Dynaudnorm & earwax filters

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Dynaudnorm & earwax filters

Ronak Patel
Hey guys,

I'm trying to use the dynaudnorm and earwax filters in my iPhone app, and I've noticed that the dynaudnorm filter wants to resample the audio.

Is there any way to avoid resampling the audio? What format does the filter expect the input to be?

I'm initializing the filters with the following format:

AVAudioFormat 0x600002846260:  2 ch,  44100 Hz, Float32, non-inter

I initialize the input and output frames like so:

frame?.pointee.channels = Int32(format.channelCount)
frame?.pointee.channel_layout = UInt64(av_get_default_channel_layout(Int32(format.channelCount)))
frame?.pointee.sample_rate = Int32(format.sampleRate)
frame?.pointee.format = Int32(AV_SAMPLE_FMT_FLTP.rawValue)
frame?.pointee.nb_samples = Int32(maximumFrameCount)

What am I doing wrong?

Thanks,

Ronak
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Re: Dynaudnorm & earwax filters

Carl Eugen Hoyos-2
2018-12-11 1:07 GMT+01:00, Ronak <[hidden email]>:
> Hey guys,
>
> I'm trying to use the dynaudnorm and earwax filters in my iPhone app, and
> I've noticed that the dynaudnorm filter wants to resample the audio.
>
> Is there any way to avoid resampling the audio? What format does the filter
> expect the input to be?

The dynaudnorm filter only accepts planar double, the earwax filter only
packed s16.

Carl Eugen
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Re: Dynaudnorm & earwax filters

Ronak Patel


> On Dec 10, 2018, at 7:29 PM, Carl Eugen Hoyos <[hidden email]> wrote:
>
> 2018-12-11 1:07 GMT+01:00, Ronak <[hidden email]>:
>> Hey guys,
>>
>> I'm trying to use the dynaudnorm and earwax filters in my iPhone app, and
>> I've noticed that the dynaudnorm filter wants to resample the audio.
>>
>> Is there any way to avoid resampling the audio? What format does the filter
>> expect the input to be?
>
> The dynaudnorm filter only accepts planar double, the earwax filter only
> packed s16.

Ok thanks. I tried to use this filter in my iOS code; but I'm getting errors with an error code -35.

This is my code that tries to write data into the filter graph and reads it back; what am I doing wrong?


private func filterBuffer(_ inputBuffer: UnsafeMutableAudioBufferListPointer, frameCount: UInt32, outputBuffer: UnsafeMutableAudioBufferListPointer) throws {

    // copy the pointers to the audio buffer into the frame for manipulation

    // each buffer represents the audio per channel
    for index in 0..<inputBuffer.count {
      let dataByteSize = Int(inputBuffer[index].mDataByteSize)
      let buffer = inputBuffer[index].mData?.bindMemory(to: UInt8.self, capacity: dataByteSize)

      inputAudioFrame?.pointee.extended_data[index] = buffer
    }

    // write the audio frame into the audioInputContext so it can be filtered
    let writeResult = av_buffersrc_write_frame(audioInputContext, inputAudioFrame)
    if writeResult == 0 {

      // pull the filtered audio out of the audioOutputContext
      let pullResult = av_buffersink_get_frame(audioOutputContext, outputAudioFrame)
      if pullResult >= 0 {
        let filteredAudioBufferData = outputAudioFrame?.pointee.extended_data

        // copy the pointers to the filtered audio into the output buffers
        for index in 0..<outputBuffer.count {
          outputBuffer[index].mData = UnsafeMutableRawPointer(filteredAudioBufferData?[index])
        }
      } else {

        // the audio couldn't be filtered properly, throw an error
        throw PlayerError.filterFailure([:])
      }
    } else {

      // the audio couldn't be filtered properly, throw an error
      throw PlayerError.filterFailure([:])
    }
  }

Thanks,

Ronak


>
> Carl Eugen
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Re: Dynaudnorm & earwax filters

Carl Eugen Hoyos-2
2018-12-11 17:25 GMT+01:00, Ronak <[hidden email]>:

>
>
>> On Dec 10, 2018, at 7:29 PM, Carl Eugen Hoyos <[hidden email]> wrote:
>>
>> 2018-12-11 1:07 GMT+01:00, Ronak <[hidden email]>:
>>> Hey guys,
>>>
>>> I'm trying to use the dynaudnorm and earwax filters in my iPhone app, and
>>> I've noticed that the dynaudnorm filter wants to resample the audio.
>>>
>>> Is there any way to avoid resampling the audio? What format does the
>>> filter
>>> expect the input to be?
>>
>> The dynaudnorm filter only accepts planar double, the earwax filter only
>> packed s16.
>
> Ok thanks. I tried to use this filter in my iOS code; but I'm getting errors
> with an error code -35.

You may get a useful answer on the libav-user mailing list, your question
is a little off-topic here.

Carl Eugen
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Re: Dynaudnorm & earwax filters

Nicolas George
In reply to this post by Ronak Patel
Ronak (2018-12-11):
> Ok thanks. I tried to use this filter in my iOS code; but I'm getting
> errors with an error code -35.
>
> This is my code that tries to write data into the filter graph and
> reads it back; what am I doing wrong?

I do not read whatever language that is, but at the very least your code
is missing the translation error code -> error message.

Regards,

--
  Nicolas George

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Re: Dynaudnorm & earwax filters

Ronak Patel


> On Dec 12, 2018, at 8:32 AM, Nicolas George <[hidden email]> wrote:
>
> Ronak (2018-12-11):
>> Ok thanks. I tried to use this filter in my iOS code; but I'm getting
>> errors with an error code -35.
>>
>> This is my code that tries to write data into the filter graph and
>> reads it back; what am I doing wrong?
>
> I do not read whatever language that is, but at the very least your code
> is missing the translation error code -> error message.
>

I found out what my problem is; it's that the dynaudnorm filter is returning EAGAIN; which means I need to send it more PCM frames.

Now, I'm trying to integrate this filter into a real time player context; and I would like to avoid audio artifacts. I've been playing with various options that the filter has; but I can't seem to find one where it would work better in the real time context.

Does anyone know what the correct parameters would be so it works frame by frame or in a much smaller frame size so we can avoid audio artifacts?
Alternatively, is there another ffmpeg filter better suited to real time dynamic range compression or volume normalization?

> Regards,
>
> --
>  Nicolas George
> _______________________________________________
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> [hidden email]
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
> To unsubscribe, visit link above, or email
> [hidden email] with subject "unsubscribe".

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Re: Dynaudnorm & earwax filters

Paul B Mahol
On 12/12/18, Ronak <[hidden email]> wrote:

>
>
>> On Dec 12, 2018, at 8:32 AM, Nicolas George <[hidden email]> wrote:
>>
>> Ronak (2018-12-11):
>>> Ok thanks. I tried to use this filter in my iOS code; but I'm getting
>>> errors with an error code -35.
>>>
>>> This is my code that tries to write data into the filter graph and
>>> reads it back; what am I doing wrong?
>>
>> I do not read whatever language that is, but at the very least your code
>> is missing the translation error code -> error message.
>>
>
> I found out what my problem is; it's that the dynaudnorm filter is returning
> EAGAIN; which means I need to send it more PCM frames.
>
> Now, I'm trying to integrate this filter into a real time player context;
> and I would like to avoid audio artifacts. I've been playing with various
> options that the filter has; but I can't seem to find one where it would
> work better in the real time context.
>
> Does anyone know what the correct parameters would be so it works frame by
> frame or in a much smaller frame size so we can avoid audio artifacts?
> Alternatively, is there another ffmpeg filter better suited to real time
> dynamic range compression or volume normalization?
>

If you read documentation of filter options you would know.
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Re: Dynaudnorm & earwax filters

Ronak Patel

> On Dec 12, 2018, at 11:26 AM, Paul B Mahol <[hidden email]> wrote:
>
> On 12/12/18, Ronak <[hidden email]> wrote:
>>
>>
>>> On Dec 12, 2018, at 8:32 AM, Nicolas George <[hidden email]> wrote:
>>>
>>> Ronak (2018-12-11):
>>>> Ok thanks. I tried to use this filter in my iOS code; but I'm getting
>>>> errors with an error code -35.
>>>>
>>>> This is my code that tries to write data into the filter graph and
>>>> reads it back; what am I doing wrong?
>>>
>>> I do not read whatever language that is, but at the very least your code
>>> is missing the translation error code -> error message.
>>>
>>
>> I found out what my problem is; it's that the dynaudnorm filter is returning
>> EAGAIN; which means I need to send it more PCM frames.
>>
>> Now, I'm trying to integrate this filter into a real time player context;
>> and I would like to avoid audio artifacts. I've been playing with various
>> options that the filter has; but I can't seem to find one where it would
>> work better in the real time context.
>>
>> Does anyone know what the correct parameters would be so it works frame by
>> frame or in a much smaller frame size so we can avoid audio artifacts?
>> Alternatively, is there another ffmpeg filter better suited to real time
>> dynamic range compression or volume normalization?
>>
>
> If you read documentation of filter options you would know.

I already did and tried all sorts of things. I've tried options like: "f=500:g=3:m=10:n=1:b=1", "f=40:g=3:m=10:n=1:b=1". I've even tried the extreme: "f=8000:g=3:m=10:n=1:b=1"

But I still get back lots of EAGAIN.


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Re: Dynaudnorm & earwax filters

Paul B Mahol
On 12/12/18, Ronak <[hidden email]> wrote:

>
>> On Dec 12, 2018, at 11:26 AM, Paul B Mahol <[hidden email]> wrote:
>>
>> On 12/12/18, Ronak <[hidden email]> wrote:
>>>
>>>
>>>> On Dec 12, 2018, at 8:32 AM, Nicolas George <[hidden email]> wrote:
>>>>
>>>> Ronak (2018-12-11):
>>>>> Ok thanks. I tried to use this filter in my iOS code; but I'm getting
>>>>> errors with an error code -35.
>>>>>
>>>>> This is my code that tries to write data into the filter graph and
>>>>> reads it back; what am I doing wrong?
>>>>
>>>> I do not read whatever language that is, but at the very least your code
>>>> is missing the translation error code -> error message.
>>>>
>>>
>>> I found out what my problem is; it's that the dynaudnorm filter is
>>> returning
>>> EAGAIN; which means I need to send it more PCM frames.
>>>
>>> Now, I'm trying to integrate this filter into a real time player context;
>>> and I would like to avoid audio artifacts. I've been playing with various
>>> options that the filter has; but I can't seem to find one where it would
>>> work better in the real time context.
>>>
>>> Does anyone know what the correct parameters would be so it works frame
>>> by
>>> frame or in a much smaller frame size so we can avoid audio artifacts?
>>> Alternatively, is there another ffmpeg filter better suited to real time
>>> dynamic range compression or volume normalization?
>>>
>>
>> If you read documentation of filter options you would know.
>
> I already did and tried all sorts of things. I've tried options like:
> "f=500:g=3:m=10:n=1:b=1", "f=40:g=3:m=10:n=1:b=1". I've even tried the
> extreme: "f=8000:g=3:m=10:n=1:b=1"
>
> But I still get back lots of EAGAIN.

That's normal, if you insist on 0 latency look at something else.
Other players like mpv, handle it fine.
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Re: Dynaudnorm & earwax filters

Ronak Patel


> On Dec 12, 2018, at 11:36 AM, Paul B Mahol <[hidden email]> wrote:
>
> On 12/12/18, Ronak <[hidden email]> wrote:
>>
>>> On Dec 12, 2018, at 11:26 AM, Paul B Mahol <[hidden email]> wrote:
>>>
>>> On 12/12/18, Ronak <[hidden email]> wrote:
>>>>
>>>>
>>>>> On Dec 12, 2018, at 8:32 AM, Nicolas George <[hidden email]> wrote:
>>>>>
>>>>> Ronak (2018-12-11):
>>>>>> Ok thanks. I tried to use this filter in my iOS code; but I'm getting
>>>>>> errors with an error code -35.
>>>>>>
>>>>>> This is my code that tries to write data into the filter graph and
>>>>>> reads it back; what am I doing wrong?
>>>>>
>>>>> I do not read whatever language that is, but at the very least your code
>>>>> is missing the translation error code -> error message.
>>>>>
>>>>
>>>> I found out what my problem is; it's that the dynaudnorm filter is
>>>> returning
>>>> EAGAIN; which means I need to send it more PCM frames.
>>>>
>>>> Now, I'm trying to integrate this filter into a real time player context;
>>>> and I would like to avoid audio artifacts. I've been playing with various
>>>> options that the filter has; but I can't seem to find one where it would
>>>> work better in the real time context.
>>>>
>>>> Does anyone know what the correct parameters would be so it works frame
>>>> by
>>>> frame or in a much smaller frame size so we can avoid audio artifacts?
>>>> Alternatively, is there another ffmpeg filter better suited to real time
>>>> dynamic range compression or volume normalization?
>>>>
>>>
>>> If you read documentation of filter options you would know.
>>
>> I already did and tried all sorts of things. I've tried options like:
>> "f=500:g=3:m=10:n=1:b=1", "f=40:g=3:m=10:n=1:b=1". I've even tried the
>> extreme: "f=8000:g=3:m=10:n=1:b=1"
>>
>> But I still get back lots of EAGAIN.
>
> That's normal, if you insist on 0 latency look at something else.
> Other players like mpv, handle it fine.

Ok. One last thing is it seems like the filter is spitting out lots of pops and crackles when I can get it to return audio frames back out.

Do you know why that would be? I changed all my arguments to just be f="1000" since I thought my options would be causing this. But it's not.

Just in case it helps, I am sending in FLTP which is being resampled by the rwresample filter to S32. I don't think that would be a factor in this right?

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Re: Dynaudnorm & earwax filters

Paul B Mahol
On 12/12/18, Ronak <[hidden email]> wrote:

>
>
>> On Dec 12, 2018, at 11:36 AM, Paul B Mahol <[hidden email]> wrote:
>>
>> On 12/12/18, Ronak <[hidden email]> wrote:
>>>
>>>> On Dec 12, 2018, at 11:26 AM, Paul B Mahol <[hidden email]> wrote:
>>>>
>>>> On 12/12/18, Ronak <[hidden email]> wrote:
>>>>>
>>>>>
>>>>>> On Dec 12, 2018, at 8:32 AM, Nicolas George <[hidden email]> wrote:
>>>>>>
>>>>>> Ronak (2018-12-11):
>>>>>>> Ok thanks. I tried to use this filter in my iOS code; but I'm getting
>>>>>>> errors with an error code -35.
>>>>>>>
>>>>>>> This is my code that tries to write data into the filter graph and
>>>>>>> reads it back; what am I doing wrong?
>>>>>>
>>>>>> I do not read whatever language that is, but at the very least your
>>>>>> code
>>>>>> is missing the translation error code -> error message.
>>>>>>
>>>>>
>>>>> I found out what my problem is; it's that the dynaudnorm filter is
>>>>> returning
>>>>> EAGAIN; which means I need to send it more PCM frames.
>>>>>
>>>>> Now, I'm trying to integrate this filter into a real time player
>>>>> context;
>>>>> and I would like to avoid audio artifacts. I've been playing with
>>>>> various
>>>>> options that the filter has; but I can't seem to find one where it
>>>>> would
>>>>> work better in the real time context.
>>>>>
>>>>> Does anyone know what the correct parameters would be so it works frame
>>>>> by
>>>>> frame or in a much smaller frame size so we can avoid audio artifacts?
>>>>> Alternatively, is there another ffmpeg filter better suited to real
>>>>> time
>>>>> dynamic range compression or volume normalization?
>>>>>
>>>>
>>>> If you read documentation of filter options you would know.
>>>
>>> I already did and tried all sorts of things. I've tried options like:
>>> "f=500:g=3:m=10:n=1:b=1", "f=40:g=3:m=10:n=1:b=1". I've even tried the
>>> extreme: "f=8000:g=3:m=10:n=1:b=1"
>>>
>>> But I still get back lots of EAGAIN.
>>
>> That's normal, if you insist on 0 latency look at something else.
>> Other players like mpv, handle it fine.
>
> Ok. One last thing is it seems like the filter is spitting out lots of pops
> and crackles when I can get it to return audio frames back out.
>
> Do you know why that would be? I changed all my arguments to just be
> f="1000" since I thought my options would be causing this. But it's not.
>
> Just in case it helps, I am sending in FLTP which is being resampled by the
> rwresample filter to S32. I don't think that would be a factor in this
> right?
>

You should send only DBL to this filter.
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Re: Dynaudnorm & earwax filters

Ronak Patel


> On Dec 12, 2018, at 12:10 PM, Paul B Mahol <[hidden email]> wrote:
>
> On 12/12/18, Ronak <[hidden email]> wrote:
>>
>>
>>> On Dec 12, 2018, at 11:36 AM, Paul B Mahol <[hidden email]> wrote:
>>>
>>> On 12/12/18, Ronak <[hidden email]> wrote:
>>>>
>>>>> On Dec 12, 2018, at 11:26 AM, Paul B Mahol <[hidden email]> wrote:
>>>>>
>>>>> On 12/12/18, Ronak <[hidden email]> wrote:
>>>>>>
>>>>>>
>>>>>>> On Dec 12, 2018, at 8:32 AM, Nicolas George <[hidden email]> wrote:
>>>>>>>
>>>>>>> Ronak (2018-12-11):
>>>>>>>> Ok thanks. I tried to use this filter in my iOS code; but I'm getting
>>>>>>>> errors with an error code -35.
>>>>>>>>
>>>>>>>> This is my code that tries to write data into the filter graph and
>>>>>>>> reads it back; what am I doing wrong?
>>>>>>>
>>>>>>> I do not read whatever language that is, but at the very least your
>>>>>>> code
>>>>>>> is missing the translation error code -> error message.
>>>>>>>
>>>>>>
>>>>>> I found out what my problem is; it's that the dynaudnorm filter is
>>>>>> returning
>>>>>> EAGAIN; which means I need to send it more PCM frames.
>>>>>>
>>>>>> Now, I'm trying to integrate this filter into a real time player
>>>>>> context;
>>>>>> and I would like to avoid audio artifacts. I've been playing with
>>>>>> various
>>>>>> options that the filter has; but I can't seem to find one where it
>>>>>> would
>>>>>> work better in the real time context.
>>>>>>
>>>>>> Does anyone know what the correct parameters would be so it works frame
>>>>>> by
>>>>>> frame or in a much smaller frame size so we can avoid audio artifacts?
>>>>>> Alternatively, is there another ffmpeg filter better suited to real
>>>>>> time
>>>>>> dynamic range compression or volume normalization?
>>>>>>
>>>>>
>>>>> If you read documentation of filter options you would know.
>>>>
>>>> I already did and tried all sorts of things. I've tried options like:
>>>> "f=500:g=3:m=10:n=1:b=1", "f=40:g=3:m=10:n=1:b=1". I've even tried the
>>>> extreme: "f=8000:g=3:m=10:n=1:b=1"
>>>>
>>>> But I still get back lots of EAGAIN.
>>>
>>> That's normal, if you insist on 0 latency look at something else.
>>> Other players like mpv, handle it fine.
>>
>> Ok. One last thing is it seems like the filter is spitting out lots of pops
>> and crackles when I can get it to return audio frames back out.
>>
>> Do you know why that would be? I changed all my arguments to just be
>> f="1000" since I thought my options would be causing this. But it's not.
>>
>> Just in case it helps, I am sending in FLTP which is being resampled by the
>> rwresample filter to S32. I don't think that would be a factor in this
>> right?
>>
>
> You should send only DBL to this filter.

Sorry I misquoted.

[volume normalization @ 0x7fa4c860dd80] auto-inserting filter 'auto_resampler_0' between the filter 'input' and the filter 'volume normalization'
[auto_resampler_0 @ 0x7fa4c8610940] ch:2 chl:stereo fmt:fltp r:44100Hz -> ch:2 chl:stereo fmt:dblp r:44100Hz

It is being resampled to DBLP.

Besides doing a whole bunch of trial and error, are there any recommended options to use here?

I'm writing one frame of PCM audio into the filter at a time, within my playback audio graph.

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Re: Dynaudnorm & earwax filters

Paul B Mahol
On 12/12/18, Ronak <[hidden email]> wrote:

>
>
>> On Dec 12, 2018, at 12:10 PM, Paul B Mahol <[hidden email]> wrote:
>>
>> On 12/12/18, Ronak <[hidden email]> wrote:
>>>
>>>
>>>> On Dec 12, 2018, at 11:36 AM, Paul B Mahol <[hidden email]> wrote:
>>>>
>>>> On 12/12/18, Ronak <[hidden email]> wrote:
>>>>>
>>>>>> On Dec 12, 2018, at 11:26 AM, Paul B Mahol <[hidden email]> wrote:
>>>>>>
>>>>>> On 12/12/18, Ronak <[hidden email]> wrote:
>>>>>>>
>>>>>>>
>>>>>>>> On Dec 12, 2018, at 8:32 AM, Nicolas George <[hidden email]> wrote:
>>>>>>>>
>>>>>>>> Ronak (2018-12-11):
>>>>>>>>> Ok thanks. I tried to use this filter in my iOS code; but I'm
>>>>>>>>> getting
>>>>>>>>> errors with an error code -35.
>>>>>>>>>
>>>>>>>>> This is my code that tries to write data into the filter graph and
>>>>>>>>> reads it back; what am I doing wrong?
>>>>>>>>
>>>>>>>> I do not read whatever language that is, but at the very least your
>>>>>>>> code
>>>>>>>> is missing the translation error code -> error message.
>>>>>>>>
>>>>>>>
>>>>>>> I found out what my problem is; it's that the dynaudnorm filter is
>>>>>>> returning
>>>>>>> EAGAIN; which means I need to send it more PCM frames.
>>>>>>>
>>>>>>> Now, I'm trying to integrate this filter into a real time player
>>>>>>> context;
>>>>>>> and I would like to avoid audio artifacts. I've been playing with
>>>>>>> various
>>>>>>> options that the filter has; but I can't seem to find one where it
>>>>>>> would
>>>>>>> work better in the real time context.
>>>>>>>
>>>>>>> Does anyone know what the correct parameters would be so it works
>>>>>>> frame
>>>>>>> by
>>>>>>> frame or in a much smaller frame size so we can avoid audio
>>>>>>> artifacts?
>>>>>>> Alternatively, is there another ffmpeg filter better suited to real
>>>>>>> time
>>>>>>> dynamic range compression or volume normalization?
>>>>>>>
>>>>>>
>>>>>> If you read documentation of filter options you would know.
>>>>>
>>>>> I already did and tried all sorts of things. I've tried options like:
>>>>> "f=500:g=3:m=10:n=1:b=1", "f=40:g=3:m=10:n=1:b=1". I've even tried the
>>>>> extreme: "f=8000:g=3:m=10:n=1:b=1"
>>>>>
>>>>> But I still get back lots of EAGAIN.
>>>>
>>>> That's normal, if you insist on 0 latency look at something else.
>>>> Other players like mpv, handle it fine.
>>>
>>> Ok. One last thing is it seems like the filter is spitting out lots of
>>> pops
>>> and crackles when I can get it to return audio frames back out.
>>>
>>> Do you know why that would be? I changed all my arguments to just be
>>> f="1000" since I thought my options would be causing this. But it's not.
>>>
>>> Just in case it helps, I am sending in FLTP which is being resampled by
>>> the
>>> rwresample filter to S32. I don't think that would be a factor in this
>>> right?
>>>
>>
>> You should send only DBL to this filter.
>
> Sorry I misquoted.
>
> [volume normalization @ 0x7fa4c860dd80] auto-inserting filter
> 'auto_resampler_0' between the filter 'input' and the filter 'volume
> normalization'
> [auto_resampler_0 @ 0x7fa4c8610940] ch:2 chl:stereo fmt:fltp r:44100Hz ->
> ch:2 chl:stereo fmt:dblp r:44100Hz
>
> It is being resampled to DBLP.
>
> Besides doing a whole bunch of trial and error, are there any recommended
> options to use here?
>
> I'm writing one frame of PCM audio into the filter at a time, within my
> playback audio graph.
>

I can not guess, need to look at source code.
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Trimming email quotes

Carl Zwanzig
In reply to this post by Ronak Patel
Any chance folks could start editing their replies to remove the extraneous
quoting?

On 12/12/2018 10:39 AM, Ronak wrote:
>> On Dec 12, 2018, at 12:10 PM, Paul B Mahol <[hidden email]> wrote:
>> On 12/12/18, Ronak <[hidden email]> wrote:
>>>> On Dec 12, 2018, at 11:36 AM, Paul B Mahol <[hidden email]> wrote:
>>>> On 12/12/18, Ronak <[hidden email]> wrote:
>>>>>> On Dec 12, 2018, at 11:26 AM, Paul B Mahol <[hidden email]> wrote:
>>>>>> On 12/12/18, Ronak <[hidden email]> wrote:
>>>>>>>> On Dec 12, 2018, at 8:32 AM, Nicolas George <[hidden email]> wrote:
>>>>>>>> Ronak (2018-12-11):
[55 lines of quotes]
[one-line response]


And also cut the quoted footers since each list email gets a fresh one.

Thanks,

z!
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Re: Trimming email quotes

Paul B Mahol
On 12/12/18, Carl Zwanzig <[hidden email]> wrote:
> Any chance folks could start editing their replies to remove the extraneous
> quoting?
>

No, we like to waste resources.
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Re: Trimming email quotes

Reindl Harald
In reply to this post by Carl Zwanzig


Am 12.12.18 um 19:46 schrieb Carl Zwanzig:
> And also cut the quoted footers since each list email gets a fresh one

people all over the world are not capable to basically handle mail
clients, sad but true
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