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-rw-r--r--services/audioflinger/AudioFlinger.cpp150
-rw-r--r--services/audioflinger/AudioFlinger.h69
-rw-r--r--services/audioflinger/AudioMixer.cpp12
-rw-r--r--services/audioflinger/AudioPolicyService.cpp5
4 files changed, 125 insertions, 111 deletions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 09d7954..1a2fa66 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -160,7 +160,10 @@ static const char * const audio_interfaces[] = {
AudioFlinger::AudioFlinger()
: BnAudioFlinger(),
- mPrimaryHardwareDev(NULL), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1),
+ mPrimaryHardwareDev(NULL),
+ mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef()
+ mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1),
+ mMode(AUDIO_MODE_INVALID),
mBtNrecIsOff(false)
{
}
@@ -172,7 +175,6 @@ void AudioFlinger::onFirstRef()
Mutex::Autolock _l(mLock);
/* TODO: move all this work into an Init() function */
- mHardwareStatus = AUDIO_HW_IDLE;
for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
const hw_module_t *mod;
@@ -265,13 +267,10 @@ status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
result.append("Clients:\n");
for (size_t i = 0; i < mClients.size(); ++i) {
- wp<Client> wClient = mClients.valueAt(i);
- if (wClient != 0) {
- sp<Client> client = wClient.promote();
- if (client != 0) {
- snprintf(buffer, SIZE, " pid: %d\n", client->pid());
- result.append(buffer);
- }
+ sp<Client> client = mClients.valueAt(i).promote();
+ if (client != 0) {
+ snprintf(buffer, SIZE, " pid: %d\n", client->pid());
+ result.append(buffer);
}
}
@@ -971,7 +970,8 @@ void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2)
{
size_t size = mNotificationClients.size();
for (size_t i = 0; i < size; i++) {
- mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2);
+ mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle,
+ param2);
}
}
@@ -985,13 +985,19 @@ void AudioFlinger::removeClient_l(pid_t pid)
// ----------------------------------------------------------------------------
-AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device)
+AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device,
+ type_t type)
: Thread(false),
- mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0),
- mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), mStandby(false), mId(id), mExiting(false),
- mDevice(device)
+ mType(type),
+ mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0),
+ // mChannelMask
+ mChannelCount(0),
+ mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID),
+ mParamStatus(NO_ERROR),
+ mStandby(false), mId(id), mExiting(false),
+ mDevice(device),
+ mDeathRecipient(new PMDeathRecipient(this))
{
- mDeathRecipient = new PMDeathRecipient(this);
}
AudioFlinger::ThreadBase::~ThreadBase()
@@ -1372,20 +1378,24 @@ void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModu
AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output,
int id,
- uint32_t device)
- : ThreadBase(audioFlinger, id, device),
- mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), mOutput(output),
+ uint32_t device,
+ type_t type)
+ : ThreadBase(audioFlinger, id, device, type),
+ mMixBuffer(NULL), mSuspended(0), mBytesWritten(0),
+ // Assumes constructor is called by AudioFlinger with it's mLock held,
+ // but it would be safer to explicitly pass initial masterMute as parameter
+ mMasterMute(audioFlinger->masterMute_l()),
+ // mStreamTypes[] initialized in constructor body
+ mOutput(output),
+ // Assumes constructor is called by AudioFlinger with it's mLock held,
+ // but it would be safer to explicitly pass initial masterVolume as parameter
+ mMasterVolume(audioFlinger->masterVolume_l()),
mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false)
{
snprintf(mName, kNameLength, "AudioOut_%d", id);
readOutputParameters();
- // Assumes constructor is called by AudioFlinger with it's mLock held,
- // but it would be safer to explicitly pass these as parameters
- mMasterVolume = mAudioFlinger->masterVolume_l();
- mMasterMute = mAudioFlinger->masterMute_l();
-
// mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor
// There is no AUDIO_STREAM_MIN, and ++ operator does not compile
for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT;
@@ -1431,13 +1441,10 @@ status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>
result.append(buffer);
result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n");
for (size_t i = 0; i < mActiveTracks.size(); ++i) {
- wp<Track> wTrack = mActiveTracks[i];
- if (wTrack != 0) {
- sp<Track> track = wTrack.promote();
- if (track != 0) {
- track->dump(buffer, SIZE);
- result.append(buffer);
- }
+ sp<Track> track = mActiveTracks[i].promote();
+ if (track != 0) {
+ track->dump(buffer, SIZE);
+ result.append(buffer);
}
}
write(fd, result.string(), result.size());
@@ -1740,7 +1747,7 @@ void AudioFlinger::PlaybackThread::readOutputParameters()
// FIXME - Current mixer implementation only supports stereo output: Always
// Allocate a stereo buffer even if HW output is mono.
- if (mMixBuffer != NULL) delete[] mMixBuffer;
+ delete[] mMixBuffer;
mMixBuffer = new int16_t[mFrameCount * 2];
memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
@@ -1845,13 +1852,12 @@ uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs()
// ----------------------------------------------------------------------------
-AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
- : PlaybackThread(audioFlinger, output, id, device),
- mAudioMixer(NULL), mPrevMixerStatus(MIXER_IDLE)
+AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
+ int id, uint32_t device, type_t type)
+ : PlaybackThread(audioFlinger, output, id, device, type),
+ mAudioMixer(new AudioMixer(mFrameCount, mSampleRate)),
+ mPrevMixerStatus(MIXER_IDLE)
{
- mType = ThreadBase::MIXER;
- mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
-
// FIXME - Current mixer implementation only supports stereo output
if (mChannelCount == 1) {
ALOGE("Invalid audio hardware channel count");
@@ -2193,7 +2199,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
// read original volumes with volume control
float typeVolume = mStreamTypes[track->type()].volume;
float v = masterVolume * typeVolume;
- uint32_t vlr = cblk->volumeLR;
+ uint32_t vlr = cblk->getVolumeLR();
vl = vlr & 0xFFFF;
vr = vlr >> 16;
// track volumes come from shared memory, so can't be trusted and must be clamped
@@ -2461,6 +2467,8 @@ bool AudioFlinger::MixerThread::checkForNewParameters_l()
}
if (status == NO_ERROR && reconfig) {
delete mAudioMixer;
+ // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't)
+ mAudioMixer = NULL;
readOutputParameters();
mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
for (size_t i = 0; i < mTracks.size() ; i++) {
@@ -2513,9 +2521,10 @@ uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs()
// ----------------------------------------------------------------------------
AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
- : PlaybackThread(audioFlinger, output, id, device)
+ : PlaybackThread(audioFlinger, output, id, device, DIRECT)
+ // mLeftVolFloat, mRightVolFloat
+ // mLeftVolShort, mRightVolShort
{
- mType = ThreadBase::DIRECT;
}
AudioFlinger::DirectOutputThread::~DirectOutputThread()
@@ -2729,7 +2738,7 @@ bool AudioFlinger::DirectOutputThread::threadLoop()
} else {
float typeVolume = mStreamTypes[track->type()].volume;
float v = mMasterVolume * typeVolume;
- uint32_t vlr = cblk->volumeLR;
+ uint32_t vlr = cblk->getVolumeLR();
float v_clamped = v * (vlr & 0xFFFF);
if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
left = v_clamped/MAX_GAIN;
@@ -2992,10 +3001,11 @@ uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs()
// ----------------------------------------------------------------------------
-AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id)
- : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX)
+AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
+ AudioFlinger::MixerThread* mainThread, int id)
+ : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING),
+ mWaitTimeMs(UINT_MAX)
{
- mType = ThreadBase::DUPLICATING;
addOutputTrack(mainThread);
}
@@ -3244,13 +3254,17 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
: RefBase(),
mThread(thread),
mClient(client),
- mCblk(0),
+ mCblk(NULL),
+ // mBuffer
+ // mBufferEnd
mFrameCount(0),
mState(IDLE),
mClientTid(-1),
mFormat(format),
mFlags(flags & ~SYSTEM_FLAGS_MASK),
mSessionId(sessionId)
+ // mChannelCount
+ // mChannelMask
{
ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
@@ -3317,6 +3331,7 @@ AudioFlinger::ThreadBase::TrackBase::~TrackBase()
}
mCblkMemory.clear(); // and free the shared memory
if (mClient != NULL) {
+ // Client destructor must run with AudioFlinger mutex locked
Mutex::Autolock _l(mClient->audioFlinger()->mLock);
mClient.clear();
}
@@ -3468,7 +3483,7 @@ void AudioFlinger::PlaybackThread::Track::destroy()
void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
{
- uint32_t vlr = mCblk->volumeLR;
+ uint32_t vlr = mCblk->getVolumeLR();
snprintf(buffer, size, " %05d %05d %03u %03u 0x%08x %05u %04u %1d %1d %1d %05u %05u %05u 0x%08x 0x%08x 0x%08x 0x%08x\n",
mName - AudioMixer::TRACK0,
(mClient == NULL) ? getpid() : mClient->pid(),
@@ -3827,7 +3842,6 @@ AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
if (mCblk != NULL) {
mCblk->flags |= CBLK_DIRECTION_OUT;
mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
- mCblk->volumeLR = (MAX_GAIN_INT << 16) | MAX_GAIN_INT;
mOutBuffer.frameCount = 0;
playbackThread->mTracks.add(this);
ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \
@@ -4065,7 +4079,7 @@ AudioFlinger::Client::~Client()
mAudioFlinger->removeClient_l(mPid);
}
-const sp<MemoryDealer>& AudioFlinger::Client::heap() const
+sp<MemoryDealer> AudioFlinger::Client::heap() const
{
return mMemoryDealer;
}
@@ -4075,13 +4089,12 @@ const sp<MemoryDealer>& AudioFlinger::Client::heap() const
AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
const sp<IAudioFlingerClient>& client,
pid_t pid)
- : mAudioFlinger(audioFlinger), mPid(pid), mClient(client)
+ : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
{
}
AudioFlinger::NotificationClient::~NotificationClient()
{
- mClient.clear();
}
void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
@@ -4266,15 +4279,16 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
uint32_t channels,
int id,
uint32_t device) :
- ThreadBase(audioFlinger, id, device),
- mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL)
+ ThreadBase(audioFlinger, id, device, RECORD),
+ mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL),
+ // mRsmpInIndex and mInputBytes set by readInputParameters()
+ mReqChannelCount(popcount(channels)),
+ mReqSampleRate(sampleRate)
+ // mBytesRead is only meaningful while active, and so is cleared in start()
+ // (but might be better to also clear here for dump?)
{
- mType = ThreadBase::RECORD;
-
snprintf(mName, kNameLength, "AudioIn_%d", id);
- mReqChannelCount = popcount(channels);
- mReqSampleRate = sampleRate;
readInputParameters();
}
@@ -4282,10 +4296,8 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
AudioFlinger::RecordThread::~RecordThread()
{
delete[] mRsmpInBuffer;
- if (mResampler != NULL) {
- delete mResampler;
- delete[] mRsmpOutBuffer;
- }
+ delete mResampler;
+ delete[] mRsmpOutBuffer;
}
void AudioFlinger::RecordThread::onFirstRef()
@@ -4829,9 +4841,11 @@ void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
void AudioFlinger::RecordThread::readInputParameters()
{
- if (mRsmpInBuffer) delete mRsmpInBuffer;
- if (mRsmpOutBuffer) delete mRsmpOutBuffer;
- if (mResampler) delete mResampler;
+ delete mRsmpInBuffer;
+ // mRsmpInBuffer is always assigned a new[] below
+ delete mRsmpOutBuffer;
+ mRsmpOutBuffer = NULL;
+ delete mResampler;
mResampler = NULL;
mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
@@ -5243,12 +5257,8 @@ void AudioFlinger::acquireAudioSessionId(int audioSession)
return;
}
}
- AudioSessionRef *ref = new AudioSessionRef();
- ref->sessionid = audioSession;
- ref->pid = caller;
- ref->cnt = 1;
- mAudioSessionRefs.push(ref);
- ALOGV(" added new entry for %d", ref->sessionid);
+ mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
+ ALOGV(" added new entry for %d", audioSession);
}
void AudioFlinger::releaseAudioSessionId(int audioSession)
@@ -6179,7 +6189,7 @@ AudioFlinger::EffectModule::~EffectModule()
}
}
-status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle)
+status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle)
{
status_t status;
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 16048c8..4156da8 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -226,16 +226,16 @@ private:
public:
Client(const sp<AudioFlinger>& audioFlinger, pid_t pid);
virtual ~Client();
- const sp<MemoryDealer>& heap() 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;
+ const sp<AudioFlinger> mAudioFlinger;
+ const sp<MemoryDealer> mMemoryDealer;
+ const pid_t mPid;
};
// --- Notification Client ---
@@ -246,7 +246,7 @@ private:
pid_t pid);
virtual ~NotificationClient();
- sp<IAudioFlingerClient> client() { return mClient; }
+ sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; }
// IBinder::DeathRecipient
virtual void binderDied(const wp<IBinder>& who);
@@ -255,9 +255,9 @@ private:
NotificationClient(const NotificationClient&);
NotificationClient& operator = (const NotificationClient&);
- sp<AudioFlinger> mAudioFlinger;
- pid_t mPid;
- sp<IAudioFlingerClient> mClient;
+ const sp<AudioFlinger> mAudioFlinger;
+ const pid_t mPid;
+ const sp<IAudioFlingerClient> mAudioFlingerClient;
};
class TrackHandle;
@@ -277,17 +277,17 @@ private:
class ThreadBase : public Thread {
public:
- ThreadBase (const sp<AudioFlinger>& audioFlinger, int id, uint32_t device);
- virtual ~ThreadBase();
-
- enum type {
+ enum type_t {
MIXER, // Thread class is MixerThread
DIRECT, // Thread class is DirectOutputThread
DUPLICATING, // Thread class is DuplicatingThread
RECORD // Thread class is RecordThread
};
+ ThreadBase (const sp<AudioFlinger>& audioFlinger, int id, uint32_t device, type_t type);
+ virtual ~ThreadBase();
+
status_t dumpBase(int fd, const Vector<String16>& args);
status_t dumpEffectChains(int fd, const Vector<String16>& args);
@@ -367,8 +367,8 @@ private:
bool step();
void reset();
- wp<ThreadBase> mThread;
- sp<Client> mClient;
+ const wp<ThreadBase> mThread;
+ /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const
sp<IMemory> mCblkMemory;
audio_track_cblk_t* mCblk;
void* mBuffer;
@@ -377,9 +377,9 @@ private:
// we don't really need a lock for these
track_state mState;
int mClientTid;
- audio_format_t mFormat;
+ const audio_format_t mFormat;
uint32_t mFlags;
- int mSessionId;
+ const int mSessionId;
uint8_t mChannelCount;
uint32_t mChannelMask;
};
@@ -408,7 +408,7 @@ private:
};
virtual status_t initCheck() const = 0;
- int type() const { return mType; }
+ type_t type() const { return mType; }
uint32_t sampleRate() const;
int channelCount() const;
audio_format_t format() const;
@@ -530,9 +530,9 @@ private:
friend class RecordThread;
friend class RecordTrack;
- int mType;
+ const type_t mType;
Condition mWaitWorkCV;
- sp<AudioFlinger> mAudioFlinger;
+ const sp<AudioFlinger> mAudioFlinger;
uint32_t mSampleRate;
size_t mFrameCount;
uint32_t mChannelMask;
@@ -553,7 +553,7 @@ private:
char mName[kNameLength];
sp<IPowerManager> mPowerManager;
sp<IBinder> mWakeLockToken;
- sp<PMDeathRecipient> mDeathRecipient;
+ const sp<PMDeathRecipient> mDeathRecipient;
// list of suspended effects per session and per type. The first vector is
// keyed by session ID, the second by type UUID timeLow field
KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions;
@@ -671,7 +671,7 @@ private:
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; }
+ const wp<ThreadBase>& thread() { return mThread; }
private:
@@ -688,10 +688,11 @@ private:
Vector < Buffer* > mBufferQueue;
AudioBufferProvider::Buffer mOutBuffer;
bool mActive;
- DuplicatingThread* mSourceThread;
+ DuplicatingThread* const mSourceThread; // for waitTimeMs() in write()
}; // end of OutputTrack
- PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device);
+ PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id,
+ uint32_t device, type_t type);
virtual ~PlaybackThread();
virtual status_t dump(int fd, const Vector<String16>& args);
@@ -817,7 +818,8 @@ private:
MixerThread (const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output,
int id,
- uint32_t device);
+ uint32_t device,
+ type_t type = MIXER);
virtual ~MixerThread();
// Thread virtuals
@@ -917,7 +919,7 @@ private:
virtual status_t onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
private:
- sp<PlaybackThread::Track> mTrack;
+ const sp<PlaybackThread::Track> mTrack;
};
friend class Client;
@@ -1021,8 +1023,8 @@ private:
int16_t *mRsmpInBuffer;
size_t mRsmpInIndex;
size_t mInputBytes;
- int mReqChannelCount;
- uint32_t mReqSampleRate;
+ const int mReqChannelCount;
+ const uint32_t mReqSampleRate;
ssize_t mBytesRead;
};
@@ -1036,7 +1038,7 @@ private:
virtual status_t onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
private:
- sp<RecordThread::RecordTrack> mRecordTrack;
+ const sp<RecordThread::RecordTrack> mRecordTrack;
};
//--- Audio Effect Management
@@ -1105,9 +1107,9 @@ private:
int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; }
void setChain(const wp<EffectChain>& chain) { mChain = chain; }
void setThread(const wp<ThreadBase>& thread) { mThread = thread; }
- wp<ThreadBase>& thread() { return mThread; }
+ const wp<ThreadBase>& thread() { return mThread; }
- status_t addHandle(sp<EffectHandle>& handle);
+ status_t addHandle(const sp<EffectHandle>& handle);
void disconnect(const wp<EffectHandle>& handle, bool unpiniflast);
size_t removeHandle (const wp<EffectHandle>& handle);
@@ -1378,8 +1380,11 @@ mutable Mutex mLock; // mutex for process, commands and handl
};
struct AudioSessionRef {
- int sessionid;
- pid_t pid;
+ // FIXME rename parameter names when fields get "m" prefix
+ AudioSessionRef(int sessionid_, pid_t pid_) :
+ sessionid(sessionid_), pid(pid_), cnt(1) {}
+ const int sessionid;
+ const pid_t pid;
int cnt;
};
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index a8102e5..1f7d883 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -48,9 +48,10 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate)
mState.enabledTracks= 0;
mState.needsChanged = 0;
mState.frameCount = frameCount;
+ mState.hook = process__nop;
mState.outputTemp = NULL;
mState.resampleTemp = NULL;
- mState.hook = process__nop;
+ // mState.reserved
track_t* t = mState.tracks;
for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) {
t->needs = 0;
@@ -70,12 +71,13 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate)
t->enabled = 0;
t->format = 16;
t->channelMask = AUDIO_CHANNEL_OUT_STEREO;
- t->buffer.raw = 0;
t->bufferProvider = NULL;
+ t->buffer.raw = NULL;
+ // t->buffer.frameCount
t->hook = NULL;
+ t->in = NULL;
t->resampler = NULL;
t->sampleRate = mSampleRate;
- t->in = NULL;
t->mainBuffer = NULL;
t->auxBuffer = NULL;
t++;
@@ -1127,9 +1129,7 @@ void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state)
}
}
- if (buff != NULL) {
- delete [] buff;
- }
+ delete [] buff;
}
#endif
diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp
index 2df1385..7695d2b 100644
--- a/services/audioflinger/AudioPolicyService.cpp
+++ b/services/audioflinger/AudioPolicyService.cpp
@@ -649,7 +649,7 @@ AudioPolicyService::AudioCommandThread::~AudioCommandThread()
release_wake_lock(mName.string());
}
mAudioCommands.clear();
- if (mpToneGenerator != NULL) delete mpToneGenerator;
+ delete mpToneGenerator;
}
void AudioPolicyService::AudioCommandThread::onFirstRef()
@@ -682,8 +682,7 @@ bool AudioPolicyService::AudioCommandThread::threadLoop()
ToneData *data = (ToneData *)command->mParam;
ALOGV("AudioCommandThread() processing start tone %d on stream %d",
data->mType, data->mStream);
- if (mpToneGenerator != NULL)
- delete mpToneGenerator;
+ delete mpToneGenerator;
mpToneGenerator = new ToneGenerator(data->mStream, 1.0);
mpToneGenerator->startTone(data->mType);
delete data;
generated by cgit v1.1 at 2025-08-04 23:32:51 +0000