Home | History | Annotate | Download | only in media
      1 /*
      2  * Copyright (C) 2007 The Android Open Source Project
      3  *
      4  * Licensed under the Apache License, Version 2.0 (the "License");
      5  * you may not use this file except in compliance with the License.
      6  * You may obtain a copy of the License at
      7  *
      8  *      http://www.apache.org/licenses/LICENSE-2.0
      9  *
     10  * Unless required by applicable law or agreed to in writing, software
     11  * distributed under the License is distributed on an "AS IS" BASIS,
     12  * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
     13  * See the License for the specific language governing permissions and
     14  * limitations under the License.
     15  */
     16 
     17 #ifndef ANDROID_AUDIOTRACK_H
     18 #define ANDROID_AUDIOTRACK_H
     19 
     20 #include <cutils/sched_policy.h>
     21 #include <media/AudioSystem.h>
     22 #include <media/AudioTimestamp.h>
     23 #include <media/IAudioTrack.h>
     24 #include <utils/threads.h>
     25 
     26 namespace android {
     27 
     28 // ----------------------------------------------------------------------------
     29 
     30 class audio_track_cblk_t;
     31 class AudioTrackClientProxy;
     32 class StaticAudioTrackClientProxy;
     33 
     34 // ----------------------------------------------------------------------------
     35 
     36 class AudioTrack : public RefBase
     37 {
     38 public:
     39     enum channel_index {
     40         MONO   = 0,
     41         LEFT   = 0,
     42         RIGHT  = 1
     43     };
     44 
     45     /* Events used by AudioTrack callback function (callback_t).
     46      * Keep in sync with frameworks/base/media/java/android/media/AudioTrack.java NATIVE_EVENT_*.
     47      */
     48     enum event_type {
     49         EVENT_MORE_DATA = 0,        // Request to write more data to buffer.
     50                                     // If this event is delivered but the callback handler
     51                                     // does not want to write more data, the handler must explicitly
     52                                     // ignore the event by setting frameCount to zero.
     53         EVENT_UNDERRUN = 1,         // Buffer underrun occurred.
     54         EVENT_LOOP_END = 2,         // Sample loop end was reached; playback restarted from
     55                                     // loop start if loop count was not 0.
     56         EVENT_MARKER = 3,           // Playback head is at the specified marker position
     57                                     // (See setMarkerPosition()).
     58         EVENT_NEW_POS = 4,          // Playback head is at a new position
     59                                     // (See setPositionUpdatePeriod()).
     60         EVENT_BUFFER_END = 5,       // Playback head is at the end of the buffer.
     61                                     // Not currently used by android.media.AudioTrack.
     62         EVENT_NEW_IAUDIOTRACK = 6,  // IAudioTrack was re-created, either due to re-routing and
     63                                     // voluntary invalidation by mediaserver, or mediaserver crash.
     64         EVENT_STREAM_END = 7,       // Sent after all the buffers queued in AF and HW are played
     65                                     // back (after stop is called)
     66         EVENT_NEW_TIMESTAMP = 8,    // Delivered periodically and when there's a significant change
     67                                     // in the mapping from frame position to presentation time.
     68                                     // See AudioTimestamp for the information included with event.
     69     };
     70 
     71     /* Client should declare Buffer on the stack and pass address to obtainBuffer()
     72      * and releaseBuffer().  See also callback_t for EVENT_MORE_DATA.
     73      */
     74 
     75     class Buffer
     76     {
     77     public:
     78         // FIXME use m prefix
     79         size_t      frameCount;   // number of sample frames corresponding to size;
     80                                   // on input it is the number of frames desired,
     81                                   // on output is the number of frames actually filled
     82                                   // (currently ignored, but will make the primary field in future)
     83 
     84         size_t      size;         // input/output in bytes == frameCount * frameSize
     85                                   // on output is the number of bytes actually filled
     86                                   // FIXME this is redundant with respect to frameCount,
     87                                   // and TRANSFER_OBTAIN mode is broken for 8-bit data
     88                                   // since we don't define the frame format
     89 
     90         union {
     91             void*       raw;
     92             short*      i16;      // signed 16-bit
     93             int8_t*     i8;       // unsigned 8-bit, offset by 0x80
     94         };
     95     };
     96 
     97     /* As a convenience, if a callback is supplied, a handler thread
     98      * is automatically created with the appropriate priority. This thread
     99      * invokes the callback when a new buffer becomes available or various conditions occur.
    100      * Parameters:
    101      *
    102      * event:   type of event notified (see enum AudioTrack::event_type).
    103      * user:    Pointer to context for use by the callback receiver.
    104      * info:    Pointer to optional parameter according to event type:
    105      *          - EVENT_MORE_DATA: pointer to AudioTrack::Buffer struct. The callback must not write
    106      *            more bytes than indicated by 'size' field and update 'size' if fewer bytes are
    107      *            written.
    108      *          - EVENT_UNDERRUN: unused.
    109      *          - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining.
    110      *          - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames.
    111      *          - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames.
    112      *          - EVENT_BUFFER_END: unused.
    113      *          - EVENT_NEW_IAUDIOTRACK: unused.
    114      *          - EVENT_STREAM_END: unused.
    115      *          - EVENT_NEW_TIMESTAMP: pointer to const AudioTimestamp.
    116      */
    117 
    118     typedef void (*callback_t)(int event, void* user, void *info);
    119 
    120     /* Returns the minimum frame count required for the successful creation of
    121      * an AudioTrack object.
    122      * Returned status (from utils/Errors.h) can be:
    123      *  - NO_ERROR: successful operation
    124      *  - NO_INIT: audio server or audio hardware not initialized
    125      *  - BAD_VALUE: unsupported configuration
    126      */
    127 
    128     static status_t getMinFrameCount(size_t* frameCount,
    129                                      audio_stream_type_t streamType,
    130                                      uint32_t sampleRate);
    131 
    132     /* How data is transferred to AudioTrack
    133      */
    134     enum transfer_type {
    135         TRANSFER_DEFAULT,   // not specified explicitly; determine from the other parameters
    136         TRANSFER_CALLBACK,  // callback EVENT_MORE_DATA
    137         TRANSFER_OBTAIN,    // FIXME deprecated: call obtainBuffer() and releaseBuffer()
    138         TRANSFER_SYNC,      // synchronous write()
    139         TRANSFER_SHARED,    // shared memory
    140     };
    141 
    142     /* Constructs an uninitialized AudioTrack. No connection with
    143      * AudioFlinger takes place.  Use set() after this.
    144      */
    145                         AudioTrack();
    146 
    147     /* Creates an AudioTrack object and registers it with AudioFlinger.
    148      * Once created, the track needs to be started before it can be used.
    149      * Unspecified values are set to appropriate default values.
    150      * With this constructor, the track is configured for streaming mode.
    151      * Data to be rendered is supplied by write() or by the callback EVENT_MORE_DATA.
    152      * Intermixing a combination of write() and non-ignored EVENT_MORE_DATA is not allowed.
    153      *
    154      * Parameters:
    155      *
    156      * streamType:         Select the type of audio stream this track is attached to
    157      *                     (e.g. AUDIO_STREAM_MUSIC).
    158      * sampleRate:         Data source sampling rate in Hz.
    159      * format:             Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed
    160      *                     16 bits per sample).
    161      * channelMask:        Channel mask.
    162      * frameCount:         Minimum size of track PCM buffer in frames. This defines the
    163      *                     application's contribution to the
    164      *                     latency of the track. The actual size selected by the AudioTrack could be
    165      *                     larger if the requested size is not compatible with current audio HAL
    166      *                     configuration.  Zero means to use a default value.
    167      * flags:              See comments on audio_output_flags_t in <system/audio.h>.
    168      * cbf:                Callback function. If not null, this function is called periodically
    169      *                     to provide new data and inform of marker, position updates, etc.
    170      * user:               Context for use by the callback receiver.
    171      * notificationFrames: The callback function is called each time notificationFrames PCM
    172      *                     frames have been consumed from track input buffer.
    173      *                     This is expressed in units of frames at the initial source sample rate.
    174      * sessionId:          Specific session ID, or zero to use default.
    175      * transferType:       How data is transferred to AudioTrack.
    176      * threadCanCallJava:  Not present in parameter list, and so is fixed at false.
    177      */
    178 
    179                         AudioTrack( audio_stream_type_t streamType,
    180                                     uint32_t sampleRate,
    181                                     audio_format_t format,
    182                                     audio_channel_mask_t,
    183                                     int frameCount       = 0,
    184                                     audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
    185                                     callback_t cbf       = NULL,
    186                                     void* user           = NULL,
    187                                     int notificationFrames = 0,
    188                                     int sessionId        = 0,
    189                                     transfer_type transferType = TRANSFER_DEFAULT,
    190                                     const audio_offload_info_t *offloadInfo = NULL,
    191                                     int uid = -1);
    192 
    193     /* Creates an audio track and registers it with AudioFlinger.
    194      * With this constructor, the track is configured for static buffer mode.
    195      * The format must not be 8-bit linear PCM.
    196      * Data to be rendered is passed in a shared memory buffer
    197      * identified by the argument sharedBuffer, which must be non-0.
    198      * The memory should be initialized to the desired data before calling start().
    199      * The write() method is not supported in this case.
    200      * It is recommended to pass a callback function to be notified of playback end by an
    201      * EVENT_UNDERRUN event.
    202      */
    203 
    204                         AudioTrack( audio_stream_type_t streamType,
    205                                     uint32_t sampleRate,
    206                                     audio_format_t format,
    207                                     audio_channel_mask_t channelMask,
    208                                     const sp<IMemory>& sharedBuffer,
    209                                     audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
    210                                     callback_t cbf      = NULL,
    211                                     void* user          = NULL,
    212                                     int notificationFrames = 0,
    213                                     int sessionId       = 0,
    214                                     transfer_type transferType = TRANSFER_DEFAULT,
    215                                     const audio_offload_info_t *offloadInfo = NULL,
    216                                     int uid = -1);
    217 
    218     /* Terminates the AudioTrack and unregisters it from AudioFlinger.
    219      * Also destroys all resources associated with the AudioTrack.
    220      */
    221 protected:
    222                         virtual ~AudioTrack();
    223 public:
    224 
    225     /* Initialize an AudioTrack that was created using the AudioTrack() constructor.
    226      * Don't call set() more than once, or after the AudioTrack() constructors that take parameters.
    227      * Returned status (from utils/Errors.h) can be:
    228      *  - NO_ERROR: successful initialization
    229      *  - INVALID_OPERATION: AudioTrack is already initialized
    230      *  - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...)
    231      *  - NO_INIT: audio server or audio hardware not initialized
    232      * If status is not equal to NO_ERROR, don't call any other APIs on this AudioTrack.
    233      * If sharedBuffer is non-0, the frameCount parameter is ignored and
    234      * replaced by the shared buffer's total allocated size in frame units.
    235      *
    236      * Parameters not listed in the AudioTrack constructors above:
    237      *
    238      * threadCanCallJava:  Whether callbacks are made from an attached thread and thus can call JNI.
    239      */
    240             status_t    set(audio_stream_type_t streamType,
    241                             uint32_t sampleRate,
    242                             audio_format_t format,
    243                             audio_channel_mask_t channelMask,
    244                             int frameCount      = 0,
    245                             audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
    246                             callback_t cbf      = NULL,
    247                             void* user          = NULL,
    248                             int notificationFrames = 0,
    249                             const sp<IMemory>& sharedBuffer = 0,
    250                             bool threadCanCallJava = false,
    251                             int sessionId       = 0,
    252                             transfer_type transferType = TRANSFER_DEFAULT,
    253                             const audio_offload_info_t *offloadInfo = NULL,
    254                             int uid = -1);
    255 
    256     /* Result of constructing the AudioTrack. This must be checked for successful initialization
    257      * before using any AudioTrack API (except for set()), because using
    258      * an uninitialized AudioTrack produces undefined results.
    259      * See set() method above for possible return codes.
    260      */
    261             status_t    initCheck() const   { return mStatus; }
    262 
    263     /* Returns this track's estimated latency in milliseconds.
    264      * This includes the latency due to AudioTrack buffer size, AudioMixer (if any)
    265      * and audio hardware driver.
    266      */
    267             uint32_t    latency() const     { return mLatency; }
    268 
    269     /* getters, see constructors and set() */
    270 
    271             audio_stream_type_t streamType() const { return mStreamType; }
    272             audio_format_t format() const   { return mFormat; }
    273 
    274     /* Return frame size in bytes, which for linear PCM is
    275      * channelCount * (bit depth per channel / 8).
    276      * channelCount is determined from channelMask, and bit depth comes from format.
    277      * For non-linear formats, the frame size is typically 1 byte.
    278      */
    279             size_t      frameSize() const   { return mFrameSize; }
    280 
    281             uint32_t    channelCount() const { return mChannelCount; }
    282             uint32_t    frameCount() const  { return mFrameCount; }
    283 
    284     /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */
    285             sp<IMemory> sharedBuffer() const { return mSharedBuffer; }
    286 
    287     /* After it's created the track is not active. Call start() to
    288      * make it active. If set, the callback will start being called.
    289      * If the track was previously paused, volume is ramped up over the first mix buffer.
    290      */
    291             status_t        start();
    292 
    293     /* Stop a track.
    294      * In static buffer mode, the track is stopped immediately.
    295      * In streaming mode, the callback will cease being called.  Note that obtainBuffer() still
    296      * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK.
    297      * In streaming mode the stop does not occur immediately: any data remaining in the buffer
    298      * is first drained, mixed, and output, and only then is the track marked as stopped.
    299      */
    300             void        stop();
    301             bool        stopped() const;
    302 
    303     /* Flush a stopped or paused track. All previously buffered data is discarded immediately.
    304      * This has the effect of draining the buffers without mixing or output.
    305      * Flush is intended for streaming mode, for example before switching to non-contiguous content.
    306      * This function is a no-op if the track is not stopped or paused, or uses a static buffer.
    307      */
    308             void        flush();
    309 
    310     /* Pause a track. After pause, the callback will cease being called and
    311      * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works
    312      * and will fill up buffers until the pool is exhausted.
    313      * Volume is ramped down over the next mix buffer following the pause request,
    314      * and then the track is marked as paused.  It can be resumed with ramp up by start().
    315      */
    316             void        pause();
    317 
    318     /* Set volume for this track, mostly used for games' sound effects
    319      * left and right volumes. Levels must be >= 0.0 and <= 1.0.
    320      * This is the older API.  New applications should use setVolume(float) when possible.
    321      */
    322             status_t    setVolume(float left, float right);
    323 
    324     /* Set volume for all channels.  This is the preferred API for new applications,
    325      * especially for multi-channel content.
    326      */
    327             status_t    setVolume(float volume);
    328 
    329     /* Set the send level for this track. An auxiliary effect should be attached
    330      * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0.
    331      */
    332             status_t    setAuxEffectSendLevel(float level);
    333             void        getAuxEffectSendLevel(float* level) const;
    334 
    335     /* Set source sample rate for this track in Hz, mostly used for games' sound effects
    336      */
    337             status_t    setSampleRate(uint32_t sampleRate);
    338 
    339     /* Return current source sample rate in Hz, or 0 if unknown */
    340             uint32_t    getSampleRate() const;
    341 
    342     /* Enables looping and sets the start and end points of looping.
    343      * Only supported for static buffer mode.
    344      *
    345      * Parameters:
    346      *
    347      * loopStart:   loop start in frames relative to start of buffer.
    348      * loopEnd:     loop end in frames relative to start of buffer.
    349      * loopCount:   number of loops to execute. Calling setLoop() with loopCount == 0 cancels any
    350      *              pending or active loop. loopCount == -1 means infinite looping.
    351      *
    352      * For proper operation the following condition must be respected:
    353      *      loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount().
    354      *
    355      * If the loop period (loopEnd - loopStart) is too small for the implementation to support,
    356      * setLoop() will return BAD_VALUE.  loopCount must be >= -1.
    357      *
    358      */
    359             status_t    setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount);
    360 
    361     /* Sets marker position. When playback reaches the number of frames specified, a callback with
    362      * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker
    363      * notification callback.  To set a marker at a position which would compute as 0,
    364      * a workaround is to the set the marker at a nearby position such as ~0 or 1.
    365      * If the AudioTrack has been opened with no callback function associated, the operation will
    366      * fail.
    367      *
    368      * Parameters:
    369      *
    370      * marker:   marker position expressed in wrapping (overflow) frame units,
    371      *           like the return value of getPosition().
    372      *
    373      * Returned status (from utils/Errors.h) can be:
    374      *  - NO_ERROR: successful operation
    375      *  - INVALID_OPERATION: the AudioTrack has no callback installed.
    376      */
    377             status_t    setMarkerPosition(uint32_t marker);
    378             status_t    getMarkerPosition(uint32_t *marker) const;
    379 
    380     /* Sets position update period. Every time the number of frames specified has been played,
    381      * a callback with event type EVENT_NEW_POS is called.
    382      * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification
    383      * callback.
    384      * If the AudioTrack has been opened with no callback function associated, the operation will
    385      * fail.
    386      * Extremely small values may be rounded up to a value the implementation can support.
    387      *
    388      * Parameters:
    389      *
    390      * updatePeriod:  position update notification period expressed in frames.
    391      *
    392      * Returned status (from utils/Errors.h) can be:
    393      *  - NO_ERROR: successful operation
    394      *  - INVALID_OPERATION: the AudioTrack has no callback installed.
    395      */
    396             status_t    setPositionUpdatePeriod(uint32_t updatePeriod);
    397             status_t    getPositionUpdatePeriod(uint32_t *updatePeriod) const;
    398 
    399     /* Sets playback head position.
    400      * Only supported for static buffer mode.
    401      *
    402      * Parameters:
    403      *
    404      * position:  New playback head position in frames relative to start of buffer.
    405      *            0 <= position <= frameCount().  Note that end of buffer is permitted,
    406      *            but will result in an immediate underrun if started.
    407      *
    408      * Returned status (from utils/Errors.h) can be:
    409      *  - NO_ERROR: successful operation
    410      *  - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode.
    411      *  - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack
    412      *               buffer
    413      */
    414             status_t    setPosition(uint32_t position);
    415 
    416     /* Return the total number of frames played since playback start.
    417      * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz.
    418      * It is reset to zero by flush(), reload(), and stop().
    419      *
    420      * Parameters:
    421      *
    422      *  position:  Address where to return play head position.
    423      *
    424      * Returned status (from utils/Errors.h) can be:
    425      *  - NO_ERROR: successful operation
    426      *  - BAD_VALUE:  position is NULL
    427      */
    428             status_t    getPosition(uint32_t *position) const;
    429 
    430     /* For static buffer mode only, this returns the current playback position in frames
    431      * relative to start of buffer.  It is analogous to the position units used by
    432      * setLoop() and setPosition().  After underrun, the position will be at end of buffer.
    433      */
    434             status_t    getBufferPosition(uint32_t *position);
    435 
    436     /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids
    437      * rewriting the buffer before restarting playback after a stop.
    438      * This method must be called with the AudioTrack in paused or stopped state.
    439      * Not allowed in streaming mode.
    440      *
    441      * Returned status (from utils/Errors.h) can be:
    442      *  - NO_ERROR: successful operation
    443      *  - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode.
    444      */
    445             status_t    reload();
    446 
    447     /* Returns a handle on the audio output used by this AudioTrack.
    448      *
    449      * Parameters:
    450      *  none.
    451      *
    452      * Returned value:
    453      *  handle on audio hardware output
    454      */
    455             audio_io_handle_t    getOutput();
    456 
    457     /* Returns the unique session ID associated with this track.
    458      *
    459      * Parameters:
    460      *  none.
    461      *
    462      * Returned value:
    463      *  AudioTrack session ID.
    464      */
    465             int    getSessionId() const { return mSessionId; }
    466 
    467     /* Attach track auxiliary output to specified effect. Use effectId = 0
    468      * to detach track from effect.
    469      *
    470      * Parameters:
    471      *
    472      * effectId:  effectId obtained from AudioEffect::id().
    473      *
    474      * Returned status (from utils/Errors.h) can be:
    475      *  - NO_ERROR: successful operation
    476      *  - INVALID_OPERATION: the effect is not an auxiliary effect.
    477      *  - BAD_VALUE: The specified effect ID is invalid
    478      */
    479             status_t    attachAuxEffect(int effectId);
    480 
    481     /* Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames.
    482      * After filling these slots with data, the caller should release them with releaseBuffer().
    483      * If the track buffer is not full, obtainBuffer() returns as many contiguous
    484      * [empty slots for] frames as are available immediately.
    485      * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK
    486      * regardless of the value of waitCount.
    487      * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a
    488      * maximum timeout based on waitCount; see chart below.
    489      * Buffers will be returned until the pool
    490      * is exhausted, at which point obtainBuffer() will either block
    491      * or return WOULD_BLOCK depending on the value of the "waitCount"
    492      * parameter.
    493      * Each sample is 16-bit signed PCM.
    494      *
    495      * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications,
    496      * which should use write() or callback EVENT_MORE_DATA instead.
    497      *
    498      * Interpretation of waitCount:
    499      *  +n  limits wait time to n * WAIT_PERIOD_MS,
    500      *  -1  causes an (almost) infinite wait time,
    501      *   0  non-blocking.
    502      *
    503      * Buffer fields
    504      * On entry:
    505      *  frameCount  number of frames requested
    506      * After error return:
    507      *  frameCount  0
    508      *  size        0
    509      *  raw         undefined
    510      * After successful return:
    511      *  frameCount  actual number of frames available, <= number requested
    512      *  size        actual number of bytes available
    513      *  raw         pointer to the buffer
    514      */
    515 
    516     /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */
    517             status_t    obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
    518                                 __attribute__((__deprecated__));
    519 
    520 private:
    521     /* If nonContig is non-NULL, it is an output parameter that will be set to the number of
    522      * additional non-contiguous frames that are available immediately.
    523      * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(),
    524      * in case the requested amount of frames is in two or more non-contiguous regions.
    525      * FIXME requested and elapsed are both relative times.  Consider changing to absolute time.
    526      */
    527             status_t    obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
    528                                      struct timespec *elapsed = NULL, size_t *nonContig = NULL);
    529 public:
    530 
    531 //EL_FIXME to be reconciled with new obtainBuffer() return codes and control block proxy
    532 //            enum {
    533 //            NO_MORE_BUFFERS = 0x80000001,   // same name in AudioFlinger.h, ok to be different value
    534 //            TEAR_DOWN       = 0x80000002,
    535 //            STOPPED = 1,
    536 //            STREAM_END_WAIT,
    537 //            STREAM_END
    538 //        };
    539 
    540     /* Release a filled buffer of "audioBuffer->frameCount" frames for AudioFlinger to process. */
    541     // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed
    542             void        releaseBuffer(Buffer* audioBuffer);
    543 
    544     /* As a convenience we provide a write() interface to the audio buffer.
    545      * Input parameter 'size' is in byte units.
    546      * This is implemented on top of obtainBuffer/releaseBuffer. For best
    547      * performance use callbacks. Returns actual number of bytes written >= 0,
    548      * or one of the following negative status codes:
    549      *      INVALID_OPERATION   AudioTrack is configured for static buffer or streaming mode
    550      *      BAD_VALUE           size is invalid
    551      *      WOULD_BLOCK         when obtainBuffer() returns same, or
    552      *                          AudioTrack was stopped during the write
    553      *      or any other error code returned by IAudioTrack::start() or restoreTrack_l().
    554      */
    555             ssize_t     write(const void* buffer, size_t size);
    556 
    557     /*
    558      * Dumps the state of an audio track.
    559      */
    560             status_t    dump(int fd, const Vector<String16>& args) const;
    561 
    562     /*
    563      * Return the total number of frames which AudioFlinger desired but were unavailable,
    564      * and thus which resulted in an underrun.  Reset to zero by stop().
    565      */
    566             uint32_t    getUnderrunFrames() const;
    567 
    568     /* Get the flags */
    569             audio_output_flags_t getFlags() const { return mFlags; }
    570 
    571     /* Set parameters - only possible when using direct output */
    572             status_t    setParameters(const String8& keyValuePairs);
    573 
    574     /* Get parameters */
    575             String8     getParameters(const String8& keys);
    576 
    577     /* Poll for a timestamp on demand.
    578      * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs,
    579      * or if you need to get the most recent timestamp outside of the event callback handler.
    580      * Caution: calling this method too often may be inefficient;
    581      * if you need a high resolution mapping between frame position and presentation time,
    582      * consider implementing that at application level, based on the low resolution timestamps.
    583      * Returns NO_ERROR if timestamp is valid.
    584      */
    585             status_t    getTimestamp(AudioTimestamp& timestamp);
    586 
    587 protected:
    588     /* copying audio tracks is not allowed */
    589                         AudioTrack(const AudioTrack& other);
    590             AudioTrack& operator = (const AudioTrack& other);
    591 
    592     /* a small internal class to handle the callback */
    593     class AudioTrackThread : public Thread
    594     {
    595     public:
    596         AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false);
    597 
    598         // Do not call Thread::requestExitAndWait() without first calling requestExit().
    599         // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough.
    600         virtual void        requestExit();
    601 
    602                 void        pause();    // suspend thread from execution at next loop boundary
    603                 void        resume();   // allow thread to execute, if not requested to exit
    604 
    605     private:
    606                 void        pauseInternal(nsecs_t ns = 0LL);
    607                                         // like pause(), but only used internally within thread
    608 
    609         friend class AudioTrack;
    610         virtual bool        threadLoop();
    611         AudioTrack&         mReceiver;
    612         virtual ~AudioTrackThread();
    613         Mutex               mMyLock;    // Thread::mLock is private
    614         Condition           mMyCond;    // Thread::mThreadExitedCondition is private
    615         bool                mPaused;    // whether thread is requested to pause at next loop entry
    616         bool                mPausedInt; // whether thread internally requests pause
    617         nsecs_t             mPausedNs;  // if mPausedInt then associated timeout, otherwise ignored
    618         bool                mIgnoreNextPausedInt;   // whether to ignore next mPausedInt request
    619     };
    620 
    621             // body of AudioTrackThread::threadLoop()
    622             // returns the maximum amount of time before we would like to run again, where:
    623             //      0           immediately
    624             //      > 0         no later than this many nanoseconds from now
    625             //      NS_WHENEVER still active but no particular deadline
    626             //      NS_INACTIVE inactive so don't run again until re-started
    627             //      NS_NEVER    never again
    628             static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3;
    629             nsecs_t processAudioBuffer(const sp<AudioTrackThread>& thread);
    630             status_t processStreamEnd(int32_t waitCount);
    631 
    632 
    633             // caller must hold lock on mLock for all _l methods
    634 
    635             status_t createTrack_l(audio_stream_type_t streamType,
    636                                  uint32_t sampleRate,
    637                                  audio_format_t format,
    638                                  size_t frameCount,
    639                                  audio_output_flags_t flags,
    640                                  const sp<IMemory>& sharedBuffer,
    641                                  audio_io_handle_t output,
    642                                  size_t epoch);
    643 
    644             // can only be called when mState != STATE_ACTIVE
    645             void flush_l();
    646 
    647             void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount);
    648             audio_io_handle_t getOutput_l();
    649 
    650             // FIXME enum is faster than strcmp() for parameter 'from'
    651             status_t restoreTrack_l(const char *from);
    652 
    653             bool     isOffloaded() const
    654                 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; }
    655 
    656     // Next 3 fields may be changed if IAudioTrack is re-created, but always != 0
    657     sp<IAudioTrack>         mAudioTrack;
    658     sp<IMemory>             mCblkMemory;
    659     audio_track_cblk_t*     mCblk;                  // re-load after mLock.unlock()
    660 
    661     sp<AudioTrackThread>    mAudioTrackThread;
    662     float                   mVolume[2];
    663     float                   mSendLevel;
    664     mutable uint32_t        mSampleRate;            // mutable because getSampleRate() can update it.
    665     size_t                  mFrameCount;            // corresponds to current IAudioTrack
    666     size_t                  mReqFrameCount;         // frame count to request the next time a new
    667                                                     // IAudioTrack is needed
    668 
    669 
    670     // constant after constructor or set()
    671     audio_format_t          mFormat;                // as requested by client, not forced to 16-bit
    672     audio_stream_type_t     mStreamType;
    673     uint32_t                mChannelCount;
    674     audio_channel_mask_t    mChannelMask;
    675     transfer_type           mTransfer;
    676 
    677     // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data.  For 8-bit PCM data, it's
    678     // twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer.
    679     size_t                  mFrameSize;             // app-level frame size
    680     size_t                  mFrameSizeAF;           // AudioFlinger frame size
    681 
    682     status_t                mStatus;
    683 
    684     // can change dynamically when IAudioTrack invalidated
    685     uint32_t                mLatency;               // in ms
    686 
    687     // Indicates the current track state.  Protected by mLock.
    688     enum State {
    689         STATE_ACTIVE,
    690         STATE_STOPPED,
    691         STATE_PAUSED,
    692         STATE_PAUSED_STOPPING,
    693         STATE_FLUSHED,
    694         STATE_STOPPING,
    695     }                       mState;
    696 
    697     // for client callback handler
    698     callback_t              mCbf;                   // callback handler for events, or NULL
    699     void*                   mUserData;
    700 
    701     // for notification APIs
    702     uint32_t                mNotificationFramesReq; // requested number of frames between each
    703                                                     // notification callback,
    704                                                     // at initial source sample rate
    705     uint32_t                mNotificationFramesAct; // actual number of frames between each
    706                                                     // notification callback,
    707                                                     // at initial source sample rate
    708     bool                    mRefreshRemaining;      // processAudioBuffer() should refresh next 2
    709 
    710     // These are private to processAudioBuffer(), and are not protected by a lock
    711     uint32_t                mRemainingFrames;       // number of frames to request in obtainBuffer()
    712     bool                    mRetryOnPartialBuffer;  // sleep and retry after partial obtainBuffer()
    713     uint32_t                mObservedSequence;      // last observed value of mSequence
    714 
    715     sp<IMemory>             mSharedBuffer;
    716     uint32_t                mLoopPeriod;            // in frames, zero means looping is disabled
    717     uint32_t                mMarkerPosition;        // in wrapping (overflow) frame units
    718     bool                    mMarkerReached;
    719     uint32_t                mNewPosition;           // in frames
    720     uint32_t                mUpdatePeriod;          // in frames, zero means no EVENT_NEW_POS
    721 
    722     audio_output_flags_t    mFlags;
    723     int                     mSessionId;
    724     int                     mAuxEffectId;
    725 
    726     mutable Mutex           mLock;
    727 
    728     bool                    mIsTimed;
    729     int                     mPreviousPriority;          // before start()
    730     SchedPolicy             mPreviousSchedulingGroup;
    731     bool                    mAwaitBoost;    // thread should wait for priority boost before running
    732 
    733     // The proxy should only be referenced while a lock is held because the proxy isn't
    734     // multi-thread safe, especially the SingleStateQueue part of the proxy.
    735     // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock,
    736     // provided that the caller also holds an extra reference to the proxy and shared memory to keep
    737     // them around in case they are replaced during the obtainBuffer().
    738     sp<StaticAudioTrackClientProxy> mStaticProxy;   // for type safety only
    739     sp<AudioTrackClientProxy>       mProxy;         // primary owner of the memory
    740 
    741     bool                    mInUnderrun;            // whether track is currently in underrun state
    742     String8                 mName;                  // server's name for this IAudioTrack
    743     uint32_t                mPausedPosition;
    744 
    745 private:
    746     class DeathNotifier : public IBinder::DeathRecipient {
    747     public:
    748         DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { }
    749     protected:
    750         virtual void        binderDied(const wp<IBinder>& who);
    751     private:
    752         const wp<AudioTrack> mAudioTrack;
    753     };
    754 
    755     sp<DeathNotifier>       mDeathNotifier;
    756     uint32_t                mSequence;              // incremented for each new IAudioTrack attempt
    757     audio_io_handle_t       mOutput;                // cached output io handle
    758     int                     mClientUid;
    759 };
    760 
    761 class TimedAudioTrack : public AudioTrack
    762 {
    763 public:
    764     TimedAudioTrack();
    765 
    766     /* allocate a shared memory buffer that can be passed to queueTimedBuffer */
    767     status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer);
    768 
    769     /* queue a buffer obtained via allocateTimedBuffer for playback at the
    770        given timestamp.  PTS units are microseconds on the media time timeline.
    771        The media time transform (set with setMediaTimeTransform) set by the
    772        audio producer will handle converting from media time to local time
    773        (perhaps going through the common time timeline in the case of
    774        synchronized multiroom audio case) */
    775     status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts);
    776 
    777     /* define a transform between media time and either common time or
    778        local time */
    779     enum TargetTimeline {LOCAL_TIME, COMMON_TIME};
    780     status_t setMediaTimeTransform(const LinearTransform& xform,
    781                                    TargetTimeline target);
    782 };
    783 
    784 }; // namespace android
    785 
    786 #endif // ANDROID_AUDIOTRACK_H
    787