在audiotrack java层和native层还有audio hal中dump pcm

(69条消息) [RK3288][Android6.0] Audio播放时的pcm数据dump思路_Kris Fei's blog-CSDN博客

(69条消息) android framework层dump pcm数据的方法_和大伙儿去乘凉-CSDN博客

1.

dump pcm数据是音频工程师经常用的的debug手段，这里总结下在framework层dump数据的位置

dump 数据的方法，方法3的dump_out_data

FILE *fp1 = fopen("/data/audio_spk.pcm", "a+");
 if (fp1) {
         int flen = fwrite((char *)buffer, 1, bytes, fp1);
           fclose(fp1);
   }

2.AudioTrack的processAudioBuffer函数

应用层如果是callback的形式可以在此函数中dump
nsecs_t AudioTrack::processAudioBuffer()

具体dump位置如下，在EVENT_MORE_DATA的event回调之后

。。。。。。
       size_t reqSize = audioBuffer.size;
        mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
        size_t writtenSize = audioBuffer.size;
 
        // Sanity check on returned size
        if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) {
            ALOGE("EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes",
                    reqSize, ssize_t(writtenSize));
            return NS_NEVER;
        }

        FILE *fp1 = fopen("/data/cb16.pcm", "a+");
		 if (fp1) {
				fwrite((char *)audioBuffer.raw, 1, writtenSize, fp1);
				fclose(fp1);
			}


。。。。。。。

3.AudioFlinger的PlaybackThread::threadLoop_write函数内 mNormalSink->write((char *)mSinkBuffer + offset, count);之后

ssize_t AudioFlinger::PlaybackThread::threadLoop_write()
{
    LOG_HIST_TS();
    mInWrite = true;
    ssize_t bytesWritten;
    const size_t offset = mCurrentWriteLength - mBytesRemaining;
 
    // If an NBAIO sink is present, use it to write the normal mixer's submix
    if (mNormalSink != 0) {
 
        const size_t count = mBytesRemaining / mFrameSize;
 
        ATRACE_BEGIN("write");
        // update the setpoint when AudioFlinger::mScreenState changes
        uint32_t screenState = AudioFlinger::mScreenState;
        if (screenState != mScreenState) {
            mScreenState = screenState;
            MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
            if (pipe != NULL) {
                pipe->setAvgFrames((mScreenState & 1) ?
                        (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
            }
        }
        ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count);
        ATRACE_END();
        if (framesWritten > 0) {
            bytesWritten = framesWritten * mFrameSize;
#ifdef TEE_SINK
            mTee.write((char *)mSinkBuffer + offset, framesWritten);
#endif

4.hal中dump pcm数据 

out_write调 操作入参的 void* buffer 和长度 size_t bytes

static ssize_t out_write(struct audio_stream_out *stream, const void* buffer,
                         size_t bytes)
{
......
	//通过property来控制是否要dump pcm data
    property_get("media.audio.record", value, NULL);
    //
    prop_pcm = atoi(value);
    //非0的值决定要dump多少数据才停止，单位是M
    if (prop_pcm > 0) {
        dump_out_data(buffer, bytes, &prop_pcm);
    }
......


dump_out_data()实现

static void dump_out_data(const void* buffer,size_t bytes, int *size)
 {
    ALOGD("dump pcm file.");
    static FILE* fd;
    static int offset = 0;
    if(fd == NULL) {
        fd=fopen("/data/debug.pcm","wb+");
            if(fd == NULL) {
            ALOGD("DEBUG open /data/debug.pcm error =%d ,errno = %d",fd,errno);
            offset = 0;
        }
    }
    //写pcm数据
    fwrite(buffer,bytes,1,fd);
    offset += bytes;
    fflush(fd);
    //比如size是1，那么就是1Ｍ后停止dump数据
    if(offset >= (*size)*1024*1024) {
        *size = 0;
        fclose(fd);
        offset = 0;
        system("setprop media.audio.record 0");
        ALOGD("TEST playback pcmfile end");
    }
}

5.使用callback形式，在 AudioRecord::processAudioBuffer()的函数内EVENT_MORE_DATA的事件回调之后

 
        if (mRetryOnPartialBuffer) {
            mRetryOnPartialBuffer = false;
            if (avail < mRemainingFrames) {
                int64_t myns = ((mRemainingFrames - avail) *
                        1100000000LL) / mSampleRate;
                if (ns < 0 || myns < ns) {
                    ns = myns;
                }
                return ns;
            }
        }
 
        size_t reqSize = audioBuffer.size;
        mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
        size_t readSize = audioBuffer.size;
 
	
		FILE *fp1 = fopen("/data/cb16.pcm", "a+");
		if (fp1) {
			fwrite((char *)audioBuffer.raw, 1, readSize, fp1);
			fclose(fp1);
		}

另外方法

1. dump audioflinger的数据
/frameworks/av/services/audioflinger/Tracks.cpp#getNextBuffer

// AudioBufferProvider interface
status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
        AudioBufferProvider::Buffer* buffer)
{
    ServerProxy::Buffer buf;
    size_t desiredFrames = buffer->frameCount;
    buf.mFrameCount = desiredFrames;
    status_t status = mServerProxy->obtainBuffer(&buf);
    buffer->frameCount = buf.mFrameCount;
    buffer->raw = buf.mRaw;
    if (buf.mFrameCount == 0) {
        mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames);
    } else {
        mAudioTrackServerProxy->tallyUnderrunFrames(0);
    }
    //JaychouNote: dump audio flinger data begin.
        int tmpFd = ::open("/data/misc/audio/dumpTrack.pcm", O_CREAT | O_WRONLY | O_APPEND, 0777);
        if ( tmpFd < 0 ) {
            ALOGE("Tracks.cpp Fail to open dumpTrack file");
        } else {
            ::write(tmpFd, buffer->raw, buffer->frameCount * TrackBase::mChannelCount * 2);
            ::close(tmpFd);
        }
        //JaychouNote: dump audio flinger data end.
    return status;
}

dump hal层的数据

/hardware/qcom/audio/hal/audio_hw.c#out_write

static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,size_t bytes)
{
    struct stream_out *out = (struct stream_out *)stream;
    struct audio_device *adev = out->dev;
    ssize_t ret = 0;
    ......
    } else {
        if (out->pcm) {
            if (out->muted)
                memset((void *)buffer, 0, bytes);

            ALOGVV("%s: writing buffer (%d bytes) to pcm device", __func__, bytes);
        //JaychouNote: dump hal data begin.
    int tmpFd = open("/data/misc/audio/dumpTrack_hal.pcm", O_CREAT | O_WRONLY | O_APPEND, 0777);

    if ( tmpFd < 0 ) {
        ALOGE("Fail to open dumpTrack_hal file");
    } else {
        write(tmpFd, buffer, bytes);
        close(tmpFd);
    }
    //JaychouNote: dump hal data end.
            if (adev->adm_request_focus)
                adev->adm_request_focus(adev->adm_data, out->handle);

            if (out->usecase == USECASE_AUDIO_PLAYBACK_AFE_PROXY) {
                ret = pcm_mmap_write(out->pcm, (void *)buffer, bytes);
            }
            else
                ret = pcm_write(out->pcm, (void *)buffer, bytes);

            if (ret == 0)
                out->written += bytes / (out->config.channels * sizeof(short));

            if (adev->adm_abandon_focus)
                adev->adm_abandon_focus(adev->adm_data, out->handle);
        }
    }

exit:
    pthread_mutex_unlock(&out->lock);

    if (ret != 0) {
        if (out->pcm)
            ALOGE("%s: error %zu - %s", __func__, ret, pcm_get_error(out->pcm));
        out_standby(&out->stream.common);
        usleep(bytes * 1000000 / audio_stream_out_frame_size(stream) /
               out_get_sample_rate(&out->stream.common));
    }
    return bytes;
}