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/*
* 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/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 {
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,
DASH_PLAYER = 6,
};
#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
// callback mechanism for passing messages to MediaPlayer object
typedef void (*notify_callback_f)(void* cookie,
int msg, int ext1, int ext2, const Parcel *obj);
// abstract base class - use MediaPlayerInterface
class MediaPlayerBase : public RefBase
{
public:
// 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 status_t getFramesWritten(uint32_t *frameswritten) const = 0;
virtual int getSessionId() const = 0;
virtual audio_stream_type_t getAudioStreamType() const = 0;
virtual uint32_t getSampleRate() 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(); }
};
MediaPlayerBase() : mCookie(0), mNotify(0) {}
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 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) = 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(
void* cookie, notify_callback_f notifyFunc) {
Mutex::Autolock autoLock(mNotifyLock);
mCookie = cookie; mNotify = notifyFunc;
}
void sendEvent(int msg, int ext1=0, int ext2=0,
const Parcel *obj=NULL) {
notify_callback_f notifyCB;
void* cookie;
{
Mutex::Autolock autoLock(mNotifyLock);
notifyCB = mNotify;
cookie = mCookie;
}
if (notifyCB) notifyCB(cookie, msg, ext1, ext2, obj);
}
virtual status_t dump(int /* fd */, const Vector<String16>& /* args */) const {
return INVALID_OPERATION;
}
virtual status_t suspend() {
return INVALID_OPERATION;
}
virtual status_t resume() {
return INVALID_OPERATION;
}
private:
friend class MediaPlayerService;
Mutex mNotifyLock;
void* mCookie;
notify_callback_f mNotify;
};
// 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
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