summaryrefslogtreecommitdiffstats
path: root/media/libmedia
diff options
context:
space:
mode:
Diffstat (limited to 'media/libmedia')
-rw-r--r--media/libmedia/Android.mk8
-rw-r--r--media/libmedia/AudioEffect.cpp8
-rw-r--r--media/libmedia/AudioRecord.cpp64
-rw-r--r--media/libmedia/AudioSystem.cpp120
-rw-r--r--media/libmedia/AudioTrack.cpp213
-rw-r--r--media/libmedia/IAudioFlinger.cpp82
-rw-r--r--media/libmedia/IAudioFlingerClient.cpp2
-rw-r--r--media/libmedia/IAudioPolicyService.cpp69
-rw-r--r--media/libmedia/IAudioTrack.cpp24
-rw-r--r--media/libmedia/IMediaDeathNotifier.cpp2
-rw-r--r--media/libmedia/IMediaPlayer.cpp6
-rw-r--r--media/libmedia/IMediaPlayerService.cpp16
-rw-r--r--media/libmedia/IOMX.cpp7
-rw-r--r--media/libmedia/JetPlayer.cpp19
-rw-r--r--media/libmedia/MediaScannerClient.cpp1
-rw-r--r--media/libmedia/ToneGenerator.cpp39
-rw-r--r--media/libmedia/Visualizer.cpp17
-rw-r--r--media/libmedia/autodetect.cpp2
-rw-r--r--media/libmedia/fixedfft.cpp162
-rw-r--r--media/libmedia/mediametadataretriever.cpp2
-rw-r--r--media/libmedia/mediaplayer.cpp8
21 files changed, 352 insertions, 519 deletions
diff --git a/media/libmedia/Android.mk b/media/libmedia/Android.mk
index 7af4a87..23670df 100644
--- a/media/libmedia/Android.mk
+++ b/media/libmedia/Android.mk
@@ -43,13 +43,12 @@ LOCAL_SRC_FILES:= \
IEffectClient.cpp \
AudioEffect.cpp \
Visualizer.cpp \
- MemoryLeakTrackUtil.cpp \
- fixedfft.cpp.arm
+ MemoryLeakTrackUtil.cpp
LOCAL_SHARED_LIBRARIES := \
libui libcutils libutils libbinder libsonivox libicuuc libexpat \
libcamera_client libstagefright_foundation \
- libgui libdl
+ libgui libdl libaudioutils
LOCAL_WHOLE_STATIC_LIBRARY := libmedia_helper
@@ -61,6 +60,7 @@ LOCAL_C_INCLUDES := \
$(TOP)/frameworks/base/include/media/stagefright/openmax \
external/icu4c/common \
external/expat/lib \
- system/media/audio_effects/include
+ system/media/audio_effects/include \
+ system/media/audio_utils/include
include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libmedia/AudioEffect.cpp b/media/libmedia/AudioEffect.cpp
index 6639d06..a242846 100644
--- a/media/libmedia/AudioEffect.cpp
+++ b/media/libmedia/AudioEffect.cpp
@@ -342,7 +342,7 @@ void AudioEffect::binderDied()
{
ALOGW("IEffect died");
mStatus = NO_INIT;
- if (mCbf) {
+ if (mCbf != NULL) {
status_t status = DEAD_OBJECT;
mCbf(EVENT_ERROR, mUserData, &status);
}
@@ -363,7 +363,7 @@ void AudioEffect::controlStatusChanged(bool controlGranted)
mStatus = ALREADY_EXISTS;
}
}
- if (mCbf) {
+ if (mCbf != NULL) {
mCbf(EVENT_CONTROL_STATUS_CHANGED, mUserData, &controlGranted);
}
}
@@ -373,7 +373,7 @@ void AudioEffect::enableStatusChanged(bool enabled)
ALOGV("enableStatusChanged %p enabled %d mCbf %p", this, enabled, mCbf);
if (mStatus == ALREADY_EXISTS) {
mEnabled = enabled;
- if (mCbf) {
+ if (mCbf != NULL) {
mCbf(EVENT_ENABLE_STATUS_CHANGED, mUserData, &enabled);
}
}
@@ -389,7 +389,7 @@ void AudioEffect::commandExecuted(uint32_t cmdCode,
return;
}
- if (mCbf && cmdCode == EFFECT_CMD_SET_PARAM) {
+ if (mCbf != NULL && cmdCode == EFFECT_CMD_SET_PARAM) {
effect_param_t *cmd = (effect_param_t *)cmdData;
cmd->status = *(int32_t *)replyData;
mCbf(EVENT_PARAMETER_CHANGED, mUserData, cmd);
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 34a5eb7..c96bc76 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -39,9 +39,7 @@
#include <system/audio.h>
#include <cutils/bitops.h>
-
-#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
-#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
+#include <cutils/compiler.h>
namespace android {
// ---------------------------------------------------------------------------
@@ -50,7 +48,7 @@ namespace android {
status_t AudioRecord::getMinFrameCount(
int* frameCount,
uint32_t sampleRate,
- int format,
+ audio_format_t format,
int channelCount)
{
size_t size = 0;
@@ -80,14 +78,15 @@ status_t AudioRecord::getMinFrameCount(
// ---------------------------------------------------------------------------
AudioRecord::AudioRecord()
- : mStatus(NO_INIT), mSessionId(0)
+ : mStatus(NO_INIT), mSessionId(0),
+ mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
{
}
AudioRecord::AudioRecord(
- int inputSource,
+ audio_source_t inputSource,
uint32_t sampleRate,
- int format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
@@ -95,7 +94,8 @@ AudioRecord::AudioRecord(
void* user,
int notificationFrames,
int sessionId)
- : mStatus(NO_INIT), mSessionId(0)
+ : mStatus(NO_INIT), mSessionId(0),
+ mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
{
mStatus = set(inputSource, sampleRate, format, channelMask,
frameCount, flags, cbf, user, notificationFrames, sessionId);
@@ -119,9 +119,9 @@ AudioRecord::~AudioRecord()
}
status_t AudioRecord::set(
- int inputSource,
+ audio_source_t inputSource,
uint32_t sampleRate,
- int format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
@@ -148,7 +148,7 @@ status_t AudioRecord::set(
sampleRate = DEFAULT_SAMPLE_RATE;
}
// these below should probably come from the audioFlinger too...
- if (format == 0) {
+ if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
}
// validate parameters
@@ -206,11 +206,8 @@ status_t AudioRecord::set(
return status;
}
- if (cbf != 0) {
+ if (cbf != NULL) {
mClientRecordThread = new ClientRecordThread(*this, threadCanCallJava);
- if (mClientRecordThread == 0) {
- return NO_INIT;
- }
}
mStatus = NO_ERROR;
@@ -231,7 +228,7 @@ status_t AudioRecord::set(
mMarkerReached = false;
mNewPosition = 0;
mUpdatePeriod = 0;
- mInputSource = (uint8_t)inputSource;
+ mInputSource = inputSource;
mFlags = flags;
mInput = input;
AudioSystem::acquireAudioSessionId(mSessionId);
@@ -251,7 +248,7 @@ uint32_t AudioRecord::latency() const
return mLatency;
}
-int AudioRecord::format() const
+audio_format_t AudioRecord::format() const
{
return mFormat;
}
@@ -266,7 +263,7 @@ uint32_t AudioRecord::frameCount() const
return mFrameCount;
}
-int AudioRecord::frameSize() const
+size_t AudioRecord::frameSize() const
{
if (audio_is_linear_pcm(mFormat)) {
return channelCount()*audio_bytes_per_sample(mFormat);
@@ -275,9 +272,9 @@ int AudioRecord::frameSize() const
}
}
-int AudioRecord::inputSource() const
+audio_source_t AudioRecord::inputSource() const
{
- return (int)mInputSource;
+ return mInputSource;
}
// -------------------------------------------------------------------------
@@ -326,9 +323,11 @@ status_t AudioRecord::start()
cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
cblk->waitTimeMs = 0;
if (t != 0) {
- t->run("ClientRecordThread", ANDROID_PRIORITY_AUDIO);
+ t->run("ClientRecordThread", ANDROID_PRIORITY_AUDIO);
} else {
- setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_AUDIO);
+ mPreviousPriority = getpriority(PRIO_PROCESS, 0);
+ mPreviousSchedulingGroup = androidGetThreadSchedulingGroup(0);
+ androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
}
} else {
mActive = 0;
@@ -363,7 +362,8 @@ status_t AudioRecord::stop()
if (t != 0) {
t->requestExit();
} else {
- setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
+ setpriority(PRIO_PROCESS, 0, mPreviousPriority);
+ androidSetThreadSchedulingGroup(0, mPreviousSchedulingGroup);
}
}
@@ -387,7 +387,7 @@ uint32_t AudioRecord::getSampleRate()
status_t AudioRecord::setMarkerPosition(uint32_t marker)
{
- if (mCbf == 0) return INVALID_OPERATION;
+ if (mCbf == NULL) return INVALID_OPERATION;
mMarkerPosition = marker;
mMarkerReached = false;
@@ -397,7 +397,7 @@ status_t AudioRecord::setMarkerPosition(uint32_t marker)
status_t AudioRecord::getMarkerPosition(uint32_t *marker)
{
- if (marker == 0) return BAD_VALUE;
+ if (marker == NULL) return BAD_VALUE;
*marker = mMarkerPosition;
@@ -406,7 +406,7 @@ status_t AudioRecord::getMarkerPosition(uint32_t *marker)
status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod)
{
- if (mCbf == 0) return INVALID_OPERATION;
+ if (mCbf == NULL) return INVALID_OPERATION;
uint32_t curPosition;
getPosition(&curPosition);
@@ -418,7 +418,7 @@ status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod)
status_t AudioRecord::getPositionUpdatePeriod(uint32_t *updatePeriod)
{
- if (updatePeriod == 0) return BAD_VALUE;
+ if (updatePeriod == NULL) return BAD_VALUE;
*updatePeriod = mUpdatePeriod;
@@ -427,7 +427,7 @@ status_t AudioRecord::getPositionUpdatePeriod(uint32_t *updatePeriod)
status_t AudioRecord::getPosition(uint32_t *position)
{
- if (position == 0) return BAD_VALUE;
+ if (position == NULL) return BAD_VALUE;
AutoMutex lock(mLock);
*position = mCblk->user;
@@ -448,7 +448,7 @@ unsigned int AudioRecord::getInputFramesLost()
// must be called with mLock held
status_t AudioRecord::openRecord_l(
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
@@ -508,11 +508,11 @@ status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
goto start_loop_here;
while (framesReady == 0) {
active = mActive;
- if (UNLIKELY(!active)) {
+ if (CC_UNLIKELY(!active)) {
cblk->lock.unlock();
return NO_MORE_BUFFERS;
}
- if (UNLIKELY(!waitCount)) {
+ if (CC_UNLIKELY(!waitCount)) {
cblk->lock.unlock();
return WOULD_BLOCK;
}
@@ -529,7 +529,7 @@ status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
if (cblk->flags & CBLK_INVALID_MSK) {
goto create_new_record;
}
- if (__builtin_expect(result!=NO_ERROR, false)) {
+ if (CC_UNLIKELY(result != NO_ERROR)) {
cblk->waitTimeMs += waitTimeMs;
if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
ALOGW( "obtainBuffer timed out (is the CPU pegged?) "
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index f7f129c..110a294 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -35,12 +35,13 @@ sp<IAudioFlinger> AudioSystem::gAudioFlinger;
sp<AudioSystem::AudioFlingerClient> AudioSystem::gAudioFlingerClient;
audio_error_callback AudioSystem::gAudioErrorCallback = NULL;
// Cached values
-DefaultKeyedVector<int, audio_io_handle_t> AudioSystem::gStreamOutputMap(0);
+
+DefaultKeyedVector<audio_stream_type_t, audio_io_handle_t> AudioSystem::gStreamOutputMap(0);
DefaultKeyedVector<audio_io_handle_t, AudioSystem::OutputDescriptor *> AudioSystem::gOutputs(0);
-// Cached values for recording queries
+// Cached values for recording queries, all protected by gLock
uint32_t AudioSystem::gPrevInSamplingRate = 16000;
-int AudioSystem::gPrevInFormat = AUDIO_FORMAT_PCM_16_BIT;
+audio_format_t AudioSystem::gPrevInFormat = AUDIO_FORMAT_PCM_16_BIT;
int AudioSystem::gPrevInChannelCount = 1;
size_t AudioSystem::gInBuffSize = 0;
@@ -49,7 +50,7 @@ size_t AudioSystem::gInBuffSize = 0;
const sp<IAudioFlinger>& AudioSystem::get_audio_flinger()
{
Mutex::Autolock _l(gLock);
- if (gAudioFlinger.get() == 0) {
+ if (gAudioFlinger == 0) {
sp<IServiceManager> sm = defaultServiceManager();
sp<IBinder> binder;
do {
@@ -120,7 +121,7 @@ status_t AudioSystem::getMasterMute(bool* mute)
return NO_ERROR;
}
-status_t AudioSystem::setStreamVolume(int stream, float value, int output)
+status_t AudioSystem::setStreamVolume(audio_stream_type_t stream, float value, int output)
{
if (uint32_t(stream) >= AUDIO_STREAM_CNT) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
@@ -129,7 +130,7 @@ status_t AudioSystem::setStreamVolume(int stream, float value, int output)
return NO_ERROR;
}
-status_t AudioSystem::setStreamMute(int stream, bool mute)
+status_t AudioSystem::setStreamMute(audio_stream_type_t stream, bool mute)
{
if (uint32_t(stream) >= AUDIO_STREAM_CNT) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
@@ -138,7 +139,7 @@ status_t AudioSystem::setStreamMute(int stream, bool mute)
return NO_ERROR;
}
-status_t AudioSystem::getStreamVolume(int stream, float* volume, int output)
+status_t AudioSystem::getStreamVolume(audio_stream_type_t stream, float* volume, int output)
{
if (uint32_t(stream) >= AUDIO_STREAM_CNT) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
@@ -147,7 +148,7 @@ status_t AudioSystem::getStreamVolume(int stream, float* volume, int output)
return NO_ERROR;
}
-status_t AudioSystem::getStreamMute(int stream, bool* mute)
+status_t AudioSystem::getStreamMute(audio_stream_type_t stream, bool* mute)
{
if (uint32_t(stream) >= AUDIO_STREAM_CNT) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
@@ -156,9 +157,9 @@ status_t AudioSystem::getStreamMute(int stream, bool* mute)
return NO_ERROR;
}
-status_t AudioSystem::setMode(int mode)
+status_t AudioSystem::setMode(audio_mode_t mode)
{
- if (mode >= AUDIO_MODE_CNT) return BAD_VALUE;
+ if (uint32_t(mode) >= AUDIO_MODE_CNT) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
return af->setMode(mode);
@@ -203,7 +204,12 @@ int AudioSystem::logToLinear(float volume)
return volume ? 100 - int(dBConvertInverse * log(volume) + 0.5) : 0;
}
-status_t AudioSystem::getOutputSamplingRate(int* samplingRate, int streamType)
+// DEPRECATED
+status_t AudioSystem::getOutputSamplingRate(int* samplingRate, int streamType) {
+ return getOutputSamplingRate(samplingRate, (audio_stream_type_t)streamType);
+}
+
+status_t AudioSystem::getOutputSamplingRate(int* samplingRate, audio_stream_type_t streamType)
{
OutputDescriptor *outputDesc;
audio_io_handle_t output;
@@ -212,14 +218,14 @@ status_t AudioSystem::getOutputSamplingRate(int* samplingRate, int streamType)
streamType = AUDIO_STREAM_MUSIC;
}
- output = getOutput((audio_stream_type_t)streamType);
+ output = getOutput(streamType);
if (output == 0) {
return PERMISSION_DENIED;
}
gLock.lock();
outputDesc = AudioSystem::gOutputs.valueFor(output);
- if (outputDesc == 0) {
+ if (outputDesc == NULL) {
ALOGV("getOutputSamplingRate() no output descriptor for output %d in gOutputs", output);
gLock.unlock();
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
@@ -236,7 +242,12 @@ status_t AudioSystem::getOutputSamplingRate(int* samplingRate, int streamType)
return NO_ERROR;
}
-status_t AudioSystem::getOutputFrameCount(int* frameCount, int streamType)
+// DEPRECATED
+status_t AudioSystem::getOutputFrameCount(int* frameCount, int streamType) {
+ return getOutputFrameCount(frameCount, (audio_stream_type_t)streamType);
+}
+
+status_t AudioSystem::getOutputFrameCount(int* frameCount, audio_stream_type_t streamType)
{
OutputDescriptor *outputDesc;
audio_io_handle_t output;
@@ -245,14 +256,14 @@ status_t AudioSystem::getOutputFrameCount(int* frameCount, int streamType)
streamType = AUDIO_STREAM_MUSIC;
}
- output = getOutput((audio_stream_type_t)streamType);
+ output = getOutput(streamType);
if (output == 0) {
return PERMISSION_DENIED;
}
gLock.lock();
outputDesc = AudioSystem::gOutputs.valueFor(output);
- if (outputDesc == 0) {
+ if (outputDesc == NULL) {
gLock.unlock();
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
@@ -267,7 +278,7 @@ status_t AudioSystem::getOutputFrameCount(int* frameCount, int streamType)
return NO_ERROR;
}
-status_t AudioSystem::getOutputLatency(uint32_t* latency, int streamType)
+status_t AudioSystem::getOutputLatency(uint32_t* latency, audio_stream_type_t streamType)
{
OutputDescriptor *outputDesc;
audio_io_handle_t output;
@@ -276,14 +287,14 @@ status_t AudioSystem::getOutputLatency(uint32_t* latency, int streamType)
streamType = AUDIO_STREAM_MUSIC;
}
- output = getOutput((audio_stream_type_t)streamType);
+ output = getOutput(streamType);
if (output == 0) {
return PERMISSION_DENIED;
}
gLock.lock();
outputDesc = AudioSystem::gOutputs.valueFor(output);
- if (outputDesc == 0) {
+ if (outputDesc == NULL) {
gLock.unlock();
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
@@ -298,25 +309,30 @@ status_t AudioSystem::getOutputLatency(uint32_t* latency, int streamType)
return NO_ERROR;
}
-status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, int format, int channelCount,
+status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount,
size_t* buffSize)
{
+ gLock.lock();
// Do we have a stale gInBufferSize or are we requesting the input buffer size for new values
- if ((gInBuffSize == 0) || (sampleRate != gPrevInSamplingRate) || (format != gPrevInFormat)
+ size_t inBuffSize = gInBuffSize;
+ if ((inBuffSize == 0) || (sampleRate != gPrevInSamplingRate) || (format != gPrevInFormat)
|| (channelCount != gPrevInChannelCount)) {
+ gLock.unlock();
+ const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
+ if (af == 0) {
+ return PERMISSION_DENIED;
+ }
+ inBuffSize = af->getInputBufferSize(sampleRate, format, channelCount);
+ gLock.lock();
// save the request params
gPrevInSamplingRate = sampleRate;
gPrevInFormat = format;
gPrevInChannelCount = channelCount;
- gInBuffSize = 0;
- const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
- if (af == 0) {
- return PERMISSION_DENIED;
- }
- gInBuffSize = af->getInputBufferSize(sampleRate, format, channelCount);
+ gInBuffSize = inBuffSize;
}
- *buffSize = gInBuffSize;
+ gLock.unlock();
+ *buffSize = inBuffSize;
return NO_ERROR;
}
@@ -328,7 +344,7 @@ status_t AudioSystem::setVoiceVolume(float value)
return af->setVoiceVolume(value);
}
-status_t AudioSystem::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int stream)
+status_t AudioSystem::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, audio_stream_type_t stream)
{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
@@ -337,7 +353,7 @@ status_t AudioSystem::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames
stream = AUDIO_STREAM_MUSIC;
}
- return af->getRenderPosition(halFrames, dspFrames, getOutput((audio_stream_type_t)stream));
+ return af->getRenderPosition(halFrames, dspFrames, getOutput(stream));
}
unsigned int AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) {
@@ -389,7 +405,7 @@ void AudioSystem::AudioFlingerClient::binderDied(const wp<IBinder>& who) {
void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, int ioHandle, void *param2) {
ALOGV("ioConfigChanged() event %d", event);
OutputDescriptor *desc;
- uint32_t stream;
+ audio_stream_type_t stream;
if (ioHandle == 0) return;
@@ -397,8 +413,8 @@ void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, int ioHandle, v
switch (event) {
case STREAM_CONFIG_CHANGED:
- if (param2 == 0) break;
- stream = *(uint32_t *)param2;
+ if (param2 == NULL) break;
+ stream = *(audio_stream_type_t *)param2;
ALOGV("ioConfigChanged() STREAM_CONFIG_CHANGED stream %d, output %d", stream, ioHandle);
if (gStreamOutputMap.indexOfKey(stream) >= 0) {
gStreamOutputMap.replaceValueFor(stream, ioHandle);
@@ -409,7 +425,7 @@ void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, int ioHandle, v
ALOGV("ioConfigChanged() opening already existing output! %d", ioHandle);
break;
}
- if (param2 == 0) break;
+ if (param2 == NULL) break;
desc = (OutputDescriptor *)param2;
OutputDescriptor *outputDesc = new OutputDescriptor(*desc);
@@ -438,7 +454,7 @@ void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, int ioHandle, v
ALOGW("ioConfigChanged() modifying unknow output! %d", ioHandle);
break;
}
- if (param2 == 0) break;
+ if (param2 == NULL) break;
desc = (OutputDescriptor *)param2;
ALOGV("ioConfigChanged() new config for output %d samplingRate %d, format %d channels %d frameCount %d latency %d",
@@ -462,7 +478,7 @@ void AudioSystem::setErrorCallback(audio_error_callback cb) {
gAudioErrorCallback = cb;
}
-bool AudioSystem::routedToA2dpOutput(int streamType) {
+bool AudioSystem::routedToA2dpOutput(audio_stream_type_t streamType) {
switch(streamType) {
case AUDIO_STREAM_MUSIC:
case AUDIO_STREAM_VOICE_CALL:
@@ -484,7 +500,7 @@ sp<AudioSystem::AudioPolicyServiceClient> AudioSystem::gAudioPolicyServiceClient
const sp<IAudioPolicyService>& AudioSystem::get_audio_policy_service()
{
gLock.lock();
- if (gAudioPolicyService.get() == 0) {
+ if (gAudioPolicyService == 0) {
sp<IServiceManager> sm = defaultServiceManager();
sp<IBinder> binder;
do {
@@ -531,21 +547,15 @@ audio_policy_dev_state_t AudioSystem::getDeviceConnectionState(audio_devices_t d
return aps->getDeviceConnectionState(device, device_address);
}
-status_t AudioSystem::setPhoneState(int state)
+status_t AudioSystem::setPhoneState(audio_mode_t state)
{
+ if (uint32_t(state) >= AUDIO_MODE_CNT) return BAD_VALUE;
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
return aps->setPhoneState(state);
}
-status_t AudioSystem::setRingerMode(uint32_t mode, uint32_t mask)
-{
- const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
- if (aps == 0) return PERMISSION_DENIED;
- return aps->setRingerMode(mode, mask);
-}
-
status_t AudioSystem::setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
@@ -563,7 +573,7 @@ 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,
- uint32_t format,
+ audio_format_t format,
uint32_t channels,
audio_policy_output_flags_t flags)
{
@@ -621,9 +631,9 @@ void AudioSystem::releaseOutput(audio_io_handle_t output)
aps->releaseOutput(output);
}
-audio_io_handle_t AudioSystem::getInput(int inputSource,
+audio_io_handle_t AudioSystem::getInput(audio_source_t inputSource,
uint32_t samplingRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channels,
audio_in_acoustics_t acoustics,
int sessionId)
@@ -663,18 +673,22 @@ status_t AudioSystem::initStreamVolume(audio_stream_type_t stream,
return aps->initStreamVolume(stream, indexMin, indexMax);
}
-status_t AudioSystem::setStreamVolumeIndex(audio_stream_type_t stream, int index)
+status_t AudioSystem::setStreamVolumeIndex(audio_stream_type_t stream,
+ int index,
+ audio_devices_t device)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
- return aps->setStreamVolumeIndex(stream, index);
+ return aps->setStreamVolumeIndex(stream, index, device);
}
-status_t AudioSystem::getStreamVolumeIndex(audio_stream_type_t stream, int *index)
+status_t AudioSystem::getStreamVolumeIndex(audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
- return aps->getStreamVolumeIndex(stream, index);
+ return aps->getStreamVolumeIndex(stream, index, device);
}
uint32_t AudioSystem::getStrategyForStream(audio_stream_type_t stream)
@@ -723,7 +737,7 @@ status_t AudioSystem::setEffectEnabled(int id, bool enabled)
return aps->setEffectEnabled(id, enabled);
}
-status_t AudioSystem::isStreamActive(int stream, bool* state, uint32_t inPastMs)
+status_t AudioSystem::isStreamActive(audio_stream_type_t stream, bool* state, uint32_t inPastMs)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index d51cd69..8c33f41 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -1,4 +1,4 @@
-/* //device/extlibs/pv/android/AudioTrack.cpp
+/* frameworks/base/media/libmedia/AudioTrack.cpp
**
** Copyright 2007, The Android Open Source Project
**
@@ -38,12 +38,12 @@
#include <utils/Atomic.h>
#include <cutils/bitops.h>
+#include <cutils/compiler.h>
#include <system/audio.h>
#include <system/audio_policy.h>
-#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
-#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
+#include <audio_utils/primitives.h>
namespace android {
// ---------------------------------------------------------------------------
@@ -51,7 +51,7 @@ namespace android {
// static
status_t AudioTrack::getMinFrameCount(
int* frameCount,
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate)
{
int afSampleRate;
@@ -79,14 +79,15 @@ status_t AudioTrack::getMinFrameCount(
// ---------------------------------------------------------------------------
AudioTrack::AudioTrack()
- : mStatus(NO_INIT)
+ : mStatus(NO_INIT),
+ mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
{
}
AudioTrack::AudioTrack(
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate,
- int format,
+ audio_format_t format,
int channelMask,
int frameCount,
uint32_t flags,
@@ -94,7 +95,8 @@ AudioTrack::AudioTrack(
void* user,
int notificationFrames,
int sessionId)
- : mStatus(NO_INIT)
+ : mStatus(NO_INIT),
+ mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
{
mStatus = set(streamType, sampleRate, format, channelMask,
frameCount, flags, cbf, user, notificationFrames,
@@ -106,13 +108,33 @@ AudioTrack::AudioTrack(
uint32_t sampleRate,
int format,
int channelMask,
+ int frameCount,
+ uint32_t flags,
+ callback_t cbf,
+ void* user,
+ int notificationFrames,
+ int sessionId)
+ : mStatus(NO_INIT),
+ mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
+{
+ mStatus = set((audio_stream_type_t)streamType, sampleRate, (audio_format_t)format, channelMask,
+ frameCount, flags, cbf, user, notificationFrames,
+ 0, false, sessionId);
+}
+
+AudioTrack::AudioTrack(
+ audio_stream_type_t streamType,
+ uint32_t sampleRate,
+ audio_format_t format,
+ int channelMask,
const sp<IMemory>& sharedBuffer,
uint32_t flags,
callback_t cbf,
void* user,
int notificationFrames,
int sessionId)
- : mStatus(NO_INIT)
+ : mStatus(NO_INIT),
+ mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
{
mStatus = set(streamType, sampleRate, format, channelMask,
0, flags, cbf, user, notificationFrames,
@@ -139,9 +161,9 @@ AudioTrack::~AudioTrack()
}
status_t AudioTrack::set(
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate,
- int format,
+ audio_format_t format,
int channelMask,
int frameCount,
uint32_t flags,
@@ -178,7 +200,7 @@ status_t AudioTrack::set(
sampleRate = afSampleRate;
}
// these below should probably come from the audioFlinger too...
- if (format == 0) {
+ if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
}
if (channelMask == 0) {
@@ -203,8 +225,8 @@ status_t AudioTrack::set(
uint32_t channelCount = popcount(channelMask);
audio_io_handle_t output = AudioSystem::getOutput(
- (audio_stream_type_t)streamType,
- sampleRate,format, channelMask,
+ streamType,
+ sampleRate, format, channelMask,
(audio_policy_output_flags_t)flags);
if (output == 0) {
@@ -214,7 +236,7 @@ status_t AudioTrack::set(
mVolume[LEFT] = 1.0f;
mVolume[RIGHT] = 1.0f;
- mSendLevel = 0;
+ mSendLevel = 0.0f;
mFrameCount = frameCount;
mNotificationFramesReq = notificationFrames;
mSessionId = sessionId;
@@ -223,7 +245,7 @@ status_t AudioTrack::set(
// create the IAudioTrack
status_t status = createTrack_l(streamType,
sampleRate,
- (uint32_t)format,
+ format,
(uint32_t)channelMask,
frameCount,
flags,
@@ -235,23 +257,19 @@ status_t AudioTrack::set(
return status;
}
- if (cbf != 0) {
+ if (cbf != NULL) {
mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
- if (mAudioTrackThread == 0) {
- ALOGE("Could not create callback thread");
- return NO_INIT;
- }
}
mStatus = NO_ERROR;
mStreamType = streamType;
- mFormat = (uint32_t)format;
+ mFormat = format;
mChannelMask = (uint32_t)channelMask;
mChannelCount = channelCount;
mSharedBuffer = sharedBuffer;
mMuted = false;
- mActive = 0;
+ mActive = false;
mCbf = cbf;
mUserData = user;
mLoopCount = 0;
@@ -278,12 +296,12 @@ uint32_t AudioTrack::latency() const
return mLatency;
}
-int AudioTrack::streamType() const
+audio_stream_type_t AudioTrack::streamType() const
{
return mStreamType;
}
-int AudioTrack::format() const
+audio_format_t AudioTrack::format() const
{
return mFormat;
}
@@ -298,7 +316,7 @@ uint32_t AudioTrack::frameCount() const
return mCblk->frameCount;
}
-int AudioTrack::frameSize() const
+size_t AudioTrack::frameSize() const
{
if (audio_is_linear_pcm(mFormat)) {
return channelCount()*audio_bytes_per_sample(mFormat);
@@ -337,18 +355,20 @@ void AudioTrack::start()
sp <IMemory> iMem = mCblkMemory;
audio_track_cblk_t* cblk = mCblk;
- if (mActive == 0) {
+ if (!mActive) {
mFlushed = false;
- mActive = 1;
+ mActive = true;
mNewPosition = cblk->server + mUpdatePeriod;
cblk->lock.lock();
cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
cblk->waitTimeMs = 0;
android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags);
if (t != 0) {
- t->run("AudioTrackThread", ANDROID_PRIORITY_AUDIO);
+ t->run("AudioTrackThread", ANDROID_PRIORITY_AUDIO);
} else {
- setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_AUDIO);
+ mPreviousPriority = getpriority(PRIO_PROCESS, 0);
+ mPreviousSchedulingGroup = androidGetThreadSchedulingGroup(0);
+ androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
}
ALOGV("start %p before lock cblk %p", this, mCblk);
@@ -366,11 +386,12 @@ void AudioTrack::start()
cblk->lock.unlock();
if (status != NO_ERROR) {
ALOGV("start() failed");
- mActive = 0;
+ mActive = false;
if (t != 0) {
t->requestExit();
} else {
- setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
+ setpriority(PRIO_PROCESS, 0, mPreviousPriority);
+ androidSetThreadSchedulingGroup(0, mPreviousSchedulingGroup);
}
}
}
@@ -390,8 +411,8 @@ void AudioTrack::stop()
}
AutoMutex lock(mLock);
- if (mActive == 1) {
- mActive = 0;
+ if (mActive) {
+ mActive = false;
mCblk->cv.signal();
mAudioTrack->stop();
// Cancel loops (If we are in the middle of a loop, playback
@@ -408,7 +429,8 @@ void AudioTrack::stop()
if (t != 0) {
t->requestExit();
} else {
- setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
+ setpriority(PRIO_PROCESS, 0, mPreviousPriority);
+ androidSetThreadSchedulingGroup(0, mPreviousSchedulingGroup);
}
}
@@ -419,7 +441,8 @@ void AudioTrack::stop()
bool AudioTrack::stopped() const
{
- return !mActive;
+ AutoMutex lock(mLock);
+ return stopped_l();
}
void AudioTrack::flush()
@@ -451,8 +474,8 @@ void AudioTrack::pause()
{
ALOGV("pause");
AutoMutex lock(mLock);
- if (mActive == 1) {
- mActive = 0;
+ if (mActive) {
+ mActive = false;
mAudioTrack->pause();
}
}
@@ -470,7 +493,7 @@ bool AudioTrack::muted() const
status_t AudioTrack::setVolume(float left, float right)
{
- if (left > 1.0f || right > 1.0f) {
+ if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) {
return BAD_VALUE;
}
@@ -478,8 +501,7 @@ status_t AudioTrack::setVolume(float left, float right)
mVolume[LEFT] = left;
mVolume[RIGHT] = right;
- // write must be atomic
- mCblk->volumeLR = (uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000);
+ mCblk->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000));
return NO_ERROR;
}
@@ -497,14 +519,14 @@ void AudioTrack::getVolume(float* left, float* right)
status_t AudioTrack::setAuxEffectSendLevel(float level)
{
ALOGV("setAuxEffectSendLevel(%f)", level);
- if (level > 1.0f) {
+ if (level < 0.0f || level > 1.0f) {
return BAD_VALUE;
}
AutoMutex lock(mLock);
mSendLevel = level;
- mCblk->sendLevel = uint16_t(level * 0x1000);
+ mCblk->setSendLevel(level);
return NO_ERROR;
}
@@ -582,13 +604,13 @@ status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCou
status_t AudioTrack::getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCount)
{
AutoMutex lock(mLock);
- if (loopStart != 0) {
+ if (loopStart != NULL) {
*loopStart = mCblk->loopStart;
}
- if (loopEnd != 0) {
+ if (loopEnd != NULL) {
*loopEnd = mCblk->loopEnd;
}
- if (loopCount != 0) {
+ if (loopCount != NULL) {
if (mCblk->loopCount < 0) {
*loopCount = -1;
} else {
@@ -601,7 +623,7 @@ status_t AudioTrack::getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCo
status_t AudioTrack::setMarkerPosition(uint32_t marker)
{
- if (mCbf == 0) return INVALID_OPERATION;
+ if (mCbf == NULL) return INVALID_OPERATION;
mMarkerPosition = marker;
mMarkerReached = false;
@@ -611,7 +633,7 @@ status_t AudioTrack::setMarkerPosition(uint32_t marker)
status_t AudioTrack::getMarkerPosition(uint32_t *marker)
{
- if (marker == 0) return BAD_VALUE;
+ if (marker == NULL) return BAD_VALUE;
*marker = mMarkerPosition;
@@ -620,7 +642,7 @@ status_t AudioTrack::getMarkerPosition(uint32_t *marker)
status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
{
- if (mCbf == 0) return INVALID_OPERATION;
+ if (mCbf == NULL) return INVALID_OPERATION;
uint32_t curPosition;
getPosition(&curPosition);
@@ -632,7 +654,7 @@ status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod)
{
- if (updatePeriod == 0) return BAD_VALUE;
+ if (updatePeriod == NULL) return BAD_VALUE;
*updatePeriod = mUpdatePeriod;
@@ -642,9 +664,10 @@ status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod)
status_t AudioTrack::setPosition(uint32_t position)
{
AutoMutex lock(mLock);
- Mutex::Autolock _l(mCblk->lock);
- if (!stopped()) return INVALID_OPERATION;
+ if (!stopped_l()) return INVALID_OPERATION;
+
+ Mutex::Autolock _l(mCblk->lock);
if (position > mCblk->user) return BAD_VALUE;
@@ -656,7 +679,7 @@ status_t AudioTrack::setPosition(uint32_t position)
status_t AudioTrack::getPosition(uint32_t *position)
{
- if (position == 0) return BAD_VALUE;
+ if (position == NULL) return BAD_VALUE;
AutoMutex lock(mLock);
*position = mFlushed ? 0 : mCblk->server;
@@ -667,7 +690,7 @@ status_t AudioTrack::reload()
{
AutoMutex lock(mLock);
- if (!stopped()) return INVALID_OPERATION;
+ if (!stopped_l()) return INVALID_OPERATION;
flush_l();
@@ -685,7 +708,7 @@ audio_io_handle_t AudioTrack::getOutput()
// must be called with mLock held
audio_io_handle_t AudioTrack::getOutput_l()
{
- return AudioSystem::getOutput((audio_stream_type_t)mStreamType,
+ return AudioSystem::getOutput(mStreamType,
mCblk->sampleRate, mFormat, mChannelMask, (audio_policy_output_flags_t)mFlags);
}
@@ -708,9 +731,9 @@ status_t AudioTrack::attachAuxEffect(int effectId)
// must be called with mLock held
status_t AudioTrack::createTrack_l(
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
@@ -802,9 +825,7 @@ status_t AudioTrack::createTrack_l(
ALOGE("Could not get control block");
return NO_INIT;
}
- mAudioTrack.clear();
mAudioTrack = track;
- mCblkMemory.clear();
mCblkMemory = cblk;
mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
android_atomic_or(CBLK_DIRECTION_OUT, &mCblk->flags);
@@ -816,8 +837,8 @@ status_t AudioTrack::createTrack_l(
mCblk->stepUser(mCblk->frameCount);
}
- mCblk->volumeLR = (uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000);
- mCblk->sendLevel = uint16_t(mSendLevel * 0x1000);
+ mCblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000));
+ mCblk->setSendLevel(mSendLevel);
mAudioTrack->attachAuxEffect(mAuxEffectId);
mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
mCblk->waitTimeMs = 0;
@@ -829,7 +850,7 @@ status_t AudioTrack::createTrack_l(
status_t AudioTrack::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;
@@ -851,12 +872,12 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
goto start_loop_here;
while (framesAvail == 0) {
active = mActive;
- if (UNLIKELY(!active)) {
+ if (CC_UNLIKELY(!active)) {
ALOGV("Not active and NO_MORE_BUFFERS");
cblk->lock.unlock();
return NO_MORE_BUFFERS;
}
- if (UNLIKELY(!waitCount)) {
+ if (CC_UNLIKELY(!waitCount)) {
cblk->lock.unlock();
return WOULD_BLOCK;
}
@@ -865,7 +886,7 @@ status_t AudioTrack::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();
@@ -874,7 +895,7 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
if (cblk->flags & CBLK_INVALID_MSK) {
goto create_new_track;
}
- if (__builtin_expect(result!=NO_ERROR, false)) {
+ if (CC_UNLIKELY(result != NO_ERROR)) {
cblk->waitTimeMs += waitTimeMs;
if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
// timing out when a loop has been set and we have already written upto loop end
@@ -978,7 +999,7 @@ ssize_t AudioTrack::write(const void* buffer, size_t userSize)
ssize_t written = 0;
const int8_t *src = (const int8_t *)buffer;
Buffer audioBuffer;
- size_t frameSz = (size_t)frameSize();
+ size_t frameSz = frameSize();
do {
audioBuffer.frameCount = userSize/frameSz;
@@ -998,12 +1019,7 @@ ssize_t AudioTrack::write(const void* buffer, size_t userSize)
if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) {
// Divide capacity by 2 to take expansion into account
toWrite = audioBuffer.size>>1;
- // 8 to 16 bit conversion
- int count = toWrite;
- int16_t *dst = (int16_t *)(audioBuffer.i8);
- while(count--) {
- *dst++ = (int16_t)(*src++^0x80) << 8;
- }
+ memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) src, toWrite);
} else {
toWrite = audioBuffer.size;
memcpy(audioBuffer.i8, src, toWrite);
@@ -1032,10 +1048,11 @@ bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
sp <IAudioTrack> audioTrack = mAudioTrack;
sp <IMemory> iMem = mCblkMemory;
audio_track_cblk_t* cblk = mCblk;
+ bool active = mActive;
mLock.unlock();
// Manage underrun callback
- if (mActive && (cblk->framesAvailable() == cblk->frameCount)) {
+ if (active && (cblk->framesAvailable() == cblk->frameCount)) {
ALOGV("Underrun 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_UNDERRUN, mUserData, 0);
@@ -1123,19 +1140,14 @@ bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
if (writtenSize > reqSize) writtenSize = reqSize;
if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) {
- // 8 to 16 bit conversion
- const int8_t *src = audioBuffer.i8 + writtenSize-1;
- int count = writtenSize;
- int16_t *dst = audioBuffer.i16 + writtenSize-1;
- while(count--) {
- *dst-- = (int16_t)(*src--^0x80) << 8;
- }
+ // 8 to 16 bit conversion, note that source and destination are the same address
+ memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize);
writtenSize <<= 1;
}
audioBuffer.size = writtenSize;
// NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for
- // 8 bit PCM data: in this case, mCblk->frameSize is based on a sampel size of
+ // 8 bit PCM data: in this case, mCblk->frameSize is based on a sample size of
// 16 bit.
audioBuffer.frameCount = writtenSize/mCblk->frameSize;
@@ -1307,15 +1319,15 @@ void AudioTrack::AudioTrackThread::onFirstRef()
audio_track_cblk_t::audio_track_cblk_t()
: lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0),
- userBase(0), serverBase(0), buffers(0), frameCount(0),
- loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), volumeLR(0),
- sendLevel(0), flags(0)
+ userBase(0), serverBase(0), buffers(NULL), frameCount(0),
+ loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), mVolumeLR(0x10001000),
+ mSendLevel(0), flags(0)
{
}
uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount)
{
- uint32_t u = this->user;
+ uint32_t u = user;
u += frameCount;
// Ensure that user is never ahead of server for AudioRecord
@@ -1324,16 +1336,16 @@ uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount)
if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) {
bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
}
- } else if (u > this->server) {
- ALOGW("stepServer occured after track reset");
- u = this->server;
+ } else if (u > server) {
+ ALOGW("stepServer occurred after track reset");
+ u = server;
}
if (u >= userBase + this->frameCount) {
userBase += this->frameCount;
}
- this->user = u;
+ user = u;
// Clear flow control error condition as new data has been written/read to/from buffer.
if (flags & CBLK_UNDERRUN_MSK) {
@@ -1350,7 +1362,7 @@ bool audio_track_cblk_t::stepServer(uint32_t frameCount)
return false;
}
- uint32_t s = this->server;
+ uint32_t s = server;
s += frameCount;
if (flags & CBLK_DIRECTION_MSK) {
@@ -1363,9 +1375,9 @@ bool audio_track_cblk_t::stepServer(uint32_t frameCount)
// while the mixer is processing a block: in this case,
// stepServer() is called After the flush() has reset u & s and
// we have s > u
- if (s > this->user) {
- ALOGW("stepServer occured after track reset");
- s = this->user;
+ if (s > user) {
+ ALOGW("stepServer occurred after track reset");
+ s = user;
}
}
@@ -1381,7 +1393,7 @@ bool audio_track_cblk_t::stepServer(uint32_t frameCount)
serverBase += this->frameCount;
}
- this->server = s;
+ server = s;
if (!(flags & CBLK_INVALID_MSK)) {
cv.signal();
@@ -1392,7 +1404,7 @@ bool audio_track_cblk_t::stepServer(uint32_t frameCount)
void* audio_track_cblk_t::buffer(uint32_t offset) const
{
- return (int8_t *)this->buffers + (offset - userBase) * this->frameSize;
+ return (int8_t *)buffers + (offset - userBase) * frameSize;
}
uint32_t audio_track_cblk_t::framesAvailable()
@@ -1403,8 +1415,8 @@ uint32_t audio_track_cblk_t::framesAvailable()
uint32_t audio_track_cblk_t::framesAvailable_l()
{
- uint32_t u = this->user;
- uint32_t s = this->server;
+ uint32_t u = user;
+ uint32_t s = server;
if (flags & CBLK_DIRECTION_MSK) {
uint32_t limit = (s < loopStart) ? s : loopStart;
@@ -1416,8 +1428,8 @@ uint32_t audio_track_cblk_t::framesAvailable_l()
uint32_t audio_track_cblk_t::framesReady()
{
- uint32_t u = this->user;
- uint32_t s = this->server;
+ uint32_t u = user;
+ uint32_t s = server;
if (flags & CBLK_DIRECTION_MSK) {
if (u < loopEnd) {
@@ -1462,4 +1474,3 @@ bool audio_track_cblk_t::tryLock()
// -------------------------------------------------------------------------
}; // namespace android
-
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index abd491f..fc5520f 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -82,9 +82,9 @@ public:
virtual sp<IAudioTrack> createTrack(
pid_t pid,
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
@@ -97,7 +97,7 @@ public:
sp<IAudioTrack> track;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
data.writeInt32(pid);
- data.writeInt32(streamType);
+ data.writeInt32((int32_t) streamType);
data.writeInt32(sampleRate);
data.writeInt32(format);
data.writeInt32(channelMask);
@@ -131,7 +131,7 @@ public:
pid_t pid,
int input,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
@@ -188,13 +188,13 @@ public:
return reply.readInt32();
}
- virtual uint32_t format(int output) const
+ virtual audio_format_t format(int output) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
data.writeInt32(output);
remote()->transact(FORMAT, data, &reply);
- return reply.readInt32();
+ return (audio_format_t) reply.readInt32();
}
virtual size_t frameCount(int output) const
@@ -249,47 +249,47 @@ public:
return reply.readInt32();
}
- virtual status_t setStreamVolume(int stream, float value, int output)
+ virtual status_t setStreamVolume(audio_stream_type_t stream, float value, int output)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(stream);
+ data.writeInt32((int32_t) stream);
data.writeFloat(value);
data.writeInt32(output);
remote()->transact(SET_STREAM_VOLUME, data, &reply);
return reply.readInt32();
}
- virtual status_t setStreamMute(int stream, bool muted)
+ virtual status_t setStreamMute(audio_stream_type_t stream, bool muted)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(stream);
+ data.writeInt32((int32_t) stream);
data.writeInt32(muted);
remote()->transact(SET_STREAM_MUTE, data, &reply);
return reply.readInt32();
}
- virtual float streamVolume(int stream, int output) const
+ virtual float streamVolume(audio_stream_type_t stream, int output) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(stream);
+ data.writeInt32((int32_t) stream);
data.writeInt32(output);
remote()->transact(STREAM_VOLUME, data, &reply);
return reply.readFloat();
}
- virtual bool streamMute(int stream) const
+ virtual bool streamMute(audio_stream_type_t stream) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(stream);
+ data.writeInt32((int32_t) stream);
remote()->transact(STREAM_MUTE, data, &reply);
return reply.readInt32();
}
- virtual status_t setMode(int mode)
+ virtual status_t setMode(audio_mode_t mode)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -343,7 +343,7 @@ public:
remote()->transact(REGISTER_CLIENT, data, &reply);
}
- virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
+ virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -356,7 +356,7 @@ public:
virtual int openOutput(uint32_t *pDevices,
uint32_t *pSamplingRate,
- uint32_t *pFormat,
+ audio_format_t *pFormat,
uint32_t *pChannels,
uint32_t *pLatencyMs,
uint32_t flags)
@@ -364,7 +364,7 @@ public:
Parcel data, reply;
uint32_t devices = pDevices ? *pDevices : 0;
uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
- uint32_t format = pFormat ? *pFormat : 0;
+ audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT;
uint32_t channels = pChannels ? *pChannels : 0;
uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
@@ -382,7 +382,7 @@ public:
if (pDevices) *pDevices = devices;
samplingRate = reply.readInt32();
if (pSamplingRate) *pSamplingRate = samplingRate;
- format = reply.readInt32();
+ format = (audio_format_t) reply.readInt32();
if (pFormat) *pFormat = format;
channels = reply.readInt32();
if (pChannels) *pChannels = channels;
@@ -430,14 +430,14 @@ public:
virtual int openInput(uint32_t *pDevices,
uint32_t *pSamplingRate,
- uint32_t *pFormat,
+ audio_format_t *pFormat,
uint32_t *pChannels,
- uint32_t acoustics)
+ audio_in_acoustics_t acoustics)
{
Parcel data, reply;
uint32_t devices = pDevices ? *pDevices : 0;
uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
- uint32_t format = pFormat ? *pFormat : 0;
+ audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT;
uint32_t channels = pChannels ? *pChannels : 0;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -445,14 +445,14 @@ public:
data.writeInt32(samplingRate);
data.writeInt32(format);
data.writeInt32(channels);
- data.writeInt32(acoustics);
+ data.writeInt32((int32_t) acoustics);
remote()->transact(OPEN_INPUT, data, &reply);
int input = reply.readInt32();
devices = reply.readInt32();
if (pDevices) *pDevices = devices;
samplingRate = reply.readInt32();
if (pSamplingRate) *pSamplingRate = samplingRate;
- format = reply.readInt32();
+ format = (audio_format_t) reply.readInt32();
if (pFormat) *pFormat = format;
channels = reply.readInt32();
if (pChannels) *pChannels = channels;
@@ -468,11 +468,11 @@ public:
return reply.readInt32();
}
- virtual status_t setStreamOutput(uint32_t stream, int output)
+ virtual status_t setStreamOutput(audio_stream_type_t stream, int output)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(stream);
+ data.writeInt32((int32_t) stream);
data.writeInt32(output);
remote()->transact(SET_STREAM_OUTPUT, data, &reply);
return reply.readInt32();
@@ -640,7 +640,7 @@ public:
*id = tmp;
}
tmp = reply.readInt32();
- if (enabled) {
+ if (enabled != NULL) {
*enabled = tmp;
}
effect = interface_cast<IEffect>(reply.readStrongBinder());
@@ -678,7 +678,7 @@ status_t BnAudioFlinger::onTransact(
pid_t pid = data.readInt32();
int streamType = data.readInt32();
uint32_t sampleRate = data.readInt32();
- int format = data.readInt32();
+ audio_format_t format = (audio_format_t) data.readInt32();
int channelCount = data.readInt32();
size_t bufferCount = data.readInt32();
uint32_t flags = data.readInt32();
@@ -687,7 +687,7 @@ status_t BnAudioFlinger::onTransact(
int sessionId = data.readInt32();
status_t status;
sp<IAudioTrack> track = createTrack(pid,
- streamType, sampleRate, format,
+ (audio_stream_type_t) streamType, sampleRate, format,
channelCount, bufferCount, flags, buffer, output, &sessionId, &status);
reply->writeInt32(sessionId);
reply->writeInt32(status);
@@ -699,7 +699,7 @@ status_t BnAudioFlinger::onTransact(
pid_t pid = data.readInt32();
int input = data.readInt32();
uint32_t sampleRate = data.readInt32();
- int format = data.readInt32();
+ audio_format_t format = (audio_format_t) data.readInt32();
int channelCount = data.readInt32();
size_t bufferCount = data.readInt32();
uint32_t flags = data.readInt32();
@@ -762,31 +762,31 @@ status_t BnAudioFlinger::onTransact(
int stream = data.readInt32();
float volume = data.readFloat();
int output = data.readInt32();
- reply->writeInt32( setStreamVolume(stream, volume, output) );
+ reply->writeInt32( setStreamVolume((audio_stream_type_t) stream, volume, output) );
return NO_ERROR;
} break;
case SET_STREAM_MUTE: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
int stream = data.readInt32();
- reply->writeInt32( setStreamMute(stream, data.readInt32()) );
+ reply->writeInt32( setStreamMute((audio_stream_type_t) stream, data.readInt32()) );
return NO_ERROR;
} break;
case STREAM_VOLUME: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
int stream = data.readInt32();
int output = data.readInt32();
- reply->writeFloat( streamVolume(stream, output) );
+ reply->writeFloat( streamVolume((audio_stream_type_t) stream, output) );
return NO_ERROR;
} break;
case STREAM_MUTE: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
int stream = data.readInt32();
- reply->writeInt32( streamMute(stream) );
+ reply->writeInt32( streamMute((audio_stream_type_t) stream) );
return NO_ERROR;
} break;
case SET_MODE: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- int mode = data.readInt32();
+ audio_mode_t mode = (audio_mode_t) data.readInt32();
reply->writeInt32( setMode(mode) );
return NO_ERROR;
} break;
@@ -825,7 +825,7 @@ status_t BnAudioFlinger::onTransact(
case GET_INPUTBUFFERSIZE: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
uint32_t sampleRate = data.readInt32();
- int format = data.readInt32();
+ audio_format_t format = (audio_format_t) data.readInt32();
int channelCount = data.readInt32();
reply->writeInt32( getInputBufferSize(sampleRate, format, channelCount) );
return NO_ERROR;
@@ -834,7 +834,7 @@ status_t BnAudioFlinger::onTransact(
CHECK_INTERFACE(IAudioFlinger, data, reply);
uint32_t devices = data.readInt32();
uint32_t samplingRate = data.readInt32();
- uint32_t format = data.readInt32();
+ audio_format_t format = (audio_format_t) data.readInt32();
uint32_t channels = data.readInt32();
uint32_t latency = data.readInt32();
uint32_t flags = data.readInt32();
@@ -879,15 +879,15 @@ status_t BnAudioFlinger::onTransact(
CHECK_INTERFACE(IAudioFlinger, data, reply);
uint32_t devices = data.readInt32();
uint32_t samplingRate = data.readInt32();
- uint32_t format = data.readInt32();
+ audio_format_t format = (audio_format_t) data.readInt32();
uint32_t channels = data.readInt32();
- uint32_t acoutics = data.readInt32();
+ audio_in_acoustics_t acoustics = (audio_in_acoustics_t) data.readInt32();
int input = openInput(&devices,
&samplingRate,
&format,
&channels,
- acoutics);
+ acoustics);
reply->writeInt32(input);
reply->writeInt32(devices);
reply->writeInt32(samplingRate);
@@ -904,7 +904,7 @@ status_t BnAudioFlinger::onTransact(
CHECK_INTERFACE(IAudioFlinger, data, reply);
uint32_t stream = data.readInt32();
int output = data.readInt32();
- reply->writeInt32(setStreamOutput(stream, output));
+ reply->writeInt32(setStreamOutput((audio_stream_type_t) stream, output));
return NO_ERROR;
} break;
case SET_VOICE_VOLUME: {
diff --git a/media/libmedia/IAudioFlingerClient.cpp b/media/libmedia/IAudioFlingerClient.cpp
index 5a3f250..9458bc0 100644
--- a/media/libmedia/IAudioFlingerClient.cpp
+++ b/media/libmedia/IAudioFlingerClient.cpp
@@ -73,7 +73,7 @@ status_t BnAudioFlingerClient::onTransact(
CHECK_INTERFACE(IAudioFlingerClient, data, reply);
int event = data.readInt32();
int ioHandle = data.readInt32();
- void *param2 = 0;
+ void *param2 = NULL;
AudioSystem::OutputDescriptor desc;
uint32_t stream;
if (event == AudioSystem::STREAM_CONFIG_CHANGED) {
diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp
index 50b4855..99385aa4 100644
--- a/media/libmedia/IAudioPolicyService.cpp
+++ b/media/libmedia/IAudioPolicyService.cpp
@@ -33,7 +33,7 @@ enum {
SET_DEVICE_CONNECTION_STATE = IBinder::FIRST_CALL_TRANSACTION,
GET_DEVICE_CONNECTION_STATE,
SET_PHONE_STATE,
- SET_RINGER_MODE,
+ SET_RINGER_MODE, // reserved, no longer used
SET_FORCE_USE,
GET_FORCE_USE,
GET_OUTPUT,
@@ -91,7 +91,7 @@ public:
return static_cast <audio_policy_dev_state_t>(reply.readInt32());
}
- virtual status_t setPhoneState(int state)
+ virtual status_t setPhoneState(audio_mode_t state)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
@@ -100,16 +100,6 @@ public:
return static_cast <status_t> (reply.readInt32());
}
- virtual status_t setRingerMode(uint32_t mode, uint32_t mask)
- {
- Parcel data, reply;
- data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
- data.writeInt32(mode);
- data.writeInt32(mask);
- remote()->transact(SET_RINGER_MODE, data, &reply);
- return static_cast <status_t> (reply.readInt32());
- }
-
virtual status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config)
{
Parcel data, reply;
@@ -132,7 +122,7 @@ public:
virtual audio_io_handle_t getOutput(
audio_stream_type_t stream,
uint32_t samplingRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channels,
audio_policy_output_flags_t flags)
{
@@ -154,7 +144,7 @@ public:
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
data.writeInt32(output);
- data.writeInt32(stream);
+ data.writeInt32((int32_t) stream);
data.writeInt32(session);
remote()->transact(START_OUTPUT, data, &reply);
return static_cast <status_t> (reply.readInt32());
@@ -167,7 +157,7 @@ public:
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
data.writeInt32(output);
- data.writeInt32(stream);
+ data.writeInt32((int32_t) stream);
data.writeInt32(session);
remote()->transact(STOP_OUTPUT, data, &reply);
return static_cast <status_t> (reply.readInt32());
@@ -182,16 +172,16 @@ public:
}
virtual audio_io_handle_t getInput(
- int inputSource,
+ audio_source_t inputSource,
uint32_t samplingRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channels,
audio_in_acoustics_t acoustics,
int audioSession)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
- data.writeInt32(inputSource);
+ data.writeInt32((int32_t) inputSource);
data.writeInt32(samplingRate);
data.writeInt32(static_cast <uint32_t>(format));
data.writeInt32(channels);
@@ -240,21 +230,28 @@ public:
return static_cast <status_t> (reply.readInt32());
}
- virtual status_t setStreamVolumeIndex(audio_stream_type_t stream, int index)
+ virtual status_t setStreamVolumeIndex(audio_stream_type_t stream,
+ int index,
+ audio_devices_t device)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
data.writeInt32(static_cast <uint32_t>(stream));
data.writeInt32(index);
+ data.writeInt32(static_cast <uint32_t>(device));
remote()->transact(SET_STREAM_VOLUME, data, &reply);
return static_cast <status_t> (reply.readInt32());
}
- virtual status_t getStreamVolumeIndex(audio_stream_type_t stream, int *index)
+ virtual status_t getStreamVolumeIndex(audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
data.writeInt32(static_cast <uint32_t>(stream));
+ data.writeInt32(static_cast <uint32_t>(device));
+
remote()->transact(GET_STREAM_VOLUME, data, &reply);
int lIndex = reply.readInt32();
if (index) *index = lIndex;
@@ -324,11 +321,11 @@ public:
return static_cast <status_t> (reply.readInt32());
}
- virtual bool isStreamActive(int stream, uint32_t inPastMs) const
+ virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
- data.writeInt32(stream);
+ data.writeInt32((int32_t) stream);
data.writeInt32(inPastMs);
remote()->transact(IS_STREAM_ACTIVE, data, &reply);
return reply.readInt32();
@@ -394,15 +391,7 @@ status_t BnAudioPolicyService::onTransact(
case SET_PHONE_STATE: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
- reply->writeInt32(static_cast <uint32_t>(setPhoneState(data.readInt32())));
- return NO_ERROR;
- } break;
-
- case SET_RINGER_MODE: {
- CHECK_INTERFACE(IAudioPolicyService, data, reply);
- uint32_t mode = data.readInt32();
- uint32_t mask = data.readInt32();
- reply->writeInt32(static_cast <uint32_t>(setRingerMode(mode, mask)));
+ reply->writeInt32(static_cast <uint32_t>(setPhoneState((audio_mode_t) data.readInt32())));
return NO_ERROR;
} break;
@@ -427,7 +416,7 @@ status_t BnAudioPolicyService::onTransact(
audio_stream_type_t stream =
static_cast <audio_stream_type_t>(data.readInt32());
uint32_t samplingRate = data.readInt32();
- uint32_t format = data.readInt32();
+ audio_format_t format = (audio_format_t) data.readInt32();
uint32_t channels = data.readInt32();
audio_policy_output_flags_t flags =
static_cast <audio_policy_output_flags_t>(data.readInt32());
@@ -472,9 +461,9 @@ status_t BnAudioPolicyService::onTransact(
case GET_INPUT: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
- int inputSource = data.readInt32();
+ audio_source_t inputSource = (audio_source_t) data.readInt32();
uint32_t samplingRate = data.readInt32();
- uint32_t format = 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());
@@ -525,7 +514,10 @@ status_t BnAudioPolicyService::onTransact(
audio_stream_type_t stream =
static_cast <audio_stream_type_t>(data.readInt32());
int index = data.readInt32();
- reply->writeInt32(static_cast <uint32_t>(setStreamVolumeIndex(stream, index)));
+ audio_devices_t device = static_cast <audio_devices_t>(data.readInt32());
+ reply->writeInt32(static_cast <uint32_t>(setStreamVolumeIndex(stream,
+ index,
+ device)));
return NO_ERROR;
} break;
@@ -533,8 +525,9 @@ status_t BnAudioPolicyService::onTransact(
CHECK_INTERFACE(IAudioPolicyService, data, reply);
audio_stream_type_t stream =
static_cast <audio_stream_type_t>(data.readInt32());
+ audio_devices_t device = static_cast <audio_devices_t>(data.readInt32());
int index;
- status_t status = getStreamVolumeIndex(stream, &index);
+ status_t status = getStreamVolumeIndex(stream, &index, device);
reply->writeInt32(index);
reply->writeInt32(static_cast <uint32_t>(status));
return NO_ERROR;
@@ -598,9 +591,9 @@ status_t BnAudioPolicyService::onTransact(
case IS_STREAM_ACTIVE: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
- int stream = data.readInt32();
+ audio_stream_type_t stream = (audio_stream_type_t) data.readInt32();
uint32_t inPastMs = (uint32_t)data.readInt32();
- reply->writeInt32( isStreamActive(stream, inPastMs) );
+ reply->writeInt32( isStreamActive((audio_stream_type_t) stream, inPastMs) );
return NO_ERROR;
} break;
diff --git a/media/libmedia/IAudioTrack.cpp b/media/libmedia/IAudioTrack.cpp
index 0b372f3..e618619 100644
--- a/media/libmedia/IAudioTrack.cpp
+++ b/media/libmedia/IAudioTrack.cpp
@@ -46,6 +46,18 @@ public:
{
}
+ virtual sp<IMemory> getCblk() const
+ {
+ Parcel data, reply;
+ sp<IMemory> cblk;
+ data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_CBLK, data, &reply);
+ if (status == NO_ERROR) {
+ cblk = interface_cast<IMemory>(reply.readStrongBinder());
+ }
+ return cblk;
+ }
+
virtual status_t start()
{
Parcel data, reply;
@@ -88,18 +100,6 @@ public:
remote()->transact(PAUSE, data, &reply);
}
- virtual sp<IMemory> getCblk() const
- {
- Parcel data, reply;
- sp<IMemory> cblk;
- data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
- status_t status = remote()->transact(GET_CBLK, data, &reply);
- if (status == NO_ERROR) {
- cblk = interface_cast<IMemory>(reply.readStrongBinder());
- }
- return cblk;
- }
-
virtual status_t attachAuxEffect(int effectId)
{
Parcel data, reply;
diff --git a/media/libmedia/IMediaDeathNotifier.cpp b/media/libmedia/IMediaDeathNotifier.cpp
index 8525482..aeb35a5 100644
--- a/media/libmedia/IMediaDeathNotifier.cpp
+++ b/media/libmedia/IMediaDeathNotifier.cpp
@@ -36,7 +36,7 @@ IMediaDeathNotifier::getMediaPlayerService()
{
ALOGV("getMediaPlayerService");
Mutex::Autolock _l(sServiceLock);
- if (sMediaPlayerService.get() == 0) {
+ if (sMediaPlayerService == 0) {
sp<IServiceManager> sm = defaultServiceManager();
sp<IBinder> binder;
do {
diff --git a/media/libmedia/IMediaPlayer.cpp b/media/libmedia/IMediaPlayer.cpp
index 9c1e6b7..64cc919 100644
--- a/media/libmedia/IMediaPlayer.cpp
+++ b/media/libmedia/IMediaPlayer.cpp
@@ -198,11 +198,11 @@ public:
return reply.readInt32();
}
- status_t setAudioStreamType(int type)
+ status_t setAudioStreamType(audio_stream_type_t stream)
{
Parcel data, reply;
data.writeInterfaceToken(IMediaPlayer::getInterfaceDescriptor());
- data.writeInt32(type);
+ data.writeInt32((int32_t) stream);
remote()->transact(SET_AUDIO_STREAM_TYPE, data, &reply);
return reply.readInt32();
}
@@ -397,7 +397,7 @@ status_t BnMediaPlayer::onTransact(
} break;
case SET_AUDIO_STREAM_TYPE: {
CHECK_INTERFACE(IMediaPlayer, data, reply);
- reply->writeInt32(setAudioStreamType(data.readInt32()));
+ reply->writeInt32(setAudioStreamType((audio_stream_type_t) data.readInt32()));
return NO_ERROR;
} break;
case SET_LOOPING: {
diff --git a/media/libmedia/IMediaPlayerService.cpp b/media/libmedia/IMediaPlayerService.cpp
index 8e4dd04..f5b5cbd 100644
--- a/media/libmedia/IMediaPlayerService.cpp
+++ b/media/libmedia/IMediaPlayerService.cpp
@@ -78,7 +78,7 @@ public:
return interface_cast<IMediaRecorder>(reply.readStrongBinder());
}
- virtual sp<IMemory> decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, int* pFormat)
+ virtual sp<IMemory> decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat)
{
Parcel data, reply;
data.writeInterfaceToken(IMediaPlayerService::getInterfaceDescriptor());
@@ -86,11 +86,11 @@ public:
remote()->transact(DECODE_URL, data, &reply);
*pSampleRate = uint32_t(reply.readInt32());
*pNumChannels = reply.readInt32();
- *pFormat = reply.readInt32();
+ *pFormat = (audio_format_t) reply.readInt32();
return interface_cast<IMemory>(reply.readStrongBinder());
}
- virtual sp<IMemory> decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, int* pFormat)
+ virtual sp<IMemory> decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat)
{
Parcel data, reply;
data.writeInterfaceToken(IMediaPlayerService::getInterfaceDescriptor());
@@ -100,7 +100,7 @@ public:
remote()->transact(DECODE_FD, data, &reply);
*pSampleRate = uint32_t(reply.readInt32());
*pNumChannels = reply.readInt32();
- *pFormat = reply.readInt32();
+ *pFormat = (audio_format_t) reply.readInt32();
return interface_cast<IMemory>(reply.readStrongBinder());
}
@@ -148,11 +148,11 @@ status_t BnMediaPlayerService::onTransact(
const char* url = data.readCString();
uint32_t sampleRate;
int numChannels;
- int format;
+ audio_format_t format;
sp<IMemory> player = decode(url, &sampleRate, &numChannels, &format);
reply->writeInt32(sampleRate);
reply->writeInt32(numChannels);
- reply->writeInt32(format);
+ reply->writeInt32((int32_t) format);
reply->writeStrongBinder(player->asBinder());
return NO_ERROR;
} break;
@@ -163,11 +163,11 @@ status_t BnMediaPlayerService::onTransact(
int64_t length = data.readInt64();
uint32_t sampleRate;
int numChannels;
- int format;
+ audio_format_t format;
sp<IMemory> player = decode(fd, offset, length, &sampleRate, &numChannels, &format);
reply->writeInt32(sampleRate);
reply->writeInt32(numChannels);
- reply->writeInt32(format);
+ reply->writeInt32((int32_t) format);
reply->writeStrongBinder(player->asBinder());
return NO_ERROR;
} break;
diff --git a/media/libmedia/IOMX.cpp b/media/libmedia/IOMX.cpp
index d2f5f71..27c7e03 100644
--- a/media/libmedia/IOMX.cpp
+++ b/media/libmedia/IOMX.cpp
@@ -59,9 +59,10 @@ public:
: BpInterface<IOMX>(impl) {
}
- virtual bool livesLocally(pid_t pid) {
+ virtual bool livesLocally(node_id node, pid_t pid) {
Parcel data, reply;
data.writeInterfaceToken(IOMX::getInterfaceDescriptor());
+ data.writeIntPtr((intptr_t)node);
data.writeInt32(pid);
remote()->transact(LIVES_LOCALLY, data, &reply);
@@ -417,7 +418,9 @@ status_t BnOMX::onTransact(
case LIVES_LOCALLY:
{
CHECK_INTERFACE(IOMX, data, reply);
- reply->writeInt32(livesLocally((pid_t)data.readInt32()));
+ node_id node = (void *)data.readIntPtr();
+ pid_t pid = (pid_t)data.readInt32();
+ reply->writeInt32(livesLocally(node, pid));
return OK;
}
diff --git a/media/libmedia/JetPlayer.cpp b/media/libmedia/JetPlayer.cpp
index fa5b67a..8456db5 100644
--- a/media/libmedia/JetPlayer.cpp
+++ b/media/libmedia/JetPlayer.cpp
@@ -91,7 +91,7 @@ int JetPlayer::init()
mAudioTrack = new AudioTrack();
mAudioTrack->set(AUDIO_STREAM_MUSIC, //TODO parametrize this
pLibConfig->sampleRate,
- 1, // format = PCM 16bits per sample,
+ AUDIO_FORMAT_PCM_16_BIT,
(pLibConfig->numChannels == 2) ? AUDIO_CHANNEL_OUT_STEREO : AUDIO_CHANNEL_OUT_MONO,
mTrackBufferSize,
0);
@@ -100,7 +100,8 @@ int JetPlayer::init()
{
Mutex::Autolock l(mMutex);
ALOGV("JetPlayer::init(): trying to start render thread");
- createThreadEtc(renderThread, this, "jetRenderThread", ANDROID_PRIORITY_AUDIO);
+ mThread = new JetPlayerThread(this);
+ mThread->run("jetRenderThread", ANDROID_PRIORITY_AUDIO);
mCondition.wait(mMutex);
}
if (mTid > 0) {
@@ -156,12 +157,6 @@ int JetPlayer::release()
//-------------------------------------------------------------------------------------------------
-int JetPlayer::renderThread(void* p) {
-
- return ((JetPlayer*)p)->render();
-}
-
-//-------------------------------------------------------------------------------------------------
int JetPlayer::render() {
EAS_RESULT result = EAS_FAILURE;
EAS_I32 count;
@@ -173,10 +168,6 @@ int JetPlayer::render() {
// allocate render buffer
mAudioBuffer =
new EAS_PCM[pLibConfig->mixBufferSize * pLibConfig->numChannels * MIX_NUM_BUFFERS];
- if (!mAudioBuffer) {
- ALOGE("JetPlayer::render(): mAudioBuffer allocate failed");
- goto threadExit;
- }
// signal main thread that we started
{
@@ -343,8 +334,8 @@ int JetPlayer::loadFromFile(const char* path)
Mutex::Autolock lock(mMutex);
mEasJetFileLoc = (EAS_FILE_LOCATOR) malloc(sizeof(EAS_FILE));
- memset(mJetFilePath, 0, 256);
- strncpy(mJetFilePath, path, strlen(path));
+ strncpy(mJetFilePath, path, sizeof(mJetFilePath));
+ mJetFilePath[sizeof(mJetFilePath) - 1] = '\0';
mEasJetFileLoc->path = mJetFilePath;
mEasJetFileLoc->fd = 0;
diff --git a/media/libmedia/MediaScannerClient.cpp b/media/libmedia/MediaScannerClient.cpp
index 40b8188..9fe1820 100644
--- a/media/libmedia/MediaScannerClient.cpp
+++ b/media/libmedia/MediaScannerClient.cpp
@@ -173,6 +173,7 @@ void MediaScannerClient::convertValues(uint32_t encoding)
const char* source = mValues->getEntry(i);
int targetLength = len * 3 + 1;
char* buffer = new char[targetLength];
+ // don't normally check for NULL, but in this case targetLength may be large
if (!buffer)
break;
char* target = buffer;
diff --git a/media/libmedia/ToneGenerator.cpp b/media/libmedia/ToneGenerator.cpp
index 35dfbb8..e6e989d 100644
--- a/media/libmedia/ToneGenerator.cpp
+++ b/media/libmedia/ToneGenerator.cpp
@@ -751,7 +751,7 @@ const ToneGenerator::ToneDescriptor ToneGenerator::sToneDescriptors[] = {
// Used by ToneGenerator::getToneForRegion() to convert user specified supervisory tone type
// to actual tone for current region.
-const unsigned char ToneGenerator::sToneMappingTable[NUM_REGIONS-1][NUM_SUP_TONES] = {
+const unsigned char /*tone_type*/ ToneGenerator::sToneMappingTable[NUM_REGIONS-1][NUM_SUP_TONES] = {
{ // ANSI
TONE_ANSI_DIAL, // TONE_SUP_DIAL
TONE_ANSI_BUSY, // TONE_SUP_BUSY
@@ -798,7 +798,7 @@ const unsigned char ToneGenerator::sToneMappingTable[NUM_REGIONS-1][NUM_SUP_TONE
// none
//
////////////////////////////////////////////////////////////////////////////////
-ToneGenerator::ToneGenerator(int streamType, float volume, bool threadCanCallJava) {
+ToneGenerator::ToneGenerator(audio_stream_type_t streamType, float volume, bool threadCanCallJava) {
ALOGV("ToneGenerator constructor: streamType=%d, volume=%f\n", streamType, volume);
@@ -811,9 +811,9 @@ ToneGenerator::ToneGenerator(int streamType, float volume, bool threadCanCallJav
mThreadCanCallJava = threadCanCallJava;
mStreamType = streamType;
mVolume = volume;
- mpAudioTrack = 0;
- mpToneDesc = 0;
- mpNewToneDesc = 0;
+ mpAudioTrack = NULL;
+ mpToneDesc = NULL;
+ mpNewToneDesc = NULL;
// Generate tone by chunks of 20 ms to keep cadencing precision
mProcessSize = (mSamplingRate * 20) / 1000;
@@ -855,7 +855,7 @@ ToneGenerator::ToneGenerator(int streamType, float volume, bool threadCanCallJav
ToneGenerator::~ToneGenerator() {
ALOGV("ToneGenerator destructor\n");
- if (mpAudioTrack) {
+ if (mpAudioTrack != NULL) {
stopTone();
ALOGV("Delete Track: %p\n", mpAudioTrack);
delete mpAudioTrack;
@@ -878,7 +878,7 @@ ToneGenerator::~ToneGenerator() {
// none
//
////////////////////////////////////////////////////////////////////////////////
-bool ToneGenerator::startTone(int toneType, int durationMs) {
+bool ToneGenerator::startTone(tone_type toneType, int durationMs) {
bool lResult = false;
status_t lStatus;
@@ -1012,15 +1012,11 @@ bool ToneGenerator::initAudioTrack() {
if (mpAudioTrack) {
delete mpAudioTrack;
- mpAudioTrack = 0;
+ mpAudioTrack = NULL;
}
// Open audio track in mono, PCM 16bit, default sampling rate, default buffer size
mpAudioTrack = new AudioTrack();
- if (mpAudioTrack == 0) {
- ALOGE("AudioTrack allocation failed");
- goto initAudioTrack_exit;
- }
ALOGV("Create Track: %p\n", mpAudioTrack);
mpAudioTrack->set(mStreamType,
@@ -1052,7 +1048,7 @@ initAudioTrack_exit:
if (mpAudioTrack) {
ALOGV("Delete Track I: %p\n", mpAudioTrack);
delete mpAudioTrack;
- mpAudioTrack = 0;
+ mpAudioTrack = NULL;
}
return false;
@@ -1321,7 +1317,7 @@ audioCallback_EndLoop:
bool ToneGenerator::prepareWave() {
unsigned int segmentIdx = 0;
- if (!mpNewToneDesc) {
+ if (mpNewToneDesc == NULL) {
return false;
}
@@ -1353,9 +1349,6 @@ bool ToneGenerator::prepareWave() {
new ToneGenerator::WaveGenerator((unsigned short)mSamplingRate,
frequency,
TONEGEN_GAIN/lNumWaves);
- if (lpWaveGen == 0) {
- goto prepareWave_exit;
- }
mWaveGens.add(frequency, lpWaveGen);
}
frequency = mpNewToneDesc->segments[segmentIdx].waveFreq[++freqIdx];
@@ -1375,12 +1368,6 @@ bool ToneGenerator::prepareWave() {
}
return true;
-
-prepareWave_exit:
-
- clearWaveGens();
-
- return false;
}
@@ -1447,13 +1434,13 @@ void ToneGenerator::clearWaveGens() {
// none
//
////////////////////////////////////////////////////////////////////////////////
-int ToneGenerator::getToneForRegion(int toneType) {
- int regionTone;
+ToneGenerator::tone_type ToneGenerator::getToneForRegion(tone_type toneType) {
+ tone_type regionTone;
if (mRegion == CEPT || toneType < FIRST_SUP_TONE || toneType > LAST_SUP_TONE) {
regionTone = toneType;
} else {
- regionTone = sToneMappingTable[mRegion][toneType - FIRST_SUP_TONE];
+ regionTone = (tone_type) sToneMappingTable[mRegion][toneType - FIRST_SUP_TONE];
}
ALOGV("getToneForRegion, tone %d, region %d, regionTone %d", toneType, mRegion, regionTone);
diff --git a/media/libmedia/Visualizer.cpp b/media/libmedia/Visualizer.cpp
index d08ffa5..13b64e9 100644
--- a/media/libmedia/Visualizer.cpp
+++ b/media/libmedia/Visualizer.cpp
@@ -27,8 +27,7 @@
#include <cutils/bitops.h>
#include <media/Visualizer.h>
-
-extern void fixed_fft_real(int n, int32_t *v);
+#include <audio_utils/fixedfft.h>
namespace android {
@@ -54,7 +53,7 @@ Visualizer::~Visualizer()
status_t Visualizer::setEnabled(bool enabled)
{
- Mutex::Autolock _l(mLock);
+ Mutex::Autolock _l(mCaptureLock);
sp<CaptureThread> t = mCaptureThread;
if (t != 0) {
@@ -74,7 +73,7 @@ status_t Visualizer::setEnabled(bool enabled)
if (status == NO_ERROR) {
if (t != 0) {
if (enabled) {
- t->run("AudioTrackThread");
+ t->run("Visualizer");
} else {
t->requestExit();
}
@@ -93,7 +92,7 @@ status_t Visualizer::setCaptureCallBack(capture_cbk_t cbk, void* user, uint32_t
if (rate > CAPTURE_RATE_MAX) {
return BAD_VALUE;
}
- Mutex::Autolock _l(mLock);
+ Mutex::Autolock _l(mCaptureLock);
if (mEnabled) {
return INVALID_OPERATION;
@@ -115,10 +114,6 @@ status_t Visualizer::setCaptureCallBack(capture_cbk_t cbk, void* user, uint32_t
if (cbk != NULL) {
mCaptureThread = new CaptureThread(*this, rate, ((flags & CAPTURE_CALL_JAVA) != 0));
- if (mCaptureThread == 0) {
- ALOGE("Could not create callback thread");
- return NO_INIT;
- }
}
ALOGV("setCaptureCallBack() rate: %d thread %p flags 0x%08x",
rate, mCaptureThread.get(), mCaptureFlags);
@@ -133,7 +128,7 @@ status_t Visualizer::setCaptureSize(uint32_t size)
return BAD_VALUE;
}
- Mutex::Autolock _l(mLock);
+ Mutex::Autolock _l(mCaptureLock);
if (mEnabled) {
return INVALID_OPERATION;
}
@@ -235,7 +230,7 @@ status_t Visualizer::doFft(uint8_t *fft, uint8_t *waveform)
void Visualizer::periodicCapture()
{
- Mutex::Autolock _l(mLock);
+ Mutex::Autolock _l(mCaptureLock);
ALOGV("periodicCapture() %p mCaptureCallBack %p mCaptureFlags 0x%08x",
this, mCaptureCallBack, mCaptureFlags);
if (mCaptureCallBack != NULL &&
diff --git a/media/libmedia/autodetect.cpp b/media/libmedia/autodetect.cpp
index dfcc6a0..be5c3b2 100644
--- a/media/libmedia/autodetect.cpp
+++ b/media/libmedia/autodetect.cpp
@@ -16,7 +16,7 @@
#include "autodetect.h"
-typedef struct CharRange {
+struct CharRange {
uint16_t first;
uint16_t last;
};
diff --git a/media/libmedia/fixedfft.cpp b/media/libmedia/fixedfft.cpp
deleted file mode 100644
index 2b495e6..0000000
--- a/media/libmedia/fixedfft.cpp
+++ /dev/null
@@ -1,162 +0,0 @@
-/*
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-/* A Fixed point implementation of Fast Fourier Transform (FFT). Complex numbers
- * are represented by 32-bit integers, where higher 16 bits are real part and
- * lower ones are imaginary part. Few compromises are made between efficiency,
- * accuracy, and maintainability. To make it fast, arithmetic shifts are used
- * instead of divisions, and bitwise inverses are used instead of negates. To
- * keep it small, only radix-2 Cooley-Tukey algorithm is implemented, and only
- * half of the twiddle factors are stored. Although there are still ways to make
- * it even faster or smaller, it costs too much on one of the aspects.
- */
-
-#include <stdio.h>
-#include <stdint.h>
-#ifdef __arm__
-#include <machine/cpu-features.h>
-#endif
-
-#define LOG_FFT_SIZE 10
-#define MAX_FFT_SIZE (1 << LOG_FFT_SIZE)
-
-static const int32_t twiddle[MAX_FFT_SIZE / 4] = {
- 0x00008000, 0xff378001, 0xfe6e8002, 0xfda58006, 0xfcdc800a, 0xfc13800f,
- 0xfb4a8016, 0xfa81801e, 0xf9b88027, 0xf8ef8032, 0xf827803e, 0xf75e804b,
- 0xf6958059, 0xf5cd8068, 0xf5058079, 0xf43c808b, 0xf374809e, 0xf2ac80b2,
- 0xf1e480c8, 0xf11c80de, 0xf05580f6, 0xef8d8110, 0xeec6812a, 0xedff8146,
- 0xed388163, 0xec718181, 0xebab81a0, 0xeae481c1, 0xea1e81e2, 0xe9588205,
- 0xe892822a, 0xe7cd824f, 0xe7078276, 0xe642829d, 0xe57d82c6, 0xe4b982f1,
- 0xe3f4831c, 0xe3308349, 0xe26d8377, 0xe1a983a6, 0xe0e683d6, 0xe0238407,
- 0xdf61843a, 0xde9e846e, 0xdddc84a3, 0xdd1b84d9, 0xdc598511, 0xdb998549,
- 0xdad88583, 0xda1885be, 0xd95885fa, 0xd8988637, 0xd7d98676, 0xd71b86b6,
- 0xd65c86f6, 0xd59e8738, 0xd4e1877b, 0xd42487c0, 0xd3678805, 0xd2ab884c,
- 0xd1ef8894, 0xd13488dd, 0xd0798927, 0xcfbe8972, 0xcf0489be, 0xce4b8a0c,
- 0xcd928a5a, 0xccd98aaa, 0xcc218afb, 0xcb698b4d, 0xcab28ba0, 0xc9fc8bf5,
- 0xc9468c4a, 0xc8908ca1, 0xc7db8cf8, 0xc7278d51, 0xc6738dab, 0xc5c08e06,
- 0xc50d8e62, 0xc45b8ebf, 0xc3a98f1d, 0xc2f88f7d, 0xc2488fdd, 0xc198903e,
- 0xc0e990a1, 0xc03a9105, 0xbf8c9169, 0xbedf91cf, 0xbe329236, 0xbd86929e,
- 0xbcda9307, 0xbc2f9371, 0xbb8593dc, 0xbadc9448, 0xba3394b5, 0xb98b9523,
- 0xb8e39592, 0xb83c9603, 0xb7969674, 0xb6f196e6, 0xb64c9759, 0xb5a897ce,
- 0xb5059843, 0xb46298b9, 0xb3c09930, 0xb31f99a9, 0xb27f9a22, 0xb1df9a9c,
- 0xb1409b17, 0xb0a29b94, 0xb0059c11, 0xaf689c8f, 0xaecc9d0e, 0xae319d8e,
- 0xad979e0f, 0xacfd9e91, 0xac659f14, 0xabcd9f98, 0xab36a01c, 0xaaa0a0a2,
- 0xaa0aa129, 0xa976a1b0, 0xa8e2a238, 0xa84fa2c2, 0xa7bda34c, 0xa72ca3d7,
- 0xa69ca463, 0xa60ca4f0, 0xa57ea57e, 0xa4f0a60c, 0xa463a69c, 0xa3d7a72c,
- 0xa34ca7bd, 0xa2c2a84f, 0xa238a8e2, 0xa1b0a976, 0xa129aa0a, 0xa0a2aaa0,
- 0xa01cab36, 0x9f98abcd, 0x9f14ac65, 0x9e91acfd, 0x9e0fad97, 0x9d8eae31,
- 0x9d0eaecc, 0x9c8faf68, 0x9c11b005, 0x9b94b0a2, 0x9b17b140, 0x9a9cb1df,
- 0x9a22b27f, 0x99a9b31f, 0x9930b3c0, 0x98b9b462, 0x9843b505, 0x97ceb5a8,
- 0x9759b64c, 0x96e6b6f1, 0x9674b796, 0x9603b83c, 0x9592b8e3, 0x9523b98b,
- 0x94b5ba33, 0x9448badc, 0x93dcbb85, 0x9371bc2f, 0x9307bcda, 0x929ebd86,
- 0x9236be32, 0x91cfbedf, 0x9169bf8c, 0x9105c03a, 0x90a1c0e9, 0x903ec198,
- 0x8fddc248, 0x8f7dc2f8, 0x8f1dc3a9, 0x8ebfc45b, 0x8e62c50d, 0x8e06c5c0,
- 0x8dabc673, 0x8d51c727, 0x8cf8c7db, 0x8ca1c890, 0x8c4ac946, 0x8bf5c9fc,
- 0x8ba0cab2, 0x8b4dcb69, 0x8afbcc21, 0x8aaaccd9, 0x8a5acd92, 0x8a0cce4b,
- 0x89becf04, 0x8972cfbe, 0x8927d079, 0x88ddd134, 0x8894d1ef, 0x884cd2ab,
- 0x8805d367, 0x87c0d424, 0x877bd4e1, 0x8738d59e, 0x86f6d65c, 0x86b6d71b,
- 0x8676d7d9, 0x8637d898, 0x85fad958, 0x85beda18, 0x8583dad8, 0x8549db99,
- 0x8511dc59, 0x84d9dd1b, 0x84a3dddc, 0x846ede9e, 0x843adf61, 0x8407e023,
- 0x83d6e0e6, 0x83a6e1a9, 0x8377e26d, 0x8349e330, 0x831ce3f4, 0x82f1e4b9,
- 0x82c6e57d, 0x829de642, 0x8276e707, 0x824fe7cd, 0x822ae892, 0x8205e958,
- 0x81e2ea1e, 0x81c1eae4, 0x81a0ebab, 0x8181ec71, 0x8163ed38, 0x8146edff,
- 0x812aeec6, 0x8110ef8d, 0x80f6f055, 0x80def11c, 0x80c8f1e4, 0x80b2f2ac,
- 0x809ef374, 0x808bf43c, 0x8079f505, 0x8068f5cd, 0x8059f695, 0x804bf75e,
- 0x803ef827, 0x8032f8ef, 0x8027f9b8, 0x801efa81, 0x8016fb4a, 0x800ffc13,
- 0x800afcdc, 0x8006fda5, 0x8002fe6e, 0x8001ff37,
-};
-
-/* Returns the multiplication of \conj{a} and {b}. */
-static inline int32_t mult(int32_t a, int32_t b)
-{
-#if __ARM_ARCH__ >= 6
- int32_t t = b;
- __asm__("smuad %0, %0, %1" : "+r" (t) : "r" (a));
- __asm__("smusdx %0, %0, %1" : "+r" (b) : "r" (a));
- __asm__("pkhtb %0, %0, %1, ASR #16" : "+r" (t) : "r" (b));
- return t;
-#else
- return (((a >> 16) * (b >> 16) + (int16_t)a * (int16_t)b) & ~0xFFFF) |
- ((((a >> 16) * (int16_t)b - (int16_t)a * (b >> 16)) >> 16) & 0xFFFF);
-#endif
-}
-
-static inline int32_t half(int32_t a)
-{
-#if __ARM_ARCH__ >= 6
- __asm__("shadd16 %0, %0, %1" : "+r" (a) : "r" (0));
- return a;
-#else
- return ((a >> 1) & ~0x8000) | (a & 0x8000);
-#endif
-}
-
-void fixed_fft(int n, int32_t *v)
-{
- int scale = LOG_FFT_SIZE, i, p, r;
-
- for (r = 0, i = 1; i < n; ++i) {
- for (p = n; !(p & r); p >>= 1, r ^= p);
- if (i < r) {
- int32_t t = v[i];
- v[i] = v[r];
- v[r] = t;
- }
- }
-
- for (p = 1; p < n; p <<= 1) {
- --scale;
-
- for (i = 0; i < n; i += p << 1) {
- int32_t x = half(v[i]);
- int32_t y = half(v[i + p]);
- v[i] = x + y;
- v[i + p] = x - y;
- }
-
- for (r = 1; r < p; ++r) {
- int32_t w = MAX_FFT_SIZE / 4 - (r << scale);
- i = w >> 31;
- w = twiddle[(w ^ i) - i] ^ (i << 16);
- for (i = r; i < n; i += p << 1) {
- int32_t x = half(v[i]);
- int32_t y = mult(w, v[i + p]);
- v[i] = x - y;
- v[i + p] = x + y;
- }
- }
- }
-}
-
-void fixed_fft_real(int n, int32_t *v)
-{
- int scale = LOG_FFT_SIZE, m = n >> 1, i;
-
- fixed_fft(n, v);
- for (i = 1; i <= n; i <<= 1, --scale);
- v[0] = mult(~v[0], 0x80008000);
- v[m] = half(v[m]);
-
- for (i = 1; i < n >> 1; ++i) {
- int32_t x = half(v[i]);
- int32_t z = half(v[n - i]);
- int32_t y = z - (x ^ 0xFFFF);
- x = half(x + (z ^ 0xFFFF));
- y = mult(y, twiddle[i << scale]);
- v[i] = x - y;
- v[n - i] = (x + y) ^ 0xFFFF;
- }
-}
diff --git a/media/libmedia/mediametadataretriever.cpp b/media/libmedia/mediametadataretriever.cpp
index 88e269f..8d53357 100644
--- a/media/libmedia/mediametadataretriever.cpp
+++ b/media/libmedia/mediametadataretriever.cpp
@@ -35,7 +35,7 @@ sp<MediaMetadataRetriever::DeathNotifier> MediaMetadataRetriever::sDeathNotifier
const sp<IMediaPlayerService>& MediaMetadataRetriever::getService()
{
Mutex::Autolock lock(sServiceLock);
- if (sService.get() == 0) {
+ if (sService == 0) {
sp<IServiceManager> sm = defaultServiceManager();
sp<IBinder> binder;
do {
diff --git a/media/libmedia/mediaplayer.cpp b/media/libmedia/mediaplayer.cpp
index 2284927..f1c47dd 100644
--- a/media/libmedia/mediaplayer.cpp
+++ b/media/libmedia/mediaplayer.cpp
@@ -30,7 +30,7 @@
#include <gui/SurfaceTextureClient.h>
#include <media/mediaplayer.h>
-#include <media/AudioTrack.h>
+#include <media/AudioSystem.h>
#include <surfaceflinger/Surface.h>
@@ -478,7 +478,7 @@ status_t MediaPlayer::reset()
return reset_l();
}
-status_t MediaPlayer::setAudioStreamType(int type)
+status_t MediaPlayer::setAudioStreamType(audio_stream_type_t type)
{
ALOGV("MediaPlayer::setAudioStreamType");
Mutex::Autolock _l(mLock);
@@ -709,7 +709,7 @@ void MediaPlayer::notify(int msg, int ext1, int ext2, const Parcel *obj)
}
}
-/*static*/ sp<IMemory> MediaPlayer::decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, int* pFormat)
+/*static*/ sp<IMemory> MediaPlayer::decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat)
{
ALOGV("decode(%s)", url);
sp<IMemory> p;
@@ -729,7 +729,7 @@ void MediaPlayer::died()
notify(MEDIA_ERROR, MEDIA_ERROR_SERVER_DIED, 0);
}
-/*static*/ sp<IMemory> MediaPlayer::decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, int* pFormat)
+/*static*/ sp<IMemory> MediaPlayer::decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat)
{
ALOGV("decode(%d, %lld, %lld)", fd, offset, length);
sp<IMemory> p;
generated by cgit v1.1 at 2025-08-22 17:59:54 +0000