mtk android 4.4 audio framework 代码分析(未完成)
mtk android 4.4 audio framework 代码分析(未完成)，有需要的朋友可以参考下。
2/28/2015 3:01:24 PM
关于 audio_policy.conf 文件这个文件 mediatek/config/$project 下， 在 audiomtkpolicymanager.cpp 中解析，
解析出如下信息：
1） ATTACHED_OUTPUT_DEVICES_TAG "attached_output_devices" ，对应 类定义中的变量 mAttachedOutputDevices
2 ） #define DEFAULT_OUTPUT_DEVICE_TAG "default_output_device"
对应类定义中的 mDefaultOutputDevice
#define ATTACHED_INPUT_DEVICES_TAG "attached_input_devices"
对应类定义中的 mAvailableInputDevices
mHasA2dp = true; 根据文件解析出是否有此MODULE。
mHasUsb = true; 根据文件接触出是否有此MODULE。
mHasRemoteSubmix = true; 根据文件接触出是否有此MODULE。
3）
最重要的解析出 mHwModules，而这个变量的定义在audiomtkpolicymanager.h 中，Vector
上面关于文件解析的东西其实也属于本节的内容，不过还是单独出去了。
-》mpClientInterface = clientInterface; 就是APS
-》 AudioMTKPolicyManager::LoadCustomVolume
-》GetVolumeVer1ParamFromNV 从NVRAM里读取参数，这个暂且不表。
-》initializeVolumeCurves(); // 初始化VOLUME曲线，SETVOLUME时会用到，以后分析。
-》if (loadAudioPolicyConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE) != NO_ERROR)
上面已经分析了，解析配置文件。
-》mHwModules[i]->mHandle = mpClientInterface->loadHwModule(mHwModules[i]->mName);
mHandle 是 audio_module_handle_t类型，实际上是 AF中成员变量DefaultKeyedVector
原型：
uint32_t AudioMTKPolicyManager::setOutputDevice(audio_io_handle_t output,
audio_devices_t device,
bool force,
int delayMs)
-》
AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output); 根据传入参数取得 outputDesc
-》
if (outputDesc->isDuplicated()) { // 蓝牙，暂且不分析
muteWaitMs = setOutputDevice(outputDesc->mOutput1->mId, device, force, delayMs);
muteWaitMs += setOutputDevice(outputDesc->mOutput2->mId, device, force, delayMs);
return muteWaitMs;
}
-》
if (device != AUDIO_DEVICE_NONE) {
outputDesc->mDevice = device;
} // 设置 outputDesc route to 传入的 device. 软件层面上的。
-》
muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevice, delayMs);
-》
param.addInt(String8(AudioParameter::keyRouting), (int)device); 设置keyroute 的PARA,
-》
mpClientInterface->setParameters(output, param.toString(), delayMs); APS cmd thread 切换。
-》 // update stream volumes according to new device
applyStreamVolumes(output, device, delayMs);
分析见下面。
mpClientInterface->setParameters(output, param.toString(), delayMs); APS cmd thread 切换。
定义 void AudioPolicyService::setParameters(audio_io_handle_t ioHandle,
const char *keyValuePairs,
int delayMs)
{
mAudioCommandThread->parametersCommand(ioHandle, keyValuePairs,
delayMs);
}
-》 AudioCommand *command = new AudioCommand();
-》 insertCommand_l(command, delayMs);
-》 AudioCommandThread::threadLoop()
-》case SET_PARAMETERS: AudioSystem::setParameters
-》 af->setParameters
-》 thread = checkPlaybackThread_l(ioHandle); 找到 相应的 thread
-》 thread->setParameters(keyValuePairs);
-》 ThreadBase::setParameters(const String8& keyValuePairs)
-》 mNewParameters.add(keyValuePairs);
Vector mNewParameters 是 ThreadBase的成员变量。
-》 PlaybackThread::threadLoop()
-》 status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
keyValuePair.string());
mOutput 是AudioStreamOut 类型：
struct AudioStreamOut {
AudioHwDevice* const audioHwDev;
audio_stream_out_t* const stream;
audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); }
AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out) :
audioHwDev(dev), stream(out) {}
};
-》 out->stream.common.set_parameters = out_set_parameters; HAL层
-》 status_t AudioMTKStreamOut::setParameters(const String8 &keyValuePairs)
{
AudioParameter param = AudioParameter(keyValuePairs);
String8 keyRouting = String8(AudioParameter::keyRouting);
status_t status = NO_ERROR;
int devices = 0;
ALOGD("setParameters() %s", keyValuePairs.string());
if (param.getInt(keyRouting, devices) == NO_ERROR) {
param.remove(keyRouting);
dokeyRouting(devices);
mAudioResourceManager->doSetMode();
}
if (param.size()) {
status = BAD_VALUE;
}
return status;
}
-》 AudioMTKStreamOut::dokeyRouting(uint32_t new_device)
-》 mAudioResourceManager->SelectOutputDevice(new_device);
-》 AudioResourceManager::SelectOutputDevice(uint32_t new_device)
AudioResourceManager::SelectOutputDevice 分析
pre_device = mDlOutputDevice;
-》 StopOutputDevice(); // 关掉 mDlOutputDevice，
-》 mDlOutputDevice = new_device; 设置NEW device。
-》 AudioResourceManager::StartOutputDevice()
AudioResourceManager::StartOutputDevice() 分析定义：
switch (mAudioMode) {
case AUDIO_MODE_NORMAL:
case AUDIO_MODE_RINGTONE: {
TurnonAudioDevice(mDlOutputDevice);
break;
}
case AUDIO_MODE_IN_CALL:
case AUDIO_MODE_IN_CALL_2: {
TurnonAudioDeviceIncall(mDlOutputDevice);
break;
}
case AUDIO_MODE_IN_COMMUNICATION: {
TurnonAudioDevice(mDlOutputDevice);
break;
}
}
-》 AudioResourceManager::TurnonAudioDevice(unsigned int mDlOutputDevice)
-》 mAudioAnalogInstance->AnalogOpen(AudioAnalogType::DEVICE_OUT_EARPIECER,
AudioAnalogType::DEVICE_PLATFORM_MACHINE); 打开对应的设备。
-》 AudioAnalogControl::AnalogOpen
定义：
// analog open power , need to open by mux setting
status_t AudioAnalogControl::AnalogOpen(AudioAnalogType::DEVICE_TYPE DeviceType, AudioAnalogType::DEVICE_TYPE_SETTING Type_setting)
{
ALOGD("AnalogOpen DeviceType = %s", kAudioAnalogDeviceTypeName[DeviceType]);
CheckDevicePolicy((uint32*)&DeviceType,AudioAnalogType::AUDIOANALOG_DEVICE);
mBlockAttribute[DeviceType].mEnable = true;
mAudioPlatformDevice->AnalogOpen(DeviceType); 直接操作KERNEL 接口，寄存器。
mAudioMachineDevice->AnalogOpen(DeviceType);
return NO_ERROR;
}
void applyStreamVolumes(audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false); // 注意调用此传入的参数。
-》
for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
checkAndSetVolume(stream,
mStreams[stream].getVolumeIndex(device),
output,
device,
delayMs,
force);
}
-》传入的参数 StreamDescriptor mStreams[AudioSystem::NUM_STREAM_TYPES]; // stream descriptors for volume control 定义， 实际上类似于一个二维数组，STREAM是第一维， DEVICE是第二维。 上述代码实际上是取出音量的index(int).
class StreamDescriptor
{
public:
StreamDescriptor();
int getVolumeIndex(audio_devices_t device);
void dump(int fd);
int mIndexMin; // min volume index
int mIndexMax; // max volume index
KeyedVector mIndexCur; // current volume index per device
bool mCanBeMuted; // true is the stream can be muted
const VolumeCurvePoint *mVolumeCurve[DEVICE_CATEGORY_CNT];
#ifdef MTK_AUDIO
float mIndexRange;
#endif
};
AudioMTKPolicyManager::checkAndSetVolume
定义：
status_t checkAndSetVolume(int stream, int index, audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false); // 注意 后面2个参数。
-》
if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) {
ALOGV("checkAndSetVolume() stream %d muted count %d",
stream, mOutputs.valueFor(output)->mMuteCount[stream]);
return NO_ERROR;
} do not change actual stream volume if the stream is muted
-》
// do not change in call volume if bluetooth is connected and vice versa
if ((stream == AudioSystem::VOICE_CALL && mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) ||
(stream == AudioSystem::BLUETOOTH_SCO && mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO)) {
ALOGD("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm",
stream, mForceUse[AudioSystem::FOR_COMMUNICATION]);
return INVALID_OPERATION;
}
-》
float volume = computeVolume(stream, index, output, device); // 计算音量。 后面详细分析、
-》
//for VT notify tone when incoming call. it's volume will be adusted in hardware.
if((stream == AudioSystem::VOICE_CALL ||stream == AudioSystem::BLUETOOTH_SCO) && mOutputs.valueFor(output)->mRefCount[stream]!=0 && mPhoneState==AudioSystem::MODE_IN_CALL)
{
volume =1.0;
} 处理特列 MODE_IN_CALL
-》
// ALPS00554824 KH: If notifiaction is exist, FM should be mute
if ((stream == AudioSystem::FM) &&
(mOutputs.valueFor(output)->mRefCount[AudioSystem::NOTIFICATION]
|| mOutputs.valueFor(output)->mRefCount[AudioSystem::RING]
|| mOutputs.valueFor(output)->mRefCount[AudioSystem::ALARM]))
{
volume =0.0;
}
-》
if (volume != mOutputs.valueFor(output)->mCurVolume[stream] ||
force) the float value returned by computeVolume() changed
// - the force flag is set ，两者有一个条件满足则可以 调节音量。
-》
mOutputs.valueFor(output)->mCurVolume[stream] = volume;
float mCurVolume[AudioSystem::NUM_STREAM_TYPES]; // current stream volume，更新软件音量。
-》
mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, volume, output, delayMs); //aps set volume
-》
int AudioPolicyService::setStreamVolume(audio_stream_type_t stream,
float volume,
audio_io_handle_t output,
int delayMs)
{
return (int)mAudioCommandThread->volumeCommand(stream, volume,
output, delayMs);
}
-》
AudioCommandThread::volumeCommand
-》
AudioCommand *command = new AudioCommand();
-》
insertCommand_l(command, delayMs);
-》
mAudioCommands.insertAt(command, i + 1); 加入到mAudioCommands的 CMD容器中。
-》
AudioCommandThread::threadLoop
while (!mAudioCommands.isEmpty()) {
nsecs_t curTime = systemTime();
// commands are sorted by increasing time stamp: execute them from index 0 and up
if (mAudioCommands[0]->mTime <= curTime) {
// 当mAudioCommands 不为空，时间来到，
-》
CASE SET_VOLUME: {
VolumeData *data = (VolumeData *)command->mParam;
ALOGV("AudioCommandThread() processing set volume stream %d, \
volume %f, output %d", data->mStream, data->mVolume, data->mIO);
command->mStatus = AudioSystem::setStreamVolume(data->mStream, data->mVolume,data->mIO); //
-》
af->setStreamVolume(stream, value, output); 下面分析
### AudioFlinger::setStreamVolume 分析 ###
status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
audio_io_handle_t output) 定义
-》
thread = checkPlaybackThread_l(output);
-》
mStreamTypes[stream].volume = value; 设置软件音量
stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; AF中音量表示方法。
-》
thread->setStreamVolume(stream, value); 后面分析
-》
if(stream == AUDIO_STREAM_FM)
{
MTK_ALOG_D("setStreamVolume FM value = %f",value);
#if defined(MT5192_FM) || defined(MT5193_FM)
int FmVolume = (AudioSystem::logToLinear(value));
char Volume[30];
sprintf(Volume,"SetFmVolume=%d",FmVolume);
String8 Key = String8(Volume);
#else
int FmVolume = (AudioSystem::logToLinear(value)>>4);
char Volume[30];
sprintf(Volume,"SetFmVolume=%d",FmVolume);
String8 Key = String8(Volume);
#endif
audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
dev->set_parameters (dev,Key);
} 如果是stream 是FM， 则直接dev->set_parameters (dev,Key); 设置硬件VOLUME.
### PlaybackThread::setStreamVolume 分析 ###
struct stream_type_t {
stream_type_t()
: volume(1.0f),
mute(false)
{
}
float volume;
bool mute;
}; // AF中 和 PLAYBACKTHREAD 中都有个这样的。
定义 void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
{
Mutex::Autolock _l(mLock);
mStreamTypes[stream].volume = value;
}
构造函数中 mAudioManager = (AudioManager)context.getSystemService(Context.AUDIO_SERVICE);
-》
mObserver = new WiredAccessoryObserver();
-》
init 中 public String getSwitchStatePath() {
return String.format(“/sys/class/switch/%s/state”, mDevName);
}
-》
mAudioManager.setWiredDeviceConnectionState(device, state, headsetName);
-》
AudioService中 public void setWiredDeviceConnectionState(int device, int state, String name) {
synchronized (mConnectedDevices) {
int delay = checkSendBecomingNoisyIntent(device, state);
queueMsgUnderWakeLock(mAudioHandler,
MSG_SET_WIRED_DEVICE_CONNECTION_STATE,
device,
state,
name,
delay);
}
}
-》
checkSendBecomingNoisyIntent 中
sendMsg(mAudioHandler ， MSG_BROADCAST_AUDIO_BECOMING_NOISY,
SENDMSG_REPLACE,
0,
0,
null,
0);
delay = 1000;
Senmsg在handler中处理
sendBroadcastToAll(new Intent(AudioManager.ACTION_AUDIO_BECOMING_NOISY));
这个广播哪些地方有处理，举例子：MUSIC应用中
public void onReceive(Context context, Intent intent) {
String intentAction = intent.getAction();
MusicLogUtils.d("MediaButtonIntentReceiver", "intentAction " + intentAction);
if (AudioManager.ACTION_AUDIO_BECOMING_NOISY.equals(intentAction)) {
-》
i.setAction(MediaPlaybackService.SERVICECMD);
i.putExtra(MediaPlaybackService.CMDNAME, MediaPlaybackService.CMDPAUSE);
context.startService(i);
-》
在 mediaplayerservice 中处理
else if (CMDPAUSE.equals(cmd) || PAUSE_ACTION.equals(action)
|| AudioManager.ACTION_AUDIO_BECOMING_NOISY.equals(action)) {
pause();
mPausedByTransientLossOfFocus = false;
}
-》 audioservice 中
case MSG_SET_WIRED_DEVICE_CONNECTION_STATE:
onSetWiredDeviceConnectionState(msg.arg1, msg.arg2, (String)msg.obj);
-》
private void onSetWiredDeviceConnectionState(int device, int state, String name)
{
synchronized (mConnectedDevices) {
Log.d(TAG,”onSetWiredDeviceConnectionState:” + “device:” + device + “,state:” + state);
if ((state == 0) && ((device == AudioSystem.DEVICE_OUT_WIRED_HEADSET) ||(device == AudioSystem.DEVICE_OUT_WIRED_HEADPHONE))) { // 耳机
/// M: Change for sound output from device when a2dp conneted @ {
//setBluetoothA2dpOnInt(true); AudioSystem.setForceUse(AudioSystem.FOR_MEDIA,AudioSystem.FORCE_NONE);
///@}
}
boolean isUsb = ((device & AudioSystem.DEVICE_OUT_ALL_USB) != 0);
handleDeviceConnection((state == 1), device, (isUsb ? name : “”));
if (state != 0) {
if ((device == AudioSystem.DEVICE_OUT_WIRED_HEADSET) ||
(device == AudioSystem.DEVICE_OUT_WIRED_HEADPHONE)) {
setBluetoothA2dpOnInt(false);
}
if ((device & mSafeMediaVolumeDevices) != 0) {
sendMsg(mAudioHandler,
MSG_CHECK_MUSIC_ACTIVE,
SENDMSG_REPLACE,
0,
0,
null,
MUSIC_ACTIVE_POLL_PERIOD_MS);
}
}
if (!isUsb) {
sendDeviceConnectionIntent(device, state, name);
}
}
}
boolean isUsb = ((device & AudioSystem.DEVICE_OUT_ALL_USB) != 0);
handleDeviceConnection((state == 1), device, (isUsb ? name : “”));
下面是定义：
private boolean handleDeviceConnection(boolean connected, int device, String params) {
synchronized (mConnectedDevices) {
boolean isConnected = (mConnectedDevices.containsKey(device) &&
(params.isEmpty() || mConnectedDevices.get(device).equals(params)));
Log.d(TAG,"handleDeviceConnection:isConnected" + isConnected);
if (isConnected && !connected) { 、、 拔出
AudioSystem.setDeviceConnectionState(device, 设置状态
AudioSystem.DEVICE_STATE_UNAVAILABLE,
mConnectedDevices.get(device));
Log.d(TAG,"handleDeviceConnection remove:" + "connected:" + connected + ",device:" + device);
mConnectedDevices.remove(device); 移除设备。
return true;
} else if (!isConnected && connected) {
AudioSystem.setDeviceConnectionState(device,
AudioSystem.DEVICE_STATE_AVAILABLE,
params);
Log.d(TAG,"handleDeviceConnection connect:" + "connected:" + connected + ",device:" + device);
mConnectedDevices.put(new Integer(device), params);
return true;
}
}
return false;
}
-》
AudioSystem::setDeviceConnectionState
-》
aps->setDeviceConnectionState(
-》
AudioMTKPolicyManager::setDeviceConnectionState(audio_devices_t device,
AudioSystem::device_connection_state state,const char *device_address)
-》
case AudioSystem::DEVICE_STATE_AVAILABLE:
if (checkOutputsForDevice(device, state, outputs) != NO_ERROR) {
return INVALID_OPERATION;
}
ALOGD("setDeviceConnectionState() checkOutputsForDevice() returned %d outputs",
outputs.size());
// register new device as available
mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices | device); 输出设备。
checkOutputForAllStrategies(); 更新strategy
if ((state == AudioSystem::DEVICE_STATE_UNAVAILABLE) ||
(mOutputs.valueFor(outputs[i])->mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
closeOutput(outputs[i]); 关掉 output
}
updateDevicesAndOutputs();
for (size_t i = 0; i < mOutputs.size(); i++) {
// do not force device change on duplicated output because if device is 0, it will
// also force a device 0 for the two outputs it is duplicated to which may override
// a valid device selection on those outputs.
setOutputDevice(mOutputs.keyAt(i),
getNewDevice(mOutputs.keyAt(i), true /*fromCache*/),
!mOutputs.valueAt(i)->isDuplicated(),
0);
}
### AudioMTKPolicyManager::checkOutputForAllStrategies() 分析 ###
-》
调用 AudioMTKPolicyManager::checkOutputForStrategy(routing_strategy strategy)
-》
audio_devices_t oldDevice = getDeviceForStrategy(strategy, true /*fromCache*/);
audio_devices_t newDevice = getDeviceForStrategy(strategy, false /*fromCache*/);
SortedVector srcOutputs = getOutputsForDevice(oldDevice, mPreviousOutputs);
SortedVector dstOutputs = getOutputsForDevice(newDevice, mOutputs);
-》
if (desc->strategyRefCount(strategy) != 0) { //
#ifdef MTK_AUDIO //ALPS00446176 .ex: Speaker->Speaker,Don't move track and mute. Only change to dstOutputs[0]
if(dstOutputs[0]!=srcOutputs[i]) 如果现在的dstOutputs不在 srcOutputs中
{
setStrategyMute(strategy, true, srcOutputs[i]); // 立即MUTE
setStrategyMute(strategy, false, srcOutputs[i], MUTE_TIME_MS, newDevice); // 2秒后unmute newDevice
}
#else
setStrategyMute(strategy, true, srcOutputs[i]);
setStrategyMute(strategy, false, srcOutputs[i], MUTE_TIME_MS, newDevice);
#endif
}
-》
AudioMTKPolicyManager::setStrategyMute
-》
setStreamMute(stream, on, output, delayMs, device);
-》
AudioMTKPolicyManager::setStreamMute(int stream,
bool on,
audio_io_handle_t output,
int delayMs,
audio_devices_t device)
-》
if (on) { // mute on
if (outputDesc->mMuteCount[stream] == 0) {
if (streamDesc.mCanBeMuted &&
((stream != AudioSystem::ENFORCED_AUDIBLE) ||
(mForceUse[AudioSystem::FOR_SYSTEM] == AudioSystem::FORCE_NONE))) {
checkAndSetVolume(stream, 0, output, device, delayMs);
}
}
// increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored
outputDesc->mMuteCount[stream]++;
}
checkAndSetVolume // 上面已经分析过了，最后会设置音量到 af对应的threads
接上面函数继续分析
-》
// Move tracks associated to this strategy from previous output to new output
for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
if (getStrategy((AudioSystem::stream_type)i) == strategy) {
//FIXME see fixme on name change
mpClientInterface->setStreamOutput((AudioSystem::stream_type)i,
dstOutputs[0] /* ignored */);
}
}
-》
status_t AudioPolicyCompatClient::setStreamOutput(AudioSystem::stream_type stream,
audio_io_handle_t output)
{
return mServiceOps->set_stream_output(mService, (audio_stream_type_t)stream,
output);
}
-》
aps： set_stream_output
-》
af->setStreamOutput
-》
status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output)
{
Mutex::Autolock _l(mLock);
ALOGV("setStreamOutput() stream %d to output %d", stream, output);
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
thread->invalidateTracks(stream);
}
return NO_ERROR;
}
-》
PlaybackThread::cacheParameters_l()
-》
PlaybackThread： t->mCblk->cv.signal(); 发信号， 哪里在等还没找到。
uint32_t mRefCount[AudioSystem::NUM_STREAM_TYPES]; // number of streams of each type using this output
DefaultKeyedVector mPreviousOutputs;
Audioservice中重要变量：
private final HashMap mConnectedDevices = new HashMap ();
AudioTrack.java
blic AudioTrack(int streamType, int sampleRateInHz, int channelConfig, int audioFormat,
int bufferSizeInBytes, int mode, int sessionId)
-》
int initResult = native_setup(new WeakReference(this),
mStreamType, mSampleRate, mChannels, mAudioFormat,
mNativeBufferSizeInBytes, mDataLoadMode, session);
-》
android_media_AudioTrack_native_setup(JNIEnv *env, jobject thiz, jobject weak_this,
jint streamType, jint sampleRateInHertz, jint javaChannelMask,
jint audioFormat, jint buffSizeInBytes, jint memoryMode, jintArray jSession)
sp lpTrack = new AudioTrack();
-》
## AudioFlinger::openOutput 分析 ##
DefaultKeyedVector< audio_io_handle_t, sp > mPlaybackThreads;
在 AudioFlinger::openOutput 中 返回的就是这个audio_io_handle_t KEY值。
AudioFlinger::openOutput
->
outHwDev = findSuitableHwDev_l(module, *pDevices); // 找到 AudioHwDevice, 对应HAL 下的一个设备。
->
status = hwDevHal->open_output_stream(hwDevHal,
id,
*pDevices,
(audio_output_flags_t)flags,
&config,
&outStream);
outStream 是在HAL层分配的，然后返回到AF中。
-》
AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); // 用上述2个NEW一个 AudioStreamOut。
->
if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) ||
(config.format != AUDIO_FORMAT_PCM_16_BIT) ||
(config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) {
thread = new DirectOutputThread(this, output, id, *pDevices);
ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
} else {
thread = new MixerThread(this, output, id, *pDevices);
ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
附录LOG：（开机过程中）
Line 2397: 01-01 03:59:20.971207 211 211 D AudioFlinger: openOutput(), module 1 Device 2, SamplingRate 44100, Format 0x000003, Channels 3, flags 2
Line 2398: 01-01 03:59:20.971246 211 211 D AudioFlinger: openOutput(), offloadInfo 0x0 version 0xffffffff
Line 2399: 01-01 03:59:20.971291 211 211 D AudioALSAStreamManager: +openOutputStream()
Line 2888: 01-01 03:59:21.028671 211 211 D AudioALSAStreamManager: -openOutputStream(), out = 0xb7ba8250, status = 0x0, mStreamOutVector.size() = 1
Line 2889: 01-01 03:59:21.028791 211 211 D AudioFlinger: openOutput() openOutputStream returned output 0xb7ba81e0, SamplingRate 44100, Format 0x000003, Channels 3, status 0, flags 2
Line 2889: 01-01 03:59:21.028791 211 211 D AudioFlinger: openOutput() openOutputStream returned output 0xb7ba81e0, SamplingRate 44100, Format 0x000003, Channels 3, status 0, flags 2
Line 2905: 01-01 03:59:21.048650 211 211 D AudioFlinger: openOutput() created mixer output: ID 2 thread 0xb4b26008
（在 AudioFlinger 构造函数中初始化了 mNextUniqueId(1) ）
-》
AudioTrack::AudioTrack(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
int frameCount,
audio_output_flags_t flags,
callback_t cbf,
void* user,
int notificationFrames,
int sessionId,
transfer_type transferType,
const audio_offload_info_t *offloadInfo,
int uid)
: mStatus(NO_INIT),
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT),
mPausedPosition(0)
{
mStatus = set(streamType, sampleRate, format, channelMask,
frameCount, flags, cbf, user, notificationFrames,
0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType,
offloadInfo, uid);
}
}
mPlaybackThreads.add(id, thread);
-》
audio_io_handle_t output = AudioSystem::getOutput(
streamType,
sampleRate, format, channelMask,
flags,
offloadInfo);
if (cbf != NULL) {
mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
}
// create the IAudioTrack
status_t status = createTrack_l(streamType,
sampleRate,
format,
frameCount,
flags,
sharedBuffer,
output,
0 /*epoch*/);
-》
sp track = audioFlinger->createTrack(streamType,
sampleRate,
// AudioFlinger only sees 16-bit PCM
format == AUDIO_FORMAT_PCM_8_BIT ?
AUDIO_FORMAT_PCM_16_BIT : format,
mChannelMask,
frameCount,
&trackFlags,
sharedBuffer,
output,
tid,
&mSessionId,
mName,
mClientUid,
&status);
-》
track = thread->createTrack_l(client, streamType, sampleRate, format,
channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus);
-》
// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
sp AudioFlinger::PlaybackThread::createTrack_l(
const sp& client,
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
const sp& sharedBuffer,
int sessionId,
IAudioFlinger::track_flags_t *flags,
pid_t tid,
int uid,
status_t *status)
{
sp -》
uint32_t actual = AudioSystem::getStrategyForStream(t->streamType());
track = new Track(this, client, streamType, sampleRate, format,
channelMask, frameCount, sharedBuffer, sessionId, uid, *flags);
-》
AudioFlinger::PlaybackThread::Track::Track
－》
AudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
size_t frameSize, bool clientInServer = false)
: ServerProxy(cblk, buffers, frameCount, frameSize, true /isOut/, clientInServer) { }
－》
ServerProxy::ServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
size_t frameSize, bool isOut, bool clientInServer)
: Proxy(cblk, buffers, frameCount, frameSize, isOut, clientInServer),
mAvailToClient(0), mFlush(0)
{
}
然后根据这个 streamout 和ID, NEW threads. 然后ADD 到 AF中的 mPlaybackThreads。
-》hwDevHal->set_mode(hwDevHal, mMode); // 大概是 设置MODE, NORMAL , CALL MODE? 条件是 (mPrimaryHardwareDev == NULL) ，只有第一次AudioFlinger::openOutput 时会设置模式。
Audio_hw_hal.cpp 中关于
legacy_adev_open 中将
ladev->device.open_output_stream = adev_open_output_stream;
-》 adev_open_output_stream
-》
static int adev_open_output_stream(struct audio_hw_device *dev,
audio_io_handle_t handle,
audio_devices_t devices,
audio_output_flags_t flags,
struct audio_config *config,
struct audio_stream_out **stream_out)
{
struct legacy_audio_device *ladev = to_ladev(dev);
status_t status;
struct legacy_stream_out *out;
int ret;
out = (struct legacy_stream_out *)calloc(1, sizeof(*out)); 分配内存，然后由下面设备填充。
if (!out)
return -ENOMEM;
// 打开设备，填充
out->legacy_out = ladev->hwif->openOutputStreamWithFlag(devices, (int *) &config->format,
&config->channel_mask,
&config->sample_rate, &status, flags);
if (!out->legacy_out) {
ret = status;
goto err_open;
}
out->stream.common.get_sample_rate = out_get_sample_rate;
out->stream.common.set_sample_rate = out_set_sample_rate;
out->stream.common.get_buffer_size = out_get_buffer_size;
out->stream.common.get_channels = out_get_channels;
out->stream.common.get_format = out_get_format;
out->stream.common.set_format = out_set_format;
out->stream.common.standby = out_standby;
out->stream.common.dump = out_dump;
out->stream.common.set_parameters = out_set_parameters;
out->stream.common.get_parameters = out_get_parameters;
out->stream.common.add_audio_effect = out_add_audio_effect;
out->stream.common.remove_audio_effect = out_remove_audio_effect;
out->stream.get_latency = out_get_latency;
out->stream.set_volume = out_set_volume;
out->stream.write = out_write;
out->stream.get_render_position = out_get_render_position;
out->stream.get_next_write_timestamp = out_get_next_write_timestamp;
out->stream.set_callback = out_set_callback;
out->stream.get_presentation_position = out_get_presentation_position;
*stream_out = &out->stream;
return 0;
err_open:
free(out);
*stream_out = NULL;
return ret;
}
-》AudioHardwareALSA::openOutputStream
-》
android_audio_legacy::AudioStreamOut *AudioALSAHardware::openOutputStream(
uint32_t devices,
int *format,
uint32_t *channels,
uint32_t *sampleRate,
status_t *status)
{
return mStreamManager->openOutputStream(devices, format, channels, sampleRate, status);
}
/*==============================================================================
* Implementations
============================================================================/
-》
android_audio_legacy::AudioStreamOut *AudioALSAStreamManager::openOutputStream(
uint32_t devices,
int *format,
uint32_t *channels,
uint32_t *sampleRate,
status_t *status)
{
ALOGD(“+%s()”, FUNCTION);
AudioAutoTimeoutLock streamVectorAutoTimeoutLock(mStreamVectorLock);
AudioAutoTimeoutLock _l(mLock);
if (format == NULL || channels == NULL || sampleRate == NULL || status == NULL)
{
ALOGE("%s(), NULL pointer!! format = %p, channels = %p, sampleRate = %p, status = %p",
__FUNCTION__, format, channels, sampleRate, status);
if (status != NULL) { *status = INVALID_OPERATION; }
return NULL;
}
// stream out flags
const uint32_t flags = (uint32_t)(*status);
// create stream out
AudioALSAStreamOut *pAudioALSAStreamOut = new AudioALSAStreamOut();
pAudioALSAStreamOut->set(devices, format, channels, sampleRate, status, flags);
if (*status != NO_ERROR)
{
ALOGE("-%s(), set fail, return NULL", __FUNCTION__);
delete pAudioALSAStreamOut;
pAudioALSAStreamOut = NULL;
return NULL;
}
// save stream out object in vector
pAudioALSAStreamOut->setIdentity(mStreamOutIndex);
mStreamOutVector.add(mStreamOutIndex, pAudioALSAStreamOut); // 加入到streammanager 容器中。
mStreamOutIndex++;
// setup Filter for ACF/HCF/AudEnh/VibSPK // TODO Check return status of pAudioALSAStreamOut->set
AudioMTKFilterManager *pAudioFilterManagerHandler = new AudioMTKFilterManager(*sampleRate, android_audio_legacy::AudioSystem::popCount(*channels), *format, pAudioALSAStreamOut->bufferSize());
mFilterManagerVector.add(mFilterManagerNumber, pAudioFilterManagerHandler);
mFilterManagerNumber++;
ALOGD("-%s(), out = %p, status = 0x%x, mStreamOutVector.size() = %d",
__FUNCTION__, pAudioALSAStreamOut, *status, mStreamOutVector.size());
return pAudioALSAStreamOut;
}
-》audioAStreamManager::openOutputStream
-》audioALSAStreamOut::open
-》audioALSAStreamOut::open()
-》playbackHandler = mStreamManager->createPlaybackHandler(&mStreamAttributeSource);
-> // open codec driver
mHardwareResourceManager->startOutputDevice(mStreamAttributeSource->output_devices, mStreamAttributeTarget.sample_rate);