include/media/AudioTrack.h

/*
 * Copyright (C) 2007 The Android Open Source Project
 *
 * Licensed under the Apache License, Version 2.0 (the "License");
 * you may not use this file except in compliance with the License.
 * You may obtain a copy of the License at
 *
 *      http://www.apache.org/licenses/LICENSE-2.0
 *
 * Unless required by applicable law or agreed to in writing, software
 * distributed under the License is distributed on an "AS IS" BASIS,
 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
 * See the License for the specific language governing permissions and
 * limitations under the License.
 */

#ifndef ANDROID_AUDIOTRACK_H
#define ANDROID_AUDIOTRACK_H

#include <stdint.h>
#include <sys/types.h>

#include <media/IAudioFlinger.h>
#include <media/IAudioTrack.h>
#include <media/AudioSystem.h>

#include <utils/RefBase.h>
#include <utils/Errors.h>
#include <binder/IInterface.h>
#include <binder/IMemory.h>
#include <cutils/sched_policy.h>
#include <utils/threads.h>

namespace android {

// ----------------------------------------------------------------------------

class audio_track_cblk_t;

// ----------------------------------------------------------------------------

class AudioTrack : virtual public RefBase
{
public:
    enum channel_index {
        MONO   = 0,
        LEFT   = 0,
        RIGHT  = 1
    };

    /* Events used by AudioTrack callback function (audio_track_cblk_t).
     */
    enum event_type {
        EVENT_MORE_DATA = 0,        // Request to write more data to PCM buffer.
        EVENT_UNDERRUN = 1,         // PCM buffer underrun occured.
        EVENT_LOOP_END = 2,         // Sample loop end was reached; playback restarted from
                                    // loop start if loop count was not 0.
        EVENT_MARKER = 3,           // Playback head is at the specified marker position
                                    // (See setMarkerPosition()).
        EVENT_NEW_POS = 4,          // Playback head is at a new position
                                    // (See setPositionUpdatePeriod()).
        EVENT_BUFFER_END = 5        // Playback head is at the end of the buffer.
    };

    /* Client should declare Buffer on the stack and pass address to obtainBuffer()
     * and releaseBuffer().  See also callback_t for EVENT_MORE_DATA.
     */

    class Buffer
    {
    public:
        enum {
            MUTE    = 0x00000001
        };
        uint32_t    flags;        // 0 or MUTE
        audio_format_t format; // but AUDIO_FORMAT_PCM_8_BIT -> AUDIO_FORMAT_PCM_16_BIT
        // accessed directly by WebKit ANP callback
        int         channelCount; // will be removed in the future, do not use

        size_t      frameCount;   // number of sample frames corresponding to size;
                                  // on input it is the number of frames desired,
                                  // on output is the number of frames actually filled

        size_t      size;         // input/output in byte units
        union {
            void*       raw;
            short*      i16;    // signed 16-bit
            int8_t*     i8;     // unsigned 8-bit, offset by 0x80
        };
    };


    /* As a convenience, if a callback is supplied, a handler thread
     * is automatically created with the appropriate priority. This thread
     * invokes the callback when a new buffer becomes available or various conditions occur.
     * Parameters:
     *
     * event:   type of event notified (see enum AudioTrack::event_type).
     * user:    Pointer to context for use by the callback receiver.
     * info:    Pointer to optional parameter according to event type:
     *          - EVENT_MORE_DATA: pointer to AudioTrack::Buffer struct. The callback must not write
     *            more bytes than indicated by 'size' field and update 'size' if fewer bytes are
     *            written.
     *          - EVENT_UNDERRUN: unused.
     *          - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining.
     *          - EVENT_MARKER: pointer to an uint32_t containing the marker position in frames.
     *          - EVENT_NEW_POS: pointer to an uint32_t containing the new position in frames.
     *          - EVENT_BUFFER_END: unused.
     */

    typedef void (*callback_t)(int event, void* user, void *info);

    /* Returns the minimum frame count required for the successful creation of
     * an AudioTrack object.
     * Returned status (from utils/Errors.h) can be:
     *  - NO_ERROR: successful operation
     *  - NO_INIT: audio server or audio hardware not initialized
     */

     static status_t getMinFrameCount(int* frameCount,
                                      audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT,
                                      uint32_t sampleRate = 0);

    /* Constructs an uninitialized AudioTrack. No connection with
     * AudioFlinger takes place.
     */
                        AudioTrack();

    /* Creates an audio track and registers it with AudioFlinger.
     * Once created, the track needs to be started before it can be used.
     * Unspecified values are set to the audio hardware's current
     * values.
     *
     * Parameters:
     *
     * streamType:         Select the type of audio stream this track is attached to
     *                     (e.g. AUDIO_STREAM_MUSIC).
     * sampleRate:         Track sampling rate in Hz.
     * format:             Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed
     *                     16 bits per sample).
     * channelMask:        Channel mask.
     * frameCount:         Minimum size of track PCM buffer in frames. This defines the
     *                     latency of the track. The actual size selected by the AudioTrack could be
     *                     larger if the requested size is not compatible with current audio HAL
     *                     latency.  Zero means to use a default value.
     * flags:              See comments on audio_output_flags_t in <system/audio.h>.
     * cbf:                Callback function. If not null, this function is called periodically
     *                     to request new PCM data.
     * user:               Context for use by the callback receiver.
     * notificationFrames: The callback function is called each time notificationFrames PCM
     *                     frames have been consumed from track input buffer.
     * sessionId:          Specific session ID, or zero to use default.
     * threadCanCallJava:  Whether callbacks are made from an attached thread and thus can call JNI.
     *                     If not present in parameter list, then fixed at false.
     */

                        AudioTrack( audio_stream_type_t streamType,
                                    uint32_t sampleRate  = 0,
                                    audio_format_t format = AUDIO_FORMAT_DEFAULT,
                                    audio_channel_mask_t channelMask = 0,
                                    int frameCount       = 0,
                                    audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
                                    callback_t cbf       = NULL,
                                    void* user           = NULL,
                                    int notificationFrames = 0,
                                    int sessionId        = 0);

                        // DEPRECATED
                        explicit AudioTrack( int streamType,
                                    uint32_t sampleRate  = 0,
                                    int format = AUDIO_FORMAT_DEFAULT,
                                    int channelMask      = 0,
                                    int frameCount       = 0,
                                    uint32_t flags       = (uint32_t) AUDIO_OUTPUT_FLAG_NONE,
                                    callback_t cbf       = 0,
                                    void* user           = 0,
                                    int notificationFrames = 0,
                                    int sessionId        = 0);

    /* Creates an audio track and registers it with AudioFlinger. With this constructor,
     * the PCM data to be rendered by AudioTrack is passed in a shared memory buffer
     * identified by the argument sharedBuffer. This prototype is for static buffer playback.
     * PCM data must be present in memory before the AudioTrack is started.
     * The write() and flush() methods are not supported in this case.
     * It is recommended to pass a callback function to be notified of playback end by an
     * EVENT_UNDERRUN event.
     */

                        AudioTrack( audio_stream_type_t streamType,
                                    uint32_t sampleRate = 0,
                                    audio_format_t format = AUDIO_FORMAT_DEFAULT,
                                    audio_channel_mask_t channelMask = 0,
                                    const sp<IMemory>& sharedBuffer = 0,
                                    audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
                                    callback_t cbf      = NULL,
                                    void* user          = NULL,
                                    int notificationFrames = 0,
                                    int sessionId       = 0);

    /* Terminates the AudioTrack and unregisters it from AudioFlinger.
     * Also destroys all resources associated with the AudioTrack.
     */
                        ~AudioTrack();


    /* Initialize an uninitialized AudioTrack.
     * Returned status (from utils/Errors.h) can be:
     *  - NO_ERROR: successful initialization
     *  - INVALID_OPERATION: AudioTrack is already initialized
     *  - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...)
     *  - NO_INIT: audio server or audio hardware not initialized
     * */
            status_t    set(audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT,
                            uint32_t sampleRate = 0,
                            audio_format_t format = AUDIO_FORMAT_DEFAULT,
                            audio_channel_mask_t channelMask = 0,
                            int frameCount      = 0,
                            audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
                            callback_t cbf      = NULL,
                            void* user          = NULL,
                            int notificationFrames = 0,
                            const sp<IMemory>& sharedBuffer = 0,
                            bool threadCanCallJava = false,
                            int sessionId       = 0);


    /* Result of constructing the AudioTrack. This must be checked
     * before using any AudioTrack API (except for set()), because using
     * an uninitialized AudioTrack produces undefined results.
     * See set() method above for possible return codes.
     */
            status_t    initCheck() const;

    /* Returns this track's estimated latency in milliseconds.
     * This includes the latency due to AudioTrack buffer size, AudioMixer (if any)
     * and audio hardware driver.
     */
            uint32_t     latency() const;

    /* getters, see constructors and set() */

            audio_stream_type_t streamType() const;
            audio_format_t format() const;
            int         channelCount() const;
            uint32_t    frameCount() const;

    /* Return channelCount * (bit depth per channel / 8).
     * channelCount is determined from channelMask, and bit depth comes from format.
     */
            size_t      frameSize() const;

            sp<IMemory>& sharedBuffer();


    /* After it's created the track is not active. Call start() to
     * make it active. If set, the callback will start being called.
     */
            void        start();

    /* Stop a track. If set, the callback will cease being called and
     * obtainBuffer returns STOPPED. Note that obtainBuffer() still works
     * and will fill up buffers until the pool is exhausted.
     */
            void        stop();
            bool        stopped() const;

    /* Flush a stopped track. All pending buffers are discarded.
     * This function has no effect if the track is not stopped.
     */
            void        flush();

    /* Pause a track. If set, the callback will cease being called and
     * obtainBuffer returns STOPPED. Note that obtainBuffer() still works
     * and will fill up buffers until the pool is exhausted.
     */
            void        pause();

    /* Mute or unmute this track.
     * While muted, the callback, if set, is still called.
     */
            void        mute(bool);
            bool        muted() const;

    /* Set volume for this track, mostly used for games' sound effects
     * left and right volumes. Levels must be >= 0.0 and <= 1.0.
     */
            status_t    setVolume(float left, float right);
            void        getVolume(float* left, float* right) const;

    /* Set the send level for this track. An auxiliary effect should be attached
     * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0.
     */
            status_t    setAuxEffectSendLevel(float level);
            void        getAuxEffectSendLevel(float* level) const;

    /* Set sample rate for this track, mostly used for games' sound effects
     */
            status_t    setSampleRate(int sampleRate);
            uint32_t    getSampleRate() const;

    /* Enables looping and sets the start and end points of looping.
     *
     * Parameters:
     *
     * loopStart:   loop start expressed as the number of PCM frames played since AudioTrack start.
     * loopEnd:     loop end expressed as the number of PCM frames played since AudioTrack start.
     * loopCount:   number of loops to execute. Calling setLoop() with loopCount == 0 cancels any
     *              pending or active loop. loopCount = -1 means infinite looping.
     *
     * For proper operation the following condition must be respected:
     *          (loopEnd-loopStart) <= framecount()
     */
            status_t    setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount);

    /* Sets marker position. When playback reaches the number of frames specified, a callback with
     * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker
     * notification callback.
     * If the AudioTrack has been opened with no callback function associated, the operation will
     * fail.
     *
     * Parameters:
     *
     * marker:   marker position expressed in frames.
     *
     * Returned status (from utils/Errors.h) can be:
     *  - NO_ERROR: successful operation
     *  - INVALID_OPERATION: the AudioTrack has no callback installed.
     */
            status_t    setMarkerPosition(uint32_t marker);
            status_t    getMarkerPosition(uint32_t *marker) const;


    /* Sets position update period. Every time the number of frames specified has been played,
     * a callback with event type EVENT_NEW_POS is called.
     * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification
     * callback.
     * If the AudioTrack has been opened with no callback function associated, the operation will
     * fail.
     *
     * Parameters:
     *
     * updatePeriod:  position update notification period expressed in frames.
     *
     * Returned status (from utils/Errors.h) can be:
     *  - NO_ERROR: successful operation
     *  - INVALID_OPERATION: the AudioTrack has no callback installed.
     */
            status_t    setPositionUpdatePeriod(uint32_t updatePeriod);
            status_t    getPositionUpdatePeriod(uint32_t *updatePeriod) const;

    /* Sets playback head position within AudioTrack buffer. The new position is specified
     * in number of frames.
     * This method must be called with the AudioTrack in paused or stopped state.
     * Note that the actual position set is <position> modulo the AudioTrack buffer size in frames.
     * Therefore using this method makes sense only when playing a "static" audio buffer
     * as opposed to streaming.
     * The getPosition() method on the other hand returns the total number of frames played since
     * playback start.
     *
     * Parameters:
     *
     * position:  New playback head position within AudioTrack buffer.
     *
     * Returned status (from utils/Errors.h) can be:
     *  - NO_ERROR: successful operation
     *  - INVALID_OPERATION: the AudioTrack is not stopped.
     *  - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack
     *               buffer
     */
            status_t    setPosition(uint32_t position);
            status_t    getPosition(uint32_t *position);

    /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids
     * rewriting the buffer before restarting playback after a stop.
     * This method must be called with the AudioTrack in paused or stopped state.
     *
     * Returned status (from utils/Errors.h) can be:
     *  - NO_ERROR: successful operation
     *  - INVALID_OPERATION: the AudioTrack is not stopped.
     */
            status_t    reload();

    /* Returns a handle on the audio output used by this AudioTrack.
     *
     * Parameters:
     *  none.
     *
     * Returned value:
     *  handle on audio hardware output
     */
            audio_io_handle_t    getOutput();

    /* Returns the unique session ID associated with this track.
     *
     * Parameters:
     *  none.
     *
     * Returned value:
     *  AudioTrack session ID.
     */
            int    getSessionId() const;

    /* Attach track auxiliary output to specified effect. Use effectId = 0
     * to detach track from effect.
     *
     * Parameters:
     *
     * effectId:  effectId obtained from AudioEffect::id().
     *
     * Returned status (from utils/Errors.h) can be:
     *  - NO_ERROR: successful operation
     *  - INVALID_OPERATION: the effect is not an auxiliary effect.
     *  - BAD_VALUE: The specified effect ID is invalid
     */
            status_t    attachAuxEffect(int effectId);

    /* Obtains a buffer of "frameCount" frames. The buffer must be
     * filled entirely, and then released with releaseBuffer().
     * If the track is stopped, obtainBuffer() returns
     * STOPPED instead of NO_ERROR as long as there are buffers available,
     * at which point NO_MORE_BUFFERS is returned.
     * Buffers will be returned until the pool (buffercount())
     * is exhausted, at which point obtainBuffer() will either block
     * or return WOULD_BLOCK depending on the value of the "blocking"
     * parameter.
     *
     * Interpretation of waitCount:
     *  +n  limits wait time to n * WAIT_PERIOD_MS,
     *  -1  causes an (almost) infinite wait time,
     *   0  non-blocking.
     */

        enum {
            NO_MORE_BUFFERS = 0x80000001,   // same name in AudioFlinger.h, ok to be different value
            STOPPED = 1
        };

            status_t    obtainBuffer(Buffer* audioBuffer, int32_t waitCount);

    /* Release a filled buffer of "frameCount" frames for AudioFlinger to process. */
            void        releaseBuffer(Buffer* audioBuffer);

    /* As a convenience we provide a write() interface to the audio buffer.
     * This is implemented on top of obtainBuffer/releaseBuffer. For best
     * performance use callbacks. Returns actual number of bytes written >= 0,
     * or one of the following negative status codes:
     *      INVALID_OPERATION   AudioTrack is configured for shared buffer mode
     *      BAD_VALUE           size is invalid
     *      STOPPED             AudioTrack was stopped during the write
     *      NO_MORE_BUFFERS     when obtainBuffer() returns same
     *      or any other error code returned by IAudioTrack::start() or restoreTrack_l().
     */
            ssize_t     write(const void* buffer, size_t size);

    /*
     * Dumps the state of an audio track.
     */
            status_t dump(int fd, const Vector<String16>& args) const;

protected:
    /* copying audio tracks is not allowed */
                        AudioTrack(const AudioTrack& other);
            AudioTrack& operator = (const AudioTrack& other);

    /* a small internal class to handle the callback */
    class AudioTrackThread : public Thread
    {
    public:
        AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false);

        // Do not call Thread::requestExitAndWait() without first calling requestExit().
        // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough.
        virtual void        requestExit();

                void        pause();    // suspend thread from execution at next loop boundary
                void        resume();   // allow thread to execute, if not requested to exit

    private:
        friend class AudioTrack;
        virtual bool        threadLoop();
        AudioTrack& mReceiver;
        ~AudioTrackThread();
        Mutex               mMyLock;    // Thread::mLock is private
        Condition           mMyCond;    // Thread::mThreadExitedCondition is private
        bool                mPaused;    // whether thread is currently paused
    };

            // body of AudioTrackThread::threadLoop()
            bool processAudioBuffer(const sp<AudioTrackThread>& thread);

            status_t createTrack_l(audio_stream_type_t streamType,
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
                                 int frameCount,
                                 audio_output_flags_t flags,
                                 const sp<IMemory>& sharedBuffer,
                                 audio_io_handle_t output);
            void flush_l();
            status_t setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount);
            audio_io_handle_t getOutput_l();
            status_t restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart);
            bool stopped_l() const { return !mActive; }

    sp<IAudioTrack>         mAudioTrack;
    sp<IMemory>             mCblkMemory;
    sp<AudioTrackThread>    mAudioTrackThread;

    float                   mVolume[2];
    float                   mSendLevel;
    uint32_t                mFrameCount;

    audio_track_cblk_t*     mCblk;
    audio_format_t          mFormat;
    audio_stream_type_t     mStreamType;
    uint8_t                 mChannelCount;
    uint8_t                 mMuted;
    uint8_t                 mReserved;
    audio_channel_mask_t    mChannelMask;
    status_t                mStatus;
    uint32_t                mLatency;

    bool                    mActive;                // protected by mLock

    callback_t              mCbf;                   // callback handler for events, or NULL
    void*                   mUserData;
    uint32_t                mNotificationFramesReq; // requested number of frames between each
                                                    // notification callback
    uint32_t                mNotificationFramesAct; // actual number of frames between each
                                                    // notification callback
    sp<IMemory>             mSharedBuffer;
    int                     mLoopCount;
    uint32_t                mRemainingFrames;
    uint32_t                mMarkerPosition;
    bool                    mMarkerReached;
    uint32_t                mNewPosition;
    uint32_t                mUpdatePeriod;
    bool                    mFlushed; // FIXME will be made obsolete by making flush() synchronous
    audio_output_flags_t    mFlags;
    int                     mSessionId;
    int                     mAuxEffectId;
    mutable Mutex           mLock;
    status_t                mRestoreStatus;
    bool                    mIsTimed;
    int                     mPreviousPriority;          // before start()
    SchedPolicy             mPreviousSchedulingGroup;
};

class TimedAudioTrack : public AudioTrack
{
public:
    TimedAudioTrack();

    /* allocate a shared memory buffer that can be passed to queueTimedBuffer */
    status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer);

    /* queue a buffer obtained via allocateTimedBuffer for playback at the
       given timestamp.  PTS units are microseconds on the media time timeline.
       The media time transform (set with setMediaTimeTransform) set by the
       audio producer will handle converting from media time to local time
       (perhaps going through the common time timeline in the case of
       synchronized multiroom audio case) */
    status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts);

    /* define a transform between media time and either common time or
       local time */
    enum TargetTimeline {LOCAL_TIME, COMMON_TIME};
    status_t setMediaTimeTransform(const LinearTransform& xform,
                                   TargetTimeline target);
};

}; // namespace android

#endif // ANDROID_AUDIOTRACK_H