[FFmpeg-devel] [PATCH] avformat/dv : change audio format endianess to BE

[Michael Niedermayer]

On Tue, Oct 17, 2017 at 08:38:21PM +0300, Александр Слободенюк wrote:
> Hello again!
> 
> I  have checked (updated and fixed) the patch on BE machine (qemu-mips with BE linux kernel, lscpu shows big endian): the CRCs of the sound are the
> same:
> 
> (first link was get directly from http://www.mediafire.com/file/8agcdgs2f9kqlbi/submarine.dv)
> wget http://download1643.mediafireuserdownload.com/h3we3e4t26tg/8agcdgs2f9kqlbi/submarine.dv
> wget https://trac.ffmpeg.org/raw-attachment/ticket/1589/A1590.dv
> 
> ./ffmpeg -i submarine.dv audio-submarine.wav
> ./ffmpeg -i A1589.dv audio-12bit.wav
> 
> (I  had  to  extract the audio out because with video CRCs are not the
> same on BE and LE machines on master version of ffmpeg)
> 
> ./ffmpeg -i audio-submarine.wav -f crc - 2>/dev/null
> CRC=0x96b2ce8c
> 
> ./ffmpeg -i audio-12bit.wav -f crc - 2>/dev/null
> CRC=0x01c9318a
> 
> ----------
> So, this CRCs are the same for all of the variants:
> 
> - not patched ffmpeg on LE machine
> - patched ffmpeg on LE machine
> - not patched ffmpeg on BE machine
> - patched ffmpeg on BE machine
> ----------
> 
> Attaching the updated to current master patch.

>  dv.c      |   30 ++++++++++--------------------
>  dvenc.c   |    9 ++++-----
>  mov.c     |    2 +-
>  version.h |    2 +-
>  4 files changed, 16 insertions(+), 27 deletions(-)
> 9e8192ee964aaf97bc14ab13f10718cf6bfd1e45  0001-avformat-dv-change-audio-format-endianess-to-BE.patch
> From 4cdb0d8b79d91c94480c292847311e51f1b25704 Mon Sep 17 00:00:00 2001
> From: Aleksandr Slobodeniuk <alenuke at yandex.ru>
> Date: Tue, 17 Oct 2017 17:59:30 +0300

> Subject: [PATCH] avformat/dv : change audio format endianess to BE

This does not explain why this change is done.
does it fix a bug?
does it make the code faster?

The commit message should explain this

> 
> ---
>  libavformat/dv.c      | 30 ++++++++++--------------------
>  libavformat/dvenc.c   |  9 ++++-----
>  libavformat/mov.c     |  2 +-
>  libavformat/version.h |  2 +-
>  4 files changed, 16 insertions(+), 27 deletions(-)
> 
> diff --git a/libavformat/dv.c b/libavformat/dv.c
> index 06de044..df5fec0 100644
> --- a/libavformat/dv.c
> +++ b/libavformat/dv.c
> @@ -116,13 +116,13 @@ static const int dv_audio_frequency[3] = {
>   * 3. Audio is always returned as 16-bit linear samples: 12-bit nonlinear samples
>   *    are converted into 16-bit linear ones.
>   */
> -static int dv_extract_audio(const uint8_t *frame, uint8_t **ppcm,
> +static int dv_extract_audio(const uint8_t *frame, uint16_t **ppcm,
>                              const AVDVProfile *sys)
>  {
>      int size, chan, i, j, d, of, smpls, freq, quant, half_ch;
> -    uint16_t lc, rc;
> +    uint16_t lc, rc, *pcm;
>      const uint8_t *as_pack;
> -    uint8_t *pcm, ipcm;
> +    uint8_t ipcm;
>  
>      as_pack = dv_extract_pack(frame, dv_audio_source);
>      if (!as_pack)    /* No audio ? */
> @@ -177,13 +177,9 @@ static int dv_extract_audio(const uint8_t *frame, uint8_t **ppcm,
>                          if (of * 2 >= size)
>                              continue;
>  
> -                        /* FIXME: maybe we have to admit that DV is a
> -                         * big-endian PCM */
> -                        pcm[of * 2]     = frame[d + 1];
> -                        pcm[of * 2 + 1] = frame[d];
> -
> -                        if (pcm[of * 2 + 1] == 0x80 && pcm[of * 2] == 0x00)
> -                            pcm[of * 2 + 1] = 0;

> +                        pcm[of] = ((uint16_t*)frame)[d/2];

Divisions should be avoided in the inner loops as they can be slow

[...]

-- 
Michael     GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB

Awnsering whenever a program halts or runs forever is
On a turing machine, in general impossible (turings halting problem).
On any real computer, always possible as a real computer has a finite number
of states N, and will either halt in less than N cycles or never halt.
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