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-rw-r--r--media/libmedia/AudioEffect.cpp13
-rw-r--r--media/libmedia/AudioRecord.cpp229
-rw-r--r--media/libmedia/AudioSystem.cpp29
-rw-r--r--media/libmedia/AudioTrack.cpp48
-rw-r--r--media/libmedia/IAudioFlinger.cpp30
-rw-r--r--media/libmedia/IAudioPolicyService.cpp25
-rw-r--r--media/libmedia/IAudioRecord.cpp4
-rw-r--r--media/libmedia/Visualizer.cpp9
8 files changed, 190 insertions, 197 deletions
diff --git a/media/libmedia/AudioEffect.cpp b/media/libmedia/AudioEffect.cpp
index 34451ca..680604b 100644
--- a/media/libmedia/AudioEffect.cpp
+++ b/media/libmedia/AudioEffect.cpp
@@ -122,19 +122,12 @@ status_t AudioEffect::set(const effect_uuid_t *type,
mSessionId = sessionId;
memset(&mDescriptor, 0, sizeof(effect_descriptor_t));
- memcpy(&mDescriptor.type, EFFECT_UUID_NULL, sizeof(effect_uuid_t));
- memcpy(&mDescriptor.uuid, EFFECT_UUID_NULL, sizeof(effect_uuid_t));
-
- if (type != NULL) {
- memcpy(&mDescriptor.type, type, sizeof(effect_uuid_t));
- }
- if (uuid != NULL) {
- memcpy(&mDescriptor.uuid, uuid, sizeof(effect_uuid_t));
- }
+ mDescriptor.type = *(type != NULL ? type : EFFECT_UUID_NULL);
+ mDescriptor.uuid = *(uuid != NULL ? uuid : EFFECT_UUID_NULL);
mIEffectClient = new EffectClient(this);
- iEffect = audioFlinger->createEffect(getpid(), (effect_descriptor_t *)&mDescriptor,
+ iEffect = audioFlinger->createEffect(getpid(), &mDescriptor,
mIEffectClient, priority, io, mSessionId, &mStatus, &mId, &enabled);
if (iEffect == 0 || (mStatus != NO_ERROR && mStatus != ALREADY_EXISTS)) {
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 0562f8e..5060525 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -18,28 +18,18 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "AudioRecord"
-#include <stdint.h>
-#include <sys/types.h>
-
-#include <sched.h>
#include <sys/resource.h>
+#include <sys/types.h>
-#include <private/media/AudioTrackShared.h>
-
-#include <media/AudioSystem.h>
+#include <binder/IPCThreadState.h>
+#include <cutils/atomic.h>
+#include <cutils/compiler.h>
#include <media/AudioRecord.h>
-#include <media/mediarecorder.h>
-
-#include <binder/IServiceManager.h>
+#include <media/AudioSystem.h>
+#include <system/audio.h>
#include <utils/Log.h>
-#include <binder/Parcel.h>
-#include <binder/IPCThreadState.h>
-#include <utils/Timers.h>
-#include <utils/Atomic.h>
-#include <system/audio.h>
-#include <cutils/bitops.h>
-#include <cutils/compiler.h>
+#include <private/media/AudioTrackShared.h>
namespace android {
// ---------------------------------------------------------------------------
@@ -49,18 +39,23 @@ status_t AudioRecord::getMinFrameCount(
int* frameCount,
uint32_t sampleRate,
audio_format_t format,
- int channelCount)
+ audio_channel_mask_t channelMask)
{
+ if (frameCount == NULL) return BAD_VALUE;
+
+ // default to 0 in case of error
+ *frameCount = 0;
+
size_t size = 0;
- if (AudioSystem::getInputBufferSize(sampleRate, format, channelCount, &size)
+ if (AudioSystem::getInputBufferSize(sampleRate, format, channelMask, &size)
!= NO_ERROR) {
ALOGE("AudioSystem could not query the input buffer size.");
return NO_INIT;
}
if (size == 0) {
- ALOGE("Unsupported configuration: sampleRate %d, format %d, channelCount %d",
- sampleRate, format, channelCount);
+ ALOGE("Unsupported configuration: sampleRate %d, format %d, channelMask %#x",
+ sampleRate, format, channelMask);
return BAD_VALUE;
}
@@ -68,6 +63,7 @@ status_t AudioRecord::getMinFrameCount(
size <<= 1;
if (audio_is_linear_pcm(format)) {
+ int channelCount = popcount(channelMask);
size /= channelCount * audio_bytes_per_sample(format);
}
@@ -87,9 +83,8 @@ AudioRecord::AudioRecord(
audio_source_t inputSource,
uint32_t sampleRate,
audio_format_t format,
- uint32_t channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
- record_flags flags,
callback_t cbf,
void* user,
int notificationFrames,
@@ -98,7 +93,7 @@ AudioRecord::AudioRecord(
mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT)
{
mStatus = set(inputSource, sampleRate, format, channelMask,
- frameCount, flags, cbf, user, notificationFrames, sessionId);
+ frameCount, cbf, user, notificationFrames, sessionId);
}
AudioRecord::~AudioRecord()
@@ -108,9 +103,10 @@ AudioRecord::~AudioRecord()
// it is looping on buffer empty condition in obtainBuffer().
// Otherwise the callback thread will never exit.
stop();
- if (mClientRecordThread != 0) {
- mClientRecordThread->requestExitAndWait();
- mClientRecordThread.clear();
+ if (mAudioRecordThread != 0) {
+ mAudioRecordThread->requestExit(); // see comment in AudioRecord.h
+ mAudioRecordThread->requestExitAndWait();
+ mAudioRecordThread.clear();
}
mAudioRecord.clear();
IPCThreadState::self()->flushCommands();
@@ -122,9 +118,8 @@ status_t AudioRecord::set(
audio_source_t inputSource,
uint32_t sampleRate,
audio_format_t format,
- uint32_t channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
- record_flags flags,
callback_t cbf,
void* user,
int notificationFrames,
@@ -132,7 +127,7 @@ status_t AudioRecord::set(
int sessionId)
{
- ALOGV("set(): sampleRate %d, channelMask %d, frameCount %d",sampleRate, channelMask, frameCount);
+ ALOGV("set(): sampleRate %d, channelMask %#x, frameCount %d",sampleRate, channelMask, frameCount);
AutoMutex lock(mLock);
@@ -174,7 +169,6 @@ status_t AudioRecord::set(
sampleRate,
format,
channelMask,
- (audio_in_acoustics_t)flags,
mSessionId);
if (input == 0) {
ALOGE("Could not get audio input for record source %d", inputSource);
@@ -207,7 +201,8 @@ status_t AudioRecord::set(
}
if (cbf != NULL) {
- mClientRecordThread = new ClientRecordThread(*this, threadCanCallJava);
+ mAudioRecordThread = new AudioRecordThread(*this, threadCanCallJava);
+ mAudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO);
}
mStatus = NO_ERROR;
@@ -217,7 +212,7 @@ status_t AudioRecord::set(
mFrameCount = mCblk->frameCount;
mChannelCount = (uint8_t)channelCount;
mChannelMask = channelMask;
- mActive = 0;
+ mActive = false;
mCbf = cbf;
mNotificationFrames = notificationFrames;
mRemainingFrames = notificationFrames;
@@ -229,7 +224,6 @@ status_t AudioRecord::set(
mNewPosition = 0;
mUpdatePeriod = 0;
mInputSource = inputSource;
- mFlags = flags;
mInput = input;
AudioSystem::acquireAudioSessionId(mSessionId);
@@ -282,41 +276,19 @@ audio_source_t AudioRecord::inputSource() const
status_t AudioRecord::start(AudioSystem::sync_event_t event, int triggerSession)
{
status_t ret = NO_ERROR;
- sp<ClientRecordThread> t = mClientRecordThread;
+ sp<AudioRecordThread> t = mAudioRecordThread;
ALOGV("start, sync event %d trigger session %d", event, triggerSession);
- if (t != 0) {
- if (t->exitPending()) {
- if (t->requestExitAndWait() == WOULD_BLOCK) {
- ALOGE("AudioRecord::start called from thread");
- return WOULD_BLOCK;
- }
- }
- }
-
AutoMutex lock(mLock);
// acquire a strong reference on the IAudioRecord and IMemory so that they cannot be destroyed
// while we are accessing the cblk
sp<IAudioRecord> audioRecord = mAudioRecord;
sp<IMemory> iMem = mCblkMemory;
audio_track_cblk_t* cblk = mCblk;
- if (mActive == 0) {
- mActive = 1;
- pid_t tid;
- if (t != 0) {
- mReadyToRun = WOULD_BLOCK;
- t->run("AudioRecord", ANDROID_PRIORITY_AUDIO);
- tid = t->getTid(); // pid_t is unknown until run()
- ALOGV("getTid=%d", tid);
- if (tid == -1) {
- tid = 0;
- }
- // thread blocks in readyToRun()
- } else {
- tid = 0; // not gettid()
- }
+ if (!mActive) {
+ mActive = true;
cblk->lock.lock();
if (!(cblk->flags & CBLK_INVALID_MSK)) {
@@ -338,58 +310,51 @@ status_t AudioRecord::start(AudioSystem::sync_event_t event, int triggerSession)
AudioSystem::kSyncRecordStartTimeOutMs;
cblk->waitTimeMs = 0;
if (t != 0) {
- // thread unblocks in readyToRun() and returns NO_ERROR
- mReadyToRun = NO_ERROR;
- mCondition.signal();
+ t->resume();
} else {
mPreviousPriority = getpriority(PRIO_PROCESS, 0);
get_sched_policy(0, &mPreviousSchedulingGroup);
androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
}
} else {
- mActive = 0;
- // thread unblocks in readyToRun() and returns NO_INIT
- mReadyToRun = NO_INIT;
- mCondition.signal();
+ mActive = false;
}
}
return ret;
}
-status_t AudioRecord::stop()
+void AudioRecord::stop()
{
- sp<ClientRecordThread> t = mClientRecordThread;
+ sp<AudioRecordThread> t = mAudioRecordThread;
ALOGV("stop");
AutoMutex lock(mLock);
- if (mActive == 1) {
- mActive = 0;
+ if (mActive) {
+ mActive = false;
mCblk->cv.signal();
mAudioRecord->stop();
// the record head position will reset to 0, so if a marker is set, we need
// to activate it again
mMarkerReached = false;
if (t != 0) {
- t->requestExit();
+ t->pause();
} else {
setpriority(PRIO_PROCESS, 0, mPreviousPriority);
set_sched_policy(0, mPreviousSchedulingGroup);
}
}
-
- return NO_ERROR;
}
bool AudioRecord::stopped() const
{
+ AutoMutex lock(mLock);
return !mActive;
}
uint32_t AudioRecord::getSampleRate() const
{
- AutoMutex lock(mLock);
return mCblk->sampleRate;
}
@@ -397,6 +362,7 @@ status_t AudioRecord::setMarkerPosition(uint32_t marker)
{
if (mCbf == NULL) return INVALID_OPERATION;
+ AutoMutex lock(mLock);
mMarkerPosition = marker;
mMarkerReached = false;
@@ -407,6 +373,7 @@ status_t AudioRecord::getMarkerPosition(uint32_t *marker) const
{
if (marker == NULL) return BAD_VALUE;
+ AutoMutex lock(mLock);
*marker = mMarkerPosition;
return NO_ERROR;
@@ -418,6 +385,8 @@ status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod)
uint32_t curPosition;
getPosition(&curPosition);
+
+ AutoMutex lock(mLock);
mNewPosition = curPosition + updatePeriod;
mUpdatePeriod = updatePeriod;
@@ -428,6 +397,7 @@ status_t AudioRecord::getPositionUpdatePeriod(uint32_t *updatePeriod) const
{
if (updatePeriod == NULL) return BAD_VALUE;
+ AutoMutex lock(mLock);
*updatePeriod = mUpdatePeriod;
return NO_ERROR;
@@ -445,10 +415,8 @@ status_t AudioRecord::getPosition(uint32_t *position) const
unsigned int AudioRecord::getInputFramesLost() const
{
- if (mActive)
- return AudioSystem::getInputFramesLost(mInput);
- else
- return 0;
+ // no need to check mActive, because if inactive this will return 0, which is what we want
+ return AudioSystem::getInputFramesLost(mInput);
}
// -------------------------------------------------------------------------
@@ -457,7 +425,7 @@ unsigned int AudioRecord::getInputFramesLost() const
status_t AudioRecord::openRecord_l(
uint32_t sampleRate,
audio_format_t format,
- uint32_t channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
audio_io_handle_t input)
{
@@ -467,13 +435,20 @@ status_t AudioRecord::openRecord_l(
return NO_INIT;
}
+ pid_t tid = -1;
+ // FIXME see similar logic at AudioTrack
+
+ int originalSessionId = mSessionId;
sp<IAudioRecord> record = audioFlinger->openRecord(getpid(), input,
sampleRate, format,
channelMask,
frameCount,
IAudioFlinger::TRACK_DEFAULT,
+ tid,
&mSessionId,
&status);
+ ALOGE_IF(originalSessionId != 0 && mSessionId != originalSessionId,
+ "session ID changed from %d to %d", originalSessionId, mSessionId);
if (record == 0) {
ALOGE("AudioFlinger could not create record track, status: %d", status);
@@ -499,7 +474,7 @@ status_t AudioRecord::openRecord_l(
status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
{
AutoMutex lock(mLock);
- int active;
+ bool active;
status_t result = NO_ERROR;
audio_track_cblk_t* cblk = mCblk;
uint32_t framesReq = audioBuffer->frameCount;
@@ -528,7 +503,7 @@ status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
cblk->lock.unlock();
mLock.lock();
- if (mActive == 0) {
+ if (!mActive) {
return status_t(STOPPED);
}
cblk->lock.lock();
@@ -613,13 +588,13 @@ audio_io_handle_t AudioRecord::getInput_l()
mCblk->sampleRate,
mFormat,
mChannelMask,
- (audio_in_acoustics_t)mFlags,
mSessionId);
return mInput;
}
int AudioRecord::getSessionId() const
{
+ // no lock needed because session ID doesn't change after first set()
return mSessionId;
}
@@ -678,7 +653,7 @@ ssize_t AudioRecord::read(void* buffer, size_t userSize)
// -------------------------------------------------------------------------
-bool AudioRecord::processAudioBuffer(const sp<ClientRecordThread>& thread)
+bool AudioRecord::processAudioBuffer(const sp<AudioRecordThread>& thread)
{
Buffer audioBuffer;
uint32_t frames = mRemainingFrames;
@@ -690,22 +665,32 @@ bool AudioRecord::processAudioBuffer(const sp<ClientRecordThread>& thread)
sp<IAudioRecord> audioRecord = mAudioRecord;
sp<IMemory> iMem = mCblkMemory;
audio_track_cblk_t* cblk = mCblk;
+ bool active = mActive;
+ uint32_t markerPosition = mMarkerPosition;
+ uint32_t newPosition = mNewPosition;
+ uint32_t user = cblk->user;
+ // determine whether a marker callback will be needed, while locked
+ bool needMarker = !mMarkerReached && (mMarkerPosition > 0) && (user >= mMarkerPosition);
+ if (needMarker) {
+ mMarkerReached = true;
+ }
+ // determine the number of new position callback(s) that will be needed, while locked
+ uint32_t updatePeriod = mUpdatePeriod;
+ uint32_t needNewPos = updatePeriod > 0 && user >= newPosition ?
+ ((user - newPosition) / updatePeriod) + 1 : 0;
+ mNewPosition = newPosition + updatePeriod * needNewPos;
mLock.unlock();
- // Manage marker callback
- if (!mMarkerReached && (mMarkerPosition > 0)) {
- if (cblk->user >= mMarkerPosition) {
- mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition);
- mMarkerReached = true;
- }
+ // perform marker callback, while unlocked
+ if (needMarker) {
+ mCbf(EVENT_MARKER, mUserData, &markerPosition);
}
- // Manage new position callback
- if (mUpdatePeriod > 0) {
- while (cblk->user >= mNewPosition) {
- mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition);
- mNewPosition += mUpdatePeriod;
- }
+ // perform new position callback(s), while unlocked
+ for (; needNewPos > 0; --needNewPos) {
+ uint32_t temp = newPosition;
+ mCbf(EVENT_NEW_POS, mUserData, &temp);
+ newPosition += updatePeriod;
}
do {
@@ -748,10 +733,12 @@ bool AudioRecord::processAudioBuffer(const sp<ClientRecordThread>& thread)
// Manage overrun callback
- if (mActive && (cblk->framesAvailable() == 0)) {
+ if (active && (cblk->framesAvailable() == 0)) {
+ // The value of active is stale, but we are almost sure to be active here because
+ // otherwise we would have exited when obtainBuffer returned STOPPED earlier.
ALOGV("Overrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags);
if (!(android_atomic_or(CBLK_UNDERRUN_ON, &cblk->flags) & CBLK_UNDERRUN_MSK)) {
- mCbf(EVENT_OVERRUN, mUserData, 0);
+ mCbf(EVENT_OVERRUN, mUserData, NULL);
}
}
@@ -805,7 +792,7 @@ status_t AudioRecord::restoreRecord_l(audio_track_cblk_t*& cblk)
result = NO_ERROR;
cblk->lock.unlock();
}
- if (result != NO_ERROR || mActive == 0) {
+ if (result != NO_ERROR || !mActive) {
result = status_t(STOPPED);
}
}
@@ -825,23 +812,51 @@ status_t AudioRecord::restoreRecord_l(audio_track_cblk_t*& cblk)
// =========================================================================
-AudioRecord::ClientRecordThread::ClientRecordThread(AudioRecord& receiver, bool bCanCallJava)
- : Thread(bCanCallJava), mReceiver(receiver)
+AudioRecord::AudioRecordThread::AudioRecordThread(AudioRecord& receiver, bool bCanCallJava)
+ : Thread(bCanCallJava), mReceiver(receiver), mPaused(true)
+{
+}
+
+AudioRecord::AudioRecordThread::~AudioRecordThread()
+{
+}
+
+bool AudioRecord::AudioRecordThread::threadLoop()
+{
+ {
+ AutoMutex _l(mMyLock);
+ if (mPaused) {
+ mMyCond.wait(mMyLock);
+ // caller will check for exitPending()
+ return true;
+ }
+ }
+ if (!mReceiver.processAudioBuffer(this)) {
+ pause();
+ }
+ return true;
+}
+
+void AudioRecord::AudioRecordThread::requestExit()
{
+ // must be in this order to avoid a race condition
+ Thread::requestExit();
+ resume();
}
-bool AudioRecord::ClientRecordThread::threadLoop()
+void AudioRecord::AudioRecordThread::pause()
{
- return mReceiver.processAudioBuffer(this);
+ AutoMutex _l(mMyLock);
+ mPaused = true;
}
-status_t AudioRecord::ClientRecordThread::readyToRun()
+void AudioRecord::AudioRecordThread::resume()
{
- AutoMutex(mReceiver.mLock);
- while (mReceiver.mReadyToRun == WOULD_BLOCK) {
- mReceiver.mCondition.wait(mReceiver.mLock);
+ AutoMutex _l(mMyLock);
+ if (mPaused) {
+ mPaused = false;
+ mMyCond.signal();
}
- return mReceiver.mReadyToRun;
}
// -------------------------------------------------------------------------
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 4c41ba5..6e0c620 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -41,7 +41,7 @@ DefaultKeyedVector<audio_io_handle_t, AudioSystem::OutputDescriptor *> AudioSyst
// Cached values for recording queries, all protected by gLock
uint32_t AudioSystem::gPrevInSamplingRate = 16000;
audio_format_t AudioSystem::gPrevInFormat = AUDIO_FORMAT_PCM_16_BIT;
-int AudioSystem::gPrevInChannelCount = 1;
+audio_channel_mask_t AudioSystem::gPrevInChannelMask = AUDIO_CHANNEL_IN_MONO;
size_t AudioSystem::gInBuffSize = 0;
@@ -334,25 +334,25 @@ status_t AudioSystem::getLatency(audio_io_handle_t output,
return NO_ERROR;
}
-status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount,
- size_t* buffSize)
+status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
+ audio_channel_mask_t channelMask, size_t* buffSize)
{
gLock.lock();
// Do we have a stale gInBufferSize or are we requesting the input buffer size for new values
size_t inBuffSize = gInBuffSize;
if ((inBuffSize == 0) || (sampleRate != gPrevInSamplingRate) || (format != gPrevInFormat)
- || (channelCount != gPrevInChannelCount)) {
+ || (channelMask != gPrevInChannelMask)) {
gLock.unlock();
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) {
return PERMISSION_DENIED;
}
- inBuffSize = af->getInputBufferSize(sampleRate, format, channelCount);
+ inBuffSize = af->getInputBufferSize(sampleRate, format, channelMask);
gLock.lock();
// save the request params
gPrevInSamplingRate = sampleRate;
gPrevInFormat = format;
- gPrevInChannelCount = channelCount;
+ gPrevInChannelMask = channelMask;
gInBuffSize = inBuffSize;
}
@@ -449,7 +449,7 @@ void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, audio_io_handle
OutputDescriptor *outputDesc = new OutputDescriptor(*desc);
gOutputs.add(ioHandle, outputDesc);
- ALOGV("ioConfigChanged() new output samplingRate %d, format %d channels %d frameCount %d latency %d",
+ ALOGV("ioConfigChanged() new output samplingRate %d, format %d channels %#x frameCount %d latency %d",
outputDesc->samplingRate, outputDesc->format, outputDesc->channels, outputDesc->frameCount, outputDesc->latency);
} break;
case OUTPUT_CLOSED: {
@@ -471,7 +471,7 @@ void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, audio_io_handle
if (param2 == NULL) break;
desc = (const OutputDescriptor *)param2;
- ALOGV("ioConfigChanged() new config for output %d samplingRate %d, format %d channels %d frameCount %d latency %d",
+ ALOGV("ioConfigChanged() new config for output %d samplingRate %d, format %d channels %#x frameCount %d latency %d",
ioHandle, desc->samplingRate, desc->format,
desc->channels, desc->frameCount, desc->latency);
OutputDescriptor *outputDesc = gOutputs.valueAt(index);
@@ -588,12 +588,12 @@ audio_policy_forced_cfg_t AudioSystem::getForceUse(audio_policy_force_use_t usag
audio_io_handle_t AudioSystem::getOutput(audio_stream_type_t stream,
uint32_t samplingRate,
audio_format_t format,
- uint32_t channels,
+ audio_channel_mask_t channelMask,
audio_output_flags_t flags)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return 0;
- return aps->getOutput(stream, samplingRate, format, channels, flags);
+ return aps->getOutput(stream, samplingRate, format, channelMask, flags);
}
status_t AudioSystem::startOutput(audio_io_handle_t output,
@@ -624,13 +624,12 @@ void AudioSystem::releaseOutput(audio_io_handle_t output)
audio_io_handle_t AudioSystem::getInput(audio_source_t inputSource,
uint32_t samplingRate,
audio_format_t format,
- uint32_t channels,
- audio_in_acoustics_t acoustics,
+ audio_channel_mask_t channelMask,
int sessionId)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return 0;
- return aps->getInput(inputSource, samplingRate, format, channels, acoustics, sessionId);
+ return aps->getInput(inputSource, samplingRate, format, channelMask, sessionId);
}
status_t AudioSystem::startInput(audio_io_handle_t input)
@@ -695,14 +694,14 @@ audio_devices_t AudioSystem::getDevicesForStream(audio_stream_type_t stream)
return aps->getDevicesForStream(stream);
}
-audio_io_handle_t AudioSystem::getOutputForEffect(effect_descriptor_t *desc)
+audio_io_handle_t AudioSystem::getOutputForEffect(const effect_descriptor_t *desc)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
return aps->getOutputForEffect(desc);
}
-status_t AudioSystem::registerEffect(effect_descriptor_t *desc,
+status_t AudioSystem::registerEffect(const effect_descriptor_t *desc,
audio_io_handle_t io,
uint32_t strategy,
int session,
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index e5a60f5..0ca035f 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -54,6 +54,11 @@ status_t AudioTrack::getMinFrameCount(
audio_stream_type_t streamType,
uint32_t sampleRate)
{
+ if (frameCount == NULL) return BAD_VALUE;
+
+ // default to 0 in case of error
+ *frameCount = 0;
+
// FIXME merge with similar code in createTrack_l(), except we're missing
// some information here that is available in createTrack_l():
// audio_io_handle_t output
@@ -98,7 +103,7 @@ AudioTrack::AudioTrack(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
- int channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
audio_output_flags_t flags,
callback_t cbf,
@@ -131,7 +136,8 @@ AudioTrack::AudioTrack(
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT)
{
- mStatus = set((audio_stream_type_t)streamType, sampleRate, (audio_format_t)format, channelMask,
+ mStatus = set((audio_stream_type_t)streamType, sampleRate, (audio_format_t)format,
+ (audio_channel_mask_t) channelMask,
frameCount, (audio_output_flags_t)flags, cbf, user, notificationFrames,
0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId);
}
@@ -140,7 +146,7 @@ AudioTrack::AudioTrack(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
- int channelMask,
+ audio_channel_mask_t channelMask,
const sp<IMemory>& sharedBuffer,
audio_output_flags_t flags,
callback_t cbf,
@@ -181,7 +187,7 @@ status_t AudioTrack::set(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
- int channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
audio_output_flags_t flags,
callback_t cbf,
@@ -247,7 +253,7 @@ status_t AudioTrack::set(
}
if (!audio_is_output_channel(channelMask)) {
- ALOGE("Invalid channel mask");
+ ALOGE("Invalid channel mask %#x", channelMask);
return BAD_VALUE;
}
uint32_t channelCount = popcount(channelMask);
@@ -272,34 +278,29 @@ status_t AudioTrack::set(
mFlags = flags;
mCbf = cbf;
- if (cbf != NULL) {
- mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
- mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
- }
-
// create the IAudioTrack
status_t status = createTrack_l(streamType,
sampleRate,
format,
- (uint32_t)channelMask,
+ channelMask,
frameCount,
flags,
sharedBuffer,
output);
-
if (status != NO_ERROR) {
- if (mAudioTrackThread != 0) {
- mAudioTrackThread->requestExit();
- mAudioTrackThread.clear();
- }
return status;
}
+ if (cbf != NULL) {
+ mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
+ mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO);
+ }
+
mStatus = NO_ERROR;
mStreamType = streamType;
mFormat = format;
- mChannelMask = (uint32_t)channelMask;
+ mChannelMask = channelMask;
mChannelCount = channelCount;
mSharedBuffer = sharedBuffer;
mMuted = false;
@@ -367,7 +368,6 @@ sp<IMemory>& AudioTrack::sharedBuffer()
void AudioTrack::start()
{
sp<AudioTrackThread> t = mAudioTrackThread;
- status_t status = NO_ERROR;
ALOGV("start %p", this);
@@ -395,6 +395,7 @@ void AudioTrack::start()
}
ALOGV("start %p before lock cblk %p", this, mCblk);
+ status_t status = NO_ERROR;
if (!(cblk->flags & CBLK_INVALID_MSK)) {
cblk->lock.unlock();
ALOGV("mAudioTrack->start()");
@@ -744,7 +745,7 @@ status_t AudioTrack::createTrack_l(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
- uint32_t channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
audio_output_flags_t flags,
const sp<IMemory>& sharedBuffer,
@@ -1472,15 +1473,6 @@ bool AudioTrack::AudioTrackThread::threadLoop()
return true;
}
-status_t AudioTrack::AudioTrackThread::readyToRun()
-{
- return NO_ERROR;
-}
-
-void AudioTrack::AudioTrackThread::onFirstRef()
-{
-}
-
void AudioTrack::AudioTrackThread::requestExit()
{
// must be in this order to avoid a race condition
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index e8dd438..71e7c31 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -32,7 +32,7 @@ enum {
CREATE_TRACK = IBinder::FIRST_CALL_TRANSACTION,
OPEN_RECORD,
SAMPLE_RATE,
- CHANNEL_COUNT,
+ CHANNEL_COUNT, // obsolete
FORMAT,
FRAME_COUNT,
LATENCY,
@@ -86,7 +86,7 @@ public:
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
- uint32_t channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
track_flags_t flags,
const sp<IMemory>& sharedBuffer,
@@ -135,9 +135,10 @@ public:
audio_io_handle_t input,
uint32_t sampleRate,
audio_format_t format,
- uint32_t channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
track_flags_t flags,
+ pid_t tid,
int *sessionId,
status_t *status)
{
@@ -151,6 +152,7 @@ public:
data.writeInt32(channelMask);
data.writeInt32(frameCount);
data.writeInt32(flags);
+ data.writeInt32((int32_t) tid);
int lSessionId = 0;
if (sessionId != NULL) {
lSessionId = *sessionId;
@@ -182,6 +184,7 @@ public:
return reply.readInt32();
}
+#if 0
virtual int channelCount(audio_io_handle_t output) const
{
Parcel data, reply;
@@ -190,6 +193,7 @@ public:
remote()->transact(CHANNEL_COUNT, data, &reply);
return reply.readInt32();
}
+#endif
virtual audio_format_t format(audio_io_handle_t output) const
{
@@ -347,13 +351,14 @@ public:
remote()->transact(REGISTER_CLIENT, data, &reply);
}
- virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const
+ virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
+ audio_channel_mask_t channelMask) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
data.writeInt32(sampleRate);
data.writeInt32(format);
- data.writeInt32(channelCount);
+ data.writeInt32(channelMask);
remote()->transact(GET_INPUTBUFFERSIZE, data, &reply);
return reply.readInt32();
}
@@ -698,7 +703,7 @@ status_t BnAudioFlinger::onTransact(
int streamType = data.readInt32();
uint32_t sampleRate = data.readInt32();
audio_format_t format = (audio_format_t) data.readInt32();
- int channelCount = data.readInt32();
+ audio_channel_mask_t channelMask = data.readInt32();
size_t bufferCount = data.readInt32();
track_flags_t flags = (track_flags_t) data.readInt32();
sp<IMemory> buffer = interface_cast<IMemory>(data.readStrongBinder());
@@ -708,7 +713,7 @@ status_t BnAudioFlinger::onTransact(
status_t status;
sp<IAudioTrack> track = createTrack(pid,
(audio_stream_type_t) streamType, sampleRate, format,
- channelCount, bufferCount, flags, buffer, output, tid, &sessionId, &status);
+ channelMask, bufferCount, flags, buffer, output, tid, &sessionId, &status);
reply->writeInt32(sessionId);
reply->writeInt32(status);
reply->writeStrongBinder(track->asBinder());
@@ -720,13 +725,14 @@ status_t BnAudioFlinger::onTransact(
audio_io_handle_t input = (audio_io_handle_t) data.readInt32();
uint32_t sampleRate = data.readInt32();
audio_format_t format = (audio_format_t) data.readInt32();
- int channelCount = data.readInt32();
+ audio_channel_mask_t channelMask = data.readInt32();
size_t bufferCount = data.readInt32();
track_flags_t flags = (track_flags_t) data.readInt32();
+ pid_t tid = (pid_t) data.readInt32();
int sessionId = data.readInt32();
status_t status;
sp<IAudioRecord> record = openRecord(pid, input,
- sampleRate, format, channelCount, bufferCount, flags, &sessionId, &status);
+ sampleRate, format, channelMask, bufferCount, flags, tid, &sessionId, &status);
reply->writeInt32(sessionId);
reply->writeInt32(status);
reply->writeStrongBinder(record->asBinder());
@@ -737,11 +743,13 @@ status_t BnAudioFlinger::onTransact(
reply->writeInt32( sampleRate((audio_io_handle_t) data.readInt32()) );
return NO_ERROR;
} break;
+#if 0
case CHANNEL_COUNT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
reply->writeInt32( channelCount((audio_io_handle_t) data.readInt32()) );
return NO_ERROR;
} break;
+#endif
case FORMAT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
reply->writeInt32( format((audio_io_handle_t) data.readInt32()) );
@@ -846,8 +854,8 @@ status_t BnAudioFlinger::onTransact(
CHECK_INTERFACE(IAudioFlinger, data, reply);
uint32_t sampleRate = data.readInt32();
audio_format_t format = (audio_format_t) data.readInt32();
- int channelCount = data.readInt32();
- reply->writeInt32( getInputBufferSize(sampleRate, format, channelCount) );
+ audio_channel_mask_t channelMask = data.readInt32();
+ reply->writeInt32( getInputBufferSize(sampleRate, format, channelMask) );
return NO_ERROR;
} break;
case OPEN_OUTPUT: {
diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp
index 7aab8d6..31c5d27 100644
--- a/media/libmedia/IAudioPolicyService.cpp
+++ b/media/libmedia/IAudioPolicyService.cpp
@@ -123,7 +123,7 @@ public:
audio_stream_type_t stream,
uint32_t samplingRate,
audio_format_t format,
- uint32_t channels,
+ audio_channel_mask_t channelMask,
audio_output_flags_t flags)
{
Parcel data, reply;
@@ -131,7 +131,7 @@ public:
data.writeInt32(static_cast <uint32_t>(stream));
data.writeInt32(samplingRate);
data.writeInt32(static_cast <uint32_t>(format));
- data.writeInt32(channels);
+ data.writeInt32(channelMask);
data.writeInt32(static_cast <uint32_t>(flags));
remote()->transact(GET_OUTPUT, data, &reply);
return static_cast <audio_io_handle_t> (reply.readInt32());
@@ -175,8 +175,7 @@ public:
audio_source_t inputSource,
uint32_t samplingRate,
audio_format_t format,
- uint32_t channels,
- audio_in_acoustics_t acoustics,
+ audio_channel_mask_t channelMask,
int audioSession)
{
Parcel data, reply;
@@ -184,8 +183,7 @@ public:
data.writeInt32((int32_t) inputSource);
data.writeInt32(samplingRate);
data.writeInt32(static_cast <uint32_t>(format));
- data.writeInt32(channels);
- data.writeInt32(static_cast <uint32_t>(acoustics));
+ data.writeInt32(channelMask);
data.writeInt32(audioSession);
remote()->transact(GET_INPUT, data, &reply);
return static_cast <audio_io_handle_t> (reply.readInt32());
@@ -276,7 +274,7 @@ public:
return (audio_devices_t) reply.readInt32();
}
- virtual audio_io_handle_t getOutputForEffect(effect_descriptor_t *desc)
+ virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
@@ -285,7 +283,7 @@ public:
return static_cast <audio_io_handle_t> (reply.readInt32());
}
- virtual status_t registerEffect(effect_descriptor_t *desc,
+ virtual status_t registerEffect(const effect_descriptor_t *desc,
audio_io_handle_t io,
uint32_t strategy,
int session,
@@ -417,14 +415,14 @@ status_t BnAudioPolicyService::onTransact(
static_cast <audio_stream_type_t>(data.readInt32());
uint32_t samplingRate = data.readInt32();
audio_format_t format = (audio_format_t) data.readInt32();
- uint32_t channels = data.readInt32();
+ audio_channel_mask_t channelMask = data.readInt32();
audio_output_flags_t flags =
static_cast <audio_output_flags_t>(data.readInt32());
audio_io_handle_t output = getOutput(stream,
samplingRate,
format,
- channels,
+ channelMask,
flags);
reply->writeInt32(static_cast <int>(output));
return NO_ERROR;
@@ -464,15 +462,12 @@ status_t BnAudioPolicyService::onTransact(
audio_source_t inputSource = (audio_source_t) data.readInt32();
uint32_t samplingRate = data.readInt32();
audio_format_t format = (audio_format_t) data.readInt32();
- uint32_t channels = data.readInt32();
- audio_in_acoustics_t acoustics =
- static_cast <audio_in_acoustics_t>(data.readInt32());
+ audio_channel_mask_t channelMask = data.readInt32();
int audioSession = data.readInt32();
audio_io_handle_t input = getInput(inputSource,
samplingRate,
format,
- channels,
- acoustics,
+ channelMask,
audioSession);
reply->writeInt32(static_cast <int>(input));
return NO_ERROR;
diff --git a/media/libmedia/IAudioRecord.cpp b/media/libmedia/IAudioRecord.cpp
index 57a80a9..0d06e98 100644
--- a/media/libmedia/IAudioRecord.cpp
+++ b/media/libmedia/IAudioRecord.cpp
@@ -42,7 +42,7 @@ public:
{
}
- virtual status_t start(int event, int triggerSession)
+ virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioRecord::getInterfaceDescriptor());
@@ -92,7 +92,7 @@ status_t BnAudioRecord::onTransact(
} break;
case START: {
CHECK_INTERFACE(IAudioRecord, data, reply);
- int event = data.readInt32();
+ int /*AudioSystem::sync_event_t*/ event = data.readInt32();
int triggerSession = data.readInt32();
reply->writeInt32(start(event, triggerSession));
return NO_ERROR;
diff --git a/media/libmedia/Visualizer.cpp b/media/libmedia/Visualizer.cpp
index de0bf7d..8196e10 100644
--- a/media/libmedia/Visualizer.cpp
+++ b/media/libmedia/Visualizer.cpp
@@ -353,13 +353,4 @@ bool Visualizer::CaptureThread::threadLoop()
return false;
}
-status_t Visualizer::CaptureThread::readyToRun()
-{
- return NO_ERROR;
-}
-
-void Visualizer::CaptureThread::onFirstRef()
-{
-}
-
}; // namespace android
generated by cgit v1.1 at 2025-08-22 09:44:19 +0000