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-rw-r--r--services/audioflinger/AudioFlinger.cpp248
-rw-r--r--services/audioflinger/AudioFlinger.h60
-rw-r--r--services/audioflinger/AudioMixer.cpp20
-rw-r--r--services/audioflinger/AudioMixer.h3
-rw-r--r--services/audioflinger/AudioPolicyService.cpp9
-rw-r--r--services/audioflinger/AudioPolicyService.h2
-rw-r--r--services/camera/libcameraservice/CameraHardwareInterface.h94
7 files changed, 194 insertions, 242 deletions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index a6ba1a0..44df5b5 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -44,7 +44,7 @@
#include <private/media/AudioEffectShared.h>
#include <system/audio.h>
-#include <hardware/audio_hal.h>
+#include <hardware/audio.h>
#include "AudioMixer.h"
#include "AudioFlinger.h"
@@ -53,15 +53,7 @@
#include <media/EffectVisualizerApi.h>
// ----------------------------------------------------------------------------
-// the sim build doesn't have gettid
-#ifndef HAVE_GETTID
-# define gettid getpid
-#endif
-
-// ----------------------------------------------------------------------------
-
-extern const char * const gEffectLibPath;
namespace android {
@@ -87,40 +79,20 @@ static const int kDumpLockSleep = 20000;
static const nsecs_t kWarningThrottle = seconds(5);
-#define AUDIOFLINGER_SECURITY_ENABLED 1
-
// ----------------------------------------------------------------------------
static bool recordingAllowed() {
-#ifndef HAVE_ANDROID_OS
- return true;
-#endif
-#if AUDIOFLINGER_SECURITY_ENABLED
if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO"));
if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO");
return ok;
-#else
- if (!checkCallingPermission(String16("android.permission.RECORD_AUDIO")))
- LOGW("WARNING: Need to add android.permission.RECORD_AUDIO to manifest");
- return true;
-#endif
}
static bool settingsAllowed() {
-#ifndef HAVE_ANDROID_OS
- return true;
-#endif
-#if AUDIOFLINGER_SECURITY_ENABLED
if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"));
if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
return ok;
-#else
- if (!checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")))
- LOGW("WARNING: Need to add android.permission.MODIFY_AUDIO_SETTINGS to manifest");
- return true;
-#endif
}
// To collect the amplifier usage
@@ -381,8 +353,8 @@ sp<IAudioTrack> AudioFlinger::createTrack(
pid_t pid,
int streamType,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
@@ -450,7 +422,7 @@ sp<IAudioTrack> AudioFlinger::createTrack(
LOGV("createTrack() lSessionId: %d", lSessionId);
track = thread->createTrack_l(client, streamType, sampleRate, format,
- channelCount, frameCount, sharedBuffer, lSessionId, &lStatus);
+ channelMask, frameCount, sharedBuffer, lSessionId, &lStatus);
// move effect chain to this output thread if an effect on same session was waiting
// for a track to be created
@@ -498,7 +470,7 @@ int AudioFlinger::channelCount(int output) const
return thread->channelCount();
}
-int AudioFlinger::format(int output) const
+uint32_t AudioFlinger::format(int output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
@@ -937,7 +909,7 @@ int AudioFlinger::ThreadBase::channelCount() const
return (int)mChannelCount;
}
-int AudioFlinger::ThreadBase::format() const
+uint32_t AudioFlinger::ThreadBase::format() const
{
return mFormat;
}
@@ -1023,6 +995,8 @@ status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args
result.append(buffer);
snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount);
result.append(buffer);
+ snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
+ result.append(buffer);
snprintf(buffer, SIZE, "Format: %d\n", mFormat);
result.append(buffer);
snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize);
@@ -1096,7 +1070,7 @@ status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>
snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
result.append(buffer);
- result.append(" Name Clien Typ Fmt Chn Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n");
+ result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n");
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> track = mTracks[i];
if (track != 0) {
@@ -1107,7 +1081,7 @@ status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>
snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
result.append(buffer);
- result.append(" Name Clien Typ Fmt Chn Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n");
+ result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n");
for (size_t i = 0; i < mActiveTracks.size(); ++i) {
wp<Track> wTrack = mActiveTracks[i];
if (wTrack != 0) {
@@ -1193,8 +1167,8 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTra
const sp<AudioFlinger::Client>& client,
int streamType,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
@@ -1204,11 +1178,14 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTra
status_t lStatus;
if (mType == DIRECT) {
- if (sampleRate != mSampleRate || format != mFormat || channelCount != (int)mChannelCount) {
- LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelCount %d for output %p",
- sampleRate, format, channelCount, mOutput);
- lStatus = BAD_VALUE;
- goto Exit;
+ if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
+ if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
+ LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \""
+ "for output %p with format %d",
+ sampleRate, format, channelMask, mOutput, mFormat);
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
}
} else {
// Resampler implementation limits input sampling rate to 2 x output sampling rate.
@@ -1245,7 +1222,7 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTra
}
track = new Track(this, client, streamType, sampleRate, format,
- channelCount, frameCount, sharedBuffer, sessionId);
+ channelMask, frameCount, sharedBuffer, sessionId);
if (track->getCblk() == NULL || track->name() < 0) {
lStatus = NO_MEMORY;
goto Exit;
@@ -1394,7 +1371,7 @@ void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
switch (event) {
case AudioSystem::OUTPUT_OPENED:
case AudioSystem::OUTPUT_CONFIG_CHANGED:
- desc.channels = mChannels;
+ desc.channels = mChannelMask;
desc.samplingRate = mSampleRate;
desc.format = mFormat;
desc.frameCount = mFrameCount;
@@ -1414,8 +1391,8 @@ void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
void AudioFlinger::PlaybackThread::readOutputParameters()
{
mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
- mChannels = mOutput->stream->common.get_channels(&mOutput->stream->common);
- mChannelCount = (uint16_t)popcount(mChannels);
+ mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
+ mChannelCount = (uint16_t)popcount(mChannelMask);
mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
mFrameSize = (uint16_t)audio_stream_frame_size(&mOutput->stream->common);
mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
@@ -1825,7 +1802,7 @@ uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track
AudioMixer::FORMAT, (void *)track->format());
mAudioMixer->setParameter(
AudioMixer::TRACK,
- AudioMixer::CHANNEL_COUNT, (void *)track->channelCount());
+ AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
mAudioMixer->setParameter(
AudioMixer::RESAMPLE,
AudioMixer::SAMPLE_RATE,
@@ -2704,7 +2681,7 @@ void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
this,
mSampleRate,
mFormat,
- mChannelCount,
+ mChannelMask,
frameCount);
if (outputTrack->cblk() != NULL) {
thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
@@ -2772,8 +2749,8 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
const wp<ThreadBase>& thread,
const sp<Client>& client,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
@@ -2793,6 +2770,7 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
// LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
size_t size = sizeof(audio_track_cblk_t);
+ uint8_t channelCount = popcount(channelMask);
size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
if (sharedBuffer == 0) {
size += bufferSize;
@@ -2807,7 +2785,8 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
// clear all buffers
mCblk->frameCount = frameCount;
mCblk->sampleRate = sampleRate;
- mCblk->channelCount = (uint8_t)channelCount;
+ mChannelCount = channelCount;
+ mChannelMask = channelMask;
if (sharedBuffer == 0) {
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
@@ -2831,7 +2810,8 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
// clear all buffers
mCblk->frameCount = frameCount;
mCblk->sampleRate = sampleRate;
- mCblk->channelCount = (uint8_t)channelCount;
+ mChannelCount = channelCount;
+ mChannelMask = channelMask;
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
// Force underrun condition to avoid false underrun callback until first data is
@@ -2898,7 +2878,11 @@ int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
}
int AudioFlinger::ThreadBase::TrackBase::channelCount() const {
- return (int)mCblk->channelCount;
+ return (const int)mChannelCount;
+}
+
+uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const {
+ return mChannelMask;
}
void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
@@ -2910,9 +2894,9 @@ void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t f
if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) {
LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \
- server %d, serverBase %d, user %d, userBase %d, channelCount %d",
+ server %d, serverBase %d, user %d, userBase %d",
bufferStart, bufferEnd, mBuffer, mBufferEnd,
- cblk->server, cblk->serverBase, cblk->user, cblk->userBase, cblk->channelCount);
+ cblk->server, cblk->serverBase, cblk->user, cblk->userBase);
return 0;
}
@@ -2927,12 +2911,12 @@ AudioFlinger::PlaybackThread::Track::Track(
const sp<Client>& client,
int streamType,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId)
- : TrackBase(thread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer, sessionId),
+ : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId),
mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL),
mAuxEffectId(0), mHasVolumeController(false)
{
@@ -2952,7 +2936,7 @@ AudioFlinger::PlaybackThread::Track::Track(
mStreamType = streamType;
// NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
// 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
- mCblk->frameSize = audio_is_linear_pcm(format) ? channelCount * sizeof(int16_t) : sizeof(int8_t);
+ mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(int8_t);
}
}
@@ -3000,12 +2984,12 @@ void AudioFlinger::PlaybackThread::Track::destroy()
void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
{
- snprintf(buffer, size, " %05d %05d %03u %03u %03u %05u %04u %1d %1d %1d %05u %05u %05u 0x%08x 0x%08x 0x%08x 0x%08x\n",
+ snprintf(buffer, size, " %05d %05d %03u %03u 0x%08x %05u %04u %1d %1d %1d %05u %05u %05u 0x%08x 0x%08x 0x%08x 0x%08x\n",
mName - AudioMixer::TRACK0,
(mClient == NULL) ? getpid() : mClient->pid(),
mStreamType,
mFormat,
- mCblk->channelCount,
+ mChannelMask,
mSessionId,
mFrameCount,
mState,
@@ -3240,21 +3224,21 @@ AudioFlinger::RecordThread::RecordTrack::RecordTrack(
const wp<ThreadBase>& thread,
const sp<Client>& client,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
uint32_t flags,
int sessionId)
: TrackBase(thread, client, sampleRate, format,
- channelCount, frameCount, flags, 0, sessionId),
+ channelMask, frameCount, flags, 0, sessionId),
mOverflow(false)
{
if (mCblk != NULL) {
LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
if (format == AUDIO_FORMAT_PCM_16_BIT) {
- mCblk->frameSize = channelCount * sizeof(int16_t);
+ mCblk->frameSize = mChannelCount * sizeof(int16_t);
} else if (format == AUDIO_FORMAT_PCM_8_BIT) {
- mCblk->frameSize = channelCount * sizeof(int8_t);
+ mCblk->frameSize = mChannelCount * sizeof(int8_t);
} else {
mCblk->frameSize = sizeof(int8_t);
}
@@ -3334,10 +3318,10 @@ void AudioFlinger::RecordThread::RecordTrack::stop()
void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
{
- snprintf(buffer, size, " %05d %03u %03u %05d %04u %01d %05u %08x %08x\n",
+ snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n",
(mClient == NULL) ? getpid() : mClient->pid(),
mFormat,
- mCblk->channelCount,
+ mChannelMask,
mSessionId,
mFrameCount,
mState,
@@ -3353,10 +3337,10 @@ AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
const wp<ThreadBase>& thread,
DuplicatingThread *sourceThread,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount)
- : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelCount, frameCount, NULL, 0),
+ : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0),
mActive(false), mSourceThread(sourceThread)
{
@@ -3367,8 +3351,10 @@ AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
mCblk->volume[0] = mCblk->volume[1] = 0x1000;
mOutBuffer.frameCount = 0;
playbackThread->mTracks.add(this);
- LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channelCount %d mBufferEnd %p",
- mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channelCount, mBufferEnd);
+ LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \
+ "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p",
+ mCblk, mBuffer, mCblk->buffers,
+ mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd);
} else {
LOGW("Error creating output track on thread %p", playbackThread);
}
@@ -3403,7 +3389,7 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr
{
Buffer *pInBuffer;
Buffer inBuffer;
- uint32_t channelCount = mCblk->channelCount;
+ uint32_t channelCount = mChannelCount;
bool outputBufferFull = false;
inBuffer.frameCount = frames;
inBuffer.i16 = data;
@@ -3688,8 +3674,8 @@ sp<IAudioRecord> AudioFlinger::openRecord(
pid_t pid,
int input,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
uint32_t flags,
int *sessionId,
@@ -3738,7 +3724,7 @@ sp<IAudioRecord> AudioFlinger::openRecord(
}
// create new record track. The record track uses one track in mHardwareMixerThread by convention.
recordTrack = new RecordThread::RecordTrack(thread, client, sampleRate,
- format, channelCount, frameCount, flags, lSessionId);
+ format, channelMask, frameCount, flags, lSessionId);
}
if (recordTrack->getCblk() == NULL) {
// remove local strong reference to Client before deleting the RecordTrack so that the Client
@@ -4086,7 +4072,7 @@ status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
if (mActiveTrack != 0) {
result.append("Active Track:\n");
- result.append(" Clien Fmt Chn Session Buf S SRate Serv User\n");
+ result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n");
mActiveTrack->dump(buffer, SIZE);
result.append(buffer);
@@ -4240,7 +4226,7 @@ void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
switch (event) {
case AudioSystem::INPUT_OPENED:
case AudioSystem::INPUT_CONFIG_CHANGED:
- desc.channels = mChannels;
+ desc.channels = mChannelMask;
desc.samplingRate = mSampleRate;
desc.format = mFormat;
desc.frameCount = mFrameCount;
@@ -4263,8 +4249,8 @@ void AudioFlinger::RecordThread::readInputParameters()
mResampler = 0;
mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
- mChannels = mInput->stream->common.get_channels(&mInput->stream->common);
- mChannelCount = (uint16_t)popcount(mChannels);
+ mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
+ mChannelCount = (uint16_t)popcount(mChannelMask);
mFormat = mInput->stream->common.get_format(&mInput->stream->common);
mFrameSize = (uint16_t)audio_stream_frame_size(&mInput->stream->common);
mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);
@@ -4636,32 +4622,6 @@ int AudioFlinger::nextUniqueId_l()
// ----------------------------------------------------------------------------
-status_t AudioFlinger::loadEffectLibrary(const char *libPath, int *handle)
-{
- // check calling permissions
- if (!settingsAllowed()) {
- return PERMISSION_DENIED;
- }
- // only allow libraries loaded from /system/lib/soundfx for now
- if (strncmp(gEffectLibPath, libPath, strlen(gEffectLibPath)) != 0) {
- return PERMISSION_DENIED;
- }
-
- Mutex::Autolock _l(mLock);
- return EffectLoadLibrary(libPath, handle);
-}
-
-status_t AudioFlinger::unloadEffectLibrary(int handle)
-{
- // check calling permissions
- if (!settingsAllowed()) {
- return PERMISSION_DENIED;
- }
-
- Mutex::Autolock _l(mLock);
- return EffectUnloadLibrary(handle);
-}
-
status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects)
{
Mutex::Autolock _l(mLock);
@@ -4697,7 +4657,6 @@ sp<IEffect> AudioFlinger::createEffect(pid_t pid,
{
status_t lStatus = NO_ERROR;
sp<EffectHandle> handle;
- effect_interface_t itfe;
effect_descriptor_t desc;
sp<Client> client;
wp<Client> wclient;
@@ -5535,19 +5494,19 @@ status_t AudioFlinger::EffectModule::configure()
// TODO: handle configuration of effects replacing track process
if (thread->channelCount() == 1) {
- channels = CHANNEL_MONO;
+ channels = AUDIO_CHANNEL_OUT_MONO;
} else {
- channels = CHANNEL_STEREO;
+ channels = AUDIO_CHANNEL_OUT_STEREO;
}
if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
- mConfig.inputCfg.channels = CHANNEL_MONO;
+ mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO;
} else {
mConfig.inputCfg.channels = channels;
}
mConfig.outputCfg.channels = channels;
- mConfig.inputCfg.format = SAMPLE_FORMAT_PCM_S15;
- mConfig.outputCfg.format = SAMPLE_FORMAT_PCM_S15;
+ mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
+ mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
mConfig.inputCfg.samplingRate = thread->sampleRate();
mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate;
mConfig.inputCfg.bufferProvider.cookie = NULL;
@@ -5792,11 +5751,6 @@ status_t AudioFlinger::EffectModule::setDevice(uint32_t device)
Mutex::Autolock _l(mLock);
status_t status = NO_ERROR;
if ((mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) {
- // convert device bit field from AudioSystem to EffectApi format.
- device = deviceAudioSystemToEffectApi(device);
- if (device == 0) {
- return BAD_VALUE;
- }
status_t cmdStatus;
uint32_t size = sizeof(status_t);
status = (*mEffectInterface)->command(mEffectInterface,
@@ -5817,17 +5771,12 @@ status_t AudioFlinger::EffectModule::setMode(uint32_t mode)
Mutex::Autolock _l(mLock);
status_t status = NO_ERROR;
if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) {
- // convert audio mode from AudioSystem to EffectApi format.
- int effectMode = modeAudioSystemToEffectApi(mode);
- if (effectMode < 0) {
- return BAD_VALUE;
- }
status_t cmdStatus;
uint32_t size = sizeof(status_t);
status = (*mEffectInterface)->command(mEffectInterface,
EFFECT_CMD_SET_AUDIO_MODE,
sizeof(int),
- &effectMode,
+ &mode,
&size,
&cmdStatus);
if (status == NO_ERROR) {
@@ -5837,53 +5786,6 @@ status_t AudioFlinger::EffectModule::setMode(uint32_t mode)
return status;
}
-// update this table when AudioSystem::audio_devices or audio_device_e (in EffectApi.h) are modified
-const uint32_t AudioFlinger::EffectModule::sDeviceConvTable[] = {
- DEVICE_EARPIECE, // AUDIO_DEVICE_OUT_EARPIECE
- DEVICE_SPEAKER, // AUDIO_DEVICE_OUT_SPEAKER
- DEVICE_WIRED_HEADSET, // case AUDIO_DEVICE_OUT_WIRED_HEADSET
- DEVICE_WIRED_HEADPHONE, // AUDIO_DEVICE_OUT_WIRED_HEADPHONE
- DEVICE_BLUETOOTH_SCO, // AUDIO_DEVICE_OUT_BLUETOOTH_SCO
- DEVICE_BLUETOOTH_SCO_HEADSET, // AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET
- DEVICE_BLUETOOTH_SCO_CARKIT, // AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT
- DEVICE_BLUETOOTH_A2DP, // AUDIO_DEVICE_OUT_BLUETOOTH_A2DP
- DEVICE_BLUETOOTH_A2DP_HEADPHONES, // AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES
- DEVICE_BLUETOOTH_A2DP_SPEAKER, // AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER
- DEVICE_AUX_DIGITAL // AUDIO_DEVICE_OUT_AUX_DIGITAL
-};
-
-uint32_t AudioFlinger::EffectModule::deviceAudioSystemToEffectApi(uint32_t device)
-{
- uint32_t deviceOut = 0;
- while (device) {
- const uint32_t i = 31 - __builtin_clz(device);
- device &= ~(1 << i);
- if (i >= sizeof(sDeviceConvTable)/sizeof(uint32_t)) {
- LOGE("device conversion error for AudioSystem device 0x%08x", device);
- return 0;
- }
- deviceOut |= (uint32_t)sDeviceConvTable[i];
- }
- return deviceOut;
-}
-
-// update this table when AudioSystem::audio_mode or audio_mode_e (in EffectApi.h) are modified
-const uint32_t AudioFlinger::EffectModule::sModeConvTable[] = {
- AUDIO_EFFECT_MODE_NORMAL, // AUDIO_MODE_NORMAL
- AUDIO_EFFECT_MODE_RINGTONE, // AUDIO_MODE_RINGTONE
- AUDIO_EFFECT_MODE_IN_CALL, // AUDIO_MODE_IN_CALL
- AUDIO_EFFECT_MODE_IN_CALL // AUDIO_MODE_IN_COMMUNICATION, same conversion as for AUDIO_MODE_IN_CALL
-};
-
-int AudioFlinger::EffectModule::modeAudioSystemToEffectApi(uint32_t mode)
-{
- int modeOut = -1;
- if (mode < sizeof(sModeConvTable) / sizeof(uint32_t)) {
- modeOut = (int)sModeConvTable[mode];
- }
- return modeOut;
-}
-
status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
@@ -5915,7 +5817,7 @@ status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2],
mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]);
result.append(buffer);
- snprintf(buffer, SIZE, "\t\t- apiVersion: %04X\n\t\t- flags: %08X\n",
+ snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n",
mDescriptor.apiVersion,
mDescriptor.flags);
result.append(buffer);
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 39314ad..1fad987 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -39,7 +39,7 @@
#include <binder/MemoryDealer.h>
#include <system/audio.h>
-#include <hardware/audio_hal.h>
+#include <hardware/audio.h>
#include "AudioBufferProvider.h"
@@ -76,8 +76,8 @@ public:
pid_t pid,
int streamType,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
@@ -87,7 +87,7 @@ public:
virtual uint32_t sampleRate(int output) const;
virtual int channelCount(int output) const;
- virtual int format(int output) const;
+ virtual uint32_t format(int output) const;
virtual size_t frameCount(int output) const;
virtual uint32_t latency(int output) const;
@@ -147,10 +147,6 @@ public:
virtual int newAudioSessionId();
- virtual status_t loadEffectLibrary(const char *libPath, int *handle);
-
- virtual status_t unloadEffectLibrary(int handle);
-
virtual status_t queryNumberEffects(uint32_t *numEffects);
virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor);
@@ -193,8 +189,8 @@ public:
pid_t pid,
int input,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
uint32_t flags,
int *sessionId,
@@ -305,8 +301,8 @@ private:
TrackBase(const wp<ThreadBase>& thread,
const sp<Client>& client,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
@@ -333,12 +329,14 @@ private:
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) = 0;
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
- int format() const {
+ uint32_t format() const {
return mFormat;
}
int channelCount() const ;
+ uint32_t channelMask() const;
+
int sampleRate() const;
void* getBuffer(uint32_t offset, uint32_t frames) const;
@@ -364,9 +362,11 @@ private:
// we don't really need a lock for these
int mState;
int mClientTid;
- uint8_t mFormat;
+ uint32_t mFormat;
uint32_t mFlags;
int mSessionId;
+ uint8_t mChannelCount;
+ uint32_t mChannelMask;
};
class ConfigEvent {
@@ -379,7 +379,7 @@ private:
uint32_t sampleRate() const;
int channelCount() const;
- int format() const;
+ uint32_t format() const;
size_t frameCount() const;
void wakeUp() { mWaitWorkCV.broadcast(); }
void exit();
@@ -410,10 +410,10 @@ private:
sp<AudioFlinger> mAudioFlinger;
uint32_t mSampleRate;
size_t mFrameCount;
- uint32_t mChannels;
+ uint32_t mChannelMask;
uint16_t mChannelCount;
uint16_t mFrameSize;
- int mFormat;
+ uint32_t mFormat;
Condition mParamCond;
Vector<String8> mNewParameters;
status_t mParamStatus;
@@ -446,8 +446,8 @@ private:
const sp<Client>& client,
int streamType,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId);
@@ -534,8 +534,8 @@ private:
OutputTrack( const wp<ThreadBase>& thread,
DuplicatingThread *sourceThread,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount);
~OutputTrack();
@@ -587,8 +587,8 @@ private:
const sp<AudioFlinger::Client>& client,
int streamType,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
@@ -833,8 +833,8 @@ private:
RecordTrack(const wp<ThreadBase>& thread,
const sp<Client>& client,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
uint32_t flags,
int sessionId);
@@ -1002,14 +1002,6 @@ private:
status_t start_l();
status_t stop_l();
- // update this table when AudioSystem::audio_devices or audio_device_e (in EffectApi.h) are modified
- static const uint32_t sDeviceConvTable[];
- static uint32_t deviceAudioSystemToEffectApi(uint32_t device);
-
- // update this table when AudioSystem::audio_mode or audio_mode_e (in EffectApi.h) are modified
- static const uint32_t sModeConvTable[];
- static int modeAudioSystemToEffectApi(uint32_t mode);
-
Mutex mLock; // mutex for process, commands and handles list protection
wp<ThreadBase> mThread; // parent thread
wp<EffectChain> mChain; // parent effect chain
@@ -1017,7 +1009,7 @@ private:
int mSessionId; // audio session ID
effect_descriptor_t mDescriptor;// effect descriptor received from effect engine
effect_config_t mConfig; // input and output audio configuration
- effect_interface_t mEffectInterface; // Effect module C API
+ effect_handle_t mEffectInterface; // Effect module C API
status_t mStatus; // initialization status
uint32_t mState; // current activation state (effect_state)
Vector< wp<EffectHandle> > mHandles; // list of client handles
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 50dcda7..6e9319d 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -26,6 +26,10 @@
#include <utils/Errors.h>
#include <utils/Log.h>
+#include <cutils/bitops.h>
+
+#include <system/audio.h>
+
#include "AudioMixer.h"
namespace android {
@@ -61,6 +65,7 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate)
t->channelCount = 2;
t->enabled = 0;
t->format = 16;
+ t->channelMask = AUDIO_CHANNEL_OUT_STEREO;
t->buffer.raw = 0;
t->bufferProvider = 0;
t->hook = 0;
@@ -180,13 +185,18 @@ status_t AudioMixer::setParameter(int target, int name, void *value)
switch (target) {
case TRACK:
- if (name == CHANNEL_COUNT) {
- if ((uint32_t(valueInt) <= MAX_NUM_CHANNELS) && (valueInt)) {
- if (mState.tracks[ mActiveTrack ].channelCount != valueInt) {
- mState.tracks[ mActiveTrack ].channelCount = valueInt;
- LOGV("setParameter(TRACK, CHANNEL_COUNT, %d)", valueInt);
+ if (name == 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;
+ LOGV("setParameter(TRACK, CHANNEL_MASK, %x)", mask);
invalidateState(1<<mActiveTrack);
+ return NO_ERROR;
}
+ } else {
return NO_ERROR;
}
}
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index 88408a7..75c9170 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -61,7 +61,7 @@ public:
// set Parameter names
// for target TRACK
- CHANNEL_COUNT = 0x4000,
+ CHANNEL_MASK = 0x4000,
FORMAT = 0x4001,
MAIN_BUFFER = 0x4002,
AUX_BUFFER = 0x4003,
@@ -150,6 +150,7 @@ private:
uint8_t enabled : 1;
uint8_t reserved0 : 3;
uint8_t format;
+ uint32_t channelMask;
AudioBufferProvider* bufferProvider;
mutable AudioBufferProvider::Buffer buffer;
diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp
index ef8d957..47ca3a0 100644
--- a/services/audioflinger/AudioPolicyService.cpp
+++ b/services/audioflinger/AudioPolicyService.cpp
@@ -36,15 +36,8 @@
#include <hardware/hardware.h>
#include <system/audio.h>
+#include <system/audio_policy.h>
#include <hardware/audio_policy.h>
-#include <hardware/audio_policy_hal.h>
-
-// ----------------------------------------------------------------------------
-// the sim build doesn't have gettid
-
-#ifndef HAVE_GETTID
-# define gettid getpid
-#endif
namespace android {
diff --git a/services/audioflinger/AudioPolicyService.h b/services/audioflinger/AudioPolicyService.h
index d9b5ada..b830120 100644
--- a/services/audioflinger/AudioPolicyService.h
+++ b/services/audioflinger/AudioPolicyService.h
@@ -23,8 +23,8 @@
#include <binder/BinderService.h>
#include <system/audio.h>
+#include <system/audio_policy.h>
#include <hardware/audio_policy.h>
-#include <hardware/audio_policy_hal.h>
namespace android {
diff --git a/services/camera/libcameraservice/CameraHardwareInterface.h b/services/camera/libcameraservice/CameraHardwareInterface.h
index f9fa30e..a3749cf 100644
--- a/services/camera/libcameraservice/CameraHardwareInterface.h
+++ b/services/camera/libcameraservice/CameraHardwareInterface.h
@@ -438,18 +438,23 @@ private:
}
static void __data_cb(int32_t msg_type,
- const camera_memory_t *data,
+ const camera_memory_t *data, unsigned int index,
void *user)
{
LOGV("%s", __FUNCTION__);
CameraHardwareInterface *__this =
static_cast<CameraHardwareInterface *>(user);
sp<CameraHeapMemory> mem(static_cast<CameraHeapMemory *>(data->handle));
- __this->mDataCb(msg_type, mem, __this->mCbUser);
+ if (index >= mem->mNumBufs) {
+ LOGE("%s: invalid buffer index %d, max allowed is %d", __FUNCTION__,
+ index, mem->mNumBufs);
+ return;
+ }
+ __this->mDataCb(msg_type, mem->mBuffers[index], __this->mCbUser);
}
static void __data_cb_timestamp(nsecs_t timestamp, int32_t msg_type,
- const camera_memory_t *data,
+ const camera_memory_t *data, unsigned index,
void *user)
{
LOGV("%s", __FUNCTION__);
@@ -459,38 +464,85 @@ private:
// drop all references, it will be destroyed (as well as the enclosed
// MemoryHeapBase.
sp<CameraHeapMemory> mem(static_cast<CameraHeapMemory *>(data->handle));
- __this->mDataCbTimestamp(timestamp, msg_type, mem, __this->mCbUser);
+ if (index >= mem->mNumBufs) {
+ LOGE("%s: invalid buffer index %d, max allowed is %d", __FUNCTION__,
+ index, mem->mNumBufs);
+ return;
+ }
+ __this->mDataCbTimestamp(timestamp, msg_type, mem->mBuffers[index], __this->mCbUser);
}
// This is a utility class that combines a MemoryHeapBase and a MemoryBase
// in one. Since we tend to use them in a one-to-one relationship, this is
// handy.
- class CameraHeapMemory : public MemoryBase {
+ class CameraHeapMemory : public RefBase {
public:
- CameraHeapMemory(size_t size) :
- MemoryBase(new MemoryHeapBase(size), 0, size)
+ CameraHeapMemory(int fd, size_t buf_size, uint_t num_buffers = 1) :
+ mBufSize(buf_size),
+ mNumBufs(num_buffers)
+ {
+ mHeap = new MemoryHeapBase(fd, buf_size * num_buffers);
+ commonInitialization();
+ }
+
+ CameraHeapMemory(size_t buf_size, uint_t num_buffers = 1) :
+ mBufSize(buf_size),
+ mNumBufs(num_buffers)
{
- handle.data = getHeap()->base();
- handle.size = size;
+ mHeap = new MemoryHeapBase(buf_size * num_buffers);
+ commonInitialization();
+ }
+
+ void commonInitialization()
+ {
+ handle.data = mHeap->base();
+ handle.size = mBufSize * mNumBufs;
handle.handle = this;
+
+ mBuffers = new sp<MemoryBase>[mNumBufs];
+ for (uint_t i = 0; i < mNumBufs; i++)
+ mBuffers[i] = new MemoryBase(mHeap,
+ i * mBufSize,
+ mBufSize);
+
+ handle.release = __put_memory;
}
+ virtual ~CameraHeapMemory()
+ {
+ delete [] mBuffers;
+ }
+
+ size_t mBufSize;
+ uint_t mNumBufs;
+ sp<MemoryHeapBase> mHeap;
+ sp<MemoryBase> *mBuffers;
+
camera_memory_t handle;
};
- static camera_memory_t* __get_memory(size_t size,
- void *user __attribute__((unused)))
+ static camera_memory_t* __get_memory(int fd, size_t buf_size, uint_t num_bufs,
+ void *user __attribute__((unused)))
{
- // We allocate the object here, but we do not assign it to a strong
- // pointer yet. The HAL will pass it back to us via the data callback
- // or the data-timestamp callback, and from there on we will wrap it
- // within a strong pointer.
-
- CameraHeapMemory *mem = new CameraHeapMemory(size);
+ CameraHeapMemory *mem;
+ if (fd < 0)
+ mem = new CameraHeapMemory(buf_size, num_bufs);
+ else
+ mem = new CameraHeapMemory(fd, buf_size, num_bufs);
+ mem->incStrong(mem);
return &mem->handle;
}
+ static void __put_memory(camera_memory_t *data)
+ {
+ if (!data)
+ return;
+
+ CameraHeapMemory *mem = static_cast<CameraHeapMemory *>(data->handle);
+ mem->decStrong(mem);
+ }
+
static ANativeWindow *__to_anw(void *user)
{
CameraHardwareInterface *__this =
@@ -500,7 +552,7 @@ private:
#define anw(n) __to_anw(((struct camera_preview_window *)n)->user)
static int __dequeue_buffer(struct preview_stream_ops* w,
- buffer_handle_t** buffer)
+ buffer_handle_t** buffer, int *stride)
{
int rc;
ANativeWindow *a = anw(w);
@@ -508,8 +560,10 @@ private:
rc = a->dequeueBuffer(a, &anb);
if (!rc) {
rc = a->lockBuffer(a, anb);
- if (!rc)
+ if (!rc) {
*buffer = &anb->handle;
+ *stride = anb->stride;
+ }
else
a->cancelBuffer(a, anb);
}
@@ -541,7 +595,7 @@ private:
static int __set_buffer_count(struct preview_stream_ops* w, int count)
{
ANativeWindow *a = anw(w);
- return native_window_set_buffer_count(a, count);
+ return native_window_set_buffer_count(a, count);
}
static int __set_buffers_geometry(struct preview_stream_ops* w,
generated by cgit v1.1 at 2025-11-04 15:26:19 +0000