AudioBufferProvider::Buffer* buffer, int64_t pts) { - audio_track_cblk_t* cblk = this->cblk(); - uint32_t framesReady; - uint32_t framesReq = buffer->frameCount; - - // Check if last stepServer failed, try to step now - if (mStepServerFailed) { - // FIXME When called by fast mixer, this takes a mutex with tryLock(). - // Since the fast mixer is higher priority than client callback thread, - // it does not result in priority inversion for client. - // But a non-blocking solution would be preferable to avoid - // fast mixer being unable to tryLock(), and - // to avoid the extra context switches if the client wakes up, - // discovers the mutex is locked, then has to wait for fast mixer to unlock. - if (!step()) goto getNextBuffer_exit; - ALOGV("stepServer recovered"); - mStepServerFailed = false; + ServerProxy::Buffer buf; + size_t desiredFrames = buffer->frameCount; + buf.mFrameCount = desiredFrames; + status_t status = mServerProxy->obtainBuffer(&buf); + buffer->frameCount = buf.mFrameCount; + buffer->raw = buf.mRaw; + if (buf.mFrameCount == 0) { + // only implemented so far for normal tracks, not fast tracks + mCblk->u.mStreaming.mUnderrunFrames += desiredFrames; + // FIXME also wake futex so that underrun is noticed more quickly + (void) android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); } - - // FIXME Same as above - framesReady = mServerProxy->framesReady(); - - if (CC_LIKELY(framesReady)) { - uint32_t s = cblk->server; - uint32_t bufferEnd = cblk->serverBase + mFrameCount; - - bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; - if (framesReq > framesReady) { - framesReq = framesReady; - } - if (framesReq > bufferEnd - s) { - framesReq = bufferEnd - s; - } - - buffer->raw = getBuffer(s, framesReq); - buffer->frameCount = framesReq; - return NO_ERROR; - } - -getNextBuffer_exit: - buffer->raw = NULL; - buffer->frameCount = 0; - ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); - return NOT_ENOUGH_DATA; + return status; } // Note that framesReady() takes a mutex on the control block using tryLock(). @@ -576,7 +519,7 @@ getNextBuffer_exit: // the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. // FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. size_t AudioFlinger::PlaybackThread::Track::framesReady() const { - return mServerProxy->framesReady(); + return mAudioTrackServerProxy->framesReady(); } // Don't call for fast tracks; the framesReady() could result in priority inversion @@ -732,7 +675,6 @@ void AudioFlinger::PlaybackThread::Track::reset() // Force underrun condition to avoid false underrun callback until first data is // written to buffer android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags); - android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); mFillingUpStatus = FS_FILLING; mResetDone = true; if (mState == FLUSHED) { @@ -833,7 +775,7 @@ uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() { // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); - uint32_t vlr = mServerProxy->getVolumeLR(); + uint32_t vlr = mAudioTrackServerProxy->getVolumeLR(); uint32_t vl = vlr & 0xFFFF; uint32_t vr = vlr >> 16; // track volumes come from shared memory, so can't be trusted and must be clamped @@ -870,9 +812,12 @@ status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp& void AudioFlinger::PlaybackThread::Track::invalidate() { - // FIXME should use proxy - android_atomic_or(CBLK_INVALID, &mCblk->flags); - mCblk->cv.signal(); + // FIXME should use proxy, and needs work + audio_track_cblk_t* cblk = mCblk; + android_atomic_or(CBLK_INVALID, &cblk->flags); + android_atomic_release_store(0x40000000, &cblk->mFutex); + // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE + (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX); mIsInvalid = true; } @@ -1418,6 +1363,8 @@ AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( mClientProxy->setVolumeLR((uint32_t(uint16_t(0x1000)) << 16) | uint16_t(0x1000)); mClientProxy->setSendLevel(0.0); mClientProxy->setSampleRate(sampleRate); + mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize, + true /*clientInServer*/); } else { ALOGW("Error creating output track on thread %p", playbackThread); } @@ -1498,9 +1445,10 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr if (mOutBuffer.frameCount == 0) { mOutBuffer.frameCount = pInBuffer->frameCount; nsecs_t startTime = systemTime(); - if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { - ALOGV("OutputTrack::write() %p thread %p no more output buffers", this, - mThread.unsafe_get()); + status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs); + if (status != NO_ERROR) { + ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this, + mThread.unsafe_get(), status); outputBufferFull = true; break; } @@ -1515,7 +1463,10 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); - mClientProxy->stepUser(outFrames); + Proxy::Buffer buf; + buf.mFrameCount = outFrames; + buf.mRaw = NULL; + mClientProxy->releaseBuffer(&buf); pInBuffer->frameCount -= outFrames; pInBuffer->i16 += outFrames * channelCount; mOutBuffer.frameCount -= outFrames; @@ -1559,8 +1510,10 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr // If no more buffers are pending, fill output track buffer to make sure it is started // by output mixer. if (frames == 0 && mBufferQueue.size() == 0) { - if (mCblk->user < mFrameCount) { - frames = mFrameCount - mCblk->user; + // FIXME borken, replace by getting framesReady() from proxy + size_t user = 0; // was mCblk->user + if (user < mFrameCount) { + frames = mFrameCount - user; pInBuffer = new Buffer; pInBuffer->mBuffer = new int16_t[frames * channelCount]; pInBuffer->frameCount = frames; @@ -1578,46 +1531,17 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) { - audio_track_cblk_t* cblk = mCblk; - uint32_t framesReq = buffer->frameCount; - - ALOGVV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); - buffer->frameCount = 0; - - size_t framesAvail; - { - Mutex::Autolock _l(cblk->lock); - - // read the server count again - while (!(framesAvail = mClientProxy->framesAvailable_l())) { - if (CC_UNLIKELY(!mActive)) { - ALOGV("Not active and NO_MORE_BUFFERS"); - return NO_MORE_BUFFERS; - } - status_t result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); - if (result != NO_ERROR) { - return NO_MORE_BUFFERS; - } - } - } - - if (framesReq > framesAvail) { - framesReq = framesAvail; - } - - uint32_t u = cblk->user; - uint32_t bufferEnd = cblk->userBase + mFrameCount; - - if (framesReq > bufferEnd - u) { - framesReq = bufferEnd - u; - } - - buffer->frameCount = framesReq; - buffer->raw = mClientProxy->buffer(u); - return NO_ERROR; + ClientProxy::Buffer buf; + buf.mFrameCount = buffer->frameCount; + struct timespec timeout; + timeout.tv_sec = waitTimeMs / 1000; + timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000; + status_t status = mClientProxy->obtainBuffer(&buf, &timeout); + buffer->frameCount = buf.mFrameCount; + buffer->raw = buf.mRaw; + return status; } - void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() { size_t size = mBufferQueue.size(); @@ -1688,6 +1612,11 @@ AudioFlinger::RecordThread::RecordTrack::RecordTrack( mOverflow(false) { ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); + if (mCblk != NULL) { + mAudioRecordServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, + mFrameSize); + mServerProxy = mAudioRecordServerProxy; + } } AudioFlinger::RecordThread::RecordTrack::~RecordTrack() @@ -1699,42 +1628,16 @@ AudioFlinger::RecordThread::RecordTrack::~RecordTrack() status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) { - audio_track_cblk_t* cblk = this->cblk(); - uint32_t framesAvail; - uint32_t framesReq = buffer->frameCount; - - // Check if last stepServer failed, try to step now - if (mStepServerFailed) { - if (!step()) { - goto getNextBuffer_exit; - } - ALOGV("stepServer recovered"); - mStepServerFailed = false; + ServerProxy::Buffer buf; + buf.mFrameCount = buffer->frameCount; + status_t status = mServerProxy->obtainBuffer(&buf); + buffer->frameCount = buf.mFrameCount; + buffer->raw = buf.mRaw; + if (buf.mFrameCount == 0) { + // FIXME also wake futex so that overrun is noticed more quickly + (void) android_atomic_or(CBLK_OVERRUN, &mCblk->flags); } - - // FIXME lock is not actually held, so overrun is possible - framesAvail = mServerProxy->framesAvailableIn_l(); - - if (CC_LIKELY(framesAvail)) { - uint32_t s = cblk->server; - uint32_t bufferEnd = cblk->serverBase + mFrameCount; - - if (framesReq > framesAvail) { - framesReq = framesAvail; - } - if (framesReq > bufferEnd - s) { - framesReq = bufferEnd - s; - } - - buffer->raw = getBuffer(s, framesReq); - buffer->frameCount = framesReq; - return NO_ERROR; - } - -getNextBuffer_exit: - buffer->raw = NULL; - buffer->frameCount = 0; - return NOT_ENOUGH_DATA; + return status; } status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, @@ -1790,12 +1693,12 @@ void AudioFlinger::RecordThread::RecordTrack::destroy() /*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) { - result.append(" Clien Fmt Chn mask Session Step S Serv User FrameCount\n"); + result.append(" Clien Fmt Chn mask Session Step S Serv FrameCount\n"); } void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) { - snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %08x %08x %05d\n", + snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %08x %05d\n", (mClient == 0) ? getpid_cached : mClient->pid(), mFormat, mChannelMask, @@ -1803,7 +1706,6 @@ void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) mStepCount, mState, mCblk->server, - mCblk->user, mFrameCount); } -- cgit v1.1