/*
 * 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_MEDIAPLAYERINTERFACE_H
#define ANDROID_MEDIAPLAYERINTERFACE_H

#ifdef __cplusplus

#include <sys/types.h>
#include <utils/Errors.h>
#include <utils/KeyedVector.h>
#include <utils/String8.h>
#include <utils/RefBase.h>

#include <media/mediaplayer.h>
#include <media/AudioResamplerPublic.h>
#include <media/AudioSystem.h>
#include <media/AudioTimestamp.h>
#include <media/AVSyncSettings.h>
#include <media/BufferingSettings.h>
#include <media/Metadata.h>

// Fwd decl to make sure everyone agrees that the scope of struct sockaddr_in is
// global, and not in android::
struct sockaddr_in;

namespace android {

class DataSource;
class Parcel;
class Surface;
class IGraphicBufferProducer;

template<typename T> class SortedVector;

enum player_type {
    STAGEFRIGHT_PLAYER = 3,
    NU_PLAYER = 4,
    // Test players are available only in the 'test' and 'eng' builds.
    // The shared library with the test player is passed passed as an
    // argument to the 'test:' url in the setDataSource call.
    TEST_PLAYER = 5,
};


#define DEFAULT_AUDIOSINK_BUFFERCOUNT 4
#define DEFAULT_AUDIOSINK_BUFFERSIZE 1200
#define DEFAULT_AUDIOSINK_SAMPLERATE 44100

// when the channel mask isn't known, use the channel count to derive a mask in AudioSink::open()
#define CHANNEL_MASK_USE_CHANNEL_ORDER 0

// duration below which we do not allow deep audio buffering
#define AUDIO_SINK_MIN_DEEP_BUFFER_DURATION_US 5000000

// abstract base class - use MediaPlayerInterface
class MediaPlayerBase : public RefBase
{
public:
    // callback mechanism for passing messages to MediaPlayer object
    class Listener : public RefBase {
    public:
        virtual void notify(int msg, int ext1, int ext2, const Parcel *obj) = 0;
        virtual ~Listener() {}
    };

    // AudioSink: abstraction layer for audio output
    class AudioSink : public RefBase {
    public:
        enum cb_event_t {
            CB_EVENT_FILL_BUFFER,   // Request to write more data to buffer.
            CB_EVENT_STREAM_END,    // Sent after all the buffers queued in AF and HW are played
                                    // back (after stop is called)
            CB_EVENT_TEAR_DOWN      // The AudioTrack was invalidated due to use case change:
                                    // Need to re-evaluate offloading options
        };

        // Callback returns the number of bytes actually written to the buffer.
        typedef size_t (*AudioCallback)(
                AudioSink *audioSink, void *buffer, size_t size, void *cookie,
                        cb_event_t event);

        virtual             ~AudioSink() {}
        virtual bool        ready() const = 0; // audio output is open and ready
        virtual ssize_t     bufferSize() const = 0;
        virtual ssize_t     frameCount() const = 0;
        virtual ssize_t     channelCount() const = 0;
        virtual ssize_t     frameSize() const = 0;
        virtual uint32_t    latency() const = 0;
        virtual float       msecsPerFrame() const = 0;
        virtual status_t    getPosition(uint32_t *position) const = 0;
        virtual status_t    getTimestamp(AudioTimestamp &ts) const = 0;
        virtual int64_t     getPlayedOutDurationUs(int64_t nowUs) const = 0;
        virtual status_t    getFramesWritten(uint32_t *frameswritten) const = 0;
        virtual audio_session_t getSessionId() const = 0;
        virtual audio_stream_type_t getAudioStreamType() const = 0;
        virtual uint32_t    getSampleRate() const = 0;
        virtual int64_t     getBufferDurationInUs() const = 0;

        // If no callback is specified, use the "write" API below to submit
        // audio data.
        virtual status_t    open(
                uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
                audio_format_t format=AUDIO_FORMAT_PCM_16_BIT,
                int bufferCount=DEFAULT_AUDIOSINK_BUFFERCOUNT,
                AudioCallback cb = NULL,
                void *cookie = NULL,
                audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
                const audio_offload_info_t *offloadInfo = NULL,
                bool doNotReconnect = false,
                uint32_t suggestedFrameCount = 0) = 0;

        virtual status_t    start() = 0;

        /* Input parameter |size| is in byte units stored in |buffer|.
         * Data is copied over and actual number of bytes written (>= 0)
         * is returned, or no data is copied and a negative status code
         * is returned (even when |blocking| is true).
         * When |blocking| is false, AudioSink will immediately return after
         * part of or full |buffer| is copied over.
         * When |blocking| is true, AudioSink will wait to copy the entire
         * buffer, unless an error occurs or the copy operation is
         * prematurely stopped.
         */
        virtual ssize_t     write(const void* buffer, size_t size, bool blocking = true) = 0;

        virtual void        stop() = 0;
        virtual void        flush() = 0;
        virtual void        pause() = 0;
        virtual void        close() = 0;

        virtual status_t    setPlaybackRate(const AudioPlaybackRate& rate) = 0;
        virtual status_t    getPlaybackRate(AudioPlaybackRate* rate /* nonnull */) = 0;
        virtual bool        needsTrailingPadding() { return true; }

        virtual status_t    setParameters(const String8& /* keyValuePairs */) { return NO_ERROR; }
        virtual String8     getParameters(const String8& /* keys */) { return String8::empty(); }

        virtual VolumeShaper::Status applyVolumeShaper(
                                    const sp<VolumeShaper::Configuration>& configuration,
                                    const sp<VolumeShaper::Operation>& operation);
        virtual sp<VolumeShaper::State> getVolumeShaperState(int id);
    };

                        MediaPlayerBase() {}
    virtual             ~MediaPlayerBase() {}
    virtual status_t    initCheck() = 0;
    virtual bool        hardwareOutput() = 0;

    virtual status_t    setUID(uid_t /* uid */) {
        return INVALID_OPERATION;
    }

    virtual status_t    setDataSource(
            const sp<IMediaHTTPService> &httpService,
            const char *url,
            const KeyedVector<String8, String8> *headers = NULL) = 0;

    virtual status_t    setDataSource(int fd, int64_t offset, int64_t length) = 0;

    virtual status_t    setDataSource(const sp<IStreamSource>& /* source */) {
        return INVALID_OPERATION;
    }

    virtual status_t    setDataSource(const sp<DataSource>& /* source */) {
        return INVALID_OPERATION;
    }

    // pass the buffered IGraphicBufferProducer to the media player service
    virtual status_t    setVideoSurfaceTexture(
                                const sp<IGraphicBufferProducer>& bufferProducer) = 0;

    virtual status_t    getDefaultBufferingSettings(
                                BufferingSettings* buffering /* nonnull */) {
        *buffering = BufferingSettings();
        return OK;
    }
    virtual status_t    setBufferingSettings(const BufferingSettings& /* buffering */) {
        return OK;
    }

    virtual status_t    prepare() = 0;
    virtual status_t    prepareAsync() = 0;
    virtual status_t    start() = 0;
    virtual status_t    stop() = 0;
    virtual status_t    pause() = 0;
    virtual bool        isPlaying() = 0;
    virtual status_t    setPlaybackSettings(const AudioPlaybackRate& rate) {
        // by default, players only support setting rate to the default
        if (!isAudioPlaybackRateEqual(rate, AUDIO_PLAYBACK_RATE_DEFAULT)) {
            return BAD_VALUE;
        }
        return OK;
    }
    virtual status_t    getPlaybackSettings(AudioPlaybackRate* rate /* nonnull */) {
        *rate = AUDIO_PLAYBACK_RATE_DEFAULT;
        return OK;
    }
    virtual status_t    setSyncSettings(const AVSyncSettings& sync, float /* videoFps */) {
        // By default, players only support setting sync source to default; all other sync
        // settings are ignored. There is no requirement for getters to return set values.
        if (sync.mSource != AVSYNC_SOURCE_DEFAULT) {
            return BAD_VALUE;
        }
        return OK;
    }
    virtual status_t    getSyncSettings(
                                AVSyncSettings* sync /* nonnull */, float* videoFps /* nonnull */) {
        *sync = AVSyncSettings();
        *videoFps = -1.f;
        return OK;
    }
    virtual status_t    seekTo(
            int msec, MediaPlayerSeekMode mode = MediaPlayerSeekMode::SEEK_PREVIOUS_SYNC) = 0;
    virtual status_t    getCurrentPosition(int *msec) = 0;
    virtual status_t    getDuration(int *msec) = 0;
    virtual status_t    reset() = 0;
    virtual status_t    setLooping(int loop) = 0;
    virtual player_type playerType() = 0;
    virtual status_t    setParameter(int key, const Parcel &request) = 0;
    virtual status_t    getParameter(int key, Parcel *reply) = 0;

    // default no-op implementation of optional extensions
    virtual status_t setRetransmitEndpoint(const struct sockaddr_in* /* endpoint */) {
        return INVALID_OPERATION;
    }
    virtual status_t getRetransmitEndpoint(struct sockaddr_in* /* endpoint */) {
        return INVALID_OPERATION;
    }
    virtual status_t setNextPlayer(const sp<MediaPlayerBase>& /* next */) {
        return OK;
    }

    // Invoke a generic method on the player by using opaque parcels
    // for the request and reply.
    //
    // @param request Parcel that is positioned at the start of the
    //                data sent by the java layer.
    // @param[out] reply Parcel to hold the reply data. Cannot be null.
    // @return OK if the call was successful.
    virtual status_t    invoke(const Parcel& request, Parcel *reply) = 0;

    // The Client in the MetadataPlayerService calls this method on
    // the native player to retrieve all or a subset of metadata.
    //
    // @param ids SortedList of metadata ID to be fetch. If empty, all
    //            the known metadata should be returned.
    // @param[inout] records Parcel where the player appends its metadata.
    // @return OK if the call was successful.
    virtual status_t    getMetadata(const media::Metadata::Filter& /* ids */,
                                    Parcel* /* records */) {
        return INVALID_OPERATION;
    };

    void        setNotifyCallback(
            const sp<Listener> &listener) {
        Mutex::Autolock autoLock(mNotifyLock);
        mListener = listener;
    }

    void        sendEvent(int msg, int ext1=0, int ext2=0,
                          const Parcel *obj=NULL) {
        sp<Listener> listener;
        {
            Mutex::Autolock autoLock(mNotifyLock);
            listener = mListener;
        }

        if (listener != NULL) {
            listener->notify(msg, ext1, ext2, obj);
        }
    }

    virtual status_t dump(int /* fd */, const Vector<String16>& /* args */) const {
        return INVALID_OPERATION;
    }

    // Modular DRM
    virtual status_t prepareDrm(const uint8_t /* uuid */[16], const Vector<uint8_t>& /* drmSessionId */) {
        return INVALID_OPERATION;
    }
    virtual status_t releaseDrm() {
        return INVALID_OPERATION;
    }

private:
    friend class MediaPlayerService;

    Mutex               mNotifyLock;
    sp<Listener>        mListener;
};

// Implement this class for media players that use the AudioFlinger software mixer
class MediaPlayerInterface : public MediaPlayerBase
{
public:
    virtual             ~MediaPlayerInterface() { }
    virtual bool        hardwareOutput() { return false; }
    virtual void        setAudioSink(const sp<AudioSink>& audioSink) { mAudioSink = audioSink; }
protected:
    sp<AudioSink>       mAudioSink;
};

// Implement this class for media players that output audio directly to hardware
class MediaPlayerHWInterface : public MediaPlayerBase
{
public:
    virtual             ~MediaPlayerHWInterface() {}
    virtual bool        hardwareOutput() { return true; }
    virtual status_t    setVolume(float leftVolume, float rightVolume) = 0;
    virtual status_t    setAudioStreamType(audio_stream_type_t streamType) = 0;
};

}; // namespace android

#endif // __cplusplus


#endif // ANDROID_MEDIAPLAYERINTERFACE_H
