前幾天在看stream type的時候，調用函數AudioSystem::getOutput的地方並沒有繼續往下看。
今天深入看看。
*****************************************源碼*************************************************
status_t AudioTrack::set(
int streamType,
uint32_t sampleRate,
int format,
int channels,
int frameCount,
uint32_t flags,
callback_t cbf,
void* user,
int notificationFrames,
const sp<IMemory>& sharedBuffer,
bool threadCanCallJava,
int sessionId)
{
LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
if (mAudioTrack != 0) {
LOGE("Track already in use");
return INVALID_OPERATION;
}
int afSampleRate;
if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
return NO_INIT;
}
uint32_t afLatency;
if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
return NO_INIT;
}
// handle default values first.
if (streamType == AudioSystem::DEFAULT) {
streamType = AudioSystem::MUSIC;
}
if (sampleRate == 0) {
sampleRate = afSampleRate;
}
// these below should probably come from the audioFlinger too…
if (format == 0) {
format = AudioSystem::PCM_16_BIT;
}
if (channels == 0) {
channels = AudioSystem::CHANNEL_OUT_STEREO;
}
// validate parameters
if (!AudioSystem::isValidFormat(format)) {
LOGE("Invalid format");
return BAD_VALUE;
}
// force direct flag if format is not linear PCM
if (!AudioSystem::isLinearPCM(format)) {
flags |= AudioSystem::OUTPUT_FLAG_DIRECT;
}
if (!AudioSystem::isOutputChannel(channels)) {
LOGE("Invalid channel mask");
return BAD_VALUE;
}
uint32_t channelCount = AudioSystem::popCount(channels);
audio_io_handle_t output = AudioSystem::getOutput((AudioSystem::stream_type)streamType,
sampleRate, format, channels, (AudioSystem::output_flags)flags);
if (output == 0) {
LOGE("Could not get audio output for stream type %d", streamType);
return BAD_VALUE;
}
mVolume[LEFT] = 1.0f;
mVolume[RIGHT] = 1.0f;
mSendLevel = 0;
mFrameCount = frameCount;
mNotificationFramesReq = notificationFrames;
mSessionId = sessionId;
mAuxEffectId = 0;
// create the IAudioTrack
status_t status = createTrack(streamType, sampleRate, format, channelCount,
frameCount, flags, sharedBuffer, output, true);
if (status != NO_ERROR) {
return status;
}
if (cbf != 0) {
mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
if (mAudioTrackThread == 0) {
LOGE("Could not create callback thread");
return NO_INIT;
}
}
mStatus = NO_ERROR;
mStreamType = streamType;
mFormat = format;
mChannels = channels;
mChannelCount = channelCount;
mSharedBuffer = sharedBuffer;
mMuted = false;
mActive = 0;
mCbf = cbf;
mUserData = user;
mLoopCount = 0;
mMarkerPosition = 0;
mMarkerReached = false;
mNewPosition = 0;
mUpdatePeriod = 0;
mFlags = flags;
return NO_ERROR;
}
**********************************************************************************************
#######################說明################################
+++++++++++++++++++++++++AudioSystem::getOutput+++++++++++++++++++++++++++++++++++++++
audio_io_handle_t AudioSystem::getOutput(stream_type stream,
uint32_t samplingRate,
uint32_t format,
uint32_t channels,
output_flags flags)
{
audio_io_handle_t output = 0;
// Do not use stream to output map cache if the direct output
// flag is set or if we are likely to use a direct output
// (e.g voice call stream @ 8kHz could use BT SCO device and be routed to
// a direct output on some platforms).
// TODO: the output cache and stream to output mapping implementation needs to
// be reworked for proper operation with direct outputs. This code is too specific
// to the first use case we want to cover (Voice Recognition and Voice Dialer over
// Bluetooth SCO
if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) == 0 &&
((stream != AudioSystem::VOICE_CALL && stream != AudioSystem::BLUETOOTH_SCO) ||
channels != AudioSystem::CHANNEL_OUT_MONO ||
(samplingRate != 8000 && samplingRate != 16000))) {
Mutex::Autolock _l(gLock);
output = AudioSystem::gStreamOutputMap.valueFor(stream);
LOGV_IF((output != 0), "getOutput() read %d from cache for stream %d", output, stream);
}
if (output == 0) {
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return 0;
output = aps->getOutput(stream, samplingRate, format, channels, flags);
if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) == 0) {
Mutex::Autolock _l(gLock);
AudioSystem::gStreamOutputMap.add(stream, output);
}
+++++++++++++++++++++++++++++AudioPolicyService::getOutput+++++++++++++++++++++++++++++++++++
audio_io_handle_t AudioPolicyService::getOutput(AudioSystem::stream_type stream,
uint32_t samplingRate,
uint32_t format,
uint32_t channels,
AudioSystem::output_flags flags)
{
if (mpPolicyManager == NULL) {
return 0;
}
LOGV("getOutput() tid %d", gettid());
Mutex::Autolock _l(mLock);
return mpPolicyManager->getOutput(stream, samplingRate, format, channels, flags);
+++++++++++++++++++++++++++++AudioPolicyManagerBase::getOutput+++++++++++++++++++++++++++++++++++
audio_io_handle_t AudioPolicyManagerBase::getOutput(AudioSystem::stream_type stream,
uint32_t samplingRate,
uint32_t format,
uint32_t channels,
AudioSystem::output_flags flags)
{
audio_io_handle_t output = 0;
uint32_t latency = 0;
// 此處根據stream獲取策略
routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
// 這兒再根據策略獲取device
// 這句代碼和上句代碼結合起來，就是根據stream的類型，獲取對應的device
uint32_t device = getDeviceForStrategy(strategy);
LOGV("getOutput() stream %d, samplingRate %d, format %d, channels %x, flags %x", stream, samplingRate, format, channels, flags);
#ifdef AUDIO_POLICY_TEST
// 定義瞭AUDIO_POLICY_TEST的時候，會去打開一個output
if (mCurOutput != 0) {
LOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channels %x, mDirectOutput %d",
mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
if (mTestOutputs[mCurOutput] == 0) {
LOGV("getOutput() opening test output");
AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
outputDesc->mDevice = mTestDevice;
outputDesc->mSamplingRate = mTestSamplingRate;
outputDesc->mFormat = mTestFormat;
outputDesc->mChannels = mTestChannels;
outputDesc->mLatency = mTestLatencyMs;
outputDesc->mFlags = (AudioSystem::output_flags)(mDirectOutput ? AudioSystem::OUTPUT_FLAG_DIRECT : 0);
outputDesc->mRefCount[stream] = 0;
mTestOutputs[mCurOutput] = mpClientInterface->openOutput(&outputDesc->mDevice,
&outputDesc->mSamplingRate,
&outputDesc->mFormat,
&outputDesc->mChannels,
&outputDesc->mLatency,
outputDesc->mFlags);
if (mTestOutputs[mCurOutput]) {
AudioParameter outputCmd = AudioParameter();
outputCmd.addInt(String8("set_id"),mCurOutput);
mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString());
addOutput(mTestOutputs[mCurOutput], outputDesc);
}
}
return mTestOutputs[mCurOutput];
}
#endif //AUDIO_POLICY_TEST
// Direct模式的話，會去打開一個output
// open a direct output if required by specified parameters
if (needsDirectOuput(stream, samplingRate, format, channels, flags, device)) {
LOGV("getOutput() opening direct output device %x", device);
AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
outputDesc->mDevice = device;
outputDesc->mSamplingRate = samplingRate;
outputDesc->mFormat = format;
outputDesc->mChannels = channels;
outputDesc->mLatency = 0;
outputDesc->mFlags = (AudioSystem::output_flags)(flags | AudioSystem::OUTPUT_FLAG_DIRECT);
outputDesc->mRefCount[stream] = 0;
output = mpClientInterface->openOutput(&outputDesc->mDevice,
&outputDesc->mSamplingRate,
&outputDesc->mFormat,
&outputDesc->mChannels,
&outputDesc->mLatency,
outputDesc->mFlags);
// only accept an output with the requeted parameters
if (output == 0 ||
(samplingRate != 0 && samplingRate != outputDesc->mSamplingRate) ||
(format != 0 && format != outputDesc->mFormat) ||
(channels != 0 && channels != outputDesc->mChannels)) {
LOGV("getOutput() failed opening direct output: samplingRate %d, format %d, channels %d",
samplingRate, format, channels);
if (output != 0) {
mpClientInterface->closeOutput(output);
}
delete outputDesc;
return 0;
}
addOutput(output, outputDesc);
return output;
}
if (channels != 0 && channels != AudioSystem::CHANNEL_OUT_MONO &&
channels != AudioSystem::CHANNEL_OUT_STEREO) {
return 0;
}
// open a non direct output
// stream模式的話，直接取成員變量的值
// 下面看看用於取值的這幾個成員變量賦值的地方
// get which output is suitable for the specified stream. The actual routing change will happen
// when startOutput() will be called
uint32_t a2dpDevice = device & AudioSystem::DEVICE_OUT_ALL_A2DP;
if (AudioSystem::popCount((AudioSystem::audio_devices)device) == 2) {
#ifdef WITH_A2DP
if (a2dpUsedForSonification() && a2dpDevice != 0) {
// if playing on 2 devices among which one is A2DP, use duplicated output
LOGV("getOutput() using duplicated output");
LOGW_IF((mA2dpOutput == 0), "getOutput() A2DP device in multiple %x selected but A2DP output not opened", device);
// mA2dpOutput和mDuplicatedOutput的賦值都在函數AudioPolicyManagerBase::handleA2dpConnection中
output = mDuplicatedOutput;
+++++++++++++++++++++++++++++AudioPolicyManagerBase::handleA2dpConnection+++++++++++++++++++++++++++++++++++
status_t AudioPolicyManagerBase::handleA2dpConnection(AudioSystem::audio_devices device,
const char *device_address)
{
// when an A2DP device is connected, open an A2DP and a duplicated output
LOGV("opening A2DP output for device %s", device_address);
AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
outputDesc->mDevice = device;
// 這個函數在下面有看
mA2dpOutput = mpClientInterface->openOutput(&outputDesc->mDevice,
&outputDesc->mSamplingRate,
&outputDesc->mFormat,
&outputDesc->mChannels,
&outputDesc->mLatency,
outputDesc->mFlags);
if (mA2dpOutput) {
// add A2DP output descriptor
addOutput(mA2dpOutput, outputDesc);
//TODO: configure audio effect output stage here
// set initial stream volume for A2DP device
applyStreamVolumes(mA2dpOutput, device);
if (a2dpUsedForSonification()) {
// mA2dpOutput和mHardwareOutput是已經打開的兩個output
mDuplicatedOutput = mpClientInterface->openDuplicateOutput(mA2dpOutput, mHardwareOutput);
+++++++++++++++++++++++++++++AudioPolicyService::openDuplicateOutput+++++++++++++++++++++++++++++++++++
audio_io_handle_t AudioPolicyService::openDuplicateOutput(audio_io_handle_t output1,
audio_io_handle_t output2)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
if (af == 0) {
LOGW("openDuplicateOutput() could not get AudioFlinger");
return 0;
}
return af->openDuplicateOutput(output1, output2);
+++++++++++++++++++++++++++++++AudioFlinger::openDuplicateOutput+++++++++++++++++++++++++++++++++
int AudioFlinger::openDuplicateOutput(int output1, int output2)
{
Mutex::Autolock _l(mLock);
MixerThread *thread1 = checkMixerThread_l(output1);
MixerThread *thread2 = checkMixerThread_l(output2);
++++++++++++++++++++++++++++++AudioFlinger::checkMixerThread_l++++++++++++++++++++++++++++++++++
// checkMixerThread_l() must be called with AudioFlinger::mLock held
AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const
{
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread != NULL) {
if (thread->type() == PlaybackThread::DIRECT) {
thread = NULL;
}
}
+++++++++++++++++++++++++++++++AudioFlinger::checkPlaybackThread_l+++++++++++++++++++++++++++++++++
// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const
{
PlaybackThread *thread = NULL;
if (mPlaybackThreads.indexOfKey(output) >= 0) {
// AudioFlinger::openOutput函數中會向mPlaybackThreads中追加成員
thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get();
}
return thread;
}
——————————-AudioFlinger::checkPlaybackThread_l———————————
return (MixerThread *)thread;
}
——————————AudioFlinger::checkMixerThread_l———————————-
if (thread1 == NULL || thread2 == NULL) {
LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2);
return 0;
}
int id = nextUniqueId();
// 選一個thread作為主thread，來創建DuplicatingThread
DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
++++++++++++++++++++++++++++AudioFlinger::DuplicatingThread::DuplicatingThread++++++++++++++++++++++++++++++++++++
AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id)
: MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX)
{
mType = PlaybackThread::DUPLICATING;
addOutputTrack(mainThread);
+++++++++++++++++++++++++++++AudioFlinger::DuplicatingThread::addOutputTrack+++++++++++++++++++++++++++++++++++
void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
{
int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
this,
mSampleRate,
mFormat,
mChannelCount,
frameCount);
if (outputTrack->cblk() != NULL) {
thread->setStreamVolume(AudioSystem::NUM_STREAM_TYPES, 1.0f);
mOutputTracks.add(outputTrack);
LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
updateWaitTime();
}
+++++++++++++++++++++++++++++AudioFlinger::PlaybackThread::OutputTrack::OutputTrack+++++++++++++++++++++++++++++++++++
AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
const wp<ThreadBase>& thread,
DuplicatingThread *sourceThread,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount)
: Track(thread, NULL, AudioSystem::NUM_STREAM_TYPES, sampleRate, format, channelCount, frameCount, NULL, 0),
mActive(false), mSourceThread(sourceThread)
{
PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get();
if (mCblk != NULL) {
mCblk->flags |= CBLK_DIRECTION_OUT;
mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
mCblk->volume[0] = mCblk->volume[1] = 0x1000;
mOutBuffer.frameCount = 0;
playbackThread->mTracks.add(this);
LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channelCount %d mBufferEnd %p",
mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channelCount, mBufferEnd);
} else {
LOGW("Error creating output track on thread %p", playbackThread);
}
}
—————————–AudioFlinger::PlaybackThread::OutputTrack::OutputTrack———————————–
}
—————————–AudioFlinger::DuplicatingThread::addOutputTrack———————————–
}
—————————-AudioFlinger::DuplicatingThread::DuplicatingThread————————————
// 再將另外一個thread也添加進來
thread->addOutputTrack(thread2);
mPlaybackThreads.add(id, thread);
// notify client processes of the new output creation
thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
return id;
}
——————————-AudioFlinger::openDuplicateOutput———————————
}
—————————–AudioPolicyService::openDuplicateOutput———————————–
}
if (mDuplicatedOutput != 0 ||
!a2dpUsedForSonification()) {
// If both A2DP and duplicated outputs are open, send device address to A2DP hardware
// interface
AudioParameter param;
param.add(String8("a2dp_sink_address"), String8(device_address));
mpClientInterface->setParameters(mA2dpOutput, param.toString());
mA2dpDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
if (a2dpUsedForSonification()) {
// add duplicated output descriptor
AudioOutputDescriptor *dupOutputDesc = new AudioOutputDescriptor();
dupOutputDesc->mOutput1 = mOutputs.valueFor(mHardwareOutput);
dupOutputDesc->mOutput2 = mOutputs.valueFor(mA2dpOutput);
dupOutputDesc->mSamplingRate = outputDesc->mSamplingRate;
dupOutputDesc->mFormat = outputDesc->mFormat;
dupOutputDesc->mChannels = outputDesc->mChannels;
dupOutputDesc->mLatency = outputDesc->mLatency;
addOutput(mDuplicatedOutput, dupOutputDesc);
applyStreamVolumes(mDuplicatedOutput, device);
}
} else {
LOGW("getOutput() could not open duplicated output for %d and %d",
mHardwareOutput, mA2dpOutput);
mpClientInterface->closeOutput(mA2dpOutput);
mOutputs.removeItem(mA2dpOutput);
mA2dpOutput = 0;
delete outputDesc;
return NO_INIT;
}
} else {
LOGW("setDeviceConnectionState() could not open A2DP output for device %x", device);
delete outputDesc;
return NO_INIT;
}
AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput);
if (!a2dpUsedForSonification()) {
// mute music on A2DP output if a notification or ringtone is playing
uint32_t refCount = hwOutputDesc->strategyRefCount(STRATEGY_SONIFICATION);
for (uint32_t i = 0; i < refCount; i++) {
setStrategyMute(STRATEGY_MEDIA, true, mA2dpOutput);
}
}
mA2dpSuspended = false;
return NO_ERROR;
}
—————————–AudioPolicyManagerBase::handleA2dpConnection———————————–
} else
#endif
{
// if playing on 2 devices among which none is A2DP, use hardware output
output = mHardwareOutput;
+++++++++++++++++++++++++++++++AudioPolicyManagerBase::AudioPolicyManagerBase+++++++++++++++++++++++++++++++++
mHardwareOutput在AudioPolicyManagerBase的構造函數中被賦值。
AudioPolicyManagerBase::AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface)
:
#ifdef AUDIO_POLICY_TEST
Thread(false),
#endif //AUDIO_POLICY_TEST
mPhoneState(AudioSystem::MODE_NORMAL), mRingerMode(0),
mMusicStopTime(0), mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f),
mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0),
mA2dpSuspended(false)
{
mpClientInterface = clientInterface;
for (int i = 0; i < AudioSystem::NUM_FORCE_USE; i++) {
mForceUse[i] = AudioSystem::FORCE_NONE;
}
// devices available by default are speaker, ear piece and microphone
mAvailableOutputDevices = AudioSystem::DEVICE_OUT_EARPIECE |
AudioSystem::DEVICE_OUT_SPEAKER;
mAvailableInputDevices = AudioSystem::DEVICE_IN_BUILTIN_MIC;
#ifdef WITH_A2DP
mA2dpOutput = 0;
mDuplicatedOutput = 0;
mA2dpDeviceAddress = String8("");
#endif
mScoDeviceAddress = String8("");
// open hardware output
AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
// 從下面的這個賦值可見，一般情況下，如果不使用A2DP的話，我們用的device其實是固定的：DEVICE_OUT_SPEAKER
outputDesc->mDevice = (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER;
// 從AudioPolicyManagerBase對象被創建的地方可知，mpClientInterface其實是AudioPolicyService對象。
mHardwareOutput = mpClientInterface->openOutput(&outputDesc->mDevice,
&outputDesc->mSamplingRate,
&outputDesc->mFormat,
&outputDesc->mChannels,
&outputDesc->mLatency,
outputDesc->mFlags);
if (mHardwareOutput == 0) {
LOGE("Failed to initialize hardware output stream, samplingRate: %d, format %d, channels %d",
outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannels);
} else {
addOutput(mHardwareOutput, outputDesc);
setOutputDevice(mHardwareOutput, (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER, true);
//TODO: configure audio effect output stage here
}
+++++++++++++++++++++++++++++++AudioPolicyService::openOutput+++++++++++++++++++++++++++++++++
audio_io_handle_t AudioPolicyService::openOutput(uint32_t *pDevices,
uint32_t *pSamplingRate,
uint32_t *pFormat,
uint32_t *pChannels,
uint32_t *pLatencyMs,
AudioSystem::output_flags flags)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
if (af == 0) {
LOGW("openOutput() could not get AudioFlinger");
return 0;
}
return af->openOutput(pDevices,
pSamplingRate,
(uint32_t *)pFormat,
pChannels,
pLatencyMs,
flags);
++++++++++++++++++++++++++++++AudioFlinger::openOutput++++++++++++++++++++++++++++++++++
int AudioFlinger::openOutput(uint32_t *pDevices,
uint32_t *pSamplingRate,
uint32_t *pFormat,
uint32_t *pChannels,
uint32_t *pLatencyMs,
uint32_t flags)
{
status_t status;
PlaybackThread *thread = NULL;
mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
uint32_t format = pFormat ? *pFormat : 0;
uint32_t channels = pChannels ? *pChannels : 0;
uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
LOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x",
pDevices ? *pDevices : 0,
samplingRate,
format,
channels,
flags);
if (pDevices == NULL || *pDevices == 0) {
return 0;
}
Mutex::Autolock _l(mLock);
// 調用HAL層的接口，打開一個output stream。
// device類型，被作為參數傳給瞭HAL
// 這個函數前面已經看過，此處隻把與device類型相關的東東再復習一下
++++++++++++++++++++++++++++AudioHardwareALSA::openOutputStream++++++++++++++++++++++++++++++++++++
AudioStreamOut *
AudioHardwareALSA::openOutputStream(uint32_t devices,
int *format,
uint32_t *channels,
uint32_t *sampleRate,
status_t *status)
{
…
// 從device 類型的定義可知，每種device類型占一個bit
// 從函數getDeviceForStrategy的實現可知，我們根據stream類型最終取得的device肯定隻有一種
// 此處檢查device類型是不是隻占瞭一個bit
if (devices & (devices – 1)) {
if (status) *status = err;
LOGD("openOutputStream called with bad devices");
return out;
}
// Find the appropriate alsa device
// ALSAHandleList中的內容其實是從一個數組中copy過來
// 數組中隻是列出瞭一些初始信息，很多有用的東東是在後面賦值的
// 結構體中一個重要的東東module(alsa_device_t * )是在調用函數s_open中賦值的
+++++++++++++++++++++++++++++_defaults+++++++++++++++++++++++++++++++++++
static alsa_handle_t _defaults[] = {
{
module : 0,
devices : IMX51_OUT_DEFAULT,
curDev : 0,
curMode : 0,
handle : 0,
format : SND_PCM_FORMAT_S16_LE, // AudioSystem::PCM_16_BIT
channels : 2,
sampleRate : DEFAULT_SAMPLE_RATE,
latency : 200000, // Desired Delay in usec
bufferSize : 6144, // Desired Number of samples
modPrivate : (void *)&setDefaultControls,
},
{
module : 0,
devices : IMX51_IN_DEFAULT,
curDev : 0,
curMode : 0,
handle : 0,
format : SND_PCM_FORMAT_S16_LE, // AudioSystem::PCM_16_BIT
channels : 2,
sampleRate : DEFAULT_SAMPLE_RATE,
latency : 250000, // Desired Delay in usec
bufferSize : 6144, // Desired Number of samples
modPrivate : (void *)&setDefaultControls,
},
};
—————————–_defaults———————————–
for(ALSAHandleList::iterator it = mDeviceList.begin();
it != mDeviceList.end(); ++it)
// 根據device類型，找到占的坑
if (it->devices & devices) {
// 在調s_open時，仍然將device類型作為參數傳入
err = mALSADevice->open(&(*it), devices, mode());
if (err) break;
+++++++++++++++++++++++++++++s_open+++++++++++++++++++++++++++++++++++
static status_t s_open(alsa_handle_t *handle, uint32_t devices, int mode)
{
…
// device類型是在找device名稱的時候
const char *devName = deviceName(handle, devices, mode, 1);
+++++++++++++++++++++++++++++deviceName+++++++++++++++++++++++++++++++++++
//card_device =0, return the card name, card_device=1, return the card device name
const char *deviceName(alsa_handle_t *alsa_handle, uint32_t device, int mode, int card_device)
{
…
property_get("ro.HDMI_AUDIO_OUTPUT", value, "");
// device類型在此處作判斷用
// 因此，device name的取得，很大程度上說依賴於配置文件的
if((device & AudioSystem::DEVICE_OUT_WIRED_HDMI) && havespdifdevice && (strcmp(value, "1") == 0))
{
return spdifcardname;
}else if(havesgtldevice)
{
return sgtlcardname;
}
return "default";
}
—————————–deviceName———————————–
// The PCM stream is opened in blocking mode, per ALSA defaults. The
// AudioFlinger seems to assume blocking mode too, so asynchronous mode
// should not be used.
// 根據上面得到的device name，打開一個alsa設備
int err = snd_pcm_open(&handle->handle, devName, direction(handle), 0);
if (err < 0) {
LOGE("Failed to Initialize any ALSA %s device: %s", stream, strerror(err));
return NO_INIT;
}
…
// 將device類型保存到上面占地坑中
handle->curDev = devices;
handle->curMode = mode;
return err;
}
—————————–s_open———————————–
if (devices & AudioSystem::DEVICE_OUT_WIRED_HDMI){
strcpy(mCurCard ,SPDIF);
mMixer = mMixerSpdif;
} else {
strcpy(mCurCard,SGTL5000);
mMixer = mMixerSgtl5000;
}
out = new AudioStreamOutALSA(this, &(*it));
err = out->set(format, channels, sampleRate);
break;
}
if (status) *status = err;
return out;
}
—————————-AudioHardwareALSA::openOutputStream————————————
AudioStreamOut *output = mAudioHardware->openOutputStream(*pDevices,
(int *)&format,
&channels,
&samplingRate,
&status);
LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
output,
samplingRate,
format,
channels,
status);
mHardwareStatus = AUDIO_HW_IDLE;
if (output != 0) {
int id = nextUniqueId();
if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
(format != AudioSystem::PCM_16_BIT) ||
(channels != AudioSystem::CHANNEL_OUT_STEREO)) {
// 如果是direct模式，創建一個DirectOutputThread
thread = new DirectOutputThread(this, output, id, *pDevices);
LOGV("openOutput() created direct output: ID %d thread %p", id, thread);
} else {
// stream模式的話，創建一個MixerThread
thread = new MixerThread(this, output, id, *pDevices);
LOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
#ifdef LVMX
unsigned bitsPerSample =
(format == AudioSystem::PCM_16_BIT) ? 16 :
((format == AudioSystem::PCM_8_BIT) ? 8 : 0);
unsigned channelCount = (channels == AudioSystem::CHANNEL_OUT_STEREO) ? 2 : 1;
int audioOutputType = LifeVibes::threadIdToAudioOutputType(thread->id());
LifeVibes::init_aot(audioOutputType, samplingRate, bitsPerSample, channelCount);
LifeVibes::setDevice(audioOutputType, *pDevices);
#endif
}
// 將上面創建的thread追加到playback thread列表，並將其與一個unique id關聯起來
mPlaybackThreads.add(id, thread);
if (pSamplingRate) *pSamplingRate = samplingRate;
if (pFormat) *pFormat = format;
if (pChannels) *pChannels = channels;
if (pLatencyMs) *pLatencyMs = thread->latency();
// notify client processes of the new output creation
thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
return id;
}
return 0;
}
——————————AudioFlinger::openOutput———————————-
}
——————————-AudioPolicyService::openOutput———————————
updateDeviceForStrategy();
#ifdef AUDIO_POLICY_TEST
AudioParameter outputCmd = AudioParameter();
outputCmd.addInt(String8("set_id"), 0);
mpClientInterface->setParameters(mHardwareOutput, outputCmd.toString());
mTestDevice = AudioSystem::DEVICE_OUT_SPEAKER;
mTestSamplingRate = 44100;
mTestFormat = AudioSystem::PCM_16_BIT;
mTestChannels = AudioSystem::CHANNEL_OUT_STEREO;
mTestLatencyMs = 0;
mCurOutput = 0;
mDirectOutput = false;
for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
mTestOutputs[i] = 0;
}
const size_t SIZE = 256;
char buffer[SIZE];
snprintf(buffer, SIZE, "AudioPolicyManagerTest");
run(buffer, ANDROID_PRIORITY_AUDIO);
#endif //AUDIO_POLICY_TEST
}
AudioPolicyManagerBase對象是在AudioPolicyService的構造函數中創建的。
++++++++++++++++++++++++++++++AudioPolicyService::AudioPolicyService++++++++++++++++++++++++++++++++++
AudioPolicyService::AudioPolicyService()
: BnAudioPolicyService() , mpPolicyManager(NULL)
{
char value[PROPERTY_VALUE_MAX];
// start tone playback thread
mTonePlaybackThread = new AudioCommandThread(String8(""));
// start audio commands thread
mAudioCommandThread = new AudioCommandThread(String8("ApmCommandThread"));
#if (defined GENERIC_AUDIO) || (defined AUDIO_POLICY_TEST)
mpPolicyManager = new AudioPolicyManagerBase(this);
LOGV("build for GENERIC_AUDIO – using generic audio policy");
#else
// if running in emulation – use the emulator driver
if (property_get("ro.kernel.qemu", value, 0)) {
LOGV("Running in emulation – using generic audio policy");
mpPolicyManager = new AudioPolicyManagerBase(this);
}
else {
LOGV("Using hardware specific audio policy");
// 正常的話，應該是走到這兒來。
// 這個函數以前看過，實際使用的應該是ALSA中的AudioPlicyManager
// Alsa中的AudioPolicyManager繼承自AudioPolicyManagerBase，
// 並且基本上沒做什麼變化
mpPolicyManager = createAudioPolicyManager(this);
}
#endif
// load properties
property_get("ro.camera.sound.forced", value, "0");
mpPolicyManager->setSystemProperty("ro.camera.sound.forced", value);
}
——————————AudioPolicyService::AudioPolicyService———————————-
——————————-AudioPolicyManagerBase::AudioPolicyManagerBase———————————
}
LOGV("getOutput() using output %d for 2 devices %x", output, device);
} else {
#ifdef WITH_A2DP
if (a2dpDevice != 0) {
// if playing on A2DP device, use a2dp output
LOGW_IF((mA2dpOutput == 0), "getOutput() A2DP device %x selected but A2DP output not opened", device);
output = mA2dpOutput;
} else
#endif
{
// if playing on not A2DP device, use hardware output
output = mHardwareOutput;
}
}
LOGW_IF((output ==0), "getOutput() could not find output for stream %d, samplingRate %d, format %d, channels %x, flags %x",
stream, samplingRate, format, channels, flags);
return output;
}
—————————–AudioPolicyManagerBase::getOutput———————————–
}
—————————–AudioPolicyService::getOutput———————————–
}
return output;
}
————————-AudioSystem::getOutput—————————————
###########################################################
&&&&&&&&&&&&&&&&&&&&&&&總結&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&
根據stream type找到對應的strategy。
根據strategy找到對應的device type。
根據device type找到對應的alsa_handle_t對象。
根據配置和device type找到設備名稱，並打開設備獲取一個snd_pcm_t對象。
將snd_pcm_t對象保存到alsa_handle_t對象對象中。
我們之前已經看過write操作，
其實就是往snd_pcm_t對象的stopped_areas或者running_areas成員中copy數據；
或者直接調用snd_pcm_t對象的fast_ops的writei函數寫數據。
&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&
摘自：江風的專欄
