On Mon, Jan 02, 2012 at 04:54:52PM +0530, Shitiz Garg wrote:
> --- a/libavcodec/dca.c
> +++ b/libavcodec/dca.c
> @@ -129,26 +129,42 @@ static const int dca_ext_audio_descr_mask[] = {
>  
> +    AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_SIDE_LEFT |
> +               AV_CH_SIDE_RIGHT,                                            
> ///< 5, C + L + R + SL + SR
> +
> +    AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |
> +               AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER,    
> ///< 6, CL + CR + L + R + SL + SR
> +
> +    AV_CH_LAYOUT_STEREO | AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT |
> +               AV_CH_FRONT_CENTER | AV_CH_BACK_CENTER,                      
> ///< 6, C + L + R + LR + RR + OV
> +
> +    AV_CH_FRONT_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER | 
> AV_CH_FRONT_LEFT_OF_CENTER |
> +               AV_CH_BACK_CENTER | AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT,      
> ///< 6, CF + CR + LF + RF + LR + RR
> +
> +    AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER | 
> AV_CH_FRONT_RIGHT_OF_CENTER |
> +               AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT,    
> ///< 7, CL + C + CR + L + R + SL + SR
> +
> +    AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER | 
> AV_CH_LAYOUT_STEREO |
> +               AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT | AV_CH_BACK_LEFT |
> +               AV_CH_BACK_RIGHT, ///< 8, CL + CR + L + R + SL1 + SL2 + SR1 + 
> SR2
> +
> +    AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER | 
> AV_CH_FRONT_RIGHT_OF_CENTER |
> +               AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_BACK_CENTER |
> +               AV_CH_SIDE_RIGHT, ///< 8, CL + C + CR + L + R + SL + S + SR

Why do you indent by a random amount (11) of spaces here?

> @@ -416,13 +432,16 @@ static av_cold void dca_init_vlcs(void)
>  
>      for (i = 0; i < 10; i++)
> -        for (j = 0; j < 7; j++){
> -            if (!bitalloc_codes[i][j]) break;
> -            dca_smpl_bitalloc[i+1].offset = bitalloc_offsets[i];
> -            dca_smpl_bitalloc[i+1].wrap = 1 + (j > 4);
> -            dca_smpl_bitalloc[i+1].vlc[j].table = 
> &dca_table[dca_vlc_offs[c]];
> -            dca_smpl_bitalloc[i+1].vlc[j].table_allocated = dca_vlc_offs[c + 
> 1] - dca_vlc_offs[c];
> -            init_vlc(&dca_smpl_bitalloc[i+1].vlc[j], bitalloc_maxbits[i][j],
> +        for (j = 0; j < 7; j++) {
> +            if (!bitalloc_codes[i][j])
> +                break;
> +            dca_smpl_bitalloc[i + 1].offset                 = 
> bitalloc_offsets[i];
> +            dca_smpl_bitalloc[i + 1].wrap                   = 1 + (j > 4);
> +            dca_smpl_bitalloc[i + 1].vlc[j].table           = 
> &dca_table[dca_vlc_offs[c]];
> +            dca_smpl_bitalloc[i + 1].vlc[j].table_allocated =
> +                                        dca_vlc_offs[c + 1] - 
> dca_vlc_offs[c];

Random indentation again.

IMO don't break the line at all or break after '-'.

> @@ -461,10 +480,10 @@ static int dca_parse_audio_coding_header(DCAContext * 
> s, int base_channel)
> -    get_array(&s->gb, s->joint_intensity + base_channel,     
> s->prim_channels - base_channel, 3);
> -    get_array(&s->gb, s->transient_huffman + base_channel,   
> s->prim_channels - base_channel, 2);
> +    get_array(&s->gb, s->joint_intensity + base_channel    , 
> s->prim_channels - base_channel, 3);
> +    get_array(&s->gb, s->transient_huffman + base_channel  , 
> s->prim_channels - base_channel, 2);
>      get_array(&s->gb, s->scalefactor_huffman + base_channel, 
> s->prim_channels - base_channel, 3);
> -    get_array(&s->gb, s->bitalloc_huffman + base_channel,    
> s->prim_channels - base_channel, 3);
> +    get_array(&s->gb, s->bitalloc_huffman + base_channel   , 
> s->prim_channels - base_channel, 3);

correct before

> @@ -792,7 +818,9 @@ static int dca_subframe_header(DCAContext * s, int 
> base_channel, int block_index
>      if (!base_channel && s->lfe) {
>          /* LFE samples */
>          int lfe_samples = 2 * s->lfe * (4 + block_index);
> -        int lfe_end_sample = 2 * s->lfe * (4 + block_index + 
> s->subsubframes[s->current_subframe]);
> +        int lfe_end_sample =
> +                2 * s->lfe * (4 + block_index + 
> s->subsubframes[s->current_subframe]);

Random again - just leave the line as-is.

> @@ -1119,20 +1159,21 @@ static int dca_subsubframe(DCAContext * s, int 
> base_channel, int block_index)
>  
> -                        size = abits_sizes[abits-1];
> -                        levels = abits_levels[abits-1];
> +                        size = abits_sizes[abits - 1];
> +                        levels = abits_levels[abits - 1];

align

> @@ -1373,17 +1416,17 @@ static int dca_exss_mask2count(int mask)
>  {
>      /* count bits that mean speaker pairs twice */
>      return av_popcount(mask)
> -        + av_popcount(mask & (
> -            DCA_EXSS_CENTER_LEFT_RIGHT
> -          | DCA_EXSS_FRONT_LEFT_RIGHT
> -          | DCA_EXSS_FRONT_HIGH_LEFT_RIGHT
> -          | DCA_EXSS_WIDE_LEFT_RIGHT
> -          | DCA_EXSS_SIDE_LEFT_RIGHT
> -          | DCA_EXSS_SIDE_HIGH_LEFT_RIGHT
> -          | DCA_EXSS_SIDE_REAR_LEFT_RIGHT
> -          | DCA_EXSS_REAR_LEFT_RIGHT
> -          | DCA_EXSS_REAR_HIGH_LEFT_RIGHT
> -          ));
> +           + av_popcount(mask & (
> +                               DCA_EXSS_CENTER_LEFT_RIGHT
> +                             | DCA_EXSS_FRONT_LEFT_RIGHT
> +                             | DCA_EXSS_FRONT_HIGH_LEFT_RIGHT
> +                             | DCA_EXSS_WIDE_LEFT_RIGHT
> +                             | DCA_EXSS_SIDE_LEFT_RIGHT
> +                             | DCA_EXSS_SIDE_HIGH_LEFT_RIGHT
> +                             | DCA_EXSS_SIDE_REAR_LEFT_RIGHT
> +                             | DCA_EXSS_REAR_LEFT_RIGHT
> +                             | DCA_EXSS_REAR_HIGH_LEFT_RIGHT
> +                             ));

Random again and don't leave the empty parentheses on lines of their own.

> @@ -1483,28 +1525,28 @@ static int dca_exss_parse_asset_header(DCAContext *s)
>  
>      drc_code_present = get_bits1(&s->gb);
>      if (drc_code_present)
> -        get_bits(&s->gb, 8); // drc code
> +        get_bits(&s->gb, 8);  // drc code
>  
>      if (get_bits1(&s->gb))
> -        skip_bits(&s->gb, 5); // dialog normalization code
> +        skip_bits(&s->gb, 5);  // dialog normalization code
>  
>      if (drc_code_present && embedded_stereo)
> -        get_bits(&s->gb, 8); // drc stereo code
> +        get_bits(&s->gb, 8);  // drc stereo code
>  
>      if (s->mix_metadata && get_bits1(&s->gb)) {
>          skip_bits(&s->gb, 1); // external mix
>          skip_bits(&s->gb, 6); // post mix gain code
>  
>          if (get_bits(&s->gb, 2) != 3) // mixer drc code
> -            skip_bits(&s->gb, 3); // drc limit
> +            skip_bits(&s->gb, 3);  // drc limit
>          else
> -            skip_bits(&s->gb, 8); // custom drc code
> +            skip_bits(&s->gb, 8);  // custom drc code
>  
>          if (get_bits1(&s->gb)) // channel specific scaling
>              for (i = 0; i < s->num_mix_configs; i++)
> -                skip_bits_long(&s->gb, s->mix_config_num_ch[i] * 6); // 
> scale codes
> +                skip_bits_long(&s->gb, s->mix_config_num_ch[i] * 6);  // 
> scale codes
>          else
> -            skip_bits_long(&s->gb, s->num_mix_configs * 6); // scale codes
> +            skip_bits_long(&s->gb, s->num_mix_configs * 6);  // scale codes

Why?

> @@ -1778,15 +1818,15 @@ static int dca_decode_frame(AVCodecContext *avctx, 
> void *data,
>          s->profile = FF_PROFILE_DTS_ES;
>  
>      /* check for ExSS (HD part) */
> -    if (s->dca_buffer_size - s->frame_size > 32
> -        && get_bits_long(&s->gb, 32) == DCA_HD_MARKER)
> +    if (s->dca_buffer_size - s->frame_size > 32 &&
> +                get_bits_long(&s->gb, 32) == DCA_HD_MARKER)

random - not that there was a need to change in the first place ...

> @@ -1809,7 +1849,7 @@ static int dca_decode_frame(AVCodecContext *avctx, void 
> *data,
>          }
>  
>          if (channels > !!s->lfe &&
> -            s->channel_order_tab[channels - 1 - !!s->lfe] < 0)
> +                s->channel_order_tab[channels - 1 - !!s->lfe] < 0)
>              return AVERROR_INVALIDDATA;

more

> @@ -1853,10 +1893,10 @@ static int dca_decode_frame(AVCodecContext *avctx, 
> void *data,
>  
> -        if((s->source_pcm_res & 1) && s->xch_present) {
> -            float* back_chan = s->samples + 
> s->channel_order_tab[s->xch_base_channel] * 256;
> -            float* lt_chan   = s->samples + 
> s->channel_order_tab[s->xch_base_channel - 2] * 256;
> -            float* rt_chan   = s->samples + 
> s->channel_order_tab[s->xch_base_channel - 1] * 256;
> +        if ((s->source_pcm_res & 1) && s->xch_present) {
> +            float *back_chan = s->samples + 
> s->channel_order_tab[s->xch_base_channel] * 256;
> +            float *lt_chan   = s->samples + 
> s->channel_order_tab[s->xch_base_channel - 2] * 256;
> +            float *rt_chan   = s->samples + 
> s->channel_order_tab[s->xch_base_channel - 1] * 256;

nit: Align the '*'.

> @@ -1947,17 +1986,18 @@ static const AVProfile profiles[] = {
>  
>  AVCodec ff_dca_decoder = {
> -    .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
> -    .sample_fmts = (const enum AVSampleFormat[]) {
> -        AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
> -    },
> +    .capabilities       = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
> +    .sample_fmts        = (const enum AVSampleFormat[]) {
> +                                AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16,
> +                                AV_SAMPLE_FMT_NONE
> +                          },

more randomness

Diego
_______________________________________________
libav-devel mailing list
[email protected]
https://lists.libav.org/mailman/listinfo/libav-devel

Reply via email to