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replicant-frameworks_native/libs/audioflinger/AudioFlinger.h
Eric Laurent 4edfe75018 Fix issue 2553359: Pandora does not work well with Passion deskdock / Cardock. 
The problem is due to a too big difference between the buffer size used at the hardware interface and at the A2DP interface.
When no resampling occurs we don't notice problems but the timing is very tight. As soon as resampling is activated, the AudioTrack underruns.
This is because the AudioTrack buffers are not resized when moving the AudioTrack from hardware to A2DP output.
The AudioTrack buffers are calculated based on a hardware output buffer size of 3072 bytes. Which is much less than the A2DP output buffer size (10240).

The solution consists in creating new tracks with new buffers in AudioFlinger when the A2DP output is opened
instead of just transfering active tracks from hardware output mixer thread to the new A2DP output mixer thread.
To avoid synchronization issues between mixer threads and client processes, this is done by invalidating tracks
by setting a flag in their control block and having AudioTrack release the handle on this track (IAudioTrack)
and create a new IAudioTrack when this flag is detected next time obtainBuffer() or start() is executed.

AudioFlinger modifications:
- invalidate the tracks when setStreamOutput() is called
- make sure that notifications of output opening/closing and change of stream type to output mapping are sent synchronously to client process.
This is necessary so that AudioSystem has the new stream to output mapping when the AudioTrack detects the invalidate flag in the client process.
Previously their were sent when the corresponding thread loop was executed.

AudioTrack modifications:
- move frame count calculation and verification from set() to createTrack() so that is is updated every time a new IAudioTrack is created.
- detect track invalidate flag in obtainBuffer() and start() and create a new IAudioTrack.

AudioTrackShared modifications
- group all flags (out, flowControlFlag, forceReady...) into a single bit filed to save space.

Change-Id: I9ac26b6192230627d35084e1449640caaf7d56ee
2010-05-17 02:23:47 -07:00

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/* //device/include/server/AudioFlinger/AudioFlinger.h
**
** Copyright 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_AUDIO_FLINGER_H
#define ANDROID_AUDIO_FLINGER_H
#include <stdint.h>
#include <sys/types.h>
#include <limits.h>
#include <media/IAudioFlinger.h>
#include <media/IAudioFlingerClient.h>
#include <media/IAudioTrack.h>
#include <media/IAudioRecord.h>
#include <media/AudioTrack.h>
#include <utils/Atomic.h>
#include <utils/Errors.h>
#include <utils/threads.h>
#include <binder/MemoryDealer.h>
#include <utils/SortedVector.h>
#include <utils/Vector.h>
#include <hardware_legacy/AudioHardwareInterface.h>
#include "AudioBufferProvider.h"
namespace android {
class audio_track_cblk_t;
class AudioMixer;
class AudioBuffer;
class AudioResampler;
// ----------------------------------------------------------------------------
#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
// ----------------------------------------------------------------------------
static const nsecs_t kStandbyTimeInNsecs = seconds(3);
class AudioFlinger : public BnAudioFlinger
{
public:
static void instantiate();
virtual status_t dump(int fd, const Vector<String16>& args);
// IAudioFlinger interface
virtual sp<IAudioTrack> createTrack(
pid_t pid,
int streamType,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
int output,
status_t *status);
virtual uint32_t sampleRate(int output) const;
virtual int channelCount(int output) const;
virtual int format(int output) const;
virtual size_t frameCount(int output) const;
virtual uint32_t latency(int output) const;
virtual status_t setMasterVolume(float value);
virtual status_t setMasterMute(bool muted);
virtual float masterVolume() const;
virtual bool masterMute() const;
virtual status_t setStreamVolume(int stream, float value, int output);
virtual status_t setStreamMute(int stream, bool muted);
virtual float streamVolume(int stream, int output) const;
virtual bool streamMute(int stream) const;
virtual status_t setMode(int mode);
virtual status_t setMicMute(bool state);
virtual bool getMicMute() const;
virtual bool isStreamActive(int stream) const;
virtual status_t setParameters(int ioHandle, const String8& keyValuePairs);
virtual String8 getParameters(int ioHandle, const String8& keys);
virtual void registerClient(const sp<IAudioFlingerClient>& client);
virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount);
virtual unsigned int getInputFramesLost(int ioHandle);
virtual int openOutput(uint32_t *pDevices,
uint32_t *pSamplingRate,
uint32_t *pFormat,
uint32_t *pChannels,
uint32_t *pLatencyMs,
uint32_t flags);
virtual int openDuplicateOutput(int output1, int output2);
virtual status_t closeOutput(int output);
virtual status_t suspendOutput(int output);
virtual status_t restoreOutput(int output);
virtual int openInput(uint32_t *pDevices,
uint32_t *pSamplingRate,
uint32_t *pFormat,
uint32_t *pChannels,
uint32_t acoustics);
virtual status_t closeInput(int input);
virtual status_t setStreamOutput(uint32_t stream, int output);
virtual status_t setVoiceVolume(float volume);
virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output);
enum hardware_call_state {
AUDIO_HW_IDLE = 0,
AUDIO_HW_INIT,
AUDIO_HW_OUTPUT_OPEN,
AUDIO_HW_OUTPUT_CLOSE,
AUDIO_HW_INPUT_OPEN,
AUDIO_HW_INPUT_CLOSE,
AUDIO_HW_STANDBY,
AUDIO_HW_SET_MASTER_VOLUME,
AUDIO_HW_GET_ROUTING,
AUDIO_HW_SET_ROUTING,
AUDIO_HW_GET_MODE,
AUDIO_HW_SET_MODE,
AUDIO_HW_GET_MIC_MUTE,
AUDIO_HW_SET_MIC_MUTE,
AUDIO_SET_VOICE_VOLUME,
AUDIO_SET_PARAMETER,
};
// record interface
virtual sp<IAudioRecord> openRecord(
pid_t pid,
int input,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
uint32_t flags,
status_t *status);
virtual status_t onTransact(
uint32_t code,
const Parcel& data,
Parcel* reply,
uint32_t flags);
private:
AudioFlinger();
virtual ~AudioFlinger();
// Internal dump utilites.
status_t dumpPermissionDenial(int fd, const Vector<String16>& args);
status_t dumpClients(int fd, const Vector<String16>& args);
status_t dumpInternals(int fd, const Vector<String16>& args);
// --- Client ---
class Client : public RefBase {
public:
Client(const sp<AudioFlinger>& audioFlinger, pid_t pid);
virtual ~Client();
const sp<MemoryDealer>& heap() const;
pid_t pid() const { return mPid; }
sp<AudioFlinger> audioFlinger() { return mAudioFlinger; }
private:
Client(const Client&);
Client& operator = (const Client&);
sp<AudioFlinger> mAudioFlinger;
sp<MemoryDealer> mMemoryDealer;
pid_t mPid;
};
// --- Notification Client ---
class NotificationClient : public IBinder::DeathRecipient {
public:
NotificationClient(const sp<AudioFlinger>& audioFlinger,
const sp<IAudioFlingerClient>& client,
pid_t pid);
virtual ~NotificationClient();
sp<IAudioFlingerClient> client() { return mClient; }
// IBinder::DeathRecipient
virtual void binderDied(const wp<IBinder>& who);
private:
NotificationClient(const NotificationClient&);
NotificationClient& operator = (const NotificationClient&);
sp<AudioFlinger> mAudioFlinger;
pid_t mPid;
sp<IAudioFlingerClient> mClient;
};
class TrackHandle;
class RecordHandle;
class RecordThread;
class PlaybackThread;
class MixerThread;
class DirectOutputThread;
class DuplicatingThread;
class Track;
class RecordTrack;
class ThreadBase : public Thread {
public:
ThreadBase (const sp<AudioFlinger>& audioFlinger, int id);
virtual ~ThreadBase();
status_t dumpBase(int fd, const Vector<String16>& args);
// base for record and playback
class TrackBase : public AudioBufferProvider, public RefBase {
public:
enum track_state {
IDLE,
TERMINATED,
STOPPED,
RESUMING,
ACTIVE,
PAUSING,
PAUSED
};
enum track_flags {
STEPSERVER_FAILED = 0x01, // StepServer could not acquire cblk->lock mutex
SYSTEM_FLAGS_MASK = 0x0000ffffUL,
// The upper 16 bits are used for track-specific flags.
};
TrackBase(const wp<ThreadBase>& thread,
const sp<Client>& client,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer);
~TrackBase();
virtual status_t start() = 0;
virtual void stop() = 0;
sp<IMemory> getCblk() const;
audio_track_cblk_t* cblk() const { return mCblk; }
protected:
friend class ThreadBase;
friend class RecordHandle;
friend class PlaybackThread;
friend class RecordThread;
friend class MixerThread;
friend class DirectOutputThread;
TrackBase(const TrackBase&);
TrackBase& operator = (const TrackBase&);
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) = 0;
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
int format() const {
return mFormat;
}
int channelCount() const ;
int sampleRate() const;
void* getBuffer(uint32_t offset, uint32_t frames) const;
bool isStopped() const {
return mState == STOPPED;
}
bool isTerminated() const {
return mState == TERMINATED;
}
bool step();
void reset();
wp<ThreadBase> mThread;
sp<Client> mClient;
sp<IMemory> mCblkMemory;
audio_track_cblk_t* mCblk;
void* mBuffer;
void* mBufferEnd;
uint32_t mFrameCount;
// we don't really need a lock for these
int mState;
int mClientTid;
uint8_t mFormat;
uint32_t mFlags;
};
class ConfigEvent {
public:
ConfigEvent() : mEvent(0), mParam(0) {}
int mEvent;
int mParam;
};
uint32_t sampleRate() const;
int channelCount() const;
int format() const;
size_t frameCount() const;
void wakeUp() { mWaitWorkCV.broadcast(); }
void exit();
virtual bool checkForNewParameters_l() = 0;
virtual status_t setParameters(const String8& keyValuePairs);
virtual String8 getParameters(const String8& keys) = 0;
virtual void audioConfigChanged_l(int event, int param = 0) = 0;
void sendConfigEvent(int event, int param = 0);
void sendConfigEvent_l(int event, int param = 0);
void processConfigEvents();
int id() const { return mId;}
bool standby() { return mStandby; }
mutable Mutex mLock;
protected:
friend class Track;
friend class TrackBase;
friend class PlaybackThread;
friend class MixerThread;
friend class DirectOutputThread;
friend class DuplicatingThread;
friend class RecordThread;
friend class RecordTrack;
Condition mWaitWorkCV;
sp<AudioFlinger> mAudioFlinger;
uint32_t mSampleRate;
size_t mFrameCount;
uint32_t mChannels;
uint16_t mChannelCount;
uint16_t mFrameSize;
int mFormat;
Condition mParamCond;
Vector<String8> mNewParameters;
status_t mParamStatus;
Vector<ConfigEvent *> mConfigEvents;
bool mStandby;
int mId;
bool mExiting;
};
// --- PlaybackThread ---
class PlaybackThread : public ThreadBase {
public:
enum type {
MIXER,
DIRECT,
DUPLICATING
};
enum mixer_state {
MIXER_IDLE,
MIXER_TRACKS_ENABLED,
MIXER_TRACKS_READY
};
// playback track
class Track : public TrackBase {
public:
Track( const wp<ThreadBase>& thread,
const sp<Client>& client,
int streamType,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
const sp<IMemory>& sharedBuffer);
~Track();
void dump(char* buffer, size_t size);
virtual status_t start();
virtual void stop();
void pause();
void flush();
void destroy();
void mute(bool);
void setVolume(float left, float right);
int name() const {
return mName;
}
int type() const {
return mStreamType;
}
protected:
friend class ThreadBase;
friend class AudioFlinger;
friend class TrackHandle;
friend class PlaybackThread;
friend class MixerThread;
friend class DirectOutputThread;
Track(const Track&);
Track& operator = (const Track&);
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
bool isMuted() { return mMute; }
bool isPausing() const {
return mState == PAUSING;
}
bool isPaused() const {
return mState == PAUSED;
}
bool isReady() const;
void setPaused() { mState = PAUSED; }
void reset();
bool isOutputTrack() const {
return (mStreamType == AudioSystem::NUM_STREAM_TYPES);
}
// we don't really need a lock for these
float mVolume[2];
volatile bool mMute;
// FILLED state is used for suppressing volume ramp at begin of playing
enum {FS_FILLING, FS_FILLED, FS_ACTIVE};
mutable uint8_t mFillingUpStatus;
int8_t mRetryCount;
sp<IMemory> mSharedBuffer;
bool mResetDone;
int mStreamType;
int mName;
}; // end of Track
// playback track
class OutputTrack : public Track {
public:
class Buffer: public AudioBufferProvider::Buffer {
public:
int16_t *mBuffer;
};
OutputTrack( const wp<ThreadBase>& thread,
DuplicatingThread *sourceThread,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount);
~OutputTrack();
virtual status_t start();
virtual void stop();
bool write(int16_t* data, uint32_t frames);
bool bufferQueueEmpty() { return (mBufferQueue.size() == 0) ? true : false; }
bool isActive() { return mActive; }
wp<ThreadBase>& thread() { return mThread; }
private:
status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs);
void clearBufferQueue();
// Maximum number of pending buffers allocated by OutputTrack::write()
static const uint8_t kMaxOverFlowBuffers = 10;
Vector < Buffer* > mBufferQueue;
AudioBufferProvider::Buffer mOutBuffer;
bool mActive;
DuplicatingThread* mSourceThread;
}; // end of OutputTrack
PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id);
virtual ~PlaybackThread();
virtual status_t dump(int fd, const Vector<String16>& args);
// Thread virtuals
virtual status_t readyToRun();
virtual void onFirstRef();
virtual uint32_t latency() const;
virtual status_t setMasterVolume(float value);
virtual status_t setMasterMute(bool muted);
virtual float masterVolume() const;
virtual bool masterMute() const;
virtual status_t setStreamVolume(int stream, float value);
virtual status_t setStreamMute(int stream, bool muted);
virtual float streamVolume(int stream) const;
virtual bool streamMute(int stream) const;
bool isStreamActive(int stream) const;
sp<Track> createTrack_l(
const sp<AudioFlinger::Client>& client,
int streamType,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
const sp<IMemory>& sharedBuffer,
status_t *status);
AudioStreamOut* getOutput() { return mOutput; }
virtual int type() const { return mType; }
void suspend() { mSuspended++; }
void restore() { if (mSuspended) mSuspended--; }
bool isSuspended() { return (mSuspended != 0); }
virtual String8 getParameters(const String8& keys);
virtual void audioConfigChanged_l(int event, int param = 0);
virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames);
struct stream_type_t {
stream_type_t()
: volume(1.0f),
mute(false)
{
}
float volume;
bool mute;
};
protected:
int mType;
int16_t* mMixBuffer;
int mSuspended;
int mBytesWritten;
bool mMasterMute;
SortedVector< wp<Track> > mActiveTracks;
virtual int getTrackName_l() = 0;
virtual void deleteTrackName_l(int name) = 0;
virtual uint32_t activeSleepTimeUs() = 0;
virtual uint32_t idleSleepTimeUs() = 0;
private:
friend class AudioFlinger;
friend class OutputTrack;
friend class Track;
friend class TrackBase;
friend class MixerThread;
friend class DirectOutputThread;
friend class DuplicatingThread;
PlaybackThread(const Client&);
PlaybackThread& operator = (const PlaybackThread&);
status_t addTrack_l(const sp<Track>& track);
void destroyTrack_l(const sp<Track>& track);
void readOutputParameters();
virtual status_t dumpInternals(int fd, const Vector<String16>& args);
status_t dumpTracks(int fd, const Vector<String16>& args);
SortedVector< sp<Track> > mTracks;
// mStreamTypes[] uses 1 additionnal stream type internally for the OutputTrack used by DuplicatingThread
stream_type_t mStreamTypes[AudioSystem::NUM_STREAM_TYPES + 1];
AudioStreamOut* mOutput;
float mMasterVolume;
nsecs_t mLastWriteTime;
int mNumWrites;
int mNumDelayedWrites;
bool mInWrite;
};
class MixerThread : public PlaybackThread {
public:
MixerThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id);
virtual ~MixerThread();
// Thread virtuals
virtual bool threadLoop();
void invalidateTracks(int streamType);
virtual bool checkForNewParameters_l();
virtual status_t dumpInternals(int fd, const Vector<String16>& args);
protected:
uint32_t prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove);
virtual int getTrackName_l();
virtual void deleteTrackName_l(int name);
virtual uint32_t activeSleepTimeUs();
virtual uint32_t idleSleepTimeUs();
AudioMixer* mAudioMixer;
};
class DirectOutputThread : public PlaybackThread {
public:
DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id);
~DirectOutputThread();
// Thread virtuals
virtual bool threadLoop();
virtual bool checkForNewParameters_l();
protected:
virtual int getTrackName_l();
virtual void deleteTrackName_l(int name);
virtual uint32_t activeSleepTimeUs();
virtual uint32_t idleSleepTimeUs();
private:
float mLeftVolume;
float mRightVolume;
};
class DuplicatingThread : public MixerThread {
public:
DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, int id);
~DuplicatingThread();
// Thread virtuals
virtual bool threadLoop();
void addOutputTrack(MixerThread* thread);
void removeOutputTrack(MixerThread* thread);
uint32_t waitTimeMs() { return mWaitTimeMs; }
protected:
virtual uint32_t activeSleepTimeUs();
private:
bool outputsReady(SortedVector< sp<OutputTrack> > &outputTracks);
void updateWaitTime();
SortedVector < sp<OutputTrack> > mOutputTracks;
uint32_t mWaitTimeMs;
};
PlaybackThread *checkPlaybackThread_l(int output) const;
MixerThread *checkMixerThread_l(int output) const;
RecordThread *checkRecordThread_l(int input) const;
float streamVolumeInternal(int stream) const { return mStreamTypes[stream].volume; }
void audioConfigChanged_l(int event, int ioHandle, void *param2);
friend class AudioBuffer;
class TrackHandle : public android::BnAudioTrack {
public:
TrackHandle(const sp<PlaybackThread::Track>& track);
virtual ~TrackHandle();
virtual status_t start();
virtual void stop();
virtual void flush();
virtual void mute(bool);
virtual void pause();
virtual void setVolume(float left, float right);
virtual sp<IMemory> getCblk() const;
virtual status_t onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
private:
sp<PlaybackThread::Track> mTrack;
};
friend class Client;
friend class PlaybackThread::Track;
void removeClient_l(pid_t pid);
void removeNotificationClient(pid_t pid);
// record thread
class RecordThread : public ThreadBase, public AudioBufferProvider
{
public:
// record track
class RecordTrack : public TrackBase {
public:
RecordTrack(const wp<ThreadBase>& thread,
const sp<Client>& client,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
uint32_t flags);
~RecordTrack();
virtual status_t start();
virtual void stop();
bool overflow() { bool tmp = mOverflow; mOverflow = false; return tmp; }
bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; }
void dump(char* buffer, size_t size);
private:
friend class AudioFlinger;
friend class RecordThread;
RecordTrack(const RecordTrack&);
RecordTrack& operator = (const RecordTrack&);
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
bool mOverflow;
};
RecordThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamIn *input,
uint32_t sampleRate,
uint32_t channels,
int id);
~RecordThread();
virtual bool threadLoop();
virtual status_t readyToRun() { return NO_ERROR; }
virtual void onFirstRef();
status_t start(RecordTrack* recordTrack);
void stop(RecordTrack* recordTrack);
status_t dump(int fd, const Vector<String16>& args);
AudioStreamIn* getInput() { return mInput; }
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
virtual bool checkForNewParameters_l();
virtual String8 getParameters(const String8& keys);
virtual void audioConfigChanged_l(int event, int param = 0);
void readInputParameters();
virtual unsigned int getInputFramesLost();
private:
RecordThread();
AudioStreamIn *mInput;
sp<RecordTrack> mActiveTrack;
Condition mStartStopCond;
AudioResampler *mResampler;
int32_t *mRsmpOutBuffer;
int16_t *mRsmpInBuffer;
size_t mRsmpInIndex;
size_t mInputBytes;
int mReqChannelCount;
uint32_t mReqSampleRate;
ssize_t mBytesRead;
};
class RecordHandle : public android::BnAudioRecord {
public:
RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack);
virtual ~RecordHandle();
virtual status_t start();
virtual void stop();
virtual sp<IMemory> getCblk() const;
virtual status_t onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
private:
sp<RecordThread::RecordTrack> mRecordTrack;
};
friend class RecordThread;
friend class PlaybackThread;
mutable Mutex mLock;
DefaultKeyedVector< pid_t, wp<Client> > mClients;
mutable Mutex mHardwareLock;
AudioHardwareInterface* mAudioHardware;
mutable int mHardwareStatus;
DefaultKeyedVector< int, sp<PlaybackThread> > mPlaybackThreads;
PlaybackThread::stream_type_t mStreamTypes[AudioSystem::NUM_STREAM_TYPES];
float mMasterVolume;
bool mMasterMute;
DefaultKeyedVector< int, sp<RecordThread> > mRecordThreads;
DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients;
int mNextThreadId;
#ifdef LVMX
int mLifeVibesClientPid;
#endif
};
// ----------------------------------------------------------------------------
}; // namespace android
#endif // ANDROID_AUDIO_FLINGER_H
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