前言

此篇是对《Android音频框架之一详解audioPolicy流程及HAL驱动加载》的延续，此系列博文是记录在Android7.1系统即以后版本实现
内录音功能。
当用户使用 AudioRecord 录音 API 时，

private void init() {int minBufferSize = AudioRecord.getMinBufferSize(kSampleRate, kChannelMode,kEncodeFormat);mRecord = new AudioRecord(MediaRecorder.AudioSource.REMOTE_SUBMIX,kSampleRate, kChannelMode, kEncodeFormat, minBufferSize * 2);  //>REMOTE_SUBMIXLog.d(TAG,"Create AudiRecord ...");}

Android系统的启动流程是什么呢？
本篇通过源码走读方式，来还原这个过程。

环境说明

源码:android7.1 版本，硬件平台 RK3288-box。

step 1:

用户在 app 中使用 new AudioRecord() API接口时，第一步调用 LOCAL_MODULE:= libandroid_runtime 模块注册的 JNI 接口,
AudioRecord注册的接口如下：

@ frameworks/base/core/jni/android_media_AudioRecord.cpp
static const JNINativeMethod gMethods[] = {// name,               signature,  funcPtr{"native_start",         "(II)I",    (void *)android_media_AudioRecord_start},{"native_stop",          "()V",    (void *)android_media_AudioRecord_stop},{"native_setup",         "(Ljava/lang/Object;Ljava/lang/Object;[IIIII[ILjava/lang/String;J)I",(void *)android_media_AudioRecord_setup},      //> 设置 AudioRecord 参数{"native_finalize",      "()V",    (void *)android_media_AudioRecord_finalize},{"native_release",       "()V",    (void *)android_media_AudioRecord_release},{"native_read_in_byte_array","([BIIZ)I",(void *)android_media_AudioRecord_readInArray<jbyteArray>},{"native_read_in_short_array","([SIIZ)I",(void *)android_media_AudioRecord_readInArray<jshortArray>},{"native_read_in_float_array","([FIIZ)I",(void *)android_media_AudioRecord_readInArray<jfloatArray>},{"native_read_in_direct_buffer","(Ljava/lang/Object;IZ)I",(void *)android_media_AudioRecord_readInDirectBuffer},{"native_get_buffer_size_in_frames","()I", (void *)android_media_AudioRecord_get_buffer_size_in_frames},{"native_set_marker_pos","(I)I",   (void *)android_media_AudioRecord_set_marker_pos},{"native_get_marker_pos","()I",    (void *)android_media_AudioRecord_get_marker_pos},{"native_set_pos_update_period","(I)I",   (void *)android_media_AudioRecord_set_pos_update_period},{"native_get_pos_update_period","()I",    (void *)android_media_AudioRecord_get_pos_update_period},{"native_get_min_buff_size","(III)I",   (void *)android_media_AudioRecord_get_min_buff_size},{"native_setInputDevice", "(I)Z", (void *)android_media_AudioRecord_setInputDevice},{"native_getRoutedDeviceId", "()I", (void *)android_media_AudioRecord_getRoutedDeviceId},{"native_enableDeviceCallback", "()V", (void *)android_media_AudioRecord_enableDeviceCallback},{"native_disableDeviceCallback", "()V",(void *)android_media_AudioRecord_disableDeviceCallback},{"native_get_timestamp", "(Landroid/media/AudioTimestamp;I)I",(void *)android_media_AudioRecord_get_timestamp},
};

此函数源码如下：

@ frameworks/base/core/jni/android_media_AudioRecord.cpp
static jint
android_media_AudioRecord_setup(JNIEnv *env, jobject thiz, jobject weak_this,jobject jaa, jintArray jSampleRate, jint channelMask, jint channelIndexMask,jint audioFormat, jint buffSizeInBytes, jintArray jSession, jstring opPackageName,jlong nativeRecordInJavaObj)
{jclass clazz = env->GetObjectClass(thiz);if (nativeRecordInJavaObj == 0) { //> create AudioRecords instance and set parameteraudio_format_t format = audioFormatToNative(audioFormat);lpRecorder = new AudioRecord(String16(opPackageNameStr.c_str()));     //> new AudioRecord() 见: step 2const status_t status = lpRecorder->set(paa->source,sampleRateInHertz,format,        // word length, PCMlocalChanMask,frameCount,recorderCallback,// callback_tlpCallbackData,// void* user0,             // notificationFrames,true,          // threadCanCallJavasessionId,AudioRecord::TRANSFER_DEFAULT,flags,-1, -1,        // default uid, pidpaa);} else {lpRecorder = (AudioRecord*)nativeRecordInJavaObj;lpCallbackData = new audiorecord_callback_cookie;lpCallbackData->audioRecord_class = (jclass)env->NewGlobalRef(clazz);// we use a weak reference so the AudioRecord object can be garbage collected.lpCallbackData->audioRecord_ref = env->NewGlobalRef(weak_this);lpCallbackData->busy = false;}setAudioRecord(env, thiz, lpRecorder);                   //> 调用 cpp 的 设置 audioRecord 接口函数env->SetLongField(thiz, javaAudioRecordFields.nativeCallbackCookie, (jlong)lpCallbackData);return (jint) AUDIO_JAVA_SUCCESS;
}

step 2:

下面的源码都是 C++ 的库源码内容。

@ frameworks/av/media/libmedia/AudioRecord.cpp
AudioRecord::AudioRecord(){mStatus = set(inputSource, sampleRate, format, channelMask, frameCount, cbf, user,notificationFrames, false /*threadCanCallJava*/, sessionId, transferType, flags,uid, pid, pAttributes);
}

setp 3:

设置 inputSource 参数,如果系统未建立 AudioRecordThread 线程，就创建此线程。

status_t AudioRecord::set(){if (cbf != NULL) {mAudioRecordThread = new AudioRecordThread(*this, threadCanCallJava);  // 创建线程mAudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO);// thread begins in paused state, and will not reference us until start()}status_t status = openRecord_l(0 /*epoch*/, mOpPackageName);               // 打开 record 见：step 4ALOGD("Debug_dump_info: %s,%d set() runStatus:%d ", __func__, __LINE__, status);if (status != NO_ERROR) {if (mAudioRecordThread != 0) {mAudioRecordThread->requestExit();   // see comment in AudioRecord.hmAudioRecordThread->requestExitAndWait();mAudioRecordThread.clear();}return status;}AudioSystem::acquireAudioSessionId(mSessionId, -1);return NO_ERROR;
}

setp 4:

获取 inputSource 源参数和采样率后，根据参数调用 audioFlinger->openRecord() 函数，创建audioRecord实例。

status_t AudioRecord::openRecord_l(const Modulo<uint32_t> &epoch, const String16& opPackageName)
{const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();status = AudioSystem::getInputForAttr(&mAttributes, &input,              //> 获取输入源 参数mSessionId,// FIXME compare to AudioTrackmClientPid,mClientUid,mSampleRate, mFormat, mChannelMask,mFlags, mSelectedDeviceId);//> error happen there status = AudioSystem::getSamplingRate(input, &afSampleRate);sp<IAudioRecord> record = audioFlinger->openRecord(input,mSampleRate,mFormat,mChannelMask,opPackageName,&temp,&flags,mClientPid,tid,mClientUid,&mSessionId,&notificationFrames,iMem,bufferMem,&status);// update proxymProxy = new AudioRecordClientProxy(cblk, buffers, mFrameCount, mFrameSize);mProxy->setEpoch(epoch);mProxy->setMinimum(mNotificationFramesAct);mDeathNotifier = new DeathNotifier(this);IInterface::asBinder(mAudioRecord)->linkToDeath(mDeathNotifier, this);return NO_ERROR;
}

setp 5:

函数 getInputForAttr() 根据用户APP中 new AudioRecord() 传递过来参数,检索 audio_policy_configuration.xml 文件；

@ frameworks/av/media/libmedia/AudioSystems.cppstatus_t AudioSystem::getInputForAttr(const audio_attributes_t *attr,audio_io_handle_t *input,audio_session_t session,pid_t pid,uid_t uid,uint32_t samplingRate,audio_format_t format,audio_channel_mask_t channelMask,audio_input_flags_t flags,audio_port_handle_t selectedDeviceId)
{const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();if (aps == 0) return NO_INIT;return aps->getInputForAttr(attr, input, session, pid, uid,samplingRate, format, channelMask, flags, selectedDeviceId);         //> 调用源码 见：step 6部分
}// establish binder interface to AudioPolicy service
const sp<IAudioPolicyService> AudioSystem::get_audio_policy_service()
{sp<IAudioPolicyService> ap;sp<AudioPolicyServiceClient> apc;{Mutex::Autolock _l(gLockAPS);if (gAudioPolicyService == 0) {sp<IServiceManager> sm = defaultServiceManager();sp<IBinder> binder;do {binder = sm->getService(String16("media.audio_policy"));if (binder != 0)break;ALOGW("AudioPolicyService not published, waiting...");usleep(500000); // 0.5 s} while (true);if (gAudioPolicyServiceClient == NULL) {gAudioPolicyServiceClient = new AudioPolicyServiceClient();}binder->linkToDeath(gAudioPolicyServiceClient);gAudioPolicyService = interface_cast<IAudioPolicyService>(binder);LOG_ALWAYS_FATAL_IF(gAudioPolicyService == 0);apc = gAudioPolicyServiceClient;}ap = gAudioPolicyService;}if (apc != 0) {ap->registerClient(apc);}return ap;
}

在此解释 AudioPolicyServiceClient 概念，此客户端是相对于 Android 系统启动的 AudioServcie 服务,此服务可以为多个用户 app 提供服务，
而 app 中操作 AudioRecord 时，需要通过 client 端来操作，服务端与客户端是通过 event队列实现，所有程序中可见事件发送相关代码。

step 6:

@ frameworks/av/services/audiopolicy/service/AudiopolicyInterfaceImpl.cppstatus_t AudioPolicyService::getInputForAttr(const audio_attributes_t *attr,audio_io_handle_t *input,audio_session_t session,pid_t pid,uid_t uid,uint32_t samplingRate,audio_format_t format,audio_channel_mask_t channelMask,audio_input_flags_t flags,audio_port_handle_t selectedDeviceId)
{if ((attr->source == AUDIO_SOURCE_HOTWORD) && !captureHotwordAllowed()) {return BAD_VALUE;}{Mutex::Autolock _l(mLock);// the audio_in_acoustics_t parameter is ignored by get_input()status = mAudioPolicyManager->getInputForAttr(attr, input, session, uid,        //> 实例的方法，见: step 7 部分samplingRate, format, channelMask,flags, selectedDeviceId,&inputType);                    audioPolicyEffects = mAudioPolicyEffects;if (status == NO_ERROR) {// enforce permission (if any) required for each type of inputswitch (inputType) {case AudioPolicyInterface::API_INPUT_LEGACY:break;case AudioPolicyInterface::API_INPUT_TELEPHONY_RX:// FIXME: use the same permission as for remote submix for now.case AudioPolicyInterface::API_INPUT_MIX_CAPTURE:                          //> 用户初始化 AudioSource.REMOTE_SUBMIX 类型if (!captureAudioOutputAllowed(pid, uid)) {                            //> 安卓系统未授权 REMOTE_SUBMIX 具有录音功能//ALOGE("getInputForAttr() permission denied: capture not allowed");//status = PERMISSION_DENIED;}break;case AudioPolicyInterface::API_INPUT_MIX_EXT_POLICY_REROUTE:if (!modifyAudioRoutingAllowed()) {ALOGE("getInputForAttr() permission denied: modify audio routing not allowed");status = PERMISSION_DENIED;}break;case AudioPolicyInterface::API_INPUT_INVALID:default:LOG_ALWAYS_FATAL("getInputForAttr() encountered an invalid input type %d",(int)inputType);}}if (status != NO_ERROR) {if (status == PERMISSION_DENIED) {mAudioPolicyManager->releaseInput(*input, session);}return status;}}}

通过在源码中打印日志信息，追踪至此授权问题，日志内容如下：

 230 E AudioPolicyIntefaceImpl: Debug_dump_info: getInputForAttr() permission denied: capture not allowed

失败的日志内容仅保留这一条了⁼, 因本人已经把此功能调试成功，留下来都是成功的日志，错误过程分析日志就丢了。很抱歉。
封闭上面的权限代码后， new AudioRecond() 过程就成功了。网上很多博文都有描述，就能够实现内录音功能。
笔者实践测试，可以录音，可还是录不到系统声音内容。
具体原因我还没有搞清楚，可能是平台间的 audio_policy_configuration.xml 配置内容问题。我的配置内容如下:


<?xml version="1.0" encoding="UTF-8" standalone="yes"?><audioPolicyConfiguration version="1.0" xmlns:xi="http://www.w3.org/2001/XInclude"><!-- version section contains a “version” tag in the form “major.minor” e.g version=”1.0” --><!-- Global configuration Decalaration --><globalConfiguration speaker_drc_enabled="true"/><modules><!-- Primary Audio HAL --><module name="primary" halVersion="2.5"><attachedDevices><item>Speaker</item><item>Remote Submix Out</item><item>Built-In Mic</item><item>Remote Submix In</item></attachedDevices><defaultOutputDevice>Speaker</defaultOutputDevice><!-- “mixPorts”: listing all output and input streams exposed by the audio HAL --><mixPorts><mixPort name="primary output" role="source" flags="AUDIO_OUTPUT_FLAG_PRIMARY"><profile name="" format="AUDIO_FORMAT_PCM_16_BIT"samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/></mixPort><mixPort name="primary input" role="sink"><profile name="" format="AUDIO_FORMAT_PCM_16_BIT"samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"channelMasks="AUDIO_CHANNEL_IN_STEREO"/></mixPort><!-- r_submix --><mixPort name="r_submix output" role="source"><profile name="" format="AUDIO_FORMAT_PCM_16_BIT"samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/></mixPort><mixPort name="r_submix input" role="sink"><profile name="" format="AUDIO_FORMAT_PCM_16_BIT"samplingRates="44100,48000" channelMasks="AUDIO_CHANNEL_IN_STEREO"/></mixPort></mixPorts><!-- “devicePorts”: a list of device descriptors for all input and output devices accessible via this module. --><devicePorts><!-- Output devices declaration, i.e. Sink DEVICE PORT --><devicePort tagName="Speaker" role="sink" type="AUDIO_DEVICE_OUT_SPEAKER" address=""><profile name="" format="AUDIO_FORMAT_PCM_16_BIT"samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/><gains><gain name="gain_1" mode="AUDIO_GAIN_MODE_JOINT"minValueMB="-8400"maxValueMB="4000"defaultValueMB="0"stepValueMB="100"/></gains></devicePort><devicePort tagName="Built-In Mic" type="AUDIO_DEVICE_IN_BUILTIN_MIC" role="source"><profile name="" format="AUDIO_FORMAT_PCM_16_BIT"samplingRates="44100,48000"channelMasks="AUDIO_CHANNEL_IN_STEREO"/></devicePort><!-- add remote deviceport by ljb --><devicePort tagName="Remote Submix Out" role="sink" type="AUDIO_DEVICE_OUT_REMOTE_SUBMIX"><profile name="" format="AUDIO_FORMAT_PCM_16_BIT"samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/></devicePort><devicePort tagName="Remote Submix In" type="AUDIO_DEVICE_IN_REMOTE_SUBMIX" role="source"><profile name="" format="AUDIO_FORMAT_PCM_16_BIT"samplingRates="44100,48000" channelMasks="AUDIO_CHANNEL_IN_STEREO"/></devicePort></devicePorts><!-- route declaration, i.e. list all available sources for a given sink --><routes><route type="mix" sink="Speaker"sources="primary output"/>          <route type="mix" sink="primary input"sources="Built-In Mic"/><!-- r_submix route --><route type="mix" sink="Remote Submix Out"sources="r_submix output"/><route type="mix" sink="r_submix input"sources="Remote Submix In"/></routes></module></modules><!-- End of Modules section --><!-- Volume section --><xi:include href="audio_policy_volumes.xml"/><xi:include href="default_volume_tables.xml"/><!-- End of Volume section --></audioPolicyConfiguration>

如有哪位大神指导具体差异，请留言指点，在此先谢谢了。
我是基于这个配置文件测试的，录制不到系统声音内容，下一篇我在《Android音频框架之三用户录音启动流程源码走读 startRecord 》
中，把如何修改 TinyAlsaHAL 的源码，来实现录制安卓系统声音具体方法。

step 7

接下来源码是根据用户参数，具体获取 inputSource 参数过程，我们简单走读一下源码。

@ frameworks/av/services/audiopilicy/managerdefault/AudioPolicyManager.cpp
status_t AudioPolicyManager::getInputForAttr(const audio_attributes_t *attr,audio_io_handle_t *input,audio_session_t session,uid_t uid,uint32_t samplingRate,audio_format_t format,audio_channel_mask_t channelMask,audio_input_flags_t flags,audio_port_handle_t selectedDeviceId,input_type_t *inputType)
{ALOGV("getInputForAttr() source %d, samplingRate %d, format %d, channelMask %x,""session %d, flags %#x",attr->source, samplingRate, format, channelMask, session, flags);*input = AUDIO_IO_HANDLE_NONE;*inputType = API_INPUT_INVALID;audio_devices_t device;// handle legacy remote submix case where the address was not always specifiedString8 address = String8("");audio_source_t inputSource = attr->source;audio_source_t halInputSource;AudioMix *policyMix = NULL;if (inputSource == AUDIO_SOURCE_DEFAULT) {inputSource = AUDIO_SOURCE_MIC;}halInputSource = inputSource;// Explicit routing?sp<DeviceDescriptor> deviceDesc;for (size_t i = 0; i < mAvailableInputDevices.size(); i++) {if (mAvailableInputDevices[i]->getId() == selectedDeviceId) {deviceDesc = mAvailableInputDevices[i];break;}}mInputRoutes.addRoute(session, SessionRoute::STREAM_TYPE_NA, inputSource, deviceDesc, uid);if (inputSource == AUDIO_SOURCE_REMOTE_SUBMIX &&strncmp(attr->tags, "addr=", strlen("addr=")) == 0) {status_t ret = mPolicyMixes.getInputMixForAttr(*attr, &policyMix);if (ret != NO_ERROR) {return ret;}*inputType = API_INPUT_MIX_EXT_POLICY_REROUTE;device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;address = String8(attr->tags + strlen("addr="));} else {device = getDeviceAndMixForInputSource(inputSource, &policyMix);                            //>  根据输入源获取 Mixer 信息 见: step 7.1if (device == AUDIO_DEVICE_NONE) {ALOGW("getInputForAttr() could not find device for source %d", inputSource);return BAD_VALUE;}if (policyMix != NULL) {address = policyMix->mDeviceAddress;if (policyMix->mMixType == MIX_TYPE_RECORDERS) {// there is an external policy, but this input is attached to a mix of recorders,// meaning it receives audio injected into the framework, so the recorder doesn't// know about it and is therefore considered "legacy"*inputType = API_INPUT_LEGACY;} else {// recording a mix of players defined by an external policy, we're rerouting for// an external policy*inputType = API_INPUT_MIX_EXT_POLICY_REROUTE;}} else if (audio_is_remote_submix_device(device)) {address = String8("0");*inputType = API_INPUT_MIX_CAPTURE;} else if (device == AUDIO_DEVICE_IN_TELEPHONY_RX) {*inputType = API_INPUT_TELEPHONY_RX;} else {*inputType = API_INPUT_LEGACY;}}*input = getInputForDevice(device, address, session, uid, inputSource,samplingRate, format, channelMask, flags,policyMix);                                    //> 根据输入源获取设备信息: step 7.2if (*input == AUDIO_IO_HANDLE_NONE) {mInputRoutes.removeRoute(session);return INVALID_OPERATION;}ALOGV("getInputForAttr() returns input type = %d", *inputType);return NO_ERROR;
}

step 7.1

因配置文件中没有此部分内容，程序未检索出可用信息。

audio_devices_t AudioPolicyManager::getDeviceAndMixForInputSource(audio_source_t inputSource,AudioMix **policyMix)
{ALOGD("Debug_dump_info: %s,%d ", __func__, __LINE__);audio_devices_t availableDeviceTypes = mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN;audio_devices_t selectedDeviceFromMix =mPolicyMixes.getDeviceAndMixForInputSource(inputSource, availableDeviceTypes, policyMix);ALOGD("Debug_dump_info: %s,%d ", __func__, __LINE__);if (selectedDeviceFromMix != AUDIO_DEVICE_NONE) {return selectedDeviceFromMix;}return getDeviceForInputSource(inputSource);
}audio_devices_t AudioPolicyManager::getDeviceForInputSource(audio_source_t inputSource)
{ALOGD("Debug_dump_info: %s , %d mInputRoutes.size:%d ", __func__, __LINE__,mInputRoutes.size());size_t routeIndex = 0;for (/*size_t routeIndex = 0*/; routeIndex < mInputRoutes.size(); routeIndex++) {sp<SessionRoute> route = mInputRoutes.valueAt(routeIndex);if (inputSource == route->mSource && route->isActive()) {return route->mDeviceDescriptor->type();}}ALOGD("Debug_dump_info: %s , %d routeIndex:%d ", __func__, __LINE__, routeIndex);return mEngine->getDeviceForInputSource(inputSource);                              //> 根据输入源检查设备权限 见: step 7.3
}

step 7.2:

根据输入源为 8 检索出对应的声卡和设备编号内容。

audio_io_handle_t AudioPolicyManager::getInputForDevice(audio_devices_t device,String8 address,audio_session_t session,uid_t uid,audio_source_t inputSource,uint32_t samplingRate,audio_format_t format,audio_channel_mask_t channelMask,audio_input_flags_t flags,AudioMix *policyMix)
{audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;audio_source_t halInputSource = inputSource;bool isSoundTrigger = false;ALOGD("Debug_dump_info: %s : %d  ", __func__, __LINE__);// find a compatible input profile (not necessarily identical in parameters)sp<IOProfile> profile;// samplingRate and flags may be updated by getInputProfileuint32_t profileSamplingRate = (samplingRate == 0) ? SAMPLE_RATE_HZ_DEFAULT : samplingRate;audio_format_t profileFormat = format;audio_channel_mask_t profileChannelMask = channelMask;audio_input_flags_t profileFlags = flags;for (;;) {profile = getInputProfile(device, address,profileSamplingRate, profileFormat, profileChannelMask,profileFlags);if (profile != 0) {break; // success} else if (profileFlags & AUDIO_INPUT_FLAG_RAW) {profileFlags = (audio_input_flags_t) (profileFlags & ~AUDIO_INPUT_FLAG_RAW); // retry} else if (profileFlags != AUDIO_INPUT_FLAG_NONE) {profileFlags = AUDIO_INPUT_FLAG_NONE; // retry} else { // failALOGW("Debug_dump_info: getInputForDevice() could not find profile for device 0x%X,""samplingRate %u, format %#x, channelMask 0x%X, flags %#x",device, samplingRate, format, channelMask, flags);return input;}}// Pick input sampling rate if not specified by clientif (samplingRate == 0) {samplingRate = profileSamplingRate;}if (profile->getModuleHandle() == 0) {ALOGE("Debug_dump_info: getInputForAttr(): HW module %s not opened", profile->getModuleName());return input;}sp<AudioSession> audioSession = new AudioSession(session,inputSource,format,samplingRate,channelMask,flags,uid,isSoundTrigger,policyMix, mpClientInterface);audio_config_t config = AUDIO_CONFIG_INITIALIZER;config.sample_rate = profileSamplingRate;config.channel_mask = profileChannelMask;config.format = profileFormat;ALOGD("Debug_dump_info: %s : %d  ", __func__, __LINE__);status_t status = mpClientInterface->openInput(profile->getModuleHandle(),&input,&config,&device,address,halInputSource,profileFlags);ALOGD("Debug_dump_info: %s : %d  ", __func__, __LINE__);// only accept input with the exact requested set of parametersif (status != NO_ERROR || input == AUDIO_IO_HANDLE_NONE ||(profileSamplingRate != config.sample_rate) ||!audio_formats_match(profileFormat, config.format) ||(profileChannelMask != config.channel_mask)) {ALOGW("Debug_dump_info: getInputForAttr() failed opening input: samplingRate %d"", format %d, channelMask %x",samplingRate, format, channelMask);if (input != AUDIO_IO_HANDLE_NONE) {mpClientInterface->closeInput(input);}return AUDIO_IO_HANDLE_NONE;}ALOGD("Debug_dump_info: %s : %d  ", __func__, __LINE__);sp<AudioInputDescriptor> inputDesc = new AudioInputDescriptor(profile);inputDesc->mSamplingRate = profileSamplingRate;inputDesc->mFormat = profileFormat;inputDesc->mChannelMask = profileChannelMask;inputDesc->mDevice = device;inputDesc->mPolicyMix = policyMix;inputDesc->addAudioSession(session, audioSession);addInput(input, inputDesc);mpClientInterface->onAudioPortListUpdate();ALOGD("Debug_dump_info: %s : %d  ", __func__, __LINE__);return input;
}

step 7.3

@ frameworks/av/services/audiopilicy/engineconfigable/src/Engine.hclass ManagerInterfaceImpl : public AudioPolicyManagerInterface{public:ManagerInterfaceImpl(Engine *policyEngine): mPolicyEngine(policyEngine) {}virtual android::status_t initCheck(){return mPolicyEngine->initCheck();}virtual void setObserver(AudioPolicyManagerObserver *observer){mPolicyEngine->setObserver(observer);}virtual audio_devices_t getDeviceForInputSource(audio_source_t inputSource) const     //> call function{return mPolicyEngine->getPropertyForKey<audio_devices_t, audio_source_t>(inputSource);}virtual audio_devices_t getDeviceForStrategy(routing_strategy stategy) const;virtual routing_strategy getStrategyForStream(audio_stream_type_t stream){return mPolicyEngine->getPropertyForKey<routing_strategy, audio_stream_type_t>(stream);}private:/* Copy facilities are put private to disable copy. */Engine(const Engine &object);Engine &operator=(const Engine &object);template <typename Property, typename Key>Property getPropertyForKey(Key key) const;template <typename Property, typename Key>bool setPropertyForKey(const Property &property, const Key &key);}@ frameworks/av/services/audiopilicy/engineconfigable/src/Engine.cpp
template <typename Property, typename Key>
Property Engine::getPropertyForKey(Key key) const               //> permission capture allowed
{Element<Key> *element = getFromCollection<Key>(key);if (element == NULL) {ALOGE("%s: Element not found within collection", __FUNCTION__);return static_cast<Property>(0);}return element->template get<Property>();
}

至此，用户新建 AudioRecord 实例时，安卓系统流程就如代码走读中描述，我们简单总结如下：
1> . 安卓系统启动时、加载 libandroid_runtime 库系统，也是安卓系统核心框架；
2> . 系统启动文件 init.rc 中，加载 AudioRecordService 服务，系统后台就运行着 AudioRecord 服务；
3> . 用户新建 AudioRecord、调用系统 jni 接口函数，启动安卓系统库中创建 AudioRecord过程;
4> . 系统根据音频源类型 REMOTE_SUBMIX、在配置文件生成对象中，获取新建设备相关属性；
获取到设备相关参数后、并没有在此部分新建设备，而是在 StartAudioRecord 过程中，创建的设备，
也就是笔者接下来分享内容中。

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Android音频框架之二用户录音启动流程源码走读

前言

环境说明

step 1:

step 2:

setp 3:

setp 4:

setp 5:

step 6:

step 7

step 7.1

step 7.2:

step 7.3

Android音频框架之二用户录音启动流程源码走读相关推荐

最新文章

热门文章

Android音频框架之二 用户录音启动流程源码走读

前言

环境说明

step 1:

step 2:

setp 3:

setp 4:

setp 5:

step 6:

step 7

step 7.1

step 7.2:

step 7.3

Android音频框架之二 用户录音启动流程源码走读相关推荐

最新文章

热门文章

Android音频框架之二用户录音启动流程源码走读

Android音频框架之二用户录音启动流程源码走读相关推荐