Update comments
We no longer put the filename at start of file.
Change-Id: Ic435b159a23105681e3d4a6cb1ac097bc853302e
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index da99620..1916ac5 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -131,7 +131,7 @@
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:
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 11db81b..0f2c323 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -58,8 +58,8 @@
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 @@
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 @@
/* 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 @@
*/
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 @@
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 @@
};
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 @@
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 @@
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
diff --git a/include/media/EffectsFactoryApi.h b/include/media/EffectsFactoryApi.h
index df83995..65c26f4 100644
--- a/include/media/EffectsFactoryApi.h
+++ b/include/media/EffectsFactoryApi.h
@@ -87,7 +87,7 @@
// 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 @@
//
// 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);
diff --git a/include/media/IAudioTrack.h b/include/media/IAudioTrack.h
index b346722..e4772a1 100644
--- a/include/media/IAudioTrack.h
+++ b/include/media/IAudioTrack.h
@@ -46,12 +46,12 @@
/* 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 @@
/* 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;
diff --git a/include/media/MemoryLeakTrackUtil.h b/include/media/MemoryLeakTrackUtil.h
index 290b748..ac0f6b2 100644
--- a/include/media/MemoryLeakTrackUtil.h
+++ b/include/media/MemoryLeakTrackUtil.h
@@ -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);
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 23226c0..af2db93 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -62,16 +62,23 @@
// 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 @@
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
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 2518921..db18b36 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -1,4 +1,4 @@
-/* frameworks/base/media/libmedia/AudioTrack.cpp
+/*
**
** Copyright 2007, The Android Open Source Project
**
@@ -792,7 +792,7 @@
}
}
} 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 @@
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 @@
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 @@
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 @@
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) {
diff --git a/media/libmedia/IAudioTrack.cpp b/media/libmedia/IAudioTrack.cpp
index 3724642..a7958de 100644
--- a/media/libmedia/IAudioTrack.cpp
+++ b/media/libmedia/IAudioTrack.cpp
@@ -1,4 +1,4 @@
-/* //device/extlibs/pv/android/IAudioTrack.cpp
+/*
**
** Copyright 2007, The Android Open Source Project
**
diff --git a/media/libmedia/ToneGenerator.cpp b/media/libmedia/ToneGenerator.cpp
index 6cb10aa..54eb98a 100644
--- a/media/libmedia/ToneGenerator.cpp
+++ b/media/libmedia/ToneGenerator.cpp
@@ -791,7 +791,7 @@
// 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:
diff --git a/media/libmedia/mediaplayer.cpp b/media/libmedia/mediaplayer.cpp
index f1c47dd..250425b 100644
--- a/media/libmedia/mediaplayer.cpp
+++ b/media/libmedia/mediaplayer.cpp
@@ -1,4 +1,4 @@
-/* mediaplayer.cpp
+/*
**
** Copyright 2006, The Android Open Source Project
**
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index fb163d1..131371e 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1,4 +1,4 @@
-/* //device/include/server/AudioFlinger/AudioFlinger.cpp
+/*
**
** Copyright 2007, The Android Open Source Project
**
@@ -621,6 +621,7 @@
}
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 @@
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 @@
{
// 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 @@
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 @@
}
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 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;
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index aa0ee76..48a23fa 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -1,4 +1,4 @@
-/* //device/include/server/AudioFlinger/AudioFlinger.h
+/*
**
** Copyright 2007, The Android Open Source Project
**
@@ -290,6 +290,8 @@
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 @@
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:
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 @@
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 @@
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 @@
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 @@
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 @@
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; }
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 191520a..cb7678b 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -1,4 +1,4 @@
-/* //device/include/server/AudioFlinger/AudioMixer.cpp
+/*
**
** Copyright 2007, The Android Open Source Project
**
@@ -961,7 +961,12 @@
// 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);
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index c709686..c956918 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -1,4 +1,4 @@
-/* //device/include/server/AudioFlinger/AudioMixer.h
+/*
**
** Copyright 2007, The Android Open Source Project
**