? ? ? ? AudioRecord create相关代码流程、重点关键接口、参数解析以及相关log share分析比对。
1.frameworks/base/media/java/android/media/AudioRecord.java
参数解析：
audioSource: MediaRecorder.AudioSource.MIC，详细定义需check MediaRecorder.AudioSource;
sampleRateInHz：默认采样率，单位Hz，44100Hz是唯一能保证在所有设备上工作的采样率；
channelConfig： 描述音频通道设置，AudioFormat.CHANNEL_CONFIGURATION_MONO保证能在所有设备上工作；
audioFormat：音频数据保证支持此格式，这里设置为AudioFormat.ENCODING_16BIT；
bufferSizeInBytes：在录制过程中,音频数据写入缓冲区的字节数,getMinVufferSize()获取到的值。
?主要的解析过程在native_setup函数中处理，需要注意的是区别于不同audio调用的session Id.
@SystemApi
public AudioRecord(AudioAttributes attributes, AudioFormat format, int bufferSizeInBytes,
int sessionId) throws IllegalArgumentException {
mRecordingState = RECORDSTATE_STOPPED;
...
///*add by shengjie for parse AudioRecorder
//1.REMOTE_SUBMIX这个音源是咋么处理的，没有跟，之后有项目需要的话可以跟一下
//*/
// is this AudioRecord using REMOTE_SUBMIX at full volume?
if (attributes.getCapturePreset() == MediaRecorder.AudioSource.REMOTE_SUBMIX) {
final AudioAttributes.Builder filteredAttr = new AudioAttributes.Builder();
final Iterator<String> tagsIter = attributes.getTags().iterator();
while (tagsIter.hasNext()) {
final String tag = tagsIter.next();
if (tag.equalsIgnoreCase(SUBMIX_FIXED_VOLUME)) {
mIsSubmixFullVolume = true;
Log.v(TAG, "Will record from REMOTE_SUBMIX at full fixed volume");
} else { // SUBMIX_FIXED_VOLUME: is not to be propagated to the native layers
filteredAttr.addTag(tag);
}
}
filteredAttr.setInternalCapturePreset(attributes.getCapturePreset());
mAudioAttributes = filteredAttr.build();
} else {
mAudioAttributes = attributes;
}
...
///*add by shengjie for parse AudioRecord
//2.audioParamCheck对参数进行合法性检查，赋值mRecordSource、mSampleRate、mAudioFormat
//*/
audioParamCheck(attributes.getCapturePreset(), rate, encoding);
///*add by shengjie for parse AudioRecord
//3.获取声道数以及声道掩码，单声道掩码为0x10，双声道掩码为0x0c；
//*/
if ((format.getPropertySetMask()
& AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_INDEX_MASK) != 0) {
mChannelIndexMask = format.getChannelIndexMask();
mChannelCount = format.getChannelCount();
}
if ((format.getPropertySetMask()
& AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_MASK) != 0) {
mChannelMask = getChannelMaskFromLegacyConfig(format.getChannelMask(), false);
mChannelCount = format.getChannelCount();
} else if (mChannelIndexMask == 0) {
mChannelMask = getChannelMaskFromLegacyConfig(AudioFormat.CHANNEL_IN_DEFAULT, false);
mChannelCount = AudioFormat.channelCountFromInChannelMask(mChannelMask);
}
audioBuffSizeCheck(bufferSizeInBytes);
int[] sampleRate = new int[] {mSampleRate};
///*add by shengjie for parse AudioRecord
//一个Session就是一个会话，每个会话都有一个独一无二的Id来标识。该Id的最终管理在AudioFlinger中。
//一个会话可以被多个AudioTrack对象和MediaPlayer共用。
//共用一个Session的AudioTrack和MediaPlayer共享相同的AudioEffect（音效）。
//*/
int[] session = new int[1];
session[0] = sessionId;
//TODO: update native initialization when information about hardware init failure
// due to capture device already open is available.
int initResult = native_setup( new WeakReference<AudioRecord>(this),
mAudioAttributes, sampleRate, mChannelMask, mChannelIndexMask,
mAudioFormat, mNativeBufferSizeInBytes,
session, getCurrentOpPackageName(), 0 /*nativeRecordInJavaObj*/);
if (initResult != SUCCESS) {
loge("Error code "+initResult+" when initializing native AudioRecord object.");
return; // with mState == STATE_UNINITIALIZED
}
mSampleRate = sampleRate[0];
mSessionId = session[0];
mState = STATE_INITIALIZED;
}?
2.frameworks/base/core/jni/android_media_AudioRecord.cpp
主要工作：调用AudioRecord.cpp的set函数接口，并设置
? ? audiorecord_callback_cookie *lpCallbackData回调用于传输音频数据buffer。
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)
{
//ALOGV(">> Entering android_media_AudioRecord_setup");
//ALOGV("sampleRate=%d, audioFormat=%d, channel mask=%x, buffSizeInBytes=%d "
// "nativeRecordInJavaObj=0x%llX",
// sampleRateInHertz, audioFormat, channelMask, buffSizeInBytes, nativeRecordInJavaObj);
...
sp<AudioRecord> lpRecorder = 0;
audiorecord_callback_cookie *lpCallbackData = NULL;
// if we pass in an existing *Native* AudioRecord, we don't need to create/initialize one.
if (nativeRecordInJavaObj == 0) {
if (jaa == 0) {
ALOGE("Error creating AudioRecord: invalid audio attributes");
return (jint) AUDIO_JAVA_ERROR;
}
if (jSampleRate == 0) {
ALOGE("Error creating AudioRecord: invalid sample rates");
return (jint) AUDIO_JAVA_ERROR;
}
///*add by shengjie for parse AudioRecord setup
//1.got sample Rate from new AlsaRecord(44100hz/16khz)
jint elements[1];
env->GetIntArrayRegion(jSampleRate, 0, 1, elements);
int sampleRateInHertz = elements[0];
//*/
// channel index mask takes priority over channel position masks.
if (channelIndexMask) {
// Java channel index masks need the representation bits set.
localChanMask = audio_channel_mask_from_representation_and_bits(
AUDIO_CHANNEL_REPRESENTATION_INDEX,
channelIndexMask);
}
// Java channel position masks map directly to the native definition
if (!audio_is_input_channel(localChanMask)) {
ALOGE("Error creating AudioRecord: channel mask %#x is not valid.", localChanMask);
return (jint) AUDIORECORD_ERROR_SETUP_INVALIDCHANNELMASK;
}
///*add by shengjie for parse AudioRecord setup
//2.got channel count from new AlsaRecord(1/2/4 ch)
uint32_t channelCount = audio_channel_count_from_in_mask(localChanMask);
//*/
// compare the format against the Java constants
///*add by shengjie for parse AudioRecord setup
//3.got audio format from new AlsaRecord(16bit/32bit)
audio_format_t format = audioFormatToNative(audioFormat);
if (format == AUDIO_FORMAT_INVALID) {
ALOGE("Error creating AudioRecord: unsupported audio format %d.", audioFormat);
return (jint) AUDIORECORD_ERROR_SETUP_INVALIDFORMAT;
}
//*/
///*add by shengjie for parse AudioRecord setup
size_t bytesPerSample = audio_bytes_per_sample(format);
if (buffSizeInBytes == 0) {
ALOGE("Error creating AudioRecord: frameCount is 0.");
return (jint) AUDIORECORD_ERROR_SETUP_ZEROFRAMECOUNT;
}
//4.1 每个采样帧大小:所有通道数 * 每一帧所占字节数
size_t frameSize = channelCount * bytesPerSample;
//4.2 采样帧数量：最小缓冲区大小 / 每个采样帧大小
size_t frameCount = buffSizeInBytes / frameSize;
//*/
ScopedUtfChars opPackageNameStr(env, opPackageName);
// create an uninitialized AudioRecord object
lpRecorder = new AudioRecord(String16(opPackageNameStr.c_str()));
// read the AudioAttributes values
auto paa = JNIAudioAttributeHelper::makeUnique();
jint jStatus = JNIAudioAttributeHelper::nativeFromJava(env, jaa, paa.get());
if (jStatus != (jint)AUDIO_JAVA_SUCCESS) {
return jStatus;
}
ALOGV("AudioRecord_setup for source=%d tags=%s flags=%08x", paa->source, paa->tags, paa->flags);
audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE;
if (paa->flags & AUDIO_FLAG_HW_HOTWORD) {
flags = AUDIO_INPUT_FLAG_HW_HOTWORD;
}
// create the callback information:
// this data will be passed with every AudioRecord callback
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.
///*add by shengjie for parse AudioRecord setup
//5.AudioRecord.java的指针绑定到lpCallbackData回调数据中，这样就能把数据通过回调的方式通知到上层
lpCallbackData->audioRecord_ref = env->NewGlobalRef(weak_this);
lpCallbackData->busy = false;
//*/
///*add by shengjie for AudioRecord setup
//6.调用AudioRecord.cpp的set函数，flags，他的类型为audio_input_flags_t，定义在audio-base.h中，作为音频输入的标志，此处为AUDIO_INPUT_FLAG_NONE，没有做特殊设定
const status_t status = lpRecorder->set(paa->source,
sampleRateInHertz,
format, // word length, PCM
localChanMask,
frameCount,
recorderCallback,// callback_t
lpCallbackData,// void* user
0, // notificationFrames,
true, // threadCanCallJava
sessionId,
AudioRecord::TRANSFER_DEFAULT,
flags,
-1, -1, // default uid, pid
paa.get());
//*/
...
// Set caller name so it can be logged in destructor.
// MediaMetricsConstants.h: AMEDIAMETRICS_PROP_CALLERNAME_VALUE_JAVA
lpRecorder->setCallerName("java");
} else { // end if nativeRecordInJavaObj == 0)
lpRecorder = (AudioRecord*)nativeRecordInJavaObj;
// TODO: We need to find out which members of the Java AudioRecord might need to be
...
// create the callback information:
// this data will be passed with every AudioRecord callback
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;
}
nSession = (jint *) env->GetPrimitiveArrayCritical(jSession, NULL);
if (nSession == NULL) {
ALOGE("Error creating AudioRecord: Error retrieving session id pointer");
goto native_init_failure;
}
// read the audio session ID back from AudioRecord in case a new session was created during set()
nSession[0] = lpRecorder->getSessionId();
env->ReleasePrimitiveArrayCritical(jSession, nSession, 0);
nSession = NULL;
...
///*add by shengjie for parse AudioRecord setup
//7.把lpRecorder对象以及lpCallbackData回调保存到javaAudioRecordFields的相应字段中
// save our newly created C++ AudioRecord in the "nativeRecorderInJavaObj" field
// of the Java object
setAudioRecord(env, thiz, lpRecorder);
// save our newly created callback information in the "nativeCallbackCookie" field
// of the Java object (in mNativeCallbackCookie) so we can free the memory in finalize()
env->SetLongField(thiz, javaAudioRecordFields.nativeCallbackCookie, (jlong)lpCallbackData);
//*/
return (jint) AUDIO_JAVA_SUCCESS;
...///*Error Handler Process*///...
}system/media/audio/include/system/audio-base.h?
typedef enum {
AUDIO_INPUT_FLAG_NONE = 0x0,
AUDIO_INPUT_FLAG_FAST = 0x1,
AUDIO_INPUT_FLAG_HW_HOTWORD = 0x2,
AUDIO_INPUT_FLAG_RAW = 0x4,
AUDIO_INPUT_FLAG_SYNC = 0x8,
AUDIO_INPUT_FLAG_MMAP_NOIRQ = 0x10,
AUDIO_INPUT_FLAG_VOIP_TX = 0x20,
AUDIO_INPUT_FLAG_HW_AV_SYNC = 0x40,
AUDIO_INPUT_FLAG_DIRECT = 0x80,
} audio_input_flags_t;
?
3.frameworks/av/media/libaudioclient/AudioRecord.cpp
解析相关的音频参数，主要是创建AudioRecordThread和createRecord_l
status_t AudioRecord::set(
audio_source_t inputSource,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
callback_t cbf,
void* user,
uint32_t notificationFrames,
bool threadCanCallJava,
audio_session_t sessionId,
transfer_type transferType,
audio_input_flags_t flags,
uid_t uid,
pid_t pid,
const audio_attributes_t* pAttributes,
audio_port_handle_t selectedDeviceId,
audio_microphone_direction_t selectedMicDirection,
float microphoneFieldDimension)
{
status_t status = NO_ERROR;
uint32_t channelCount;
pid_t callingPid;
pid_t myPid;
// Note mPortId is not valid until the track is created, so omit mPortId in ALOG for set.
ALOGV("%s(): inputSource %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
"notificationFrames %u, sessionId %d, transferType %d, flags %#x, opPackageName %s "
"uid %d, pid %d",
__func__,
inputSource, sampleRate, format, channelMask, frameCount, notificationFrames,
sessionId, transferType, flags, String8(mOpPackageName).string(), uid, pid);
mTracker.reset(new RecordingActivityTracker());
mSelectedDeviceId = selectedDeviceId;
mSelectedMicDirection = selectedMicDirection;
mSelectedMicFieldDimension = microphoneFieldDimension;
...
///*add by shengjie for parse AudioRecord set
//1.get trabsferType TRANSFER_SYNC/TRANSFER_CALLBACK
mTransfer = transferType;
//*/
...
if (pAttributes == NULL) {
mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
mAttributes.source = inputSource;
if (inputSource == AUDIO_SOURCE_VOICE_COMMUNICATION
|| inputSource == AUDIO_SOURCE_CAMCORDER) {
mAttributes.flags |= AUDIO_FLAG_CAPTURE_PRIVATE;
}
} else {
// stream type shouldn't be looked at, this track has audio attributes
memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t));
ALOGV("%s(): Building AudioRecord with attributes: source=%d flags=0x%x tags=[%s]",
__func__, mAttributes.source, mAttributes.flags, mAttributes.tags);
}
mSampleRate = sampleRate;
// these below should probably come from the audioFlinger too...
if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
}
// validate parameters
// AudioFlinger capture only supports linear PCM
if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) {
ALOGE("%s(): Format %#x is not linear pcm", __func__, format);
status = BAD_VALUE;
goto exit;
}
mFormat = format;
if (!audio_is_input_channel(channelMask)) {
ALOGE("%s(): Invalid channel mask %#x", __func__, channelMask);
status = BAD_VALUE;
goto exit;
}
mChannelMask = channelMask;
channelCount = audio_channel_count_from_in_mask(channelMask);
mChannelCount = channelCount;
if (audio_is_linear_pcm(format)) {
mFrameSize = channelCount * audio_bytes_per_sample(format);
} else {
mFrameSize = sizeof(uint8_t);
}
// mFrameCount is initialized in createRecord_l
mReqFrameCount = frameCount;
mNotificationFramesReq = notificationFrames;
// mNotificationFramesAct is initialized in createRecord_l
mSessionId = sessionId;
ALOGV("%s(): mSessionId %d", __func__, mSessionId);
callingPid = IPCThreadState::self()->getCallingPid();
myPid = getpid();
if (uid == AUDIO_UID_INVALID || (callingPid != myPid)) {
mClientUid = IPCThreadState::self()->getCallingUid();
} else {
mClientUid = uid;
}
if (pid == -1 || (callingPid != myPid)) {
mClientPid = callingPid;
} else {
mClientPid = pid;
}
mOrigFlags = mFlags = flags;
mCbf = cbf;
///*add by shengjie for parse AudioRecord set
//2.new AudioRecordThread to handle releated process
if (cbf != NULL) {
mAudioRecordThread = new AudioRecordThread(*this);
mAudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO);
// thread begins in paused state, and will not reference us until start()
}
//*/
// create the IAudioRecord
///*add by shengjie for parse AudioRecord set
//3.use createRecord_l to invork audio related devices/input parameters
{
AutoMutex lock(mLock);
status = createRecord_l(0 /*epoch*/, mOpPackageName);
}
//*/
ALOGV("%s(%d): status %d", __func__, mPortId, status);
if (status != NO_ERROR) {
if (mAudioRecordThread != 0) {
mAudioRecordThread->requestExit(); // see comment in AudioRecord.h
mAudioRecordThread->requestExitAndWait();
mAudioRecordThread.clear();
}
goto exit;
}
mUserData = user;
// TODO: add audio hardware input latency here
mLatency = (1000LL * mFrameCount) / mSampleRate;
mMarkerPosition = 0;
mMarkerReached = false;
mNewPosition = 0;
mUpdatePeriod = 0;
AudioSystem::acquireAudioSessionId(mSessionId, mClientPid, mClientUid);
mSequence = 1;
mObservedSequence = mSequence;
mInOverrun = false;
mFramesRead = 0;
mFramesReadServerOffset = 0;
///*error handler process*///
}?createRecord_l方法具体实现如下：
重要接口:AudioFliger.cpp createRecord_l后续会进行一系列参数的解析与分发；
// must be called with mLock held
status_t AudioRecord::createRecord_l(const Modulo<uint32_t> &epoch, const String16& opPackageName)
{
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
IAudioFlinger::CreateRecordInput input;
IAudioFlinger::CreateRecordOutput output;
sp<media::IAudioRecord> record;
...
///*loding input related parameters*/
///*add by shengjie for parse AudioRecord createRecord_l
//1.IAudioFliger.cpp通过Binder进程间通信调用AudioFliger.cpp的createRecord,承担包括音频input device构建getInputForAttr、createRecordTrack_l、getOrphanEffectChain_l等所有功能。
record = audioFlinger->createRecord(input,
output,
&status);
//*/
...
// Starting address of buffers in shared memory.
// The buffers are either immediately after the control block,
// or in a separate area at discretion of server.
void *buffers;
if (output.buffers == 0) {
buffers = cblk + 1;
} else {
// TODO: Using unsecurePointer() has some associated security pitfalls
// (see declaration for details).
// Either document why it is safe in this case or address the
// issue (e.g. by copying).
buffers = output.buffers->unsecurePointer();
if (buffers == NULL) {
ALOGE("%s(%d): Could not get buffer pointer", __func__, mPortId);
status = NO_INIT;
goto exit;
}
}
// invariant that mAudioRecord != 0 is true only after set() returns successfully
if (mAudioRecord != 0) {
IInterface::asBinder(mAudioRecord)->unlinkToDeath(mDeathNotifier, this);
mDeathNotifier.clear();
}
mAudioRecord = record;
mCblkMemory = output.cblk;
mBufferMemory = output.buffers;
IPCThreadState::self()->flushCommands();
mCblk = cblk;
...
// update proxy
mProxy = new AudioRecordClientProxy(cblk, buffers, mFrameCount, mFrameSize);
mProxy->setEpoch(epoch);
mProxy->setMinimum(mNotificationFramesAct);
...
}?
4.frameworks/av/media/libaudioclient/IAudioFlinger.cpp
virtual sp<media::IAudioRecord> createRecord(const CreateRecordInput& input,
CreateRecordOutput& output,
status_t *status)
{
Parcel data, reply;
sp<media::IAudioRecord> record;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
if (status == nullptr) {
return record;
}
input.writeToParcel(&data);
///*add by shengjie for IAudioFlinger createRecord
//通过Binder进程间通信拿到share buffer
status_t lStatus = remote()->transact(CREATE_RECORD, data, &reply);
if (lStatus != NO_ERROR) {
ALOGE("createRecord transaction error %d", lStatus);
*status = DEAD_OBJECT;
return record;
}
//*/
*status = reply.readInt32();
if (*status != NO_ERROR) {
ALOGE("createRecord returned error %d", *status);
return record;
}
record = interface_cast<media::IAudioRecord>(reply.readStrongBinder());
if (record == 0) {
ALOGE("createRecord returned a NULL IAudioRecord with status OK");
*status = DEAD_OBJECT;
return record;
}
output.readFromParcel(&reply);
return record;
}?
5.frameworks/av/services/audioflinger/AudioFlinger.cpp
getInputForAttr:设置audio input相关信息对应的audio device
checkRecordThread_l:创建recordThread去做recording相关的配置
getOrphanEffectChain_l:下发对应的不同增益
sp<media::IAudioRecord> AudioFlinger::createRecord(const CreateRecordInput& input,
CreateRecordOutput& output,
status_t *status)
{
sp<RecordThread::RecordTrack> recordTrack;
sp<RecordHandle> recordHandle;
sp<Client> client;
status_t lStatus;
audio_session_t sessionId = input.sessionId;
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
...
///*add by shengjie for parse AudioFlinger createRecord
//parse input related info and check all audio devices info
lStatus = AudioSystem::getInputForAttr(&input.attr, &output.inputId,
input.riid,
sessionId,
// FIXME compare to AudioTrack
clientPid,
clientUid,
input.opPackageName,
&input.config,
output.flags, &output.selectedDeviceId, &portId);
//*/
{
Mutex::Autolock _l(mLock);
RecordThread *thread = checkRecordThread_l(output.inputId);
if (thread == NULL) {
ALOGE("createRecord() checkRecordThread_l failed, input handle %d", output.inputId);
lStatus = BAD_VALUE;
goto Exit;
}
...
///*add by shengjie for AudioFlinger createRecord
//2.create Record Tract to check recording audio buffer and data
recordTrack = thread->createRecordTrack_l(client, input.attr, &output.sampleRate,
input.config.format, input.config.channel_mask,
&output.frameCount, sessionId,
&output.notificationFrameCount,
callingPid, clientUid, &output.flags,
input.clientInfo.clientTid,
&lStatus, portId,
input.opPackageName);
//*/
...
// Check if one effect chain was awaiting for an AudioRecord to be created on this
// session and move it to this thread.
///*add by shengjie for parse AudioFlinger createRecord
//3.enable audio input channer related effect
sp<EffectChain> chain = getOrphanEffectChain_l(sessionId);
//*/
if (chain != 0) {
Mutex::Autolock _l(thread->mLock);
thread->addEffectChain_l(chain);
}
break;
}
// End of retry loop.
// The lack of indentation is deliberate, to reduce code churn and ease merges.
}
output.cblk = recordTrack->getCblk();
output.buffers = recordTrack->getBuffers();
output.portId = portId;
// return handle to client
recordHandle = new RecordHandle(recordTrack);
...
*status = lStatus;
return recordHandle;
}?
总结：
? ? ? ? 下面的文章会对AudioFlinger createRecord里面涉及的getInputForAttr、checkRecordThread_l、getOrphanEffectChain_l接口进行更详细的解析。
? ? ? ? 对流程的追踪，目的是能在这套流程下能随便插入一些项目中关于音频麦克阵列、数据处理的一些特殊配置及要求，能在整体宏观的session、client管控下另外走一些特殊的通道满足项目需求。