AudioTrack.h revision 5aab59a2bd0a2cd80240ffd66c1b963b5fe06d65
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 <stdint.h>
21#include <sys/types.h>
22
23#include <media/IAudioFlinger.h>
24#include <media/IAudioTrack.h>
25#include <media/AudioSystem.h>
26
27#include <utils/RefBase.h>
28#include <utils/Errors.h>
29#include <binder/IInterface.h>
30#include <binder/IMemory.h>
31#include <cutils/sched_policy.h>
32#include <utils/threads.h>
33
34namespace android {
35
36// ----------------------------------------------------------------------------
37
38class audio_track_cblk_t;
39
40// ----------------------------------------------------------------------------
41
42class AudioTrack : virtual public RefBase
43{
44public:
45    enum channel_index {
46        MONO   = 0,
47        LEFT   = 0,
48        RIGHT  = 1
49    };
50
51    /* Events used by AudioTrack callback function (audio_track_cblk_t).
52     * Keep in sync with frameworks/base/media/java/android/media/AudioTrack.java NATIVE_EVENT_*.
53     */
54    enum event_type {
55        EVENT_MORE_DATA = 0,        // Request to write more data to PCM buffer.
56        EVENT_UNDERRUN = 1,         // PCM buffer underrun occurred.
57        EVENT_LOOP_END = 2,         // Sample loop end was reached; playback restarted from
58                                    // loop start if loop count was not 0.
59        EVENT_MARKER = 3,           // Playback head is at the specified marker position
60                                    // (See setMarkerPosition()).
61        EVENT_NEW_POS = 4,          // Playback head is at a new position
62                                    // (See setPositionUpdatePeriod()).
63        EVENT_BUFFER_END = 5        // Playback head is at the end of the buffer.
64    };
65
66    /* Client should declare Buffer on the stack and pass address to obtainBuffer()
67     * and releaseBuffer().  See also callback_t for EVENT_MORE_DATA.
68     */
69
70    class Buffer
71    {
72    public:
73        size_t      frameCount;   // number of sample frames corresponding to size;
74                                  // on input it is the number of frames desired,
75                                  // on output is the number of frames actually filled
76
77        size_t      size;         // input/output in byte units
78        union {
79            void*       raw;
80            short*      i16;    // signed 16-bit
81            int8_t*     i8;     // unsigned 8-bit, offset by 0x80
82        };
83    };
84
85
86    /* As a convenience, if a callback is supplied, a handler thread
87     * is automatically created with the appropriate priority. This thread
88     * invokes the callback when a new buffer becomes available or various conditions occur.
89     * Parameters:
90     *
91     * event:   type of event notified (see enum AudioTrack::event_type).
92     * user:    Pointer to context for use by the callback receiver.
93     * info:    Pointer to optional parameter according to event type:
94     *          - EVENT_MORE_DATA: pointer to AudioTrack::Buffer struct. The callback must not write
95     *            more bytes than indicated by 'size' field and update 'size' if fewer bytes are
96     *            written.
97     *          - EVENT_UNDERRUN: unused.
98     *          - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining.
99     *          - EVENT_MARKER: pointer to an uint32_t containing the marker position in frames.
100     *          - EVENT_NEW_POS: pointer to an uint32_t containing the new position in frames.
101     *          - EVENT_BUFFER_END: unused.
102     */
103
104    typedef void (*callback_t)(int event, void* user, void *info);
105
106    /* Returns the minimum frame count required for the successful creation of
107     * an AudioTrack object.
108     * Returned status (from utils/Errors.h) can be:
109     *  - NO_ERROR: successful operation
110     *  - NO_INIT: audio server or audio hardware not initialized
111     */
112
113     static status_t getMinFrameCount(int* frameCount,
114                                      audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT,
115                                      uint32_t sampleRate = 0);
116
117    /* Constructs an uninitialized AudioTrack. No connection with
118     * AudioFlinger takes place.
119     */
120                        AudioTrack();
121
122    /* Creates an AudioTrack object and registers it with AudioFlinger.
123     * Once created, the track needs to be started before it can be used.
124     * Unspecified values are set to the audio hardware's current
125     * values.
126     *
127     * Parameters:
128     *
129     * streamType:         Select the type of audio stream this track is attached to
130     *                     (e.g. AUDIO_STREAM_MUSIC).
131     * sampleRate:         Track sampling rate in Hz.
132     * format:             Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed
133     *                     16 bits per sample).
134     * channelMask:        Channel mask.
135     * frameCount:         Minimum size of track PCM buffer in frames. This defines the
136     *                     application's contribution to the
137     *                     latency of the track. The actual size selected by the AudioTrack could be
138     *                     larger if the requested size is not compatible with current audio HAL
139     *                     latency.  Zero means to use a default value.
140     * flags:              See comments on audio_output_flags_t in <system/audio.h>.
141     * cbf:                Callback function. If not null, this function is called periodically
142     *                     to provide new PCM data.
143     * user:               Context for use by the callback receiver.
144     * notificationFrames: The callback function is called each time notificationFrames PCM
145     *                     frames have been consumed from track input buffer.
146     * sessionId:          Specific session ID, or zero to use default.
147     * threadCanCallJava:  Whether callbacks are made from an attached thread and thus can call JNI.
148     *                     If not present in parameter list, then fixed at false.
149     */
150
151                        AudioTrack( audio_stream_type_t streamType,
152                                    uint32_t sampleRate  = 0,
153                                    audio_format_t format = AUDIO_FORMAT_DEFAULT,
154                                    audio_channel_mask_t channelMask = 0,
155                                    int frameCount       = 0,
156                                    audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
157                                    callback_t cbf       = NULL,
158                                    void* user           = NULL,
159                                    int notificationFrames = 0,
160                                    int sessionId        = 0);
161
162    /* Creates an audio track and registers it with AudioFlinger. With this constructor,
163     * the PCM data to be rendered by AudioTrack is passed in a shared memory buffer
164     * identified by the argument sharedBuffer. This prototype is for static buffer playback.
165     * PCM data must be present in memory before the AudioTrack is started.
166     * The write() and flush() methods are not supported in this case.
167     * It is recommended to pass a callback function to be notified of playback end by an
168     * EVENT_UNDERRUN event.
169     */
170
171                        AudioTrack( audio_stream_type_t streamType,
172                                    uint32_t sampleRate = 0,
173                                    audio_format_t format = AUDIO_FORMAT_DEFAULT,
174                                    audio_channel_mask_t channelMask = 0,
175                                    const sp<IMemory>& sharedBuffer = 0,
176                                    audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
177                                    callback_t cbf      = NULL,
178                                    void* user          = NULL,
179                                    int notificationFrames = 0,
180                                    int sessionId       = 0);
181
182    /* Terminates the AudioTrack and unregisters it from AudioFlinger.
183     * Also destroys all resources associated with the AudioTrack.
184     */
185                        ~AudioTrack();
186
187
188    /* Initialize an uninitialized AudioTrack.
189     * Returned status (from utils/Errors.h) can be:
190     *  - NO_ERROR: successful initialization
191     *  - INVALID_OPERATION: AudioTrack is already initialized
192     *  - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...)
193     *  - NO_INIT: audio server or audio hardware not initialized
194     */
195            status_t    set(audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT,
196                            uint32_t sampleRate = 0,
197                            audio_format_t format = AUDIO_FORMAT_DEFAULT,
198                            audio_channel_mask_t channelMask = 0,
199                            int frameCount      = 0,
200                            audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
201                            callback_t cbf      = NULL,
202                            void* user          = NULL,
203                            int notificationFrames = 0,
204                            const sp<IMemory>& sharedBuffer = 0,
205                            bool threadCanCallJava = false,
206                            int sessionId       = 0);
207
208
209    /* Result of constructing the AudioTrack. This must be checked
210     * before using any AudioTrack API (except for set()), because using
211     * an uninitialized AudioTrack produces undefined results.
212     * See set() method above for possible return codes.
213     */
214            status_t    initCheck() const;
215
216    /* Returns this track's estimated latency in milliseconds.
217     * This includes the latency due to AudioTrack buffer size, AudioMixer (if any)
218     * and audio hardware driver.
219     */
220            uint32_t     latency() const;
221
222    /* getters, see constructors and set() */
223
224            audio_stream_type_t streamType() const;
225            audio_format_t format() const;
226            int         channelCount() const;
227            uint32_t    frameCount() const;
228
229    /* Return channelCount * (bit depth per channel / 8).
230     * channelCount is determined from channelMask, and bit depth comes from format.
231     */
232            size_t      frameSize() const { return mFrameSize; }
233
234            sp<IMemory>& sharedBuffer();
235
236
237    /* After it's created the track is not active. Call start() to
238     * make it active. If set, the callback will start being called.
239     */
240            void        start();
241
242    /* Stop a track. If set, the callback will cease being called and
243     * obtainBuffer returns STOPPED. Note that obtainBuffer() still works
244     * and will fill up buffers until the pool is exhausted.
245     */
246            void        stop();
247            bool        stopped() const;
248
249    /* Flush a stopped track. All pending buffers are discarded.
250     * This function has no effect if the track is not stopped.
251     */
252            void        flush();
253
254    /* Pause a track. If set, the callback will cease being called and
255     * obtainBuffer returns STOPPED. Note that obtainBuffer() still works
256     * and will fill up buffers until the pool is exhausted.
257     */
258            void        pause();
259
260    /* Mute or unmute this track.
261     * While muted, the callback, if set, is still called.
262     */
263            void        mute(bool);
264            bool        muted() const;
265
266    /* Set volume for this track, mostly used for games' sound effects
267     * left and right volumes. Levels must be >= 0.0 and <= 1.0.
268     * This is the older API.  New applications should use setVolume(float) when possible.
269     */
270            status_t    setVolume(float left, float right);
271
272    /* Set volume for all channels.  This is the preferred API for new applications,
273     * especially for multi-channel content.
274     */
275            status_t    setVolume(float volume);
276
277    /* Set the send level for this track. An auxiliary effect should be attached
278     * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0.
279     */
280            status_t    setAuxEffectSendLevel(float level);
281            void        getAuxEffectSendLevel(float* level) const;
282
283    /* Set sample rate for this track in Hz, mostly used for games' sound effects
284     */
285            status_t    setSampleRate(int sampleRate);
286            uint32_t    getSampleRate() const;
287
288    /* Enables looping and sets the start and end points of looping.
289     *
290     * Parameters:
291     *
292     * loopStart:   loop start expressed as the number of PCM frames played since AudioTrack start.
293     * loopEnd:     loop end expressed as the number of PCM frames played since AudioTrack start.
294     * loopCount:   number of loops to execute. Calling setLoop() with loopCount == 0 cancels any
295     *              pending or active loop. loopCount = -1 means infinite looping.
296     *
297     * For proper operation the following condition must be respected:
298     *          (loopEnd-loopStart) <= framecount()
299     */
300            status_t    setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount);
301
302    /* Sets marker position. When playback reaches the number of frames specified, a callback with
303     * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker
304     * notification callback.
305     * If the AudioTrack has been opened with no callback function associated, the operation will
306     * fail.
307     *
308     * Parameters:
309     *
310     * marker:   marker position expressed in frames.
311     *
312     * Returned status (from utils/Errors.h) can be:
313     *  - NO_ERROR: successful operation
314     *  - INVALID_OPERATION: the AudioTrack has no callback installed.
315     */
316            status_t    setMarkerPosition(uint32_t marker);
317            status_t    getMarkerPosition(uint32_t *marker) const;
318
319
320    /* Sets position update period. Every time the number of frames specified has been played,
321     * a callback with event type EVENT_NEW_POS is called.
322     * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification
323     * callback.
324     * If the AudioTrack has been opened with no callback function associated, the operation will
325     * fail.
326     *
327     * Parameters:
328     *
329     * updatePeriod:  position update notification period expressed in frames.
330     *
331     * Returned status (from utils/Errors.h) can be:
332     *  - NO_ERROR: successful operation
333     *  - INVALID_OPERATION: the AudioTrack has no callback installed.
334     */
335            status_t    setPositionUpdatePeriod(uint32_t updatePeriod);
336            status_t    getPositionUpdatePeriod(uint32_t *updatePeriod) const;
337
338    /* Sets playback head position within AudioTrack buffer. The new position is specified
339     * in number of frames.
340     * This method must be called with the AudioTrack in paused or stopped state.
341     * Note that the actual position set is <position> modulo the AudioTrack buffer size in frames.
342     * Therefore using this method makes sense only when playing a "static" audio buffer
343     * as opposed to streaming.
344     * The getPosition() method on the other hand returns the total number of frames played since
345     * playback start.
346     *
347     * Parameters:
348     *
349     * position:  New playback head position within AudioTrack buffer.
350     *
351     * Returned status (from utils/Errors.h) can be:
352     *  - NO_ERROR: successful operation
353     *  - INVALID_OPERATION: the AudioTrack is not stopped.
354     *  - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack
355     *               buffer
356     */
357            status_t    setPosition(uint32_t position);
358            status_t    getPosition(uint32_t *position);
359
360    /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids
361     * rewriting the buffer before restarting playback after a stop.
362     * This method must be called with the AudioTrack in paused or stopped state.
363     *
364     * Returned status (from utils/Errors.h) can be:
365     *  - NO_ERROR: successful operation
366     *  - INVALID_OPERATION: the AudioTrack is not stopped.
367     */
368            status_t    reload();
369
370    /* Returns a handle on the audio output used by this AudioTrack.
371     *
372     * Parameters:
373     *  none.
374     *
375     * Returned value:
376     *  handle on audio hardware output
377     */
378            audio_io_handle_t    getOutput();
379
380    /* Returns the unique session ID associated with this track.
381     *
382     * Parameters:
383     *  none.
384     *
385     * Returned value:
386     *  AudioTrack session ID.
387     */
388            int    getSessionId() const;
389
390    /* Attach track auxiliary output to specified effect. Use effectId = 0
391     * to detach track from effect.
392     *
393     * Parameters:
394     *
395     * effectId:  effectId obtained from AudioEffect::id().
396     *
397     * Returned status (from utils/Errors.h) can be:
398     *  - NO_ERROR: successful operation
399     *  - INVALID_OPERATION: the effect is not an auxiliary effect.
400     *  - BAD_VALUE: The specified effect ID is invalid
401     */
402            status_t    attachAuxEffect(int effectId);
403
404    /* Obtains a buffer of "frameCount" frames. The buffer must be
405     * filled entirely, and then released with releaseBuffer().
406     * If the track is stopped, obtainBuffer() returns
407     * STOPPED instead of NO_ERROR as long as there are buffers available,
408     * at which point NO_MORE_BUFFERS is returned.
409     * Buffers will be returned until the pool
410     * is exhausted, at which point obtainBuffer() will either block
411     * or return WOULD_BLOCK depending on the value of the "blocking"
412     * parameter.
413     *
414     * Interpretation of waitCount:
415     *  +n  limits wait time to n * WAIT_PERIOD_MS,
416     *  -1  causes an (almost) infinite wait time,
417     *   0  non-blocking.
418     *
419     * Buffer fields
420     * On entry:
421     *  frameCount  number of frames requested
422     * After error return:
423     *  frameCount  0
424     *  size        0
425     * After successful return:
426     *  frameCount  actual number of frames available, <= number requested
427     *  size        actual number of bytes available
428     *  raw         pointer to the buffer
429     */
430
431        enum {
432            NO_MORE_BUFFERS = 0x80000001,   // same name in AudioFlinger.h, ok to be different value
433            STOPPED = 1
434        };
435
436            status_t    obtainBuffer(Buffer* audioBuffer, int32_t waitCount);
437
438    /* Release a filled buffer of "frameCount" frames for AudioFlinger to process. */
439            void        releaseBuffer(Buffer* audioBuffer);
440
441    /* As a convenience we provide a write() interface to the audio buffer.
442     * This is implemented on top of obtainBuffer/releaseBuffer. For best
443     * performance use callbacks. Returns actual number of bytes written >= 0,
444     * or one of the following negative status codes:
445     *      INVALID_OPERATION   AudioTrack is configured for shared buffer mode
446     *      BAD_VALUE           size is invalid
447     *      STOPPED             AudioTrack was stopped during the write
448     *      NO_MORE_BUFFERS     when obtainBuffer() returns same
449     *      or any other error code returned by IAudioTrack::start() or restoreTrack_l().
450     */
451            ssize_t     write(const void* buffer, size_t size);
452
453    /*
454     * Dumps the state of an audio track.
455     */
456            status_t dump(int fd, const Vector<String16>& args) const;
457
458protected:
459    /* copying audio tracks is not allowed */
460                        AudioTrack(const AudioTrack& other);
461            AudioTrack& operator = (const AudioTrack& other);
462
463    /* a small internal class to handle the callback */
464    class AudioTrackThread : public Thread
465    {
466    public:
467        AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false);
468
469        // Do not call Thread::requestExitAndWait() without first calling requestExit().
470        // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough.
471        virtual void        requestExit();
472
473                void        pause();    // suspend thread from execution at next loop boundary
474                void        resume();   // allow thread to execute, if not requested to exit
475
476    private:
477        friend class AudioTrack;
478        virtual bool        threadLoop();
479        AudioTrack& mReceiver;
480        ~AudioTrackThread();
481        Mutex               mMyLock;    // Thread::mLock is private
482        Condition           mMyCond;    // Thread::mThreadExitedCondition is private
483        bool                mPaused;    // whether thread is currently paused
484    };
485
486            // body of AudioTrackThread::threadLoop()
487            bool processAudioBuffer(const sp<AudioTrackThread>& thread);
488
489            // caller must hold lock on mLock for all _l methods
490            status_t createTrack_l(audio_stream_type_t streamType,
491                                 uint32_t sampleRate,
492                                 audio_format_t format,
493                                 audio_channel_mask_t channelMask,
494                                 int frameCount,
495                                 audio_output_flags_t flags,
496                                 const sp<IMemory>& sharedBuffer,
497                                 audio_io_handle_t output);
498            void flush_l();
499            status_t setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount);
500            audio_io_handle_t getOutput_l();
501            status_t restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart);
502            bool stopped_l() const { return !mActive; }
503
504    sp<IAudioTrack>         mAudioTrack;
505    sp<IMemory>             mCblkMemory;
506    sp<AudioTrackThread>    mAudioTrackThread;
507
508    float                   mVolume[2];
509    float                   mSendLevel;
510    uint32_t                mFrameCount;
511
512    audio_track_cblk_t*     mCblk;                  // re-load after mLock.unlock()
513    void*                   mBuffers;               // starting address of buffers in shared memory
514    audio_format_t          mFormat;
515    audio_stream_type_t     mStreamType;
516    uint8_t                 mChannelCount;
517    uint8_t                 mMuted;
518    uint8_t                 mReserved;
519    audio_channel_mask_t    mChannelMask;
520
521                // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data.
522                // For 8-bit PCM data, mFrameSizeAF is
523                // twice as large because data is expanded to 16-bit before being stored in buffer.
524    size_t                  mFrameSize;             // app-level frame size
525    size_t                  mFrameSizeAF;           // AudioFlinger frame size
526
527    status_t                mStatus;
528    uint32_t                mLatency;
529
530    bool                    mActive;                // protected by mLock
531
532    callback_t              mCbf;                   // callback handler for events, or NULL
533    void*                   mUserData;              // for client callback handler
534
535    // for notification APIs
536    uint32_t                mNotificationFramesReq; // requested number of frames between each
537                                                    // notification callback
538    uint32_t                mNotificationFramesAct; // actual number of frames between each
539                                                    // notification callback
540    sp<IMemory>             mSharedBuffer;
541    int                     mLoopCount;
542    uint32_t                mRemainingFrames;
543    uint32_t                mMarkerPosition;        // in frames
544    bool                    mMarkerReached;
545    uint32_t                mNewPosition;           // in frames
546    uint32_t                mUpdatePeriod;          // in frames
547
548    bool                    mFlushed; // FIXME will be made obsolete by making flush() synchronous
549    audio_output_flags_t    mFlags;
550    int                     mSessionId;
551    int                     mAuxEffectId;
552
553    // When locking both mLock and mCblk->lock, must lock in this order to avoid deadlock:
554    //      1. mLock
555    //      2. mCblk->lock
556    // It is OK to lock only mCblk->lock.
557    mutable Mutex           mLock;
558
559    bool                    mIsTimed;
560    int                     mPreviousPriority;          // before start()
561    SchedPolicy             mPreviousSchedulingGroup;
562};
563
564class TimedAudioTrack : public AudioTrack
565{
566public:
567    TimedAudioTrack();
568
569    /* allocate a shared memory buffer that can be passed to queueTimedBuffer */
570    status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer);
571
572    /* queue a buffer obtained via allocateTimedBuffer for playback at the
573       given timestamp.  PTS units are microseconds on the media time timeline.
574       The media time transform (set with setMediaTimeTransform) set by the
575       audio producer will handle converting from media time to local time
576       (perhaps going through the common time timeline in the case of
577       synchronized multiroom audio case) */
578    status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts);
579
580    /* define a transform between media time and either common time or
581       local time */
582    enum TargetTimeline {LOCAL_TIME, COMMON_TIME};
583    status_t setMediaTimeTransform(const LinearTransform& xform,
584                                   TargetTimeline target);
585};
586
587}; // namespace android
588
589#endif // ANDROID_AUDIOTRACK_H
590