[FFmpeg-devel] [PATCH] adpcm-ima-wav header and codec

[Timofei V. Bondarenko]

Michael Niedermayer wrote:
> On Mon, Oct 15, 2007 at 07:55:29PM +0400, Timofei wrote:
>> Hi.
>>
>> I found a few bugs in the adpcm-ima codec.
>> This format is used in smv video by sigmatel-based mp3 players.
>>
>> 1) The field 'bytes-per-second' in WAV header is set incorrectly.
>> This breaks seeking on hardware player.
>> I found the bitrate value used in calculations comes from commandline and 
>> does not reflect reality.
>> The adpcm-bitrate.patch puts correct bitrate in avctx.
>> This fixes wav header, but the ffmpeg program still prints wrong bitrate
>> during encoding.

[...]

>> --- ffmpeg-svn/libavcodec/adpcm.c	2007-10-12 12:21:16.000000000 +0400
>> +++ ffmpeg/libavcodec/adpcm.c	2007-10-12 19:01:53.000000000 +0400
>> @@ -166,6 +166,7 @@
>>                                                               /* and we have 4 bytes per channel overhead */
>>          avctx->block_align = BLKSIZE;
>>          /* seems frame_size isn't taken into account... have to buffer the samples :-( */
>> +	avctx->bit_rate = avctx->sample_rate * avctx->block_align * 8 / avctx->frame_size;
> 
> tabs are forbidden in ffmpeg svn
> and
>      /**
>      * the average bitrate
>      * - encoding: Set by user; unused for constant quantizer encoding.
>      * - decoding: Set by libavcodec. 0 or some bitrate if this info is available in the stream.
>      */
>     int bit_rate;
> 
> that is its set by the user not by libavcodec acording to API

Well. How to fix the header then?
Is it better to put one more nail into avformat/riff.c (see attachment).
And one more for printing correct bitrate.

>> --- ffmpeg-svn/libavcodec/adpcm.c	2007-10-12 12:21:16.000000000 +0400
>> +++ ffmpeg/libavcodec/adpcm.c	2007-10-12 19:01:53.000000000 +0400
>> @@ -448,19 +449,17 @@
>>          break;
>>      case CODEC_ID_ADPCM_IMA_WAV:
>>          n = avctx->frame_size / 8;
>> -            c->status[0].prev_sample = (signed short)samples[0]; /* XXX */
>> +            c->status[0].prev_sample = *samples++;
>>  /*            c->status[0].step_index = 0; *//* XXX: not sure how to init the state machine */
>>              bytestream_put_le16(&dst, c->status[0].prev_sample);
>>              *dst++ = (unsigned char)c->status[0].step_index;
>>              *dst++ = 0; /* unknown */
>> -            samples++;
> 
> this change is purely cosmetic and belongs in a seperate patch

Agree. See adpcm-codec2.patch

> 
> 
> [...]
> 
>> --- ffmpeg-svn/libavcodec/adpcm.c	2007-10-12 12:21:16.000000000 +0400
>> +++ ffmpeg/libavcodec/adpcm.c	2007-10-12 19:01:53.000000000 +0400
>> @@ -483,17 +483,17 @@
> 
> looks ok

BTW these changes break regression tests. It is intentional. Should i do 
something about?

> 
> 
>> --- ffmpeg-svn/libavcodec/adpcm.c	2007-10-12 12:21:16.000000000 +0400
>> +++ ffmpeg/libavcodec/adpcm.c	2007-10-12 19:01:53.000000000 +0400
>> @@ -220,6 +220,16 @@ static inline unsigned char adpcm_ima_co
>>      return nibble;
>>  }
>>  
>> +static void inline adpcm_ima_init_step_index(ADPCMChannelStatus *c, short next_sample)
> 
> i dont think that this should be inline, or what is the speed gain from that?
> 

Right. Look at another try.

Regards.
	Tim.
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