[EMAIL PROTECTED] wrote:
Hi, Greg,

Our DSP decoder/device are on the same physical device. We push encoded frames into the device but cannot get decoded PCMs back. So the decoder class (Decode() method) buffers each encoded frames, feeds back dummy PCMs audio services. When AudioDevice::Write() is called, we push a proper amount of buffered encoded frames to physical device.
                Decoder::Decode() ----> Audio Service ----> Device::Write()

The mismatch happens between the total bytes coming into AudioDevice::Write() and the total bytes Decoder::Decode() feeds back (equal to the total frames Decoder buffered). There are moments that encoded frames is running out while Device::Write() still gets called with dummy PCMs pushed down. I am trying to understand in what cases this mismatch will happen -- so far we are testing AMR-NB and this happens quite frequently during streaming.


You should not be using incoming PCM data to meter your flow of decoded audio
data. The audio stream is the master source of the timeline. So, you should
just be pushing audio data into your audio device as fast as it can consume
it (which will be the samplerate*channels*bits/sample).

You audio device code then must provide the master timeline information
via the OnTimeSync() calls into the core on a regular basis. This timeline
information is then sent out to all the other renderers in the system to
provide them with the current timeline. That is how video renderers know
what frame to blt.

So, just ignore and throw away all the dummy PCM data coming into the
audio device code (CHXAudioDevice) and provide the timeline information
via OnTimeSync() (and GetCurrentAudioTime()) and all should work fine.

--greg.






Thanks,

Gang Shen

-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of ext Greg
Wright
Sent: Tuesday, August 16, 2005 1:54 PM
To: Shen Gang.1 (Nokia-TP-MSW/Dallas)
Cc: [email protected]
Subject: Re: [Audio-dev] HXAudioSession & CHXAudioStream


[EMAIL PROTECTED] wrote:

Hi, Greg,

You are right. I am using that guide to code a HW audio device. I don't have problems with the dummy PCMs illustrated in that guide. As you know, HW audio device/decoder buffers encoded audio frames and feedbacks correspondent dummy PCMs. I did exactly like that. Later, I found that HXAudioSession pushes down more PCMs to device than the dummy PCMs decoder feed backed. It is not clear to me what is case HXAudioSession will insert silent PCMs into the stream. Maybe packet loss? If so, how does HW audio device know that PCMs pushed down is actually faked silence? -- this is very important for HW device/decoders for they don't accept silent PCMs.


All dummy PCMs send through the audio services should be silence
(in case they get mixed or faded with other streams). Each dummy
PCM should also be thrown away. I assume that the real audio data
is being send directly from your DSP decoder directly to the audio
physical device. Is that correct? Or, are you decoding in the DSP
and then sending *real* audio PCM data back to the renderer, which
in turn, sends it to the audio services?


Also, can you tell me how you know there are *more* PCM then
what your decoder provided? Are you mesuring by bytes or by
number of chunks sent (Write())? Your decoder will decode in
the native format of the coded audio right? It is possible that
the media engine is resampling the PCM data to match what ever
format the audio device was opened up at. For example, the audio
could be 44.1K but the audio device could only open up at 16K.
That will result in a different amount of data be written the
the CHXAudioDevice code. You can take a look at:

HX_RESULT CHXAudioSession::GetDeviceFormat()



You mentioned as "there are a few places in hxaudses.cpp where the PCM data is silenced and/or inserted into the audio stream". Would you please explain those places? It seems silent PCMs are generated in CHXAudioStream, which is controlled by HXAudioSession. Is it right?


Look in hxaudstr_net.cpp for CAudioSvcSampleConverter::silence() and the
following code chunks in hxaudses.cpp:

                     /* If the mixer buffer was not used, make sure it is 
initialized
                      * to silence since we pass it to post process hooks
                      */
                     if (!bIsMixBufferDirty)
                     {
//{FILE* f1 = ::fopen("e:\\audioses.txt", "a+"); ::fprintf(f1, "%lu\t%p\tsilence in 
mix buffer\n", HX_GET_BETTERTICKCOUNT(), this);::fclose(f1);}
                         ::memset(pMixBuffer, 0, 
HX_SAFESIZE_T(m_ulBytesPerGran));
                     }


             /* did we ever write to the session buffer ? */
             if (m_pPlayerList->GetCount() > 1 && !m_bSessionBufferDirty)
             {
                 HXLOGL4(HXLOG_ADEV, "CHXAudioSession[%p]::PlayAudio(): silence in 
session buffer", this);
                 ::memset(pSessionBuf, 0, HX_SAFESIZE_T(m_ulBytesPerGran));
             }

--greg.



Thanks,

Gang Shen










-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of ext Greg
Wright
Sent: Monday, August 15, 2005 3:45 PM
To: Shen Gang.1 (Nokia-TP-MSW/Dallas)
Cc: [email protected]
Subject: Re: [Audio-dev] HXAudioSession & CHXAudioStream


[EMAIL PROTECTED] wrote:


Hi,

When I program a DSP audio device, I found something interesting.
When no data in the buffer (CHXAudioStream), HXAudioSession still pushes fake PCMs down to audio device(::CheckToPlayMoreAudio). Those PCMs are generated by CHXAudioStream. Since DSP device has to buffer every encoded frame in a seperate queue, it is disturbing to receive ::Write() with extra PCMs. Although I made a workaround for that, I am wondering why HXAudioSession & CHXAudioStream are designed in that way. Could anyone kindly explain a little bit?

Thanks,

Gang Shen


Could you explain a little more about what it is you are doing
exactly? From the above it sounds like you have a hardware decoder
for some audio stream. Is that correct? If so, have you read
the "Hardware Decoder Integration Guide":

    https://client.helixcommunity.org/2004/devdocs/dsp_inte

It will talk a bit about where some of this 'dummy PCM' can come
from and why it is used.

If the above is not the case, then there are a few places in
hxaudses.cpp where the PCM data is silenced and/or inserted into
the audio stream. I would need to know more about your specific
playback scenario to tell you more however.

Let me know if you have any other questions,
--greg.


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