Andorid QQ for TV麦克调试问题
1.Android系统默认采样率是8000Hz；
2.QQ应用采样率是44100Hz或16000Hz；
3.QQ for TV应用会在真正开始语音聊天前，测试最佳采样率(即先会执行打开和关闭设备的一个测试；注意：有些驱动对快速打开关闭会有异常；这时可以考虑在HAL设置时间戳，当调用HAL关闭和打开函数的时间间隔不超过1S时、不执行关闭动作；完后记录状态，再次打开时也不再执行打开动作)；
4.重采样由AudioFlinger服务完成，不需要驱动和HAL模块参与。一、流程
E/AudioRecord( 2019): Could not get audio input for record source 1
E/AudioRecord-JNI( 2019): Error creating AudioRecord instance: initialization check failed.
E/AudioRecord-Java( 2019): [ android.media.AudioRecord ] Error code -20 when initializing native AudioRecord object.
如上错误是当前mic的配置文件中不存在应用设置的采样率、采样精度或声道。
frameworks/base/media/java/android/media/AudioRecord.java
public AudioRecord(int audioSource, int sampleRateInHz, int channelConfig, int audioFormat,
int bufferSizeInBytes)
throws IllegalArgumentException {
int initResult = native_setup( new WeakReference(this),
mRecordSource, mSampleRate, mChannels, mAudioFormat, mNativeBufferSizeInBytes,
session);
if (initResult != SUCCESS) {
loge("Error code "+initResult+" when initializing native AudioRecord object.");
return; // with mState == STATE_UNINITIALIZED
}
} {"native_setup", "(Ljava/lang/Object;IIIII[I)I",
(void *)android_media_AudioRecord_setup},
static int
android_media_AudioRecord_setup(JNIEnv *env, jobject thiz, jobject weak_this,
jint source, jint sampleRateInHertz, jint channels,
jint audioFormat, jint buffSizeInBytes, jintArray jSession){
sp lpRecorder = new AudioRecord();
lpRecorder->set((audio_source_t) source,
sampleRateInHertz,
format, // word length, PCM
channels,
frameCount,
recorderCallback,// callback_t
lpCallbackData,// void* user
0, // notificationFrames,
true, // threadCanCallJava)
sessionId);
if (lpRecorder->initCheck() != NO_ERROR) {
ALOGE("Error creating AudioRecord instance: initialization check failed.");
goto native_init_failure;
}
} status_t AudioRecord::set(
audio_source_t inputSource,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
int frameCount,
callback_t cbf,
void* user,
int notificationFrames,
bool threadCanCallJava,
int sessionId){
audio_io_handle_t input = AudioSystem::getInput(inputSource,
sampleRate,
format,
channelMask,
mSessionId);
if (input == 0) {
ALOGE("Could not get audio input for record source %d", inputSource);
return BAD_VALUE;
}
}
audio_io_handle_t AudioSystem::getInput(audio_source_t inputSource,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
int sessionId){
const sp& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return 0;
return aps->getInput(inputSource, samplingRate, format, channelMask, sessionId);
} frameworks/av/services/audioflinger/AudioPolicyService.cpp
audio_io_handle_t AudioPolicyService::getInput(audio_source_t inputSource,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
int audioSession){
audio_io_handle_t input = mpAudioPolicy->get_input(mpAudioPolicy, inputSource, samplingRate,
format, channelMask, (audio_in_acoustics_t) 0);
//rc = hw_get_module(AUDIO_POLICY_HARDWARE_MODULE_ID, &module);
//rc = audio_policy_dev_open(module, &mpAudioPolicyDev);
}static audio_io_handle_t ap_get_input(struct audio_policy *pol, audio_source_t inputSource,
uint32_t sampling_rate,
audio_format_t format,
audio_channel_mask_t channelMask,
audio_in_acoustics_t acoustics){
struct legacy_audio_policy *lap = to_lap(pol);
return lap->apm->getInput((int) inputSource, sampling_rate, (int) format, channelMask,
(AudioSystem::audio_in_acoustics)acoustics);
}audio_io_handle_t AudioPolicyManagerBase::getInput(int inputSource,
uint32_t samplingRate,
uint32_t format,
uint32_t channelMask,
AudioSystem::audio_in_acoustics acoustics){
audio_devices_t device = getDeviceForInputSource(inputSource);
//查询当前麦克
IOProfile *profile = getInputProfile(device,
samplingRate,
format,
channelMask);
//根据audio_policy.conf中记录当前麦克信息，看是否支持采样率、采样位数以及声道。
if (profile == NULL) {
ALOGW("getInput() could not find profile for device %04x, samplingRate %d, format %d,"
"channelMask %04x",
device, samplingRate, format, channelMask);
return 0;
}
input = mpClientInterface->openInput(profile->mModule->mHandle,
&inputDesc->mDevice,
&inputDesc->mSamplingRate,
&inputDesc->mFormat,
&inputDesc->mChannelMask);
}frameworks/av/services/audioflinger/AudioPolicyService.cpp
open_input_on_module : aps_open_input_on_module,
static audio_io_handle_t aps_open_input_on_module(void *service,
audio_module_handle_t module,
audio_devices_t *pDevices,
uint32_t *pSamplingRate,
audio_format_t *pFormat,
audio_channel_mask_t *pChannelMask){
sp af = AudioSystem::get_audio_flinger();
if (af == 0) {
ALOGW("%s: could not get AudioFlinger", __func__);
return 0;
}
return af->openInput(module, pDevices, pSamplingRate, pFormat, pChannelMask);
} audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
audio_devices_t *pDevices,
uint32_t *pSamplingRate,
audio_format_t *pFormat,
audio_channel_mask_t *pChannelMask){
status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
&inStream);status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
&inStream);
if (status == NO_ERROR && inStream != NULL) {
thread = new RecordThread(this,
input,
reqSamplingRate,
reqChannels,
id,
device);
}
}
AudioFlinger::RecordThread::RecordThread(const sp& audioFlinger,
AudioStreamIn *input,
uint32_t sampleRate,
audio_channel_mask_t channelMask,
audio_io_handle_t id,
audio_devices_t device) :
ThreadBase(audioFlinger, id, AUDIO_DEVICE_NONE, device, RECORD),
mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL),
// mRsmpInIndex and mInputBytes set by readInputParameters()
mReqChannelCount(popcount(channelMask)),
mReqSampleRate(sampleRate)
// mBytesRead is only meaningful while active, and so is cleared in start()
// (but might be better to also clear here for dump?){
readInputParameters()
}
void AudioFlinger::RecordThread::readInputParameters(){
mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2){
mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
mResampler->setSampleRate(mSampleRate);
}
//重采样！！！
}
bool AudioFlinger::RecordThread::threadLoop(){
if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
//正常从HAL获取数据
}
else{
nsecs_t now = systemTime();
if ((now - lastWarning) > kWarningThrottleNs) {
ALOGW("RecordThread: buffer overflow");
lastWarning = now;
}
}
} W/AudioFlinger( 918): RecordThread: buffer overflow
W/AudioRecord( 2238): obtainBuffer timed out (is the CPU pegged?) user=0002f260, server=0002f260
应该考虑buffer溢出；尤其是第二个问题，应该考虑底层音频数据有问题。
frameworks/base/media/java/android/media/AudioRecord.java
public int read(byte[] audioData, int offsetInBytes, int sizeInBytes) {
if (mState != STATE_INITIALIZED) {
return ERROR_INVALID_OPERATION;
}
if ( (audioData == null) || (offsetInBytes < 0 ) || (sizeInBytes < 0)
|| (offsetInBytes + sizeInBytes > audioData.length)) {
return ERROR_BAD_VALUE;
}
return native_read_in_byte_array(audioData, offsetInBytes, sizeInBytes);
}static jint android_media_AudioRecord_readInByteArray(JNIEnv *env, jobject thiz,
jbyteArray javaAudioData,
jint offsetInBytes, jint sizeInBytes) {
sp lpRecorder = getAudioRecord(env, thiz);
ssize_t readSize = lpRecorder->read(recordBuff + offsetInBytes,
sizeInBytes > (jint)recorderBuffSize ?
(jint)recorderBuffSize : sizeInBytes );
} ssize_t AudioRecord::read(void* buffer, size_t userSize){
status_t err = obtainBuffer(&audioBuffer, ((2 * MAX_RUN_TIMEOUT_MS) / WAIT_PERIOD_MS));
}
status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount){
cblk->waitTimeMs += waitTimeMs;
if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
ALOGW( "obtainBuffer timed out (is the CPU pegged?) "
"user=%08x, server=%08x", cblk->user, cblk->server);
}
audioBuffer->raw = (int8_t*)cblk->buffer(u);
//cblk->buffer(u)即是AudioFlinger中mActiveTrack->getNextBuffer(&buffer)读取的HAL数据
}