Merge "audio effects: rename configure command"
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
index d0d9ca6..3539e3f 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
@@ -376,7 +376,7 @@
bool tooLate = (mVideoLateByUs > 40000);
if (tooLate) {
- ALOGV("video late by %lld us (%.2f secs)", lateByUs, lateByUs / 1E6);
+ ALOGV("video late by %lld us (%.2f secs)", mVideoLateByUs, mVideoLateByUs / 1E6);
} else {
ALOGV("rendering video at media time %.2f secs", mediaTimeUs / 1E6);
}
diff --git a/media/libstagefright/SampleTable.cpp b/media/libstagefright/SampleTable.cpp
index 8e790fc..3e287fa 100644
--- a/media/libstagefright/SampleTable.cpp
+++ b/media/libstagefright/SampleTable.cpp
@@ -632,6 +632,13 @@
break;
}
}
+ if (left == mNumSyncSamples) {
+ if (flags == kFlagAfter) {
+ LOGE("tried to find a sync frame after the last one: %d", left);
+ return ERROR_OUT_OF_RANGE;
+ }
+ left = left - 1;
+ }
// Now ssi[left] is the sync sample index just before (or at)
// start_sample_index.
diff --git a/media/libstagefright/SurfaceMediaSource.cpp b/media/libstagefright/SurfaceMediaSource.cpp
index 51a7541..2f807d0 100644
--- a/media/libstagefright/SurfaceMediaSource.cpp
+++ b/media/libstagefright/SurfaceMediaSource.cpp
@@ -338,7 +338,9 @@
(uint32_t(buffer->height) != h) ||
(uint32_t(buffer->format) != format) ||
((uint32_t(buffer->usage) & usage) != usage)) {
- usage |= GraphicBuffer::USAGE_HW_VIDEO_ENCODER;
+ // XXX: This will be changed to USAGE_HW_VIDEO_ENCODER once driver
+ // issues with that flag get fixed.
+ usage |= GraphicBuffer::USAGE_HW_TEXTURE;
status_t error;
sp<GraphicBuffer> graphicBuffer(
mGraphicBufferAlloc->createGraphicBuffer(
diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk
index fa49592..52834db 100644
--- a/services/audioflinger/Android.mk
+++ b/services/audioflinger/Android.mk
@@ -11,9 +11,11 @@
AudioPolicyService.cpp
LOCAL_C_INCLUDES := \
- system/media/audio_effects/include
+ system/media/audio_effects/include \
+ system/media/audio_utils/include
LOCAL_SHARED_LIBRARIES := \
+ libaudioutils \
libcutils \
libutils \
libbinder \
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 2a55120..2090f1b 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -54,6 +54,8 @@
#include <audio_effects/effect_ns.h>
#include <audio_effects/effect_aec.h>
+#include <audio_utils/primitives.h>
+
#include <cpustats/ThreadCpuUsage.h>
#include <powermanager/PowerManager.h>
// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds
@@ -160,7 +162,7 @@
AudioFlinger::AudioFlinger()
: BnAudioFlinger(),
- mPrimaryHardwareDev(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1),
+ mPrimaryHardwareDev(NULL), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1),
mBtNrecIsOff(false)
{
}
@@ -1066,9 +1068,9 @@
// sendConfigEvent_l() must be called with ThreadBase::mLock held
void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param)
{
- ConfigEvent *configEvent = new ConfigEvent();
- configEvent->mEvent = event;
- configEvent->mParam = param;
+ ConfigEvent configEvent;
+ configEvent.mEvent = event;
+ configEvent.mParam = param;
mConfigEvents.add(configEvent);
ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
mWaitWorkCV.signal();
@@ -1079,15 +1081,14 @@
mLock.lock();
while(!mConfigEvents.isEmpty()) {
ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
- ConfigEvent *configEvent = mConfigEvents[0];
+ ConfigEvent configEvent = mConfigEvents[0];
mConfigEvents.removeAt(0);
// release mLock before locking AudioFlinger mLock: lock order is always
// AudioFlinger then ThreadBase to avoid cross deadlock
mLock.unlock();
mAudioFlinger->mLock.lock();
- audioConfigChanged_l(configEvent->mEvent, configEvent->mParam);
+ audioConfigChanged_l(configEvent.mEvent, configEvent.mParam);
mAudioFlinger->mLock.unlock();
- delete configEvent;
mLock.lock();
}
mLock.unlock();
@@ -1134,7 +1135,7 @@
snprintf(buffer, SIZE, " Index event param\n");
result.append(buffer);
for (size_t i = 0; i < mConfigEvents.size(); i++) {
- snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam);
+ snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam);
result.append(buffer);
}
result.append("\n");
@@ -1371,7 +1372,7 @@
int id,
uint32_t device)
: ThreadBase(audioFlinger, id, device),
- mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output),
+ mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), mOutput(output),
mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false)
{
snprintf(mName, kNameLength, "AudioOut_%d", id);
@@ -1838,7 +1839,7 @@
AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
: PlaybackThread(audioFlinger, output, id, device),
- mAudioMixer(0)
+ mAudioMixer(NULL)
{
mType = ThreadBase::MIXER;
mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
@@ -2203,7 +2204,7 @@
// XXX: these things DON'T need to be done each time
mAudioMixer->setBufferProvider(track);
- mAudioMixer->enable(AudioMixer::MIXING);
+ mAudioMixer->enable();
mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left);
mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right);
@@ -2249,7 +2250,7 @@
mixerStatus = MIXER_TRACKS_ENABLED;
}
}
- mAudioMixer->disable(AudioMixer::MIXING);
+ mAudioMixer->disable();
}
}
@@ -2461,14 +2462,6 @@
{
}
-
-static inline int16_t clamp16(int32_t sample)
-{
- if ((sample>>15) ^ (sample>>31))
- sample = 0x7FFF ^ (sample>>31);
- return sample;
-}
-
static inline
int32_t mul(int16_t in, int16_t v)
{
@@ -2772,7 +2765,7 @@
while (frameCount) {
buffer.frameCount = frameCount;
activeTrack->getNextBuffer(&buffer);
- if (UNLIKELY(buffer.raw == 0)) {
+ if (UNLIKELY(buffer.raw == NULL)) {
memset(curBuf, 0, frameCount * mFrameSize);
break;
}
@@ -3270,7 +3263,7 @@
void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
- buffer->raw = 0;
+ buffer->raw = NULL;
mFrameCount = buffer->frameCount;
step();
buffer->frameCount = 0;
@@ -3463,14 +3456,14 @@
}
buffer->raw = getBuffer(s, framesReq);
- if (buffer->raw == 0) goto getNextBuffer_exit;
+ if (buffer->raw == NULL) goto getNextBuffer_exit;
buffer->frameCount = framesReq;
return NO_ERROR;
}
getNextBuffer_exit:
- buffer->raw = 0;
+ buffer->raw = NULL;
buffer->frameCount = 0;
ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
return NOT_ENOUGH_DATA;
@@ -3711,14 +3704,14 @@
}
buffer->raw = getBuffer(s, framesReq);
- if (buffer->raw == 0) goto getNextBuffer_exit;
+ if (buffer->raw == NULL) goto getNextBuffer_exit;
buffer->frameCount = framesReq;
return NO_ERROR;
}
getNextBuffer_exit:
- buffer->raw = 0;
+ buffer->raw = NULL;
buffer->frameCount = 0;
return NOT_ENOUGH_DATA;
}
@@ -4223,7 +4216,7 @@
int id,
uint32_t device) :
ThreadBase(audioFlinger, id, device),
- mInput(input), mTrack(NULL), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0)
+ mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL)
{
mType = ThreadBase::RECORD;
@@ -4238,7 +4231,7 @@
AudioFlinger::RecordThread::~RecordThread()
{
delete[] mRsmpInBuffer;
- if (mResampler != 0) {
+ if (mResampler != NULL) {
delete mResampler;
delete[] mRsmpOutBuffer;
}
@@ -4332,7 +4325,7 @@
buffer.frameCount = mFrameCount;
if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
size_t framesOut = buffer.frameCount;
- if (mResampler == 0) {
+ if (mResampler == NULL) {
// no resampling
while (framesOut) {
size_t framesIn = mFrameCount - mRsmpInIndex;
@@ -4397,7 +4390,7 @@
// ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer()
// are 32 bit aligned which should be always true.
if (mChannelCount == 2 && mReqChannelCount == 1) {
- AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
+ ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
// the resampler always outputs stereo samples: do post stereo to mono conversion
int16_t *src = (int16_t *)mRsmpOutBuffer;
int16_t *dst = buffer.i16;
@@ -4406,7 +4399,7 @@
src += 2;
}
} else {
- AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
+ ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
}
}
@@ -4590,7 +4583,7 @@
result.append(buffer);
snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
result.append(buffer);
- snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0));
+ snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL));
result.append(buffer);
snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount);
result.append(buffer);
@@ -4625,7 +4618,7 @@
mInput->stream->common.standby(&mInput->stream->common);
usleep(kRecordThreadSleepUs);
}
- buffer->raw = 0;
+ buffer->raw = NULL;
buffer->frameCount = 0;
return NOT_ENOUGH_DATA;
}
@@ -4788,7 +4781,7 @@
if (mRsmpInBuffer) delete mRsmpInBuffer;
if (mRsmpOutBuffer) delete mRsmpOutBuffer;
if (mResampler) delete mResampler;
- mResampler = 0;
+ mResampler = NULL;
mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
@@ -6277,7 +6270,7 @@
if (isProcessEnabled()) {
// do 32 bit to 16 bit conversion for auxiliary effect input buffer
if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
- AudioMixer::ditherAndClamp(mConfig.inputCfg.buffer.s32,
+ ditherAndClamp(mConfig.inputCfg.buffer.s32,
mConfig.inputCfg.buffer.s32,
mConfig.inputCfg.buffer.frameCount/2);
}
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 6cafa7e..9707cf4 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -547,7 +547,7 @@
Condition mParamCond;
Vector<String8> mNewParameters;
status_t mParamStatus;
- Vector<ConfigEvent *> mConfigEvents;
+ Vector<ConfigEvent> mConfigEvents;
bool mStandby;
int mId;
bool mExiting;
@@ -703,7 +703,7 @@
virtual status_t readyToRun();
virtual void onFirstRef();
- virtual status_t initCheck() const { return (mOutput == 0) ? NO_INIT : NO_ERROR; }
+ virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; }
virtual uint32_t latency() const;
@@ -980,7 +980,7 @@
virtual status_t readyToRun();
virtual void onFirstRef();
- virtual status_t initCheck() const { return (mInput == 0) ? NO_INIT : NO_ERROR; }
+ virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; }
sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l(
const sp<AudioFlinger::Client>& client,
uint32_t sampleRate,
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 847f8dc..f5fdee7 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -18,6 +18,7 @@
#define LOG_TAG "AudioMixer"
//#define LOG_NDEBUG 0
+#include <assert.h>
#include <stdint.h>
#include <string.h>
#include <stdlib.h>
@@ -30,31 +31,27 @@
#include <system/audio.h>
+#include <audio_utils/primitives.h>
+
#include "AudioMixer.h"
namespace android {
-// ----------------------------------------------------------------------------
-
-static inline int16_t clamp16(int32_t sample)
-{
- if ((sample>>15) ^ (sample>>31))
- sample = 0x7FFF ^ (sample>>31);
- return sample;
-}
// ----------------------------------------------------------------------------
AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate)
: mActiveTrack(0), mTrackNames(0), mSampleRate(sampleRate)
{
+ // AudioMixer is not yet capable of multi-channel beyond stereo
+ assert(2 == MAX_NUM_CHANNELS);
mState.enabledTracks= 0;
mState.needsChanged = 0;
mState.frameCount = frameCount;
- mState.outputTemp = 0;
- mState.resampleTemp = 0;
+ mState.outputTemp = NULL;
+ mState.resampleTemp = NULL;
mState.hook = process__nop;
track_t* t = mState.tracks;
- for (int i=0 ; i<32 ; i++) {
+ for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) {
t->needs = 0;
t->volume[0] = UNITY_GAIN;
t->volume[1] = UNITY_GAIN;
@@ -73,11 +70,11 @@
t->format = 16;
t->channelMask = AUDIO_CHANNEL_OUT_STEREO;
t->buffer.raw = 0;
- t->bufferProvider = 0;
- t->hook = 0;
- t->resampler = 0;
+ t->bufferProvider = NULL;
+ t->hook = NULL;
+ t->resampler = NULL;
t->sampleRate = mSampleRate;
- t->in = 0;
+ t->in = NULL;
t->mainBuffer = NULL;
t->auxBuffer = NULL;
t++;
@@ -87,7 +84,7 @@
AudioMixer::~AudioMixer()
{
track_t* t = mState.tracks;
- for (int i=0 ; i<32 ; i++) {
+ for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) {
delete t->resampler;
t++;
}
@@ -123,129 +120,118 @@
void AudioMixer::deleteTrackName(int name)
{
name -= TRACK0;
- if (uint32_t(name) < MAX_NUM_TRACKS) {
- ALOGV("deleteTrackName(%d)", name);
- track_t& track(mState.tracks[ name ]);
- if (track.enabled != 0) {
- track.enabled = 0;
- invalidateState(1<<name);
- }
- if (track.resampler) {
- // delete the resampler
- delete track.resampler;
- track.resampler = 0;
- track.sampleRate = mSampleRate;
- invalidateState(1<<name);
- }
- track.volumeInc[0] = 0;
- track.volumeInc[1] = 0;
- mTrackNames &= ~(1<<name);
+ assert(uint32_t(name) < MAX_NUM_TRACKS);
+ ALOGV("deleteTrackName(%d)", name);
+ track_t& track(mState.tracks[ name ]);
+ if (track.enabled != 0) {
+ track.enabled = 0;
+ invalidateState(1<<name);
}
+ if (track.resampler) {
+ // delete the resampler
+ delete track.resampler;
+ track.resampler = NULL;
+ track.sampleRate = mSampleRate;
+ invalidateState(1<<name);
+ }
+ track.volumeInc[0] = 0;
+ track.volumeInc[1] = 0;
+ mTrackNames &= ~(1<<name);
}
-status_t AudioMixer::enable(int name)
+void AudioMixer::enable()
{
- switch (name) {
- case MIXING: {
- if (mState.tracks[ mActiveTrack ].enabled != 1) {
- mState.tracks[ mActiveTrack ].enabled = 1;
- ALOGV("enable(%d)", mActiveTrack);
- invalidateState(1<<mActiveTrack);
- }
- } break;
- default:
- return NAME_NOT_FOUND;
+ if (mState.tracks[ mActiveTrack ].enabled != 1) {
+ mState.tracks[ mActiveTrack ].enabled = 1;
+ ALOGV("enable(%d)", mActiveTrack);
+ invalidateState(1<<mActiveTrack);
}
- return NO_ERROR;
}
-status_t AudioMixer::disable(int name)
+void AudioMixer::disable()
{
- switch (name) {
- case MIXING: {
- if (mState.tracks[ mActiveTrack ].enabled != 0) {
- mState.tracks[ mActiveTrack ].enabled = 0;
- ALOGV("disable(%d)", mActiveTrack);
- invalidateState(1<<mActiveTrack);
- }
- } break;
- default:
- return NAME_NOT_FOUND;
+ if (mState.tracks[ mActiveTrack ].enabled != 0) {
+ mState.tracks[ mActiveTrack ].enabled = 0;
+ ALOGV("disable(%d)", mActiveTrack);
+ invalidateState(1<<mActiveTrack);
}
- return NO_ERROR;
}
-status_t AudioMixer::setActiveTrack(int track)
+void AudioMixer::setActiveTrack(int track)
{
- if (uint32_t(track-TRACK0) >= MAX_NUM_TRACKS) {
- return BAD_VALUE;
- }
- mActiveTrack = track - TRACK0;
- return NO_ERROR;
+ // this also catches track < TRACK0
+ track -= TRACK0;
+ assert(uint32_t(track) < MAX_NUM_TRACKS);
+ mActiveTrack = track;
}
-status_t AudioMixer::setParameter(int target, int name, void *value)
+void AudioMixer::setParameter(int target, int name, void *value)
{
int valueInt = (int)value;
int32_t *valueBuf = (int32_t *)value;
switch (target) {
+
case TRACK:
- if (name == CHANNEL_MASK) {
+ switch (name) {
+ case CHANNEL_MASK: {
uint32_t mask = (uint32_t)value;
if (mState.tracks[ mActiveTrack ].channelMask != mask) {
uint8_t channelCount = popcount(mask);
- if ((channelCount <= MAX_NUM_CHANNELS) && (channelCount)) {
- mState.tracks[ mActiveTrack ].channelMask = mask;
- mState.tracks[ mActiveTrack ].channelCount = channelCount;
- ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", mask);
- invalidateState(1<<mActiveTrack);
- return NO_ERROR;
- }
- } else {
- return NO_ERROR;
+ assert((channelCount <= MAX_NUM_CHANNELS) && (channelCount));
+ mState.tracks[ mActiveTrack ].channelMask = mask;
+ mState.tracks[ mActiveTrack ].channelCount = channelCount;
+ ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", mask);
+ invalidateState(1<<mActiveTrack);
}
- }
- if (name == MAIN_BUFFER) {
+ } break;
+ case MAIN_BUFFER:
if (mState.tracks[ mActiveTrack ].mainBuffer != valueBuf) {
mState.tracks[ mActiveTrack ].mainBuffer = valueBuf;
ALOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf);
invalidateState(1<<mActiveTrack);
}
- return NO_ERROR;
- }
- if (name == AUX_BUFFER) {
+ break;
+ case AUX_BUFFER:
if (mState.tracks[ mActiveTrack ].auxBuffer != valueBuf) {
mState.tracks[ mActiveTrack ].auxBuffer = valueBuf;
ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf);
invalidateState(1<<mActiveTrack);
}
- return NO_ERROR;
+ break;
+ default:
+ // bad name
+ assert(false);
}
-
break;
+
case RESAMPLE:
- if (name == SAMPLE_RATE) {
- if (valueInt > 0) {
- track_t& track = mState.tracks[ mActiveTrack ];
- if (track.setResampler(uint32_t(valueInt), mSampleRate)) {
- ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
- uint32_t(valueInt));
- invalidateState(1<<mActiveTrack);
- }
- return NO_ERROR;
+ switch (name) {
+ case SAMPLE_RATE: {
+ assert(valueInt > 0);
+ track_t& track = mState.tracks[ mActiveTrack ];
+ if (track.setResampler(uint32_t(valueInt), mSampleRate)) {
+ ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
+ uint32_t(valueInt));
+ invalidateState(1<<mActiveTrack);
}
- }
- if (name == RESET) {
+ } break;
+ case RESET: {
track_t& track = mState.tracks[ mActiveTrack ];
track.resetResampler();
invalidateState(1<<mActiveTrack);
- return NO_ERROR;
+ } break;
+ default:
+ // bad name
+ assert(false);
}
break;
+
case RAMP_VOLUME:
case VOLUME:
- if ((uint32_t(name-VOLUME0) < MAX_NUM_CHANNELS)) {
+ switch (name) {
+ case VOLUME0:
+ case VOLUME1: {
track_t& track = mState.tracks[ mActiveTrack ];
if (track.volume[name-VOLUME0] != valueInt) {
ALOGV("setParameter(VOLUME, VOLUME0/1: %04x)", valueInt);
@@ -264,8 +250,8 @@
}
invalidateState(1<<mActiveTrack);
}
- return NO_ERROR;
- } else if (name == AUXLEVEL) {
+ } break;
+ case AUXLEVEL: {
track_t& track = mState.tracks[ mActiveTrack ];
if (track.auxLevel != valueInt) {
ALOGV("setParameter(VOLUME, AUXLEVEL: %04x)", valueInt);
@@ -284,11 +270,17 @@
}
invalidateState(1<<mActiveTrack);
}
- return NO_ERROR;
+ } break;
+ default:
+ // bad name
+ assert(false);
}
break;
+
+ default:
+ // bad target
+ assert(false);
}
- return BAD_VALUE;
}
bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate)
@@ -296,7 +288,7 @@
if (value!=devSampleRate || resampler) {
if (sampleRate != value) {
sampleRate = value;
- if (resampler == 0) {
+ if (resampler == NULL) {
resampler = AudioResampler::create(
format, channelCount, devSampleRate);
}
@@ -308,12 +300,12 @@
bool AudioMixer::track_t::doesResample() const
{
- return resampler != 0;
+ return resampler != NULL;
}
void AudioMixer::track_t::resetResampler()
{
- if (resampler != 0) {
+ if (resampler != NULL) {
resampler->reset();
}
}
@@ -321,7 +313,7 @@
inline
void AudioMixer::track_t::adjustVolumeRamp(bool aux)
{
- for (int i=0 ; i<2 ; i++) {
+ for (int i=0 ; i<MAX_NUM_CHANNELS ; i++) {
if (((volumeInc[i]>0) && (((prevVolume[i]+volumeInc[i])>>16) >= volume[i])) ||
((volumeInc[i]<0) && (((prevVolume[i]+volumeInc[i])>>16) <= volume[i]))) {
volumeInc[i] = 0;
@@ -338,10 +330,9 @@
}
-status_t AudioMixer::setBufferProvider(AudioBufferProvider* buffer)
+void AudioMixer::setBufferProvider(AudioBufferProvider* buffer)
{
mState.tracks[ mActiveTrack ].bufferProvider = buffer;
- return NO_ERROR;
}
@@ -436,11 +427,11 @@
} else {
if (state->outputTemp) {
delete [] state->outputTemp;
- state->outputTemp = 0;
+ state->outputTemp = NULL;
}
if (state->resampleTemp) {
delete [] state->resampleTemp;
- state->resampleTemp = 0;
+ state->resampleTemp = NULL;
}
state->hook = process__genericNoResampling;
if (all16BitsStereoNoResample && !volumeRamp) {
@@ -485,88 +476,6 @@
}
}
-static inline
-int32_t mulAdd(int16_t in, int16_t v, int32_t a)
-{
-#if defined(__arm__) && !defined(__thumb__)
- int32_t out;
- asm( "smlabb %[out], %[in], %[v], %[a] \n"
- : [out]"=r"(out)
- : [in]"%r"(in), [v]"r"(v), [a]"r"(a)
- : );
- return out;
-#else
- return a + in * int32_t(v);
-#endif
-}
-
-static inline
-int32_t mul(int16_t in, int16_t v)
-{
-#if defined(__arm__) && !defined(__thumb__)
- int32_t out;
- asm( "smulbb %[out], %[in], %[v] \n"
- : [out]"=r"(out)
- : [in]"%r"(in), [v]"r"(v)
- : );
- return out;
-#else
- return in * int32_t(v);
-#endif
-}
-
-static inline
-int32_t mulAddRL(int left, uint32_t inRL, uint32_t vRL, int32_t a)
-{
-#if defined(__arm__) && !defined(__thumb__)
- int32_t out;
- if (left) {
- asm( "smlabb %[out], %[inRL], %[vRL], %[a] \n"
- : [out]"=r"(out)
- : [inRL]"%r"(inRL), [vRL]"r"(vRL), [a]"r"(a)
- : );
- } else {
- asm( "smlatt %[out], %[inRL], %[vRL], %[a] \n"
- : [out]"=r"(out)
- : [inRL]"%r"(inRL), [vRL]"r"(vRL), [a]"r"(a)
- : );
- }
- return out;
-#else
- if (left) {
- return a + int16_t(inRL&0xFFFF) * int16_t(vRL&0xFFFF);
- } else {
- return a + int16_t(inRL>>16) * int16_t(vRL>>16);
- }
-#endif
-}
-
-static inline
-int32_t mulRL(int left, uint32_t inRL, uint32_t vRL)
-{
-#if defined(__arm__) && !defined(__thumb__)
- int32_t out;
- if (left) {
- asm( "smulbb %[out], %[inRL], %[vRL] \n"
- : [out]"=r"(out)
- : [inRL]"%r"(inRL), [vRL]"r"(vRL)
- : );
- } else {
- asm( "smultt %[out], %[inRL], %[vRL] \n"
- : [out]"=r"(out)
- : [inRL]"%r"(inRL), [vRL]"r"(vRL)
- : );
- }
- return out;
-#else
- if (left) {
- return int16_t(inRL&0xFFFF) * int16_t(vRL&0xFFFF);
- } else {
- return int16_t(inRL>>16) * int16_t(vRL>>16);
- }
-#endif
-}
-
void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux)
{
@@ -851,19 +760,6 @@
t->in = in;
}
-void AudioMixer::ditherAndClamp(int32_t* out, int32_t const *sums, size_t c)
-{
- for (size_t i=0 ; i<c ; i++) {
- int32_t l = *sums++;
- int32_t r = *sums++;
- int32_t nl = l >> 12;
- int32_t nr = r >> 12;
- l = clamp16(nl);
- r = clamp16(nr);
- *out++ = (r<<16) | (l & 0xFFFF);
- }
-}
-
// no-op case
void AudioMixer::process__nop(state_t* state)
{
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index da9e2b5..39af103 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -47,11 +47,10 @@
enum { // names
- // track units (32 units)
+ // track units (MAX_NUM_TRACKS units)
TRACK0 = 0x1000,
- // enable/disable
- MIXING = 0x2000,
+ // 0x2000 is unused
// setParameter targets
TRACK = 0x3000,
@@ -78,19 +77,17 @@
int getTrackName();
void deleteTrackName(int name);
- status_t enable(int name);
- status_t disable(int name);
+ void enable();
+ void disable();
- status_t setActiveTrack(int track);
- status_t setParameter(int target, int name, void *value);
+ void setActiveTrack(int track);
+ void setParameter(int target, int name, void *value);
- status_t setBufferProvider(AudioBufferProvider* bufferProvider);
+ void setBufferProvider(AudioBufferProvider* bufferProvider);
void process();
uint32_t trackNames() const { return mTrackNames; }
- static void ditherAndClamp(int32_t* out, int32_t const *sums, size_t c);
-
private:
enum {
@@ -133,13 +130,13 @@
uint32_t needs;
union {
- int16_t volume[2]; // [0]3.12 fixed point
+ int16_t volume[MAX_NUM_CHANNELS]; // [0]3.12 fixed point
int32_t volumeRL;
};
- int32_t prevVolume[2];
+ int32_t prevVolume[MAX_NUM_CHANNELS];
- int32_t volumeInc[2];
+ int32_t volumeInc[MAX_NUM_CHANNELS];
int32_t auxLevel;
int32_t auxInc;
int32_t prevAuxLevel;
@@ -178,7 +175,7 @@
int32_t *outputTemp;
int32_t *resampleTemp;
int32_t reserved[2];
- track_t tracks[32]; __attribute__((aligned(32)));
+ track_t tracks[MAX_NUM_TRACKS]; __attribute__((aligned(32)));
};
int mActiveTrack;