AudioTrack.h revision b1c0993b215c5c3eebd1c6bafc22bba23d57a70b
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        enum {
74            MUTE    = 0x00000001
75        };
76        uint32_t    flags;        // 0 or MUTE
77        audio_format_t format;    // but AUDIO_FORMAT_PCM_8_BIT -> AUDIO_FORMAT_PCM_16_BIT
78        // accessed directly by WebKit ANP callback
79        int         channelCount; // will be removed in the future, do not use
80
81        size_t      frameCount;   // number of sample frames corresponding to size;
82                                  // on input it is the number of frames desired,
83                                  // on output is the number of frames actually filled
84
85        size_t      size;         // input/output in byte units
86        union {
87            void*       raw;
88            short*      i16;    // signed 16-bit
89            int8_t*     i8;     // unsigned 8-bit, offset by 0x80
90        };
91    };
92
93
94    /* As a convenience, if a callback is supplied, a handler thread
95     * is automatically created with the appropriate priority. This thread
96     * invokes the callback when a new buffer becomes available or various conditions occur.
97     * Parameters:
98     *
99     * event:   type of event notified (see enum AudioTrack::event_type).
100     * user:    Pointer to context for use by the callback receiver.
101     * info:    Pointer to optional parameter according to event type:
102     *          - EVENT_MORE_DATA: pointer to AudioTrack::Buffer struct. The callback must not write
103     *            more bytes than indicated by 'size' field and update 'size' if fewer bytes are
104     *            written.
105     *          - EVENT_UNDERRUN: unused.
106     *          - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining.
107     *          - EVENT_MARKER: pointer to an uint32_t containing the marker position in frames.
108     *          - EVENT_NEW_POS: pointer to an uint32_t containing the new position in frames.
109     *          - EVENT_BUFFER_END: unused.
110     */
111
112    typedef void (*callback_t)(int event, void* user, void *info);
113
114    /* Returns the minimum frame count required for the successful creation of
115     * an AudioTrack object.
116     * Returned status (from utils/Errors.h) can be:
117     *  - NO_ERROR: successful operation
118     *  - NO_INIT: audio server or audio hardware not initialized
119     */
120
121     static status_t getMinFrameCount(int* frameCount,
122                                      audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT,
123                                      uint32_t sampleRate = 0);
124
125    /* Constructs an uninitialized AudioTrack. No connection with
126     * AudioFlinger takes place.
127     */
128                        AudioTrack();
129
130    /* Creates an AudioTrack object and registers it with AudioFlinger.
131     * Once created, the track needs to be started before it can be used.
132     * Unspecified values are set to the audio hardware's current
133     * values.
134     *
135     * Parameters:
136     *
137     * streamType:         Select the type of audio stream this track is attached to
138     *                     (e.g. AUDIO_STREAM_MUSIC).
139     * sampleRate:         Track sampling rate in Hz.
140     * format:             Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed
141     *                     16 bits per sample).
142     * channelMask:        Channel mask.
143     * frameCount:         Minimum size of track PCM buffer in frames. This defines the
144     *                     application's contribution to the
145     *                     latency of the track. The actual size selected by the AudioTrack could be
146     *                     larger if the requested size is not compatible with current audio HAL
147     *                     latency.  Zero means to use a default value.
148     * flags:              See comments on audio_output_flags_t in <system/audio.h>.
149     * cbf:                Callback function. If not null, this function is called periodically
150     *                     to provide new PCM data.
151     * user:               Context for use by the callback receiver.
152     * notificationFrames: The callback function is called each time notificationFrames PCM
153     *                     frames have been consumed from track input buffer.
154     * sessionId:          Specific session ID, or zero to use default.
155     * threadCanCallJava:  Whether callbacks are made from an attached thread and thus can call JNI.
156     *                     If not present in parameter list, then fixed at false.
157     */
158
159                        AudioTrack( audio_stream_type_t streamType,
160                                    uint32_t sampleRate  = 0,
161                                    audio_format_t format = AUDIO_FORMAT_DEFAULT,
162                                    audio_channel_mask_t channelMask = 0,
163                                    int frameCount       = 0,
164                                    audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
165                                    callback_t cbf       = NULL,
166                                    void* user           = NULL,
167                                    int notificationFrames = 0,
168                                    int sessionId        = 0);
169
170                        // DEPRECATED
171                        explicit AudioTrack( int streamType,
172                                    uint32_t sampleRate  = 0,
173                                    int format = AUDIO_FORMAT_DEFAULT,
174                                    int channelMask      = 0,
175                                    int frameCount       = 0,
176                                    uint32_t flags       = (uint32_t) AUDIO_OUTPUT_FLAG_NONE,
177                                    callback_t cbf       = 0,
178                                    void* user           = 0,
179                                    int notificationFrames = 0,
180                                    int sessionId        = 0);
181
182    /* Creates an audio track and registers it with AudioFlinger. With this constructor,
183     * the PCM data to be rendered by AudioTrack is passed in a shared memory buffer
184     * identified by the argument sharedBuffer. This prototype is for static buffer playback.
185     * PCM data must be present in memory before the AudioTrack is started.
186     * The write() and flush() methods are not supported in this case.
187     * It is recommended to pass a callback function to be notified of playback end by an
188     * EVENT_UNDERRUN event.
189     */
190
191                        AudioTrack( audio_stream_type_t streamType,
192                                    uint32_t sampleRate = 0,
193                                    audio_format_t format = AUDIO_FORMAT_DEFAULT,
194                                    audio_channel_mask_t channelMask = 0,
195                                    const sp<IMemory>& sharedBuffer = 0,
196                                    audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
197                                    callback_t cbf      = NULL,
198                                    void* user          = NULL,
199                                    int notificationFrames = 0,
200                                    int sessionId       = 0);
201
202    /* Terminates the AudioTrack and unregisters it from AudioFlinger.
203     * Also destroys all resources associated with the AudioTrack.
204     */
205                        ~AudioTrack();
206
207
208    /* Initialize an uninitialized AudioTrack.
209     * Returned status (from utils/Errors.h) can be:
210     *  - NO_ERROR: successful initialization
211     *  - INVALID_OPERATION: AudioTrack is already initialized
212     *  - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...)
213     *  - NO_INIT: audio server or audio hardware not initialized
214     */
215            status_t    set(audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT,
216                            uint32_t sampleRate = 0,
217                            audio_format_t format = AUDIO_FORMAT_DEFAULT,
218                            audio_channel_mask_t channelMask = 0,
219                            int frameCount      = 0,
220                            audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
221                            callback_t cbf      = NULL,
222                            void* user          = NULL,
223                            int notificationFrames = 0,
224                            const sp<IMemory>& sharedBuffer = 0,
225                            bool threadCanCallJava = false,
226                            int sessionId       = 0);
227
228
229    /* Result of constructing the AudioTrack. This must be checked
230     * before using any AudioTrack API (except for set()), because using
231     * an uninitialized AudioTrack produces undefined results.
232     * See set() method above for possible return codes.
233     */
234            status_t    initCheck() const;
235
236    /* Returns this track's estimated latency in milliseconds.
237     * This includes the latency due to AudioTrack buffer size, AudioMixer (if any)
238     * and audio hardware driver.
239     */
240            uint32_t     latency() const;
241
242    /* getters, see constructors and set() */
243
244            audio_stream_type_t streamType() const;
245            audio_format_t format() const;
246            int         channelCount() const;
247            uint32_t    frameCount() const;
248
249    /* Return channelCount * (bit depth per channel / 8).
250     * channelCount is determined from channelMask, and bit depth comes from format.
251     */
252            size_t      frameSize() const;
253
254            sp<IMemory>& sharedBuffer();
255
256
257    /* After it's created the track is not active. Call start() to
258     * make it active. If set, the callback will start being called.
259     */
260            void        start();
261
262    /* Stop a track. If set, the callback will cease being called and
263     * obtainBuffer returns STOPPED. Note that obtainBuffer() still works
264     * and will fill up buffers until the pool is exhausted.
265     */
266            void        stop();
267            bool        stopped() const;
268
269    /* Flush a stopped track. All pending buffers are discarded.
270     * This function has no effect if the track is not stopped.
271     */
272            void        flush();
273
274    /* Pause a track. If set, the callback will cease being called and
275     * obtainBuffer returns STOPPED. Note that obtainBuffer() still works
276     * and will fill up buffers until the pool is exhausted.
277     */
278            void        pause();
279
280    /* Mute or unmute this track.
281     * While muted, the callback, if set, is still called.
282     */
283            void        mute(bool);
284            bool        muted() const;
285
286    /* Set volume for this track, mostly used for games' sound effects
287     * left and right volumes. Levels must be >= 0.0 and <= 1.0.
288     * This is the older API.  New applications should use setVolume(float) when possible.
289     */
290            status_t    setVolume(float left, float right);
291
292    /* Set volume for all channels.  This is the preferred API for new applications,
293     * especially for multi-channel content.
294     */
295            status_t    setVolume(float volume);
296
297    /* Set the send level for this track. An auxiliary effect should be attached
298     * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0.
299     */
300            status_t    setAuxEffectSendLevel(float level);
301            void        getAuxEffectSendLevel(float* level) const;
302
303    /* Set sample rate for this track in Hz, mostly used for games' sound effects
304     */
305            status_t    setSampleRate(int sampleRate);
306            uint32_t    getSampleRate() const;
307
308    /* Enables looping and sets the start and end points of looping.
309     *
310     * Parameters:
311     *
312     * loopStart:   loop start expressed as the number of PCM frames played since AudioTrack start.
313     * loopEnd:     loop end expressed as the number of PCM frames played since AudioTrack start.
314     * loopCount:   number of loops to execute. Calling setLoop() with loopCount == 0 cancels any
315     *              pending or active loop. loopCount = -1 means infinite looping.
316     *
317     * For proper operation the following condition must be respected:
318     *          (loopEnd-loopStart) <= framecount()
319     */
320            status_t    setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount);
321
322    /* Sets marker position. When playback reaches the number of frames specified, a callback with
323     * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker
324     * notification callback.
325     * If the AudioTrack has been opened with no callback function associated, the operation will
326     * fail.
327     *
328     * Parameters:
329     *
330     * marker:   marker position expressed in frames.
331     *
332     * Returned status (from utils/Errors.h) can be:
333     *  - NO_ERROR: successful operation
334     *  - INVALID_OPERATION: the AudioTrack has no callback installed.
335     */
336            status_t    setMarkerPosition(uint32_t marker);
337            status_t    getMarkerPosition(uint32_t *marker) const;
338
339
340    /* Sets position update period. Every time the number of frames specified has been played,
341     * a callback with event type EVENT_NEW_POS is called.
342     * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification
343     * callback.
344     * If the AudioTrack has been opened with no callback function associated, the operation will
345     * fail.
346     *
347     * Parameters:
348     *
349     * updatePeriod:  position update notification period expressed in frames.
350     *
351     * Returned status (from utils/Errors.h) can be:
352     *  - NO_ERROR: successful operation
353     *  - INVALID_OPERATION: the AudioTrack has no callback installed.
354     */
355            status_t    setPositionUpdatePeriod(uint32_t updatePeriod);
356            status_t    getPositionUpdatePeriod(uint32_t *updatePeriod) const;
357
358    /* Sets playback head position within AudioTrack buffer. The new position is specified
359     * in number of frames.
360     * This method must be called with the AudioTrack in paused or stopped state.
361     * Note that the actual position set is <position> modulo the AudioTrack buffer size in frames.
362     * Therefore using this method makes sense only when playing a "static" audio buffer
363     * as opposed to streaming.
364     * The getPosition() method on the other hand returns the total number of frames played since
365     * playback start.
366     *
367     * Parameters:
368     *
369     * position:  New playback head position within AudioTrack buffer.
370     *
371     * Returned status (from utils/Errors.h) can be:
372     *  - NO_ERROR: successful operation
373     *  - INVALID_OPERATION: the AudioTrack is not stopped.
374     *  - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack
375     *               buffer
376     */
377            status_t    setPosition(uint32_t position);
378            status_t    getPosition(uint32_t *position);
379
380    /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids
381     * rewriting the buffer before restarting playback after a stop.
382     * This method must be called with the AudioTrack in paused or stopped state.
383     *
384     * Returned status (from utils/Errors.h) can be:
385     *  - NO_ERROR: successful operation
386     *  - INVALID_OPERATION: the AudioTrack is not stopped.
387     */
388            status_t    reload();
389
390    /* Returns a handle on the audio output used by this AudioTrack.
391     *
392     * Parameters:
393     *  none.
394     *
395     * Returned value:
396     *  handle on audio hardware output
397     */
398            audio_io_handle_t    getOutput();
399
400    /* Returns the unique session ID associated with this track.
401     *
402     * Parameters:
403     *  none.
404     *
405     * Returned value:
406     *  AudioTrack session ID.
407     */
408            int    getSessionId() const;
409
410    /* Attach track auxiliary output to specified effect. Use effectId = 0
411     * to detach track from effect.
412     *
413     * Parameters:
414     *
415     * effectId:  effectId obtained from AudioEffect::id().
416     *
417     * Returned status (from utils/Errors.h) can be:
418     *  - NO_ERROR: successful operation
419     *  - INVALID_OPERATION: the effect is not an auxiliary effect.
420     *  - BAD_VALUE: The specified effect ID is invalid
421     */
422            status_t    attachAuxEffect(int effectId);
423
424    /* Obtains a buffer of "frameCount" frames. The buffer must be
425     * filled entirely, and then released with releaseBuffer().
426     * If the track is stopped, obtainBuffer() returns
427     * STOPPED instead of NO_ERROR as long as there are buffers available,
428     * at which point NO_MORE_BUFFERS is returned.
429     * Buffers will be returned until the pool
430     * is exhausted, at which point obtainBuffer() will either block
431     * or return WOULD_BLOCK depending on the value of the "blocking"
432     * parameter.
433     *
434     * Interpretation of waitCount:
435     *  +n  limits wait time to n * WAIT_PERIOD_MS,
436     *  -1  causes an (almost) infinite wait time,
437     *   0  non-blocking.
438     */
439
440        enum {
441            NO_MORE_BUFFERS = 0x80000001,   // same name in AudioFlinger.h, ok to be different value
442            STOPPED = 1
443        };
444
445            status_t    obtainBuffer(Buffer* audioBuffer, int32_t waitCount);
446
447    /* Release a filled buffer of "frameCount" frames for AudioFlinger to process. */
448            void        releaseBuffer(Buffer* audioBuffer);
449
450    /* As a convenience we provide a write() interface to the audio buffer.
451     * This is implemented on top of obtainBuffer/releaseBuffer. For best
452     * performance use callbacks. Returns actual number of bytes written >= 0,
453     * or one of the following negative status codes:
454     *      INVALID_OPERATION   AudioTrack is configured for shared buffer mode
455     *      BAD_VALUE           size is invalid
456     *      STOPPED             AudioTrack was stopped during the write
457     *      NO_MORE_BUFFERS     when obtainBuffer() returns same
458     *      or any other error code returned by IAudioTrack::start() or restoreTrack_l().
459     */
460            ssize_t     write(const void* buffer, size_t size);
461
462    /*
463     * Dumps the state of an audio track.
464     */
465            status_t dump(int fd, const Vector<String16>& args) const;
466
467protected:
468    /* copying audio tracks is not allowed */
469                        AudioTrack(const AudioTrack& other);
470            AudioTrack& operator = (const AudioTrack& other);
471
472    /* a small internal class to handle the callback */
473    class AudioTrackThread : public Thread
474    {
475    public:
476        AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false);
477
478        // Do not call Thread::requestExitAndWait() without first calling requestExit().
479        // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough.
480        virtual void        requestExit();
481
482                void        pause();    // suspend thread from execution at next loop boundary
483                void        resume();   // allow thread to execute, if not requested to exit
484
485    private:
486        friend class AudioTrack;
487        virtual bool        threadLoop();
488        AudioTrack& mReceiver;
489        ~AudioTrackThread();
490        Mutex               mMyLock;    // Thread::mLock is private
491        Condition           mMyCond;    // Thread::mThreadExitedCondition is private
492        bool                mPaused;    // whether thread is currently paused
493    };
494
495            // body of AudioTrackThread::threadLoop()
496            bool processAudioBuffer(const sp<AudioTrackThread>& thread);
497
498            status_t createTrack_l(audio_stream_type_t streamType,
499                                 uint32_t sampleRate,
500                                 audio_format_t format,
501                                 audio_channel_mask_t channelMask,
502                                 int frameCount,
503                                 audio_output_flags_t flags,
504                                 const sp<IMemory>& sharedBuffer,
505                                 audio_io_handle_t output);
506            void flush_l();
507            status_t setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount);
508            audio_io_handle_t getOutput_l();
509            status_t restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart);
510            bool stopped_l() const { return !mActive; }
511
512    sp<IAudioTrack>         mAudioTrack;
513    sp<IMemory>             mCblkMemory;
514    sp<AudioTrackThread>    mAudioTrackThread;
515
516    float                   mVolume[2];
517    float                   mSendLevel;
518    uint32_t                mFrameCount;
519
520    audio_track_cblk_t*     mCblk;
521    audio_format_t          mFormat;
522    audio_stream_type_t     mStreamType;
523    uint8_t                 mChannelCount;
524    uint8_t                 mMuted;
525    uint8_t                 mReserved;
526    audio_channel_mask_t    mChannelMask;
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    mutable Mutex           mLock;
553    status_t                mRestoreStatus;
554    bool                    mIsTimed;
555    int                     mPreviousPriority;          // before start()
556    SchedPolicy             mPreviousSchedulingGroup;
557};
558
559class TimedAudioTrack : public AudioTrack
560{
561public:
562    TimedAudioTrack();
563
564    /* allocate a shared memory buffer that can be passed to queueTimedBuffer */
565    status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer);
566
567    /* queue a buffer obtained via allocateTimedBuffer for playback at the
568       given timestamp.  PTS units are microseconds on the media time timeline.
569       The media time transform (set with setMediaTimeTransform) set by the
570       audio producer will handle converting from media time to local time
571       (perhaps going through the common time timeline in the case of
572       synchronized multiroom audio case) */
573    status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts);
574
575    /* define a transform between media time and either common time or
576       local time */
577    enum TargetTimeline {LOCAL_TIME, COMMON_TIME};
578    status_t setMediaTimeTransform(const LinearTransform& xform,
579                                   TargetTimeline target);
580};
581
582}; // namespace android
583
584#endif // ANDROID_AUDIOTRACK_H
585