/*
**
** Copyright 2008, 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_MEDIAPLAYERSERVICE_H
#define ANDROID_MEDIAPLAYERSERVICE_H
#include <arpa/inet.h>
#include <utils/threads.h>
#include <utils/Errors.h>
#include <utils/KeyedVector.h>
#include <utils/String8.h>
#include <utils/Vector.h>
#include <media/MediaPlayerInterface.h>
#include <media/Metadata.h>
#include <media/stagefright/foundation/ABase.h>
#include <system/audio.h>
namespace android {
class AudioTrack;
class IMediaRecorder;
class IMediaMetadataRetriever;
class IOMX;
class IRemoteDisplay;
class IRemoteDisplayClient;
class MediaRecorderClient;
#define CALLBACK_ANTAGONIZER 0
#if CALLBACK_ANTAGONIZER
class Antagonizer {
public:
Antagonizer(notify_callback_f cb, void* client);
void start() { mActive = true; }
void stop() { mActive = false; }
void kill();
private:
static const int interval;
Antagonizer();
static int callbackThread(void* cookie);
Mutex mLock;
Condition mCondition;
bool mExit;
bool mActive;
void* mClient;
notify_callback_f mCb;
};
#endif
class MediaPlayerService : public BnMediaPlayerService
{
class Client;
class AudioOutput : public MediaPlayerBase::AudioSink
{
class CallbackData;
public:
AudioOutput(int sessionId, int uid);
virtual ~AudioOutput();
virtual bool ready() const { return mTrack != 0; }
virtual bool realtime() const { return true; }
virtual ssize_t bufferSize() const;
virtual ssize_t frameCount() const;
virtual ssize_t channelCount() const;
virtual ssize_t frameSize() const;
virtual uint32_t latency() const;
virtual float msecsPerFrame() const;
virtual status_t getPosition(uint32_t *position) const;
virtual status_t getFramesWritten(uint32_t *frameswritten) const;
virtual int getSessionId() const;
virtual uint32_t getSampleRate() const;
virtual status_t open(
uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
audio_format_t format, int bufferCount,
AudioCallback cb, void *cookie,
audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
const audio_offload_info_t *offloadInfo = NULL);
virtual status_t start();
virtual ssize_t write(const void* buffer, size_t size);
virtual void stop();
virtual void flush();
virtual void pause();
virtual void close();
void setAudioStreamType(audio_stream_type_t streamType) {
mStreamType = streamType; }
virtual audio_stream_type_t getAudioStreamType() const { return mStreamType; }
void setVolume(float left, float right);
virtual status_t setPlaybackRatePermille(int32_t ratePermille);
status_t setAuxEffectSendLevel(float level);
status_t attachAuxEffect(int effectId);
virtual status_t dump(int fd, const Vector<String16>& args) const;
static bool isOnEmulator();
static int getMinBufferCount();
void setNextOutput(const sp<AudioOutput>& nextOutput);
void switchToNextOutput();
virtual bool needsTrailingPadding() { return mNextOutput == NULL; }
virtual status_t setParameters(const String8& keyValuePairs);
virtual String8 getParameters(const String8& keys);
private:
static void setMinBufferCount();
static void CallbackWrapper(
int event, void *me, void *info);
void deleteRecycledTrack();
sp<AudioTrack> mTrack;
sp<AudioTrack> mRecycledTrack;
sp<AudioOutput> mNextOutput;
AudioCallback mCallback;
void * mCallbackCookie;
CallbackData * mCallbackData;
uint64_t mBytesWritten;
audio_stream_type_t mStreamType;
float mLeftVolume;
float mRightVolume;
int32_t mPlaybackRatePermille;
uint32_t mSampleRateHz; // sample rate of the content, as set in open()
float mMsecsPerFrame;
int mSessionId;
int mUid;
float mSendLevel;
int mAuxEffectId;
static bool mIsOnEmulator;
static int mMinBufferCount; // 12 for emulator; otherwise 4
audio_output_flags_t mFlags;
// CallbackData is what is passed to the AudioTrack as the "user" data.
// We need to be able to target this to a different Output on the fly,
// so we can't use the Output itself for this.
class CallbackData {
public:
CallbackData(AudioOutput *cookie) {
mData = cookie;
mSwitching = false;
}
AudioOutput * getOutput() { return mData;}
void setOutput(AudioOutput* newcookie) { mData = newcookie; }
// lock/unlock are used by the callback before accessing the payload of this object
void lock() { mLock.lock(); }
void unlock() { mLock.unlock(); }
// beginTrackSwitch/endTrackSwitch are used when this object is being handed over
// to the next sink.
void beginTrackSwitch() { mLock.lock(); mSwitching = true; }
void endTrackSwitch() {
if (mSwitching) {
mLock.unlock();
}
mSwitching = false;
}
private:
AudioOutput * mData;
mutable Mutex mLock;
bool mSwitching;
DISALLOW_EVIL_CONSTRUCTORS(CallbackData);
};
}; // AudioOutput
class AudioCache : public MediaPlayerBase::AudioSink
{
public:
AudioCache(const sp<IMemoryHeap>& heap);
virtual ~AudioCache() {}
virtual bool ready() const { return (mChannelCount > 0) && (mHeap->getHeapID() > 0); }
virtual bool realtime() const { return false; }
virtual ssize_t bufferSize() const { return frameSize() * mFrameCount; }
virtual ssize_t frameCount() const { return mFrameCount; }
virtual ssize_t channelCount() const { return (ssize_t)mChannelCount; }
virtual ssize_t frameSize() const { return ssize_t(mChannelCount * ((mFormat == AUDIO_FORMAT_PCM_16_BIT)?sizeof(int16_t):sizeof(u_int8_t))); }
virtual uint32_t latency() const;
virtual float msecsPerFrame() const;
virtual status_t getPosition(uint32_t *position) const;
virtual status_t getFramesWritten(uint32_t *frameswritten) const;
virtual int getSessionId() const;
virtual uint32_t getSampleRate() const;
virtual status_t open(
uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
audio_format_t format, int bufferCount = 1,
AudioCallback cb = NULL, void *cookie = NULL,
audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
const audio_offload_info_t *offloadInfo = NULL);
virtual status_t start();
virtual ssize_t write(const void* buffer, size_t size);
virtual void stop();
virtual void flush() {}
virtual void pause() {}
virtual void close() {}
void setAudioStreamType(audio_stream_type_t streamType) {}
// stream type is not used for AudioCache
virtual audio_stream_type_t getAudioStreamType() const { return AUDIO_STREAM_DEFAULT; }
void setVolume(float left, float right) {}
virtual status_t setPlaybackRatePermille(int32_t ratePermille) { return INVALID_OPERATION; }
uint32_t sampleRate() const { return mSampleRate; }
audio_format_t format() const { return mFormat; }
size_t size() const { return mSize; }
status_t wait();
sp<IMemoryHeap> getHeap() const { return mHeap; }
static void notify(void* cookie, int msg,
int ext1, int ext2, const Parcel *obj);
virtual status_t dump(int fd, const Vector<String16>& args) const;
private:
AudioCache();
Mutex mLock;
Condition mSignal;
sp<IMemoryHeap> mHeap;
float mMsecsPerFrame;
uint16_t mChannelCount;
audio_format_t mFormat;
ssize_t mFrameCount;
uint32_t mSampleRate;
uint32_t mSize;
int mError;
bool mCommandComplete;
sp<Thread> mCallbackThread;
}; // AudioCache
public:
static void instantiate();
// IMediaPlayerService interface
virtual sp<IMediaRecorder> createMediaRecorder();
void removeMediaRecorderClient(wp<MediaRecorderClient> client);
virtual sp<IMediaMetadataRetriever> createMetadataRetriever();
virtual sp<IMediaPlayer> create(const sp<IMediaPlayerClient>& client, int audioSessionId);
virtual status_t decode(const char* url, uint32_t *pSampleRate, int* pNumChannels,
audio_format_t* pFormat,
const sp<IMemoryHeap>& heap, size_t *pSize);
virtual status_t decode(int fd, int64_t offset, int64_t length,
uint32_t *pSampleRate, int* pNumChannels,
audio_format_t* pFormat,
const sp<IMemoryHeap>& heap, size_t *pSize);
virtual sp<IOMX> getOMX();
virtual sp<ICrypto> makeCrypto();
virtual sp<IDrm> makeDrm();
virtual sp<IHDCP> makeHDCP(bool createEncryptionModule);
virtual sp<IRemoteDisplay> listenForRemoteDisplay(const sp<IRemoteDisplayClient>& client,
const String8& iface);
virtual status_t dump(int fd, const Vector<String16>& args);
virtual status_t updateProxyConfig(
const char *host, int32_t port, const char *exclusionList);
void removeClient(wp<Client> client);
// For battery usage tracking purpose
struct BatteryUsageInfo {
// how many streams are being played by one UID
int refCount;
// a temp variable to store the duration(ms) of audio codecs
// when we start a audio codec, we minus the system time from audioLastTime
// when we pause it, we add the system time back to the audioLastTime
// so after the pause, audioLastTime = pause time - start time
// if multiple audio streams are played (or recorded), then audioLastTime
// = the total playing time of all the streams
int32_t audioLastTime;
// when all the audio streams are being paused, we assign audioLastTime to
// this variable, so this value could be provided to the battery app
// in the next pullBatteryData call
int32_t audioTotalTime;
int32_t videoLastTime;
int32_t videoTotalTime;
};
KeyedVector<int, BatteryUsageInfo> mBatteryData;
enum {
SPEAKER,
OTHER_AUDIO_DEVICE,
SPEAKER_AND_OTHER,
NUM_AUDIO_DEVICES
};
struct BatteryAudioFlingerUsageInfo {
int refCount; // how many audio streams are being played
int deviceOn[NUM_AUDIO_DEVICES]; // whether the device is currently used
int32_t lastTime[NUM_AUDIO_DEVICES]; // in ms
// totalTime[]: total time of audio output devices usage
int32_t totalTime[NUM_AUDIO_DEVICES]; // in ms
};
// This varialble is used to record the usage of audio output device
// for battery app
BatteryAudioFlingerUsageInfo mBatteryAudio;
// Collect info of the codec usage from media player and media recorder
virtual void addBatteryData(uint32_t params);
// API for the Battery app to pull the data of codecs usage
virtual status_t pullBatteryData(Parcel* reply);
private:
class Client : public BnMediaPlayer {
// IMediaPlayer interface
virtual void disconnect();
virtual status_t setVideoSurfaceTexture(
const sp<IGraphicBufferProducer>& bufferProducer);
virtual status_t prepareAsync();
virtual status_t start();
virtual status_t stop();
virtual status_t pause();
virtual status_t isPlaying(bool* state);
virtual status_t seekTo(int msec);
virtual status_t getCurrentPosition(int* msec);
virtual status_t getDuration(int* msec);
virtual status_t reset();
virtual status_t setAudioStreamType(audio_stream_type_t type);
virtual status_t setLooping(int loop);
virtual status_t setVolume(float leftVolume, float rightVolume);
virtual status_t invoke(const Parcel& request, Parcel *reply);
virtual status_t setMetadataFilter(const Parcel& filter);
virtual status_t getMetadata(bool update_only,
bool apply_filter,
Parcel *reply);
virtual status_t setAuxEffectSendLevel(float level);
virtual status_t attachAuxEffect(int effectId);
virtual status_t setParameter(int key, const Parcel &request);
virtual status_t getParameter(int key, Parcel *reply);
virtual status_t setRetransmitEndpoint(const struct sockaddr_in* endpoint);
virtual status_t getRetransmitEndpoint(struct sockaddr_in* endpoint);
virtual status_t setNextPlayer(const sp<IMediaPlayer>& player);
sp<MediaPlayerBase> createPlayer(player_type playerType);
virtual status_t setDataSource(
const char *url,
const KeyedVector<String8, String8> *headers);
virtual status_t setDataSource(int fd, int64_t offset, int64_t length);
virtual status_t setDataSource(const sp<IStreamSource> &source);
sp<MediaPlayerBase> setDataSource_pre(player_type playerType);
void setDataSource_post(const sp<MediaPlayerBase>& p,
status_t status);
static void notify(void* cookie, int msg,
int ext1, int ext2, const Parcel *obj);
pid_t pid() const { return mPid; }
virtual status_t dump(int fd, const Vector<String16>& args) const;
int getAudioSessionId() { return mAudioSessionId; }
private:
friend class MediaPlayerService;
Client( const sp<MediaPlayerService>& service,
pid_t pid,
int32_t connId,
const sp<IMediaPlayerClient>& client,
int audioSessionId,
uid_t uid);
Client();
virtual ~Client();
void deletePlayer();
sp<MediaPlayerBase> getPlayer() const { Mutex::Autolock lock(mLock); return mPlayer; }
// @param type Of the metadata to be tested.
// @return true if the metadata should be dropped according to
// the filters.
bool shouldDropMetadata(media::Metadata::Type type) const;
// Add a new element to the set of metadata updated. Noop if
// the element exists already.
// @param type Of the metadata to be recorded.
void addNewMetadataUpdate(media::Metadata::Type type);
// Disconnect from the currently connected ANativeWindow.
void disconnectNativeWindow();
mutable Mutex mLock;
sp<MediaPlayerBase> mPlayer;
sp<MediaPlayerService> mService;
sp<IMediaPlayerClient> mClient;
sp<AudioOutput> mAudioOutput;
pid_t mPid;
status_t mStatus;
bool mLoop;
int32_t mConnId;
int mAudioSessionId;
uid_t mUID;
sp<ANativeWindow> mConnectedWindow;
sp<IBinder> mConnectedWindowBinder;
struct sockaddr_in mRetransmitEndpoint;
bool mRetransmitEndpointValid;
sp<Client> mNextClient;
// Metadata filters.
media::Metadata::Filter mMetadataAllow; // protected by mLock
media::Metadata::Filter mMetadataDrop; // protected by mLock
// Metadata updated. For each MEDIA_INFO_METADATA_UPDATE
// notification we try to update mMetadataUpdated which is a
// set: no duplicate.
// getMetadata clears this set.
media::Metadata::Filter mMetadataUpdated; // protected by mLock
#if CALLBACK_ANTAGONIZER
Antagonizer* mAntagonizer;
#endif
}; // Client
// ----------------------------------------------------------------------------
MediaPlayerService();
virtual ~MediaPlayerService();
mutable Mutex mLock;
SortedVector< wp<Client> > mClients;
SortedVector< wp<MediaRecorderClient> > mMediaRecorderClients;
int32_t mNextConnId;
sp<IOMX> mOMX;
sp<ICrypto> mCrypto;
};
// ----------------------------------------------------------------------------
}; // namespace android
#endif // ANDROID_MEDIAPLAYERSERVICE_H