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android / platform / frameworks / base / f1e484acb594a726fb57ad0ae4cfe902c7f35858 / . / libs / audioflinger / AudioFlinger.cpp
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/* //device/include/server/AudioFlinger/AudioFlinger.cpp
**
** Copyright 2007, 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.
*/
#define LOG_TAG "AudioFlinger"
//#define LOG_NDEBUG 0
#include <math.h>
#include <signal.h>
#include <sys/time.h>
#include <sys/resource.h>
#include <utils/IServiceManager.h>
#include <utils/Log.h>
#include <utils/Parcel.h>
#include <utils/IPCThreadState.h>
#include <utils/String16.h>
#include <utils/threads.h>
#include <cutils/properties.h>
#include <media/AudioTrack.h>
#include <media/AudioRecord.h>
#include <private/media/AudioTrackShared.h>
#include <hardware_legacy/AudioHardwareInterface.h>
#include "AudioMixer.h"
#include "AudioFlinger.h"
#ifdef WITH_A2DP
#include "A2dpAudioInterface.h"
#endif
namespace android {
//static const nsecs_t kStandbyTimeInNsecs = seconds(3);
static const unsigned long kBufferRecoveryInUsecs = 2000;
static const unsigned long kMaxBufferRecoveryInUsecs = 20000;
static const float MAX_GAIN = 4096.0f;
// retry counts for buffer fill timeout
// 50 * ~20msecs = 1 second
static const int8_t kMaxTrackRetries = 50;
static const int8_t kMaxTrackStartupRetries = 50;
#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
}
// ----------------------------------------------------------------------------
AudioFlinger::AudioFlinger()
: BnAudioFlinger(), Thread(false),
mMasterVolume(0), mMasterMute(true), mHardwareAudioMixer(0), mA2dpAudioMixer(0),
mAudioMixer(0), mAudioHardware(0), mA2dpAudioInterface(0), mHardwareOutput(0),
mA2dpOutput(0), mOutput(0), mRequestedOutput(0), mAudioRecordThread(0),
mSampleRate(0), mFrameCount(0), mChannelCount(0), mFormat(0), mMixBuffer(0),
mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mStandby(false),
mInWrite(false), mA2dpDisableCount(0), mA2dpSuppressed(false)
{
mHardwareStatus = AUDIO_HW_IDLE;
mAudioHardware = AudioHardwareInterface::create();
mHardwareStatus = AUDIO_HW_INIT;
if (mAudioHardware->initCheck() == NO_ERROR) {
// open 16-bit output stream for s/w mixer
mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
status_t status;
mHardwareOutput = mAudioHardware->openOutputStream(AudioSystem::PCM_16_BIT, 0, 0, &status);
mHardwareStatus = AUDIO_HW_IDLE;
if (mHardwareOutput) {
mHardwareAudioMixer = new AudioMixer(getOutputFrameCount(mHardwareOutput), mHardwareOutput->sampleRate());
mRequestedOutput = mHardwareOutput;
doSetOutput(mHardwareOutput);
// FIXME - this should come from settings
setMasterVolume(1.0f);
setRouting(AudioSystem::MODE_NORMAL, AudioSystem::ROUTE_SPEAKER, AudioSystem::ROUTE_ALL);
setRouting(AudioSystem::MODE_RINGTONE, AudioSystem::ROUTE_SPEAKER, AudioSystem::ROUTE_ALL);
setRouting(AudioSystem::MODE_IN_CALL, AudioSystem::ROUTE_EARPIECE, AudioSystem::ROUTE_ALL);
setMode(AudioSystem::MODE_NORMAL);
mMasterMute = false;
} else {
LOGE("Failed to initialize output stream, status: %d", status);
}
#ifdef WITH_A2DP
// Create A2DP interface
mA2dpAudioInterface = new A2dpAudioInterface();
mA2dpOutput = mA2dpAudioInterface->openOutputStream(AudioSystem::PCM_16_BIT, 0, 0, &status);
mA2dpAudioMixer = new AudioMixer(getOutputFrameCount(mA2dpOutput), mA2dpOutput->sampleRate());
// create a buffer big enough for both hardware and A2DP audio output.
size_t hwFrameCount = getOutputFrameCount(mHardwareOutput);
size_t a2dpFrameCount = getOutputFrameCount(mA2dpOutput);
size_t frameCount = (hwFrameCount > a2dpFrameCount ? hwFrameCount : a2dpFrameCount);
#else
size_t frameCount = getOutputFrameCount(mHardwareOutput);
#endif
// FIXME - Current mixer implementation only supports stereo output: Always
// Allocate a stereo buffer even if HW output is mono.
mMixBuffer = new int16_t[frameCount * 2];
memset(mMixBuffer, 0, frameCount * 2 * sizeof(int16_t));
// Start record thread
mAudioRecordThread = new AudioRecordThread(mAudioHardware);
if (mAudioRecordThread != 0) {
mAudioRecordThread->run("AudioRecordThread", PRIORITY_URGENT_AUDIO);
}
} else {
LOGE("Couldn't even initialize the stubbed audio hardware!");
}
char value[PROPERTY_VALUE_MAX];
property_get("ro.audio.silent", value, "0");
if (atoi(value)) {
LOGD("Silence is golden");
mMasterMute = true;
}
}
AudioFlinger::~AudioFlinger()
{
if (mAudioRecordThread != 0) {
mAudioRecordThread->exit();
mAudioRecordThread.clear();
}
delete mAudioHardware;
// deleting mA2dpAudioInterface also deletes mA2dpOutput;
delete mA2dpAudioInterface;
delete [] mMixBuffer;
delete mHardwareAudioMixer;
delete mA2dpAudioMixer;
}
void AudioFlinger::setOutput(AudioStreamOut* output)
{
mRequestedOutput = output;
}
void AudioFlinger::doSetOutput(AudioStreamOut* output)
{
mSampleRate = output->sampleRate();
mChannelCount = output->channelCount();
// FIXME - Current mixer implementation only supports stereo output
if (mChannelCount == 1) {
LOGE("Invalid audio hardware channel count");
}
mFormat = output->format();
mFrameCount = getOutputFrameCount(output);
mAudioMixer = (output == mA2dpOutput ? mA2dpAudioMixer : mHardwareAudioMixer);
mOutput = output;
}
size_t AudioFlinger::getOutputFrameCount(AudioStreamOut* output)
{
return output->bufferSize() / output->channelCount() / sizeof(int16_t);
}
#ifdef WITH_A2DP
bool AudioFlinger::streamDisablesA2dp(int streamType)
{
return (streamType == AudioTrack::SYSTEM ||
streamType == AudioTrack::RING ||
streamType == AudioTrack::ALARM ||
streamType == AudioTrack::NOTIFICATION);
}
void AudioFlinger::setA2dpEnabled(bool enable)
{
if (enable) {
LOGD("set output to A2DP\n");
setOutput(mA2dpOutput);
} else {
LOGD("set output to hardware audio\n");
setOutput(mHardwareOutput);
}
}
#endif // WITH_A2DP
status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
result.append("Clients:\n");
for (size_t i = 0; i < mClients.size(); ++i) {
wp<Client> wClient = mClients.valueAt(i);
if (wClient != 0) {
sp<Client> client = wClient.promote();
if (client != 0) {
snprintf(buffer, SIZE, " pid: %d\n", client->pid());
result.append(buffer);
}
}
}
write(fd, result.string(), result.size());
return NO_ERROR;
}
status_t AudioFlinger::dumpTracks(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
result.append("Tracks:\n");
result.append(" Name Clien Typ Fmt Chn Buf S M F SRate LeftV RighV Serv User\n");
for (size_t i = 0; i < mTracks.size(); ++i) {
wp<Track> wTrack = mTracks[i];
if (wTrack != 0) {
sp<Track> track = wTrack.promote();
if (track != 0) {
track->dump(buffer, SIZE);
result.append(buffer);
}
}
}
result.append("Active Tracks:\n");
result.append(" Name Clien Typ Fmt Chn Buf S M F SRate LeftV RighV Serv User\n");
for (size_t i = 0; i < mActiveTracks.size(); ++i) {
wp<Track> wTrack = mTracks[i];
if (wTrack != 0) {
sp<Track> track = wTrack.promote();
if (track != 0) {
track->dump(buffer, SIZE);
result.append(buffer);
}
}
}
write(fd, result.string(), result.size());
return NO_ERROR;
}
status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", audioMixer()->trackNames());
result.append(buffer);
snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
result.append(buffer);
snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
result.append(buffer);
snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
result.append(buffer);
snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
result.append(buffer);
snprintf(buffer, SIZE, "standby: %d\n", mStandby);
result.append(buffer);
snprintf(buffer, SIZE, "Hardware status: %d\n", mHardwareStatus);
result.append(buffer);
write(fd, result.string(), result.size());
return NO_ERROR;
}
status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "Permission Denial: "
"can't dump AudioFlinger from pid=%d, uid=%d\n",
IPCThreadState::self()->getCallingPid(),
IPCThreadState::self()->getCallingUid());
result.append(buffer);
write(fd, result.string(), result.size());
return NO_ERROR;
}
status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
{
if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
dumpPermissionDenial(fd, args);
} else {
AutoMutex lock(&mLock);
dumpClients(fd, args);
dumpTracks(fd, args);
dumpInternals(fd, args);
if (mAudioHardware) {
mAudioHardware->dumpState(fd, args);
}
}
return NO_ERROR;
}
// Thread virtuals
bool AudioFlinger::threadLoop()
{
unsigned long sleepTime = kBufferRecoveryInUsecs;
int16_t* curBuf = mMixBuffer;
Vector< sp<Track> > tracksToRemove;
size_t enabledTracks = 0;
nsecs_t standbyTime = systemTime();
do {
enabledTracks = 0;
{ // scope for the mLock
Mutex::Autolock _l(mLock);
const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
// put audio hardware into standby after short delay
if UNLIKELY(!activeTracks.size() && systemTime() > standbyTime) {
// wait until we have something to do...
LOGV("Audio hardware entering standby\n");
mHardwareStatus = AUDIO_HW_STANDBY;
if (!mStandby) {
mOutput->standby();
mStandby = true;
}
mHardwareStatus = AUDIO_HW_IDLE;
// we're about to wait, flush the binder command buffer
IPCThreadState::self()->flushCommands();
mWaitWorkCV.wait(mLock);
LOGV("Audio hardware exiting standby\n");
standbyTime = systemTime() + kStandbyTimeInNsecs;
continue;
}
// check for change in output
if (mRequestedOutput != mOutput) {
// put current output into standby mode
if (mOutput) mOutput->standby();
// change output
doSetOutput(mRequestedOutput);
}
// find out which tracks need to be processed
size_t count = activeTracks.size();
for (size_t i=0 ; i<count ; i++) {
sp<Track> t = activeTracks[i].promote();
if (t == 0) continue;
Track* const track = t.get();
audio_track_cblk_t* cblk = track->cblk();
// The first time a track is added we wait
// for all its buffers to be filled before processing it
mAudioMixer->setActiveTrack(track->name());
if (cblk->framesReady() && (track->isReady() || track->isStopped()) &&
!track->isPaused())
{
//LOGD("u=%08x, s=%08x [OK]", u, s);
// compute volume for this track
int16_t left, right;
if (track->isMuted() || mMasterMute || track->isPausing()) {
left = right = 0;
if (track->isPausing()) {
LOGV("paused(%d)", track->name());
track->setPaused();
}
} else {
float typeVolume = mStreamTypes[track->type()].volume;
float v = mMasterVolume * typeVolume;
float v_clamped = v * cblk->volume[0];
if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
left = int16_t(v_clamped);
v_clamped = v * cblk->volume[1];
if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
right = int16_t(v_clamped);
}
// XXX: these things DON'T need to be done each time
mAudioMixer->setBufferProvider(track);
mAudioMixer->enable(AudioMixer::MIXING);
int param;
if ( track->mFillingUpStatus == Track::FS_FILLED) {
// no ramp for the first volume setting
track->mFillingUpStatus = Track::FS_ACTIVE;
if (track->mState == TrackBase::RESUMING) {
track->mState = TrackBase::ACTIVE;
param = AudioMixer::RAMP_VOLUME;
} else {
param = AudioMixer::VOLUME;
}
} else {
param = AudioMixer::RAMP_VOLUME;
}
mAudioMixer->setParameter(param, AudioMixer::VOLUME0, left);
mAudioMixer->setParameter(param, AudioMixer::VOLUME1, right);
mAudioMixer->setParameter(
AudioMixer::TRACK,
AudioMixer::FORMAT, track->format());
mAudioMixer->setParameter(
AudioMixer::TRACK,
AudioMixer::CHANNEL_COUNT, track->channelCount());
mAudioMixer->setParameter(
AudioMixer::RESAMPLE,
AudioMixer::SAMPLE_RATE,
int(cblk->sampleRate));
// reset retry count
track->mRetryCount = kMaxTrackRetries;
enabledTracks++;
} else {
//LOGD("u=%08x, s=%08x [NOT READY]", u, s);
if (track->isStopped()) {
track->reset();
}
if (track->isTerminated() || track->isStopped() || track->isPaused()) {
// We have consumed all the buffers of this track.
// Remove it from the list of active tracks.
LOGV("remove(%d) from active list", track->name());
tracksToRemove.add(track);
} else {
// No buffers for this track. Give it a few chances to
// fill a buffer, then remove it from active list.
if (--(track->mRetryCount) <= 0) {
LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
tracksToRemove.add(track);
}
}
// LOGV("disable(%d)", track->name());
mAudioMixer->disable(AudioMixer::MIXING);
}
}
// remove all the tracks that need to be...
count = tracksToRemove.size();
if (UNLIKELY(count)) {
for (size_t i=0 ; i<count ; i++) {
const sp<Track>& track = tracksToRemove[i];
removeActiveTrack(track);
if (track->isTerminated()) {
mTracks.remove(track);
mAudioMixer->deleteTrackName(track->mName);
}
}
}
}
if (LIKELY(enabledTracks)) {
// mix buffers...
mAudioMixer->process(curBuf);
// output audio to hardware
mLastWriteTime = systemTime();
mInWrite = true;
size_t mixBufferSize = mFrameCount*mChannelCount*sizeof(int16_t);
mOutput->write(curBuf, mixBufferSize);
mNumWrites++;
mInWrite = false;
mStandby = false;
nsecs_t temp = systemTime();
standbyTime = temp + kStandbyTimeInNsecs;
nsecs_t delta = temp - mLastWriteTime;
nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 2;
if (delta > maxPeriod) {
LOGW("write blocked for %llu msecs", ns2ms(delta));
mNumDelayedWrites++;
}
sleepTime = kBufferRecoveryInUsecs;
} else {
// There was nothing to mix this round, which means all
// active tracks were late. Sleep a little bit to give
// them another chance. If we're too late, the audio
// hardware will zero-fill for us.
LOGV("no buffers - usleep(%lu)", sleepTime);
usleep(sleepTime);
if (sleepTime < kMaxBufferRecoveryInUsecs) {
sleepTime += kBufferRecoveryInUsecs;
}
}
// finally let go of all our tracks, without the lock held
// since we can't guarantee the destructors won't acquire that
// same lock.
tracksToRemove.clear();
} while (true);
return false;
}
status_t AudioFlinger::readyToRun()
{
if (mSampleRate == 0) {
LOGE("No working audio driver found.");
return NO_INIT;
}
LOGI("AudioFlinger's main thread ready to run.");
return NO_ERROR;
}
void AudioFlinger::onFirstRef()
{
run("AudioFlinger", ANDROID_PRIORITY_URGENT_AUDIO);
}
// IAudioFlinger interface
sp<IAudioTrack> AudioFlinger::createTrack(
pid_t pid,
int streamType,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
status_t *status)
{
sp<Track> track;
sp<TrackHandle> trackHandle;
sp<Client> client;
wp<Client> wclient;
status_t lStatus;
if (streamType >= AudioTrack::NUM_STREAM_TYPES) {
LOGE("invalid stream type");
lStatus = BAD_VALUE;
goto Exit;
}
// Resampler implementation limits input sampling rate to 2 x output sampling rate.
if (sampleRate > MAX_SAMPLE_RATE || sampleRate > mSampleRate*2) {
LOGE("Sample rate out of range: %d", sampleRate);
lStatus = BAD_VALUE;
goto Exit;
}
{
Mutex::Autolock _l(mLock);
if (mSampleRate == 0) {
LOGE("Audio driver not initialized.");
lStatus = NO_INIT;
goto Exit;
}
wclient = mClients.valueFor(pid);
if (wclient != NULL) {
client = wclient.promote();
} else {
client = new Client(this, pid);
mClients.add(pid, client);
}
track = new Track(this, client, streamType, sampleRate, format,
channelCount, frameCount, sharedBuffer);
mTracks.add(track);
trackHandle = new TrackHandle(track);
lStatus = NO_ERROR;
}
Exit:
if(status) {
*status = lStatus;
}
return trackHandle;
}
uint32_t AudioFlinger::sampleRate() const
{
return mSampleRate;
}
int AudioFlinger::channelCount() const
{
return mChannelCount;
}
int AudioFlinger::format() const
{
return mFormat;
}
size_t AudioFlinger::frameCount() const
{
return mFrameCount;
}
uint32_t AudioFlinger::latency() const
{
if (mOutput) {
return mOutput->latency();
}
else {
return 0;
}
}
status_t AudioFlinger::setMasterVolume(float value)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
// when hw supports master volume, don't scale in sw mixer
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
if (mAudioHardware->setMasterVolume(value) == NO_ERROR) {
mMasterVolume = 1.0f;
}
else {
mMasterVolume = value;
}
mHardwareStatus = AUDIO_HW_IDLE;
return NO_ERROR;
}
status_t AudioFlinger::setRouting(int mode, uint32_t routes, uint32_t mask)
{
status_t err = NO_ERROR;
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if ((mode < AudioSystem::MODE_CURRENT) || (mode >= AudioSystem::NUM_MODES)) {
LOGW("Illegal value: setRouting(%d, %u, %u)", mode, routes, mask);
return BAD_VALUE;
}
#ifdef WITH_A2DP
LOGD("setRouting %d %d %d\n", mode, routes, mask);
if (mode == AudioSystem::MODE_NORMAL &&
(mask & AudioSystem::ROUTE_BLUETOOTH_A2DP)) {
AutoMutex lock(&mLock);
bool enableA2dp = false;
if (routes & AudioSystem::ROUTE_BLUETOOTH_A2DP) {
if (mA2dpDisableCount > 0)
mA2dpSuppressed = true;
else
enableA2dp = true;
}
setA2dpEnabled(enableA2dp);
LOGD("setOutput done\n");
}
#endif
// do nothing if only A2DP routing is affected
mask &= ~AudioSystem::ROUTE_BLUETOOTH_A2DP;
if (mask) {
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_GET_ROUTING;
uint32_t r;
err = mAudioHardware->getRouting(mode, &r);
if (err == NO_ERROR) {
r = (r & ~mask) | (routes & mask);
mHardwareStatus = AUDIO_HW_SET_ROUTING;
err = mAudioHardware->setRouting(mode, r);
}
mHardwareStatus = AUDIO_HW_IDLE;
}
return err;
}
uint32_t AudioFlinger::getRouting(int mode) const
{
uint32_t routes = 0;
if ((mode >= AudioSystem::MODE_CURRENT) && (mode < AudioSystem::NUM_MODES)) {
mHardwareStatus = AUDIO_HW_GET_ROUTING;
mAudioHardware->getRouting(mode, &routes);
mHardwareStatus = AUDIO_HW_IDLE;
} else {
LOGW("Illegal value: getRouting(%d)", mode);
}
return routes;
}
status_t AudioFlinger::setMode(int mode)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if ((mode < 0) || (mode >= AudioSystem::NUM_MODES)) {
LOGW("Illegal value: setMode(%d)", mode);
return BAD_VALUE;
}
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_MODE;
status_t ret = mAudioHardware->setMode(mode);
mHardwareStatus = AUDIO_HW_IDLE;
return ret;
}
int AudioFlinger::getMode() const
{
int mode = AudioSystem::MODE_INVALID;
mHardwareStatus = AUDIO_HW_SET_MODE;
mAudioHardware->getMode(&mode);
mHardwareStatus = AUDIO_HW_IDLE;
return mode;
}
status_t AudioFlinger::setMicMute(bool state)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
status_t ret = mAudioHardware->setMicMute(state);
mHardwareStatus = AUDIO_HW_IDLE;
return ret;
}
bool AudioFlinger::getMicMute() const
{
bool state = AudioSystem::MODE_INVALID;
mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
mAudioHardware->getMicMute(&state);
mHardwareStatus = AUDIO_HW_IDLE;
return state;
}
status_t AudioFlinger::setMasterMute(bool muted)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
mMasterMute = muted;
return NO_ERROR;
}
float AudioFlinger::masterVolume() const
{
return mMasterVolume;
}
bool AudioFlinger::masterMute() const
{
return mMasterMute;
}
status_t AudioFlinger::setStreamVolume(int stream, float value)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if (uint32_t(stream) >= AudioTrack::NUM_STREAM_TYPES) {
return BAD_VALUE;
}
mStreamTypes[stream].volume = value;
status_t ret = NO_ERROR;
if (stream == AudioTrack::VOICE_CALL) {
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
ret = mAudioHardware->setVoiceVolume(value);
mHardwareStatus = AUDIO_HW_IDLE;
}
return ret;
}
status_t AudioFlinger::setStreamMute(int stream, bool muted)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if (uint32_t(stream) >= AudioTrack::NUM_STREAM_TYPES) {
return BAD_VALUE;
}
mStreamTypes[stream].mute = muted;
return NO_ERROR;
}
float AudioFlinger::streamVolume(int stream) const
{
if (uint32_t(stream) >= AudioTrack::NUM_STREAM_TYPES) {
return 0.0f;
}
return mStreamTypes[stream].volume;
}
bool AudioFlinger::streamMute(int stream) const
{
if (uint32_t(stream) >= AudioTrack::NUM_STREAM_TYPES) {
return true;
}
return mStreamTypes[stream].mute;
}
bool AudioFlinger::isMusicActive() const
{
size_t count = mActiveTracks.size();
for (size_t i = 0 ; i < count ; ++i) {
sp<Track> t = mActiveTracks[i].promote();
if (t == 0) continue;
Track* const track = t.get();
if (t->mStreamType == AudioTrack::MUSIC)
return true;
}
return false;
}
status_t AudioFlinger::setParameter(const char* key, const char* value)
{
status_t result, result2;
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_SET_PARAMETER;
result = mAudioHardware->setParameter(key, value);
if (mA2dpAudioInterface) {
result2 = mA2dpAudioInterface->setParameter(key, value);
if (result2)
result = result2;
}
mHardwareStatus = AUDIO_HW_IDLE;
return result;
}
void AudioFlinger::removeClient(pid_t pid)
{
Mutex::Autolock _l(mLock);
mClients.removeItem(pid);
}
status_t AudioFlinger::addTrack(const sp<Track>& track)
{
Mutex::Autolock _l(mLock);
// here the track could be either new, or restarted
// in both cases "unstop" the track
if (track->isPaused()) {
track->mState = TrackBase::RESUMING;
LOGV("PAUSED => RESUMING (%d)", track->name());
} else {
track->mState = TrackBase::ACTIVE;
LOGV("? => ACTIVE (%d)", track->name());
}
// set retry count for buffer fill
track->mRetryCount = kMaxTrackStartupRetries;
LOGV("mWaitWorkCV.broadcast");
mWaitWorkCV.broadcast();
if (mActiveTracks.indexOf(track) < 0) {
// the track is newly added, make sure it fills up all its
// buffers before playing. This is to ensure the client will
// effectively get the latency it requested.
track->mFillingUpStatus = Track::FS_FILLING;
track->mResetDone = false;
addActiveTrack(track);
return NO_ERROR;
}
return ALREADY_EXISTS;
}
void AudioFlinger::removeTrack(wp<Track> track, int name)
{
Mutex::Autolock _l(mLock);
sp<Track> t = track.promote();
if (t!=NULL && (t->mState <= TrackBase::STOPPED)) {
remove_track_l(track, name);
}
}
void AudioFlinger::remove_track_l(wp<Track> track, int name)
{
sp<Track> t = track.promote();
if (t!=NULL) {
t->reset();
}
audioMixer()->deleteTrackName(name);
removeActiveTrack(track);
mWaitWorkCV.broadcast();
}
void AudioFlinger::destroyTrack(const sp<Track>& track)
{
// NOTE: We're acquiring a strong reference on the track before
// acquiring the lock, this is to make sure removing it from
// mTracks won't cause the destructor to be called while the lock is
// held (note that technically, 'track' could be a reference to an item
// in mTracks, which is why we need to do this).
sp<Track> keep(track);
Mutex::Autolock _l(mLock);
track->mState = TrackBase::TERMINATED;
if (mActiveTracks.indexOf(track) < 0) {
LOGV("remove track (%d) and delete from mixer", track->name());
mTracks.remove(track);
audioMixer()->deleteTrackName(keep->name());
}
}
void AudioFlinger::addActiveTrack(const wp<Track>& t)
{
mActiveTracks.add(t);
#ifdef WITH_A2DP
// disable A2DP for certain stream types
sp<Track> track = t.promote();
if (streamDisablesA2dp(track->type())) {
if (mA2dpDisableCount++ == 0 && isA2dpEnabled()) {
setA2dpEnabled(false);
mA2dpSuppressed = true;
LOGD("mA2dpSuppressed = true\n");
}
LOGD("mA2dpDisableCount incremented to %d\n", mA2dpDisableCount);
}
#endif
}
void AudioFlinger::removeActiveTrack(const wp<Track>& t)
{
mActiveTracks.remove(t);
#ifdef WITH_A2DP
// disable A2DP for certain stream types
sp<Track> track = t.promote();
if (streamDisablesA2dp(track->type())) {
if (mA2dpDisableCount > 0) {
mA2dpDisableCount--;
if (mA2dpDisableCount == 0 && mA2dpSuppressed) {
setA2dpEnabled(true);
mA2dpSuppressed = false;
}
LOGD("mA2dpDisableCount decremented to %d\n", mA2dpDisableCount);
} else
LOGE("mA2dpDisableCount is already zero");
}
#endif
}
// ----------------------------------------------------------------------------
AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
: RefBase(),
mAudioFlinger(audioFlinger),
mMemoryDealer(new MemoryDealer(1024*1024)),
mPid(pid)
{
// 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
}
AudioFlinger::Client::~Client()
{
mAudioFlinger->removeClient(mPid);
}
const sp<MemoryDealer>& AudioFlinger::Client::heap() const
{
return mMemoryDealer;
}
// ----------------------------------------------------------------------------
AudioFlinger::TrackBase::TrackBase(
const sp<AudioFlinger>& audioFlinger,
const sp<Client>& client,
int streamType,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
const sp<IMemory>& sharedBuffer)
: RefBase(),
mAudioFlinger(audioFlinger),
mClient(client),
mStreamType(streamType),
mFrameCount(0),
mState(IDLE),
mClientTid(-1),
mFormat(format),
mFlags(0)
{
mName = audioFlinger->audioMixer()->getTrackName();
if (mName < 0) {
LOGE("no more track names availlable");
return;
}
LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
// LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
size_t size = sizeof(audio_track_cblk_t);
size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
if (sharedBuffer == 0) {
size += bufferSize;
}
mCblkMemory = client->heap()->allocate(size);
if (mCblkMemory != 0) {
mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
if (mCblk) { // construct the shared structure in-place.
new(mCblk) audio_track_cblk_t();
// clear all buffers
mCblk->frameCount = frameCount;
mCblk->sampleRate = sampleRate;
mCblk->channels = channelCount;
if (sharedBuffer == 0) {
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
// written to buffer
mCblk->flowControlFlag = 1;
} else {
mBuffer = sharedBuffer->pointer();
}
mBufferEnd = (uint8_t *)mBuffer + bufferSize;
}
} else {
LOGE("not enough memory for AudioTrack size=%u", size);
client->heap()->dump("AudioTrack");
return;
}
}
AudioFlinger::TrackBase::~TrackBase()
{
mCblk->~audio_track_cblk_t(); // destroy our shared-structure.
mCblkMemory.clear(); // and free the shared memory
mClient.clear();
}
void AudioFlinger::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
buffer->raw = 0;
mFrameCount = buffer->frameCount;
step();
buffer->frameCount = 0;
}
bool AudioFlinger::TrackBase::step() {
bool result;
audio_track_cblk_t* cblk = this->cblk();
result = cblk->stepServer(mFrameCount);
if (!result) {
LOGV("stepServer failed acquiring cblk mutex");
mFlags |= STEPSERVER_FAILED;
}
return result;
}
void AudioFlinger::TrackBase::reset() {
audio_track_cblk_t* cblk = this->cblk();
cblk->user = 0;
cblk->server = 0;
cblk->userBase = 0;
cblk->serverBase = 0;
mFlags = 0;
LOGV("TrackBase::reset");
}
sp<IMemory> AudioFlinger::TrackBase::getCblk() const
{
return mCblkMemory;
}
int AudioFlinger::TrackBase::sampleRate() const {
return mCblk->sampleRate;
}
int AudioFlinger::TrackBase::channelCount() const {
return mCblk->channels;
}
void* AudioFlinger::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
audio_track_cblk_t* cblk = this->cblk();
int16_t *bufferStart = (int16_t *)mBuffer + (offset-cblk->serverBase)*cblk->channels;
int16_t *bufferEnd = bufferStart + frames * cblk->channels;
// Check validity of returned pointer in case the track control block would have been corrupted.
if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd) {
LOGW("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \
server %d, serverBase %d, user %d, userBase %d",
bufferStart, bufferEnd, mBuffer, mBufferEnd,
cblk->server, cblk->serverBase, cblk->user, cblk->userBase);
return 0;
}
return bufferStart;
}
// ----------------------------------------------------------------------------
AudioFlinger::Track::Track(
const sp<AudioFlinger>& audioFlinger,
const sp<Client>& client,
int streamType,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
const sp<IMemory>& sharedBuffer)
: TrackBase(audioFlinger, client, streamType, sampleRate, format, channelCount, frameCount, sharedBuffer)
{
mVolume[0] = 1.0f;
mVolume[1] = 1.0f;
mMute = false;
mSharedBuffer = sharedBuffer;
}
AudioFlinger::Track::~Track()
{
wp<Track> weak(this); // never create a strong ref from the dtor
mState = TERMINATED;
mAudioFlinger->removeTrack(weak, mName);
}
void AudioFlinger::Track::destroy()
{
mAudioFlinger->destroyTrack(this);
}
void AudioFlinger::Track::dump(char* buffer, size_t size)
{
snprintf(buffer, size, " %5d %5d %3u %3u %3u %3u %1d %1d %1d %5u %5u %5u %04x %04x\n",
mName - AudioMixer::TRACK0,
mClient->pid(),
mStreamType,
mFormat,
mCblk->channels,
mFrameCount,
mState,
mMute,
mFillingUpStatus,
mCblk->sampleRate,
mCblk->volume[0],
mCblk->volume[1],
mCblk->server,
mCblk->user);
}
status_t AudioFlinger::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
{
audio_track_cblk_t* cblk = this->cblk();
uint32_t framesReady;
uint32_t framesReq = buffer->frameCount;
// Check if last stepServer failed, try to step now
if (mFlags & TrackBase::STEPSERVER_FAILED) {
if (!step()) goto getNextBuffer_exit;
LOGV("stepServer recovered");
mFlags &= ~TrackBase::STEPSERVER_FAILED;
}
framesReady = cblk->framesReady();
if (LIKELY(framesReady)) {
uint32_t s = cblk->server;
uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
if (framesReq > framesReady) {
framesReq = framesReady;
}
if (s + framesReq > bufferEnd) {
framesReq = bufferEnd - s;
}
buffer->raw = getBuffer(s, framesReq);
if (buffer->raw == 0) goto getNextBuffer_exit;
buffer->frameCount = framesReq;
return NO_ERROR;
}
getNextBuffer_exit:
buffer->raw = 0;
buffer->frameCount = 0;
return NOT_ENOUGH_DATA;
}
bool AudioFlinger::Track::isReady() const {
if (mFillingUpStatus != FS_FILLING) return true;
if (mCblk->framesReady() >= mCblk->frameCount ||
mCblk->forceReady) {
mFillingUpStatus = FS_FILLED;
mCblk->forceReady = 0;
return true;
}
return false;
}
status_t AudioFlinger::Track::start()
{
LOGV("start(%d)", mName);
mAudioFlinger->addTrack(this);
return NO_ERROR;
}
void AudioFlinger::Track::stop()
{
LOGV("stop(%d)", mName);
Mutex::Autolock _l(mAudioFlinger->mLock);
if (mState > STOPPED) {
mState = STOPPED;
// If the track is not active (PAUSED and buffers full), flush buffers
if (mAudioFlinger->mActiveTracks.indexOf(this) < 0) {
reset();
}
LOGV("(> STOPPED) => STOPPED (%d)", mName);
}
}
void AudioFlinger::Track::pause()
{
LOGV("pause(%d)", mName);
Mutex::Autolock _l(mAudioFlinger->mLock);
if (mState == ACTIVE || mState == RESUMING) {
mState = PAUSING;
LOGV("ACTIVE/RESUMING => PAUSING (%d)", mName);
}
}
void AudioFlinger::Track::flush()
{
LOGV("flush(%d)", mName);
Mutex::Autolock _l(mAudioFlinger->mLock);
if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
return;
}
// No point remaining in PAUSED state after a flush => go to
// STOPPED state
mState = STOPPED;
// NOTE: reset() will reset cblk->user and cblk->server with
// the risk that at the same time, the AudioMixer is trying to read
// data. In this case, getNextBuffer() would return a NULL pointer
// as audio buffer => the AudioMixer code MUST always test that pointer
// returned by getNextBuffer() is not NULL!
reset();
}
void AudioFlinger::Track::reset()
{
// Do not reset twice to avoid discarding data written just after a flush and before
// the audioflinger thread detects the track is stopped.
if (!mResetDone) {
TrackBase::reset();
// Force underrun condition to avoid false underrun callback until first data is
// written to buffer
mCblk->flowControlFlag = 1;
mCblk->forceReady = 0;
mFillingUpStatus = FS_FILLING;
mResetDone = true;
}
}
void AudioFlinger::Track::mute(bool muted)
{
mMute = muted;
}
void AudioFlinger::Track::setVolume(float left, float right)
{
mVolume[0] = left;
mVolume[1] = right;
}
// ----------------------------------------------------------------------------
AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::Track>& track)
: BnAudioTrack(),
mTrack(track)
{
}
AudioFlinger::TrackHandle::~TrackHandle() {
// just stop the track on deletion, associated resources
// will be freed from the main thread once all pending buffers have
// been played. Unless it's not in the active track list, in which
// case we free everything now...
mTrack->destroy();
}
status_t AudioFlinger::TrackHandle::start() {
return mTrack->start();
}
void AudioFlinger::TrackHandle::stop() {
mTrack->stop();
}
void AudioFlinger::TrackHandle::flush() {
mTrack->flush();
}
void AudioFlinger::TrackHandle::mute(bool e) {
mTrack->mute(e);
}
void AudioFlinger::TrackHandle::pause() {
mTrack->pause();
}
void AudioFlinger::TrackHandle::setVolume(float left, float right) {
mTrack->setVolume(left, right);
}
sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
return mTrack->getCblk();
}
status_t AudioFlinger::TrackHandle::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
return BnAudioTrack::onTransact(code, data, reply, flags);
}
// ----------------------------------------------------------------------------
sp<IAudioRecord> AudioFlinger::openRecord(
pid_t pid,
int streamType,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
uint32_t flags,
status_t *status)
{
sp<AudioRecordThread> thread;
sp<RecordTrack> recordTrack;
sp<RecordHandle> recordHandle;
sp<Client> client;
wp<Client> wclient;
AudioStreamIn* input = 0;
int inFrameCount;
size_t inputBufferSize;
status_t lStatus;
// check calling permissions
if (!recordingAllowed()) {
lStatus = PERMISSION_DENIED;
goto Exit;
}
if (uint32_t(streamType) >= AudioRecord::NUM_STREAM_TYPES) {
LOGE("invalid stream type");
lStatus = BAD_VALUE;
goto Exit;
}
if (sampleRate > MAX_SAMPLE_RATE) {
LOGE("Sample rate out of range");
lStatus = BAD_VALUE;
goto Exit;
}
if (mSampleRate == 0) {
LOGE("Audio driver not initialized");
lStatus = NO_INIT;
goto Exit;
}
if (mAudioRecordThread == 0) {
LOGE("Audio record thread not started");
lStatus = NO_INIT;
goto Exit;
}
// Check that audio input stream accepts requested audio parameters
inputBufferSize = mAudioHardware->getInputBufferSize(sampleRate, format, channelCount);
if (inputBufferSize == 0) {
lStatus = BAD_VALUE;
LOGE("Bad audio input parameters: sampling rate %u, format %d, channels %d", sampleRate, format, channelCount);
goto Exit;
}
// add client to list
{
Mutex::Autolock _l(mLock);
wclient = mClients.valueFor(pid);
if (wclient != NULL) {
client = wclient.promote();
} else {
client = new Client(this, pid);
mClients.add(pid, client);
}
}
// frameCount must be a multiple of input buffer size
inFrameCount = inputBufferSize/channelCount/sizeof(short);
frameCount = ((frameCount - 1)/inFrameCount + 1) * inFrameCount;
// create new record track and pass to record thread
recordTrack = new RecordTrack(this, client, streamType, sampleRate,
format, channelCount, frameCount);
// return to handle to client
recordHandle = new RecordHandle(recordTrack);
lStatus = NO_ERROR;
Exit:
if (status) {
*status = lStatus;
}
return recordHandle;
}
status_t AudioFlinger::startRecord(RecordTrack* recordTrack) {
if (mAudioRecordThread != 0) {
return mAudioRecordThread->start(recordTrack);
}
return NO_INIT;
}
void AudioFlinger::stopRecord(RecordTrack* recordTrack) {
if (mAudioRecordThread != 0) {
mAudioRecordThread->stop(recordTrack);
}
}
// ----------------------------------------------------------------------------
AudioFlinger::RecordTrack::RecordTrack(
const sp<AudioFlinger>& audioFlinger,
const sp<Client>& client,
int streamType,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount)
: TrackBase(audioFlinger, client, streamType, sampleRate, format,
channelCount, frameCount, 0),
mOverflow(false)
{
}
AudioFlinger::RecordTrack::~RecordTrack()
{
mAudioFlinger->audioMixer()->deleteTrackName(mName);
}
status_t AudioFlinger::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
{
audio_track_cblk_t* cblk = this->cblk();
uint32_t framesAvail;
uint32_t framesReq = buffer->frameCount;
// Check if last stepServer failed, try to step now
if (mFlags & TrackBase::STEPSERVER_FAILED) {
if (!step()) goto getNextBuffer_exit;
LOGV("stepServer recovered");
mFlags &= ~TrackBase::STEPSERVER_FAILED;
}
framesAvail = cblk->framesAvailable_l();
if (LIKELY(framesAvail)) {
uint32_t s = cblk->server;
uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
if (framesReq > framesAvail) {
framesReq = framesAvail;
}
if (s + framesReq > bufferEnd) {
framesReq = bufferEnd - s;
}
buffer->raw = getBuffer(s, framesReq);
if (buffer->raw == 0) goto getNextBuffer_exit;
buffer->frameCount = framesReq;
return NO_ERROR;
}
getNextBuffer_exit:
buffer->raw = 0;
buffer->frameCount = 0;
return NOT_ENOUGH_DATA;
}
status_t AudioFlinger::RecordTrack::start()
{
return mAudioFlinger->startRecord(this);
}
void AudioFlinger::RecordTrack::stop()
{
mAudioFlinger->stopRecord(this);
TrackBase::reset();
// Force overerrun condition to avoid false overrun callback until first data is
// read from buffer
mCblk->flowControlFlag = 1;
}
// ----------------------------------------------------------------------------
AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordTrack>& recordTrack)
: BnAudioRecord(),
mRecordTrack(recordTrack)
{
}
AudioFlinger::RecordHandle::~RecordHandle() {
stop();
}
status_t AudioFlinger::RecordHandle::start() {
LOGV("RecordHandle::start()");
return mRecordTrack->start();
}
void AudioFlinger::RecordHandle::stop() {
LOGV("RecordHandle::stop()");
mRecordTrack->stop();
}
sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
return mRecordTrack->getCblk();
}
status_t AudioFlinger::RecordHandle::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
return BnAudioRecord::onTransact(code, data, reply, flags);
}
// ----------------------------------------------------------------------------
AudioFlinger::AudioRecordThread::AudioRecordThread(AudioHardwareInterface* audioHardware) :
mAudioHardware(audioHardware),
mActive(false)
{
}
AudioFlinger::AudioRecordThread::~AudioRecordThread()
{
}
bool AudioFlinger::AudioRecordThread::threadLoop()
{
LOGV("AudioRecordThread: start record loop");
AudioBufferProvider::Buffer buffer;
int inBufferSize = 0;
int inFrameCount = 0;
AudioStreamIn* input = 0;
mActive = 0;
// start recording
while (!exitPending()) {
if (!mActive) {
mLock.lock();
if (!mActive && !exitPending()) {
LOGV("AudioRecordThread: loop stopping");
if (input) {
delete input;
input = 0;
}
mRecordTrack.clear();
mWaitWorkCV.wait(mLock);
LOGV("AudioRecordThread: loop starting");
if (mRecordTrack != 0) {
input = mAudioHardware->openInputStream(mRecordTrack->format(),
mRecordTrack->channelCount(),
mRecordTrack->sampleRate(),
&mStartStatus);
if (input != 0) {
inBufferSize = input->bufferSize();
inFrameCount = inBufferSize/input->frameSize();
}
} else {
mStartStatus = NO_INIT;
}
if (mStartStatus !=NO_ERROR) {
LOGW("record start failed, status %d", mStartStatus);
mActive = false;
mRecordTrack.clear();
}
mWaitWorkCV.signal();
}
mLock.unlock();
} else if (mRecordTrack != 0){
buffer.frameCount = inFrameCount;
if (LIKELY(mRecordTrack->getNextBuffer(&buffer) == NO_ERROR)) {
LOGV("AudioRecordThread read: %d frames", buffer.frameCount);
if (input->read(buffer.raw, inBufferSize) < 0) {
LOGE("Error reading audio input");
sleep(1);
}
mRecordTrack->releaseBuffer(&buffer);
mRecordTrack->overflow();
}
// client isn't retrieving buffers fast enough
else {
if (!mRecordTrack->setOverflow())
LOGW("AudioRecordThread: buffer overflow");
// Release the processor for a while before asking for a new buffer.
// This will give the application more chance to read from the buffer and
// clear the overflow.
usleep(5000);
}
}
}
if (input) {
delete input;
}
mRecordTrack.clear();
return false;
}
status_t AudioFlinger::AudioRecordThread::start(RecordTrack* recordTrack)
{
LOGV("AudioRecordThread::start");
AutoMutex lock(&mLock);
mActive = true;
// If starting the active track, just reset mActive in case a stop
// was pending and exit
if (recordTrack == mRecordTrack.get()) return NO_ERROR;
if (mRecordTrack != 0) return -EBUSY;
mRecordTrack = recordTrack;
// signal thread to start
LOGV("Signal record thread");
mWaitWorkCV.signal();
mWaitWorkCV.wait(mLock);
LOGV("Record started, status %d", mStartStatus);
return mStartStatus;
}
void AudioFlinger::AudioRecordThread::stop(RecordTrack* recordTrack) {
LOGV("AudioRecordThread::stop");
AutoMutex lock(&mLock);
if (mActive && (recordTrack == mRecordTrack.get())) {
mActive = false;
}
}
void AudioFlinger::AudioRecordThread::exit()
{
LOGV("AudioRecordThread::exit");
{
AutoMutex lock(&mLock);
requestExit();
mWaitWorkCV.signal();
}
requestExitAndWait();
}
status_t AudioFlinger::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
return BnAudioFlinger::onTransact(code, data, reply, flags);
}
// ----------------------------------------------------------------------------
void AudioFlinger::instantiate() {
defaultServiceManager()->addService(
String16("media.audio_flinger"), new AudioFlinger());
}
}; // namespace android