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We no longer put the filename at start of file.

Change-Id: Ic435b159a23105681e3d4a6cb1ac097bc853302e
This commit is contained in:
Glenn Kasten
2012-01-19 08:59:58 -08:00
parent d23d2b1662
commit b3db213eb5
19 changed files with 107 additions and 54 deletions
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2
core/jni/android_media_AudioRecord.cpp
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@@ -220,7 +220,7 @@ android_media_AudioRecord_setup(JNIEnv *env, jobject thiz, jobject weak_this,
ALOGE("Error creating AudioRecord: Error retrieving session id pointer");
goto native_init_failure;
}
// read the audio session ID back from AudioTrack in case a new session was created during set()
// read the audio session ID back from AudioRecord in case a new session was created during set()
nSession[0] = lpRecorder->getSessionId();
env->ReleasePrimitiveArrayCritical(jSession, nSession, 0);
nSession = NULL;

2
core/jni/android_media_AudioSystem.cpp
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@@ -1,4 +1,4 @@
/* //device/libs/android_runtime/android_media_AudioSystem.cpp
/*
**
** Copyright 2006, The Android Open Source Project
**

2
core/jni/android_media_ToneGenerator.cpp
View File

@@ -1,4 +1,4 @@
/* //device/libs/android_runtime/android_media_AudioSystem.cpp
/*
**
** Copyright 2008, The Android Open Source Project
**

2
include/media/AudioSystem.h
View File

@@ -131,7 +131,7 @@ public:
NUM_CONFIG_EVENTS
};
// audio output descritor used to cache output configurations in client process to avoid frequent calls
// audio output descriptor used to cache output configurations in client process to avoid frequent calls
// through IAudioFlinger
class OutputDescriptor {
public:

46
include/media/AudioTrack.h
View File

@@ -58,8 +58,8 @@ public:
EVENT_BUFFER_END = 5 // Playback head is at the end of the buffer.
};
/* Create Buffer on the stack and pass it to obtainBuffer()
* and releaseBuffer().
/* Client should declare Buffer on the stack and pass address to obtainBuffer()
* and releaseBuffer(). See also callback_t for EVENT_MORE_DATA.
*/
class Buffer
@@ -68,12 +68,16 @@ public:
enum {
MUTE = 0x00000001
};
uint32_t flags;
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;
size_t size;
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
@@ -84,15 +88,15 @@ public:
/* 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 an underrun condition occurs.
* 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 less bytes are
* written.
* 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.
@@ -225,7 +229,7 @@ public:
*/
uint32_t latency() const;
/* getters, see constructor */
/* getters, see constructors and set() */
audio_stream_type_t streamType() const;
audio_format_t format() const;
@@ -401,13 +405,19 @@ public:
status_t attachAuxEffect(int effectId);
/* Obtains a buffer of "frameCount" frames. The buffer must be
* filled entirely. If the track is stopped, obtainBuffer() returns
* 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 {
@@ -416,12 +426,19 @@ public:
};
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 lockBuffer/unlockBuffer. For best
* performance use callbacks. Return actual number of bytes written.
*
* 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);
@@ -448,6 +465,7 @@ private:
AudioTrack& mReceiver;
};
// body of AudioTrackThread::threadLoop()
bool processAudioBuffer(const sp<AudioTrackThread>& thread);
status_t createTrack_l(audio_stream_type_t streamType,
uint32_t sampleRate,
@@ -484,7 +502,7 @@ private:
bool mActive; // protected by mLock
callback_t mCbf;
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

8
include/media/EffectsFactoryApi.h
View File

@@ -87,7 +87,7 @@ int EffectQueryEffect(uint32_t index, effect_descriptor_t *pDescriptor);
// Description: Creates an effect engine of the specified type and returns an
// effect control interface on this engine. The function will allocate the
// resources for an instance of the requested effect engine and return
// a handler on the effect control interface.
// a handle on the effect control interface.
//
// Input:
// pEffectUuid: pointer to the effect uuid.
@@ -115,17 +115,17 @@ int EffectCreate(const effect_uuid_t *pEffectUuid, int32_t sessionId, int32_t io
//
// Function: EffectRelease
//
// Description: Releases the effect engine whose handler is given as argument.
// Description: Releases the effect engine whose handle is given as argument.
// All resources allocated to this particular instance of the effect are
// released.
//
// Input:
// handle: handler on the effect interface to be released.
// handle: handle on the effect interface to be released.
//
// Output:
// returned value: 0 successful operation.
// -ENODEV factory failed to initialize
// -EINVAL invalid interface handler
// -EINVAL invalid interface handle
//
////////////////////////////////////////////////////////////////////////////////
int EffectRelease(effect_handle_t handle);

8
include/media/IAudioTrack.h
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@@ -46,12 +46,12 @@ public:
/* Stop a track. If set, the callback will cease being called and
* obtainBuffer will return an error. Buffers that are already released
* will be processed, unless flush() is called.
* will continue to be processed, unless/until flush() is called.
*/
virtual void stop() = 0;
/* Flush a stopped track. All pending buffers are discarded.
* This function has no effect if the track is not stopped.
/* Flush a stopped or paused track. All pending/released buffers are discarded.
* This function has no effect if the track is not stopped or paused.
*/
virtual void flush() = 0;
@@ -62,7 +62,7 @@ public:
/* Pause a track. If set, the callback will cease being called and
* obtainBuffer will return an error. Buffers that are already released
* will be processed, unless flush() is called.
* will continue to be processed, unless/until flush() is called.
*/
virtual void pause() = 0;

2
include/media/MemoryLeakTrackUtil.h
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@@ -19,7 +19,7 @@
namespace android {
/*
* Dump the memory adddress of the calling process to the given fd.
* Dump the memory address of the calling process to the given fd.
*/
extern void dumpMemoryAddresses(int fd);

29
include/private/media/AudioTrackShared.h
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@@ -62,16 +62,23 @@ struct audio_track_cblk_t
// are in the same line of data cache.
Mutex lock; // sizeof(int)
Condition cv; // sizeof(int)
// next 4 are offsets within "buffers"
volatile uint32_t user;
volatile uint32_t server;
uint32_t userBase;
uint32_t serverBase;
// if there is a shared buffer, "buffers" is the value of pointer() for the shared
// buffer, otherwise "buffers" points immediately after the control block
void* buffers;
uint32_t frameCount;
// Cache line boundary
uint32_t loopStart;
uint32_t loopEnd;
int loopCount;
uint32_t loopEnd; // read-only for server, read/write for client
int loopCount; // read/write for client
// Channel volumes are fixed point U4.12, so 0x1000 means 1.0.
// Left channel is in [0:15], right channel is in [16:31].
@@ -82,29 +89,39 @@ private:
public:
uint32_t sampleRate;
// NOTE: audio_track_cblk_t::frameSize is not equal to AudioTrack::frameSize() for
// 8 bit PCM data: in this case, mCblk->frameSize is based on a sample size of
// 16 bit because data is converted to 16 bit before being stored in buffer
// read-only for client, server writes once at initialization and is then read-only
uint8_t frameSize; // would normally be size_t, but 8 bits is plenty
// never used
uint8_t pad1;
// used by client only
uint16_t bufferTimeoutMs; // Maximum cumulated timeout before restarting audioflinger
uint16_t waitTimeMs; // Cumulated wait time
uint16_t waitTimeMs; // Cumulated wait time, used by client only
private:
// client write-only, server read-only
uint16_t mSendLevel; // Fixed point U4.12 so 0x1000 means 1.0
public:
volatile int32_t flags;
// Cache line boundary (32 bytes)
// Since the control block is always located in shared memory, this constructor
// is only used for placement new(). It is never used for regular new() or stack.
audio_track_cblk_t();
uint32_t stepUser(uint32_t frameCount);
bool stepServer(uint32_t frameCount);
uint32_t stepUser(uint32_t frameCount); // called by client only, where
// client includes regular AudioTrack and AudioFlinger::PlaybackThread::OutputTrack
bool stepServer(uint32_t frameCount); // called by server only
void* buffer(uint32_t offset) const;
uint32_t framesAvailable();
uint32_t framesAvailable_l();
uint32_t framesReady();
uint32_t framesReady(); // called by server only
bool tryLock();
// No barriers on the following operations, so the ordering of loads/stores

2
include/utils/KeyedVector.h
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@@ -66,7 +66,7 @@ public:
ssize_t indexOfKey(const KEY& key) const;
/*!
* modifing the array
* modifying the array
*/
VALUE& editValueFor(const KEY& key);

2
media/jni/soundpool/SoundPool.cpp
View File

@@ -623,7 +623,7 @@ void SoundChannel::play(const sp<Sample>& sample, int nextChannelID, float leftV
newTrack->setVolume(leftVolume, rightVolume);
newTrack->setLoop(0, frameCount, loop);
// From now on, AudioTrack callbacks recevieved with previous toggle value will be ignored.
// From now on, AudioTrack callbacks received with previous toggle value will be ignored.
mToggle = toggle;
mAudioTrack = newTrack;
mPos = 0;

15
media/libmedia/AudioTrack.cpp
View File

@@ -1,4 +1,4 @@
/* frameworks/base/media/libmedia/AudioTrack.cpp
/*
**
** Copyright 2007, The Android Open Source Project
**
@@ -792,7 +792,7 @@ status_t AudioTrack::createTrack_l(
}
}
} else {
// Ensure that buffer alignment matches channelcount
// Ensure that buffer alignment matches channelCount
int channelCount = popcount(channelMask);
if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) {
ALOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount);
@@ -979,7 +979,8 @@ ssize_t AudioTrack::write(const void* buffer, size_t userSize)
if (mSharedBuffer != 0) return INVALID_OPERATION;
if (ssize_t(userSize) < 0) {
// sanity-check. user is most-likely passing an error code.
// Sanity-check: user is most-likely passing an error code, and it would
// make the return value ambiguous (actualSize vs error).
ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)",
buffer, userSize, userSize);
return BAD_VALUE;
@@ -1002,8 +1003,6 @@ ssize_t AudioTrack::write(const void* buffer, size_t userSize)
do {
audioBuffer.frameCount = userSize/frameSz;
// Calling obtainBuffer() with a negative wait count causes
// an (almost) infinite wait time.
status_t err = obtainBuffer(&audioBuffer, -1);
if (err < 0) {
// out of buffers, return #bytes written
@@ -1093,6 +1092,9 @@ bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
frames = mRemainingFrames;
}
// See description of waitCount parameter at declaration of obtainBuffer().
// The logic below prevents us from being stuck below at obtainBuffer()
// not being able to handle timed events (position, markers, loops).
int32_t waitCount = -1;
if (mUpdatePeriod || (!mMarkerReached && mMarkerPosition) || mLoopCount) {
waitCount = 1;
@@ -1102,9 +1104,6 @@ bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
audioBuffer.frameCount = frames;
// Calling obtainBuffer() with a wait count of 1
// limits wait time to WAIT_PERIOD_MS. This prevents from being
// stuck here not being able to handle timed events (position, markers, loops).
status_t err = obtainBuffer(&audioBuffer, waitCount);
if (err < NO_ERROR) {
if (err != TIMED_OUT) {

2
media/libmedia/IAudioTrack.cpp
View File

@@ -1,4 +1,4 @@
/* //device/extlibs/pv/android/IAudioTrack.cpp
/*
**
** Copyright 2007, The Android Open Source Project
**

2
media/libmedia/ToneGenerator.cpp
View File

@@ -791,7 +791,7 @@ const unsigned char /*tone_type*/ ToneGenerator::sToneMappingTable[NUM_REGIONS-1
// generators, instantiates output audio track.
//
// Input:
// streamType: Type of stream used for tone playback (enum AudioTrack::stream_type)
// streamType: Type of stream used for tone playback
// volume: volume applied to tone (0.0 to 1.0)
//
// Output:

2
media/libmedia/mediaplayer.cpp
View File

@@ -1,4 +1,4 @@
/* mediaplayer.cpp
/*
**
** Copyright 2006, The Android Open Source Project
**

10
services/audioflinger/AudioFlinger.cpp
View File

@@ -1,4 +1,4 @@
/* //device/include/server/AudioFlinger/AudioFlinger.cpp
/*
**
** Copyright 2007, The Android Open Source Project
**
@@ -621,6 +621,7 @@ status_t AudioFlinger::setMasterMute(bool muted)
}
Mutex::Autolock _l(mLock);
// This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger
mMasterMute = muted;
for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
mPlaybackThreads.valueAt(i)->setMasterMute(muted);
@@ -3121,6 +3122,7 @@ bool AudioFlinger::DuplicatingThread::threadLoop()
void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
{
// FIXME explain this formula
int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
this,
@@ -3392,7 +3394,7 @@ void AudioFlinger::PlaybackThread::Track::destroy()
{
// NOTE: destroyTrack_l() can remove a strong reference to this Track
// by removing it from mTracks vector, so there is a risk that this Tracks's
// desctructor is called. As the destructor needs to lock mLock,
// destructor is called. As the destructor needs to lock mLock,
// we must acquire a strong reference on this Track before locking mLock
// here so that the destructor is called only when exiting this function.
// On the other hand, as long as Track::destroy() is only called by
@@ -3998,6 +4000,7 @@ void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
: RefBase(),
mAudioFlinger(audioFlinger),
// FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
mPid(pid)
{
@@ -4659,7 +4662,7 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l()
}
if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
// do not accept frame count changes if tracks are open as the track buffer
// size depends on frame count and correct behavior would not be garantied
// size depends on frame count and correct behavior would not be guaranteed
// if frame count is changed after track creation
if (mActiveTrack != 0) {
status = INVALID_OPERATION;
@@ -6107,7 +6110,6 @@ status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle)
status_t status;
Mutex::Autolock _l(mLock);
// First handle in mHandles has highest priority and controls the effect module
int priority = handle->priority();
size_t size = mHandles.size();
sp<EffectHandle> h;

16
services/audioflinger/AudioFlinger.h
View File

@@ -1,4 +1,4 @@
/* //device/include/server/AudioFlinger/AudioFlinger.h
/*
**
** Copyright 2007, The Android Open Source Project
**
@@ -290,6 +290,8 @@ private:
enum track_state {
IDLE,
TERMINATED,
// These are order-sensitive; do not change order without reviewing the impact.
// In particular there are assumptions about > STOPPED.
STOPPED,
RESUMING,
ACTIVE,
@@ -760,6 +762,9 @@ private:
int mSuspended;
int mBytesWritten;
private:
// mMasterMute is in both PlaybackThread and in AudioFlinger. When a
// PlaybackThread needs to find out if master-muted, it checks it's local
// copy rather than the one in AudioFlinger. This optimization saves a lock.
bool mMasterMute;
protected:
SortedVector< wp<Track> > mActiveTracks;
@@ -853,6 +858,8 @@ private:
private:
void applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp);
// volumes last sent to audio HAL with stream->set_volume()
// FIXME use standard representation and names
float mLeftVolFloat;
float mRightVolFloat;
uint16_t mLeftVolShort;
@@ -900,6 +907,7 @@ private:
friend class AudioBuffer;
// server side of the client's IAudioTrack
class TrackHandle : public android::BnAudioTrack {
public:
TrackHandle(const sp<PlaybackThread::Track>& track);
@@ -1024,6 +1032,7 @@ private:
ssize_t mBytesRead;
};
// server side of the client's IAudioRecord
class RecordHandle : public android::BnAudioRecord {
public:
RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack);
@@ -1152,6 +1161,7 @@ mutable Mutex mLock; // mutex for process, commands and handl
status_t mStatus; // initialization status
effect_state mState; // current activation state
Vector< wp<EffectHandle> > mHandles; // list of client handles
// First handle in mHandles has highest priority and controls the effect module
uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after
// sending disable command.
uint32_t mDisableWaitCnt; // current process() calls count during disable period.
@@ -1377,6 +1387,7 @@ mutable Mutex mLock; // mutex for process, commands and handl
hwDev(dev), stream(in) {}
};
// for mAudioSessionRefs only
struct AudioSessionRef {
// FIXME rename parameter names when fields get "m" prefix
AudioSessionRef(int sessionid_, pid_t pid_) :
@@ -1432,10 +1443,11 @@ mutable Mutex mLock; // mutex for process, commands and handl
DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads;
DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients;
volatile int32_t mNextUniqueId;
volatile int32_t mNextUniqueId; // updated by android_atomic_inc
audio_mode_t mMode;
bool mBtNrecIsOff;
// protected by mLock
Vector<AudioSessionRef*> mAudioSessionRefs;
float masterVolume_l() const { return mMasterVolume; }

7
services/audioflinger/AudioMixer.cpp
View File

@@ -1,4 +1,4 @@
/* //device/include/server/AudioFlinger/AudioMixer.cpp
/*
**
** Copyright 2007, The Android Open Source Project
**
@@ -961,7 +961,12 @@ void AudioMixer::process__genericResampling(state_t* state)
// one track, 16 bits stereo without resampling is the most common case
void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state)
{
// This method is only called when state->enabledTracks has exactly
// one bit set. The asserts below would verify this, but are commented out
// since the whole point of this method is to optimize performance.
//assert(0 != state->enabledTracks);
const int i = 31 - __builtin_clz(state->enabledTracks);
//assert((1 << i) == state->enabledTracks);
const track_t& t = state->tracks[i];
AudioBufferProvider::Buffer& b(t.buffer);

2
services/audioflinger/AudioMixer.h
View File

@@ -1,4 +1,4 @@
/* //device/include/server/AudioFlinger/AudioMixer.h
/*
**
** Copyright 2007, The Android Open Source Project
**