Do you mean to say that aac parser adds the header to the frame..
________________________________
From: RaviY <yend...@pv.com>
To: android-framework <android-framework@googlegroups.com>
Sent: Monday, 13 July, 2009 11:49:45 PM
Subject: Re: how does omx component comes to know about the number of channels
and sampling rate of the audio clip
You can look at how this "config" information is passed from the aac
parser to the aac decoder. NOTE: The parameter values are not directly
sent to the decoder. Instead, they are sent in a buffer (called the
config buffer) which the decoder is supposed to understand and decode.
-Ravi
On Jul 13, 12:58 pm, sesha giri <sesha_giri_n...@yahoo.co.in> wrote:
> Hi Ravi,
>
> I am trying to use my own g711 decoder instead of the existing g711 decoder.
> But in case of g711 decoder the frames given to decoder does not decode the
> sampling frequecy info.
> So thats the reason by some means I wanted to pass this info from wav parser
> to decode component.
>
> --Giri
>
> ________________________________
> From: RaviY <yend...@pv.com>
> To: android-framework <android-framework@googlegroups.com>
> Sent: Monday, 13 July, 2009 7:11:20 PM
> Subject: Re: how does omx component comes to know about the number of
> channels and sampling rate of the audio clip
>
> - We don't pass that information from the parser node to the decoder
> node. The main reason being that we don't trust that information.
> - There isn't and shouldn't be any information about the bitstream
> that the decoder cannot decipher.
> - Once again, why do you have a decoder for a WAV content? Are you
> adding support for formats other than PCM, ulaw, and alaw?
>
> -Ravi
>
> On Jul 13, 1:28 am, sesha giri <sesha_giri_n...@yahoo.co.in> wrote:
>
> > Hi Ravi,
>
> > I had seen this case in pvmf_omx_audiodec_node.cpp
>
> > This is what the case(The code is from the cupcake version of Android):
> > case OMX_EventPortSettingsChanged:
> > {
> > LOGD("EventHandlerProcessing::OMX_EventPortSettingsChanged");
> > PVLOGGER_LOGMSG(PVLOGMSG_INST_LLDBG, iLogger,
> > PVLOGMSG_STACK_TRACE,
> > (0,
> > "PVMFOMXAudioDecNode::EventHandlerProcessing: OMX_EventPortSettingsChanged
> > returned from OMX component"));
>
> > // first check if dynamic reconfiguration is already in
> > progress,
> > // if so, wait until this is completed, and then initiate the
> > 2nd reconfiguration
> > if (iDynamicReconfigInProgress)
> > {
> >
> > LOGD("EventHandlerProcessing::OMX_EventPortSettingsChanged::iDynamicReconfigInProgress");
> > PVLOGGER_LOGMSG(PVLOGMSG_INST_LLDBG, iLogger,
> > PVLOGMSG_STACK_TRACE,
> > (0,
> > "PVMFOMXAudioDecNode::EventHandlerProcessing: OMX_EventPortSettingsChanged
> > returned for port %d, dynamic reconfig already in progress", aData1));
>
> > iSecondPortToReconfig = aData1;
> > iSecondPortReportedChange = true;
>
> > // check the audio sampling rate and fs right away in case
> > of output port
> > // is this output port?
> > if (iSecondPortToReconfig == iOutputPortIndex)
> > {
>
> > OMX_ERRORTYPE Err;
> > // GET the output buffer params and sizes
> > OMX_AUDIO_PARAM_PCMMODETYPE Audio_Pcm_Param;
> > Audio_Pcm_Param.nPortIndex = iOutputPortIndex; // we're
> > looking for output port params
> > Audio_Pcm_Param.nSize =
> > sizeof(OMX_AUDIO_PARAM_PCMMODETYPE);
> > Audio_Pcm_Param.nVersion.s.nVersionMajor =
> > SPECVERSIONMAJOR;
> > Audio_Pcm_Param.nVersion.s.nVersionMinor =
> > SPECVERSIONMINOR;
> > Audio_Pcm_Param.nVersion.s.nRevision = SPECREVISION;
> > Audio_Pcm_Param.nVersion.s.nStep = SPECSTEP;
>
> > Err = OMX_GetParameter(iOMXAudioDecoder,
> > OMX_IndexParamAudioPcm, &Audio_Pcm_Param);
> > if (Err != OMX_ErrorNone)
> > {
> > PVLOGGER_LOGMSG(PVLOGMSG_INST_HLDBG, iLogger,
> > PVLOGMSG_ERR,
> > (0,
> > "PVMFOMXAudioDecNode::EventHandlerProcessing() PortSettingsChanged ->
> > Cannot get component output parameters"));
>
> > SetState(EPVMFNodeError);
> > ReportErrorEvent(PVMFErrResource);
> > }
>
> > iPCMSamplingRate = Audio_Pcm_Param.nSamplingRate; //
> > can be set to 0 (if unknown)
>
> > if (iPCMSamplingRate == 0) // use default sampling rate
> > (i.e. 48000)
> > iPCMSamplingRate =
> > PVOMXAUDIODEC_DEFAULT_SAMPLINGRATE;
>
> > iNumberOfAudioChannels = Audio_Pcm_Param.nChannels;
> > // should be 1 or 2
> > if (iNumberOfAudioChannels != 1 &&
> > iNumberOfAudioChannels != 2)
> > {
> > PVLOGGER_LOGMSG(PVLOGMSG_INST_HLDBG, iLogger,
> > PVLOGMSG_ERR,
> > (0,
> > "PVMFOMXAudioDecNode::EventHandlerProcessing() PortSettingsChanged ->
> > Output parameters num channels = %d", iNumberOfAudioChannels));
>
> > SetState(EPVMFNodeError);
> > ReportErrorEvent(PVMFErrResource);
> > }
> > }
> > }
> > else
> > {
> > PVLOGGER_LOGMSG(PVLOGMSG_INST_LLDBG, iLogger,
> > PVLOGMSG_STACK_TRACE,
> > (0,
> > "PVMFOMXAudioDecNode::EventHandlerProcessing: OMX_EventPortSettingsChanged
> > returned for port %d", aData1));
>
> > iProcessingState =
> > EPVMFOMXAudioDecNodeProcessingState_PortReconfig;
> > iPortIndexForDynamicReconfig = aData1;
> > // start "discarding" data right away, don't wait
> > // check the audio sampling rate and fs right away in case
> > of output port
> > // is this output port?
> > if (iPortIndexForDynamicReconfig == iOutputPortIndex)
> > {
>
> > OMX_ERRORTYPE Err;
> > // GET the output buffer params and sizes
> > OMX_AUDIO_PARAM_PCMMODETYPE Audio_Pcm_Param;
> > Audio_Pcm_Param.nPortIndex = iOutputPortIndex; // we're
> > looking for output port params
> > Audio_Pcm_Param.nSize =
> > sizeof(OMX_AUDIO_PARAM_PCMMODETYPE);
> > Audio_Pcm_Param.nVersion.s.nVersionMajor =
> > SPECVERSIONMAJOR;
> > Audio_Pcm_Param.nVersion.s.nVersionMinor =
> > SPECVERSIONMINOR;
> > Audio_Pcm_Param.nVersion.s.nRevision = SPECREVISION;
> > Audio_Pcm_Param.nVersion.s.nStep = SPECSTEP;
>
> > Err = OMX_GetParameter(iOMXAudioDecoder,
> > OMX_IndexParamAudioPcm, &Audio_Pcm_Param);
> > if (Err != OMX_ErrorNone)
> > {
> > PVLOGGER_LOGMSG(PVLOGMSG_INST_HLDBG, iLogger,
> > PVLOGMSG_ERR,
> > (0,
> > "PVMFOMXAudioDecNode::EventHandlerProcessing() PortSettingsChanged ->
> > Cannot get component output parameters"));
>
> > SetState(EPVMFNodeError);
> > ReportErrorEvent(PVMFErrResource);
>
> > }
>
> > iPCMSamplingRate = Audio_Pcm_Param.nSamplingRate; //
> > can be set to 0 (if unknown)
>
> > if (iPCMSamplingRate == 0) // use default sampling rate
> > (i.e. 48000)
> > iPCMSamplingRate =
> > PVOMXAUDIODEC_DEFAULT_SAMPLINGRATE;
>
> > iNumberOfAudioChannels = Audio_Pcm_Param.nChannels;
> > // should be 1 or 2
> > if (iNumberOfAudioChannels != 1 &&
> > iNumberOfAudioChannels != 2)
> > {
> > PVLOGGER_LOGMSG(PVLOGMSG_INST_HLDBG, iLogger,
> > PVLOGMSG_ERR,
> > (0,
> > "PVMFOMXAudioDecNode::EventHandlerProcessing() PortSettingsChanged ->
> > Output parameters num channels = %d", iNumberOfAudioChannels));
>
> > SetState(EPVMFNodeError);
> > ReportErrorEvent(PVMFErrResource);
>
> > }
>
> > }
> > iDynamicReconfigInProgress = true;
> > }
>
> > RunIfNotReady();
> > break;
> > }//end of case OMX_PortSettingsChanged
>
> > Here if you see it gets the parameter from audio decode node which in turn
> > calls the decoder component layer
> > After geting the values it sets its own value.
>
> > But what I need is that I want to get the values from WAV Parser and set
> > the same to decoder component layer
> > Because In my case the WAV Parser is the one which has the proper info
> > which I want to Pass it on to decoder layer.
>
> > How can I do that.
>
> > --Giri
>
> > ________________________________
> > From: RaviY <yend...@pv.com>
> > To: android-framework <android-framework@googlegroups.com>
> > Sent: Saturday, 11 July, 2009 7:39:29 PM
> > Subject: Re: how does omx component comes to know about the number of
> > channels and sampling rate of the audio clip
>
> > Search for OMX_EventPortSettingsChanged.
>
> > On Jul 11, 1:12 am, sesha giri <sesha_giri_n...@yahoo.co.in> wrote:
>
> > > Hi
>
> > > I was asking how does that port reconfiguration happens.
> > > I didnt find that any where
> > > Even it happens it does not configure the sampling rate and number of
> > > chanels.
>
> > > Regarding the WAV component I am writing my own WAV component for G711.
>
> > > -Giri
>
> > > ________________________________
> > > From: RaviY <yend...@pv.com>
> > > To: android-framework <android-framework@googlegroups.com>
> > > Sent: Saturday, 11 July, 2009 9:03:06 AM
> > > Subject: Re: how does omx component comes to know about the number of
> > > channels and sampling rate of the audio clip
>
> > > I am not sure I understand the problem correctly. The parameters
> > > initialized in the OMX components are just for "initialization". IF,
> > > by change, the parameters match the actual clip, then the advantage
> > > would be that there will not be any port reconfiguration for the OMX
> > > components. That's about it.
>
> > > Also, you are talking about WAV clips, for which, there is no OMX
> > > component.
>
> > > -Ravi
>
> > > On Jul 10, 1:40 pm, sesha giri <sesha_giri_n...@yahoo.co.in> wrote:
>
> > > > Hi
>
> > > > In the omx_component file for the specific decoder there are few
> > > > initializations
> > > > In those sampling rate and number of channels are also initialized.
>
> > > > But I can see these are hard coded.
>
> ...
>
> read more ยป
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