[FFmpeg-devel] Add waveformat extensible support in wav muxer (SoC qualification task)
Michael Niedermayer
michaelni
Sun Mar 29 17:47:35 CEST 2009
On Sun, Mar 29, 2009 at 11:51:57AM +0800, zhentan feng wrote:
> Hi
>
> 2009/3/28 Michael Niedermayer <michaelni at gmx.at>
>
> > On Sat, Mar 28, 2009 at 06:47:25PM +0800, zhentan feng wrote:
> > > Hi
> > >
> > > 2009/3/28 Benjamin Larsson <banan at ludd.ltu.se>
> > >
> > > > zhentan feng wrote:
> > > >
> > > >> Hi
> > > >>
> > > >> 2009/3/27 Benjamin Larsson <banan at ludd.ltu.se>
> > > >>
> > > >>
> > > >>
> > > >>> zhentan feng wrote:
> > > >>>
> > > >>>
> > > >>>> Hi
> > > >>>>
> > > >>>> 2009/3/26 Benjamin Larsson <banan at ludd.ltu.se>
> > > >>>>
> > > >>>>
> > > >>>>
> > > >>>>> zhentan feng wrote:
> > > >>>>>
> > > >>>>>
> > > >>>>>
> > > >>>>>> Hi
> > > >>>>>>
> > > >>>>>> 2009/3/26 Michael Niedermayer <michaelni at gmx.at>
> > > >>>>>>
> > > >>>>>>
> > > >>>>>>
> > > >>>>>>
> > > >>>>>>
> > > >>>>>>> On Thu, Mar 26, 2009 at 01:19:00AM +0800, zhentan feng wrote:
> > > >>>>>>>
> > > >>>>>>>
> > > >>>>>>>
> > > >>>>>>>
> > > >>>>>>>> Hi,
> > > >>>>>>>>
> > > >>>>>>>> Here is patch for the small task of qualification tasks NO.30.
> > > >>>>>>>> Based on the work of Benjamin Larsson, I generated the pathc as
> > > >>>>>>>>
> > > >>>>>>>>
> > > >>>>>>> below.
> > > >>>
> > > >>>
> > > >>>> I think it must need further modifies.
> > > >>>>>>>>
> > > >>>>>>>>
> > > >>>>>>>>
> > > >>>>>>>>
> > > >>>>>>> yes, iam not sure if this or benjamins patch was better
> > > >>>>>>>
> > > >>>>>>>
> > > >>>>>>>
> > > >>>>>>>
> > > >>>>>>>
> > > >>>>>> I downloaded the 6-channel wav file from:
> > > >>>>>>
> > > >>>>>>
> > > >>>>>>
> > > >>>>>>
> > > >>>>>
> > > >>>
> > http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples/Microsoft/6_Channel_ID.wav
> > > >>>
> > > >>>
> > > >>>> ( and other typr wav files can acess here :
> > > >>>>>>
> > http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html
> > > >>>>>> )
> > > >>>>>>
> > > >>>>>> 1) then I run ./ffmpeg_g -i 6_Channel_ID.wav channel4.wav
> > > >>>>>>
> > > >>>>>> I get the error:
> > > >>>>>> Resampling with input channels greater than 2 unsupported.
> > > >>>>>>
> > > >>>>>> so I run another command:
> > > >>>>>> 2) ./output_example test.wav
> > > >>>>>> ./ffmpeg_g -i test.wav -ac 4 ch4.wav
> > > >>>>>>
> > > >>>>>> and enc->channels is 4, however enc->channel_layout is 0.
> > > >>>>>>
> > > >>>>>>
> > > >>>>>>
> > > >>>>>> Finally, I have 2 questions:
> > > >>>>>>
> > > >>>>>> 1) how to test the muxer works correctly?
> > > >>>>>>
> > > >>>>>>
> > > >>>>>>
> > > >>>>>>
> > > >>>>> When the muxer works correctly it should be possible to transcode a
> > wav
> > > >>>>> file that contains the wavformatextensible header.
> > > >>>>>
> > > >>>>> 2) where to specify the enc->channel_layout?
> > > >>>>> It should be filled in by ffmpeg.c.
> > > >>>>>
> > > >>>>>
> > > >>>>>
> > > >>>> I debug the commandline:
> > > >>>> ./ffmpeg_g -i test.wav -ac 4 4ch.wav
> > > >>>>
> > > >>>> and found that the enc->channel_layout value is assigned from the
> > input
> > > >>>>
> > > >>>>
> > > >>> file
> > > >>>
> > > >>>
> > > >>>> channel_layout. In this case, it is 0.
> > > >>>> and it seems that "-channel_layout" option doesn't work.
> > > >>>>
> > > >>>> 1) is it need to refresh the channel_layout value according to the
> > > >>>>
> > > >>>>
> > > >>> channels
> > > >>>
> > > >>>
> > > >>>> in ffmpeg.c or in put_wav_header()?
> > > >>>> 2) any tools to examine the output file is correct?
> > > >>>>
> > > >>>> zhentan feng
> > > >>>>
> > > >>>>
> > > >>> Hi, this is the latest version of my patch. This adds the maybe
> > correct
> > > >>> GUID values. To really test this you need to find a wav file with a
> > > >>> corrrect dwChannelMask. And then make sure that the muxer puts the
> > same
> > > >>> mask to the transcoded file. The output file should also be playable
> > in
> > > >>> windows.
> > > >>>
> > > >>> [...]
> > > >>>
> > > >>>
> > > >>
> > > >>
> > > >> 1) I download the 6 channels sample wav file form the follow link:
> > > >>
> > > >>
> > http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples/Microsoft/6_Channel_ID.wav
> > > >>
> > > >> then acoording to the new patch, I run the command:
> > > >> ./ffmpeg -i 6_Channel_ID.wav output_ch6.wav
> > > >>
> > > >> the program run normaly, here is the output result.
> > > >>
> > > >> Input #0, wav, from '6_Channel_ID.wav':
> > > >> Duration: 00:00:05.83, bitrate: 4233 kb/s
> > > >> Stream #0.0: Audio: pcm_s16le, 44100 Hz, 6 channels
> > > >> (FL|FR|FC|LFE|BL|BR), s16, 4233 kb/s
> > > >> Output #0, wav, to 'output_ch6.wav':
> > > >> Stream #0.0: Audio: pcm_s16le, 44100 Hz, 6 channels
> > > >> (FL|FR|FC|LFE|BL|BR), s16, 4233 kb/s
> > > >> Stream mapping:
> > > >> Stream #0.0 -> #0.0
> > > >> Press [q] to stop encoding
> > > >> size= 3017kB time=5.84 bitrate=4233.7kbits/s
> > > >> video:0kB audio:3017kB global headers:0kB muxing overhead 0.002201%
> > > >>
> > > >> 2) The file 6_Channel_ID.wav can be palyed by windows mediaplayer, but
> > the
> > > >> output_ch6.wav can't.
> > > >>
> > > >> I compared the two wav files and found that 6_Channel_ID.wav has 60
> > more
> > > >> bytes than output_ch6.wav
> > > >> the 60 bytes are after WAVEFORMATEXTENSIBLE sturct,followed by the
> > data.
> > > >> Except for the 60 bytes, the two files are all the same.
> > > >>
> > > >> However, I noticed that the sample website says about 6_Channel_ID.wav
> > > >> that
> > > >> :"This file has a "cue " chunk with a count of zero cue points,
> > followed
> > > >> by
> > > >> two empty cue point structures."
> > > >>
> > > >> my question is:
> > > >> are the 60 bytes cue chunk? and how to handle this?
> > > >>
> > > >> zhentan feng
> > > >> thanks
> > > >>
> > > >>
> > > >
> > > > Locate and look at the GUID in the original and the ffmpeg muxed file.
> > They
> > > > most likely differ.
> > > >
> > > > [..]
> > > >
> > >
> > > yes. it's the GUID problem.
> > > thanks.
> > > here is the new patch.
> > [...]
> > > @@ -351,10 +357,21 @@
> > > put_le16(pb, 2); /* wav_extra_size */
> > > hdrsize += 2;
> > > put_le16(pb, enc->frame_size); /* wSamplesPerBlock */
> > > - } else if(enc->extradata_size){
> > > - put_le16(pb, enc->extradata_size);
> > > + } else if(enc->extradata_size || waveformatextensible){
> > > + if (waveformatextensible) { /* write
> > WAVEFORMATEXTENSIBLE extensions */
> > > + put_le16(pb, enc->extradata_size+22); /* 22 is the
> > size of WAVEFORMATEXTENSIBLE-WAVEFORMATEX */
> > > + put_le16(pb, enc->bits_per_coded_sample); /*
> > ValidBitsPerSample || SamplesPerBlock || Reserved */
> > > + put_le32(pb, enc->channel_layout); /* dwChannelMask
> > */
> > > + put_le32(pb, enc->codec_tag); /* GUID + next 3
> > */
> > > + put_le32(pb, 0x00100000);
> > > + put_le32(pb, 0xAA000080);
> > > + put_le32(pb, 0x719B3800);
> > > + hdrsize += enc->extradata_size+22;
> > > + } else {
> > > + put_le16(pb, enc->extradata_size);
> > > + hdrsize += enc->extradata_size;
> > > + }
> >
> > what happens when waveformatextensible is 1 but one of the previous
> > (else) if is true?
> > does that fail or can that not happen at all?
> >
>
> here is new patch.
> if(waveformatextensible) is split from the if else chain.
>
>
> >
> > > put_buffer(pb, enc->extradata, enc->extradata_size);
> > > - hdrsize += enc->extradata_size;
> > > if(hdrsize&1){
> >
> > id leave that and just add 22 in the if()
>
>
> fixed.
>
>
> --
> Best wishes~
> Index: libavformat/riff.c
> ===================================================================
> --- libavformat/riff.c (revision 18184)
> +++ libavformat/riff.c (working copy)
> @@ -282,12 +282,19 @@
> int put_wav_header(ByteIOContext *pb, AVCodecContext *enc)
> {
> int bps, blkalign, bytespersec;
> + int waveformatextensible = 0;
> int hdrsize = 18;
> + int flg = 0;
>
> if(!enc->codec_tag || enc->codec_tag > 0xffff)
> return -1;
>
> - put_le16(pb, enc->codec_tag);
> + if (enc->channels > 2 && enc->channel_layout) {
> + put_le16(pb, 0xfffe);
> + waveformatextensible = 1;
> + } else
> + put_le16(pb, enc->codec_tag);
> +
waveformatextensible = enc->channels > 2 && enc->channel_layout;
if(waveformatextensible) put_le16(pb, 0xfffe);
else put_le16(pb, enc->codec_tag);
> put_le16(pb, enc->channels);
> put_le32(pb, enc->sample_rate);
> if (enc->codec_id == CODEC_ID_MP2 || enc->codec_id == CODEC_ID_MP3 || enc->codec_id == CODEC_ID_GSM_MS) {
> @@ -351,17 +358,32 @@
> put_le16(pb, 2); /* wav_extra_size */
> hdrsize += 2;
> put_le16(pb, enc->frame_size); /* wSamplesPerBlock */
> - } else if(enc->extradata_size){
> - put_le16(pb, enc->extradata_size);
> - put_buffer(pb, enc->extradata, enc->extradata_size);
> - hdrsize += enc->extradata_size;
> - if(hdrsize&1){
> - hdrsize++;
> - put_byte(pb, 0);
> + } else
> + flg = 1;
> +
> + if (waveformatextensible) { /* write WAVEFORMATEXTENSIBLE extensions */
> + put_le16(pb, enc->extradata_size+22); /* 22 is the size of WAVEFORMATEXTENSIBLE-WAVEFORMATEX */
> + put_le16(pb, enc->bits_per_coded_sample); /* ValidBitsPerSample || SamplesPerBlock || Reserved */
> + put_le32(pb, enc->channel_layout); /* dwChannelMask */
> + put_le32(pb, enc->codec_tag); /* GUID + next 3 */
> + put_le32(pb, 0x00100000);
> + put_le32(pb, 0xAA000080);
> + put_le32(pb, 0x719B3800);
> + hdrsize += 22;
> + }
> + if (flg == 1) {
> + if (enc->extradata_size){
> + put_le16(pb, enc->extradata_size);
> + hdrsize += enc->extradata_size;
> + put_buffer(pb, enc->extradata, enc->extradata_size);
> + if(hdrsize&1){
> + hdrsize++;
> + put_byte(pb, 0);
> + }
> + } else {
> + hdrsize -= 2;
> }
> - } else {
> - hdrsize -= 2;
> - }
> + }
>
> return hdrsize;
> }
looks broken when extradata_size is not 0
[...]
--
Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
Asymptotically faster algorithms should always be preferred if you have
asymptotical amounts of data
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