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From: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
To: ffmpeg-devel@ffmpeg.org
Subject: Re: [FFmpeg-devel] [PATCH] avcodec/binkaudio: add support for >2 channels dct codec
Date: Fri, 18 Mar 2022 16:03:40 +0100
Message-ID: <AS1PR01MB956477A47F15F92F1C0993438F139@AS1PR01MB9564.eurprd01.prod.exchangelabs.com> (raw)
In-Reply-To: <20220318130417.47935-1-onemda@gmail.com>

Paul B Mahol:
> As presented in .binka files.
> 
> Signed-off-by: Paul B Mahol <onemda@gmail.com>
> ---
>  libavcodec/binkaudio.c | 50 +++++++++++++++++++++++++++---------------
>  1 file changed, 32 insertions(+), 18 deletions(-)
> 
> diff --git a/libavcodec/binkaudio.c b/libavcodec/binkaudio.c
> index b4ff15beeb..54b7e22854 100644
> --- a/libavcodec/binkaudio.c
> +++ b/libavcodec/binkaudio.c
> @@ -51,13 +51,14 @@ typedef struct BinkAudioContext {
>      int version_b;          ///< Bink version 'b'
>      int first;
>      int channels;
> +    int ch_offset;
>      int frame_len;          ///< transform size (samples)
>      int overlap_len;        ///< overlap size (samples)
>      int block_size;
>      int num_bands;
>      float root;
>      unsigned int bands[26];
> -    float previous[MAX_CHANNELS][BINK_BLOCK_MAX_SIZE / 16];  ///< coeffs from previous audio block
> +    float previous[6][BINK_BLOCK_MAX_SIZE / 16];  ///< coeffs from previous audio block
>      float quant_table[96];
>      AVPacket *pkt;
>      union {
> @@ -74,6 +75,7 @@ static av_cold int decode_init(AVCodecContext *avctx)
>      int sample_rate_half;
>      int i, ret;
>      int frame_len_bits;
> +    int max_channels = avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT ? MAX_CHANNELS : 6;

If you allow up to six channels, then MAX_CHANNELS (i.e. two) needs to
be renamed.

>      int channels = avctx->ch_layout.nb_channels;
>  
>      /* determine frame length */
> @@ -85,7 +87,7 @@ static av_cold int decode_init(AVCodecContext *avctx)
>          frame_len_bits = 11;
>      }
>  
> -    if (channels < 1 || channels > MAX_CHANNELS) {
> +    if (channels < 1 || channels > max_channels) {
>          av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n", channels);
>          return AVERROR_INVALIDDATA;
>      }
> @@ -110,7 +112,7 @@ static av_cold int decode_init(AVCodecContext *avctx)
>  
>      s->frame_len     = 1 << frame_len_bits;
>      s->overlap_len   = s->frame_len / 16;
> -    s->block_size    = (s->frame_len - s->overlap_len) * s->channels;
> +    s->block_size    = (s->frame_len - s->overlap_len) * FFMIN(MAX_CHANNELS, s->channels);
>      sample_rate_half = (sample_rate + 1LL) / 2;
>      if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
>          s->root = 2.0 / (sqrt(s->frame_len) * 32768.0);
> @@ -166,7 +168,8 @@ static const uint8_t rle_length_tab[16] = {
>   * @param[out] out Output buffer (must contain s->block_size elements)
>   * @return 0 on success, negative error code on failure
>   */
> -static int decode_block(BinkAudioContext *s, float **out, int use_dct)
> +static int decode_block(BinkAudioContext *s, float **out, int use_dct,
> +                        int channels, int ch_offset)
>  {
>      int ch, i, j, k;
>      float q, quant[25];
> @@ -176,8 +179,8 @@ static int decode_block(BinkAudioContext *s, float **out, int use_dct)
>      if (use_dct)
>          skip_bits(gb, 2);
>  
> -    for (ch = 0; ch < s->channels; ch++) {
> -        FFTSample *coeffs = out[ch];
> +    for (ch = 0; ch < channels; ch++) {
> +        FFTSample *coeffs = out[ch + ch_offset];
>  
>          if (s->version_b) {
>              if (get_bits_left(gb) < 64)
> @@ -252,17 +255,17 @@ static int decode_block(BinkAudioContext *s, float **out, int use_dct)
>              s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs);
>      }
>  
> -    for (ch = 0; ch < s->channels; ch++) {
> +    for (ch = 0; ch < channels; ch++) {
>          int j;
> -        int count = s->overlap_len * s->channels;
> +        int count = s->overlap_len * channels;
>          if (!s->first) {
>              j = ch;
> -            for (i = 0; i < s->overlap_len; i++, j += s->channels)
> -                out[ch][i] = (s->previous[ch][i] * (count - j) +
> -                                      out[ch][i] *          j) / count;
> +            for (i = 0; i < s->overlap_len; i++, j += channels)
> +                out[ch + ch_offset][i] = (s->previous[ch + ch_offset][i] * (count - j) +
> +                                      out[ch + ch_offset][i] *          j) / count;
>          }
> -        memcpy(s->previous[ch], &out[ch][s->frame_len - s->overlap_len],
> -               s->overlap_len * sizeof(*s->previous[ch]));
> +        memcpy(s->previous[ch + ch_offset], &out[ch + ch_offset][s->frame_len - s->overlap_len],
> +               s->overlap_len * sizeof(*s->previous[ch + ch_offset]));
>      }
>  
>      s->first = 0;
> @@ -293,6 +296,7 @@ static int binkaudio_receive_frame(AVCodecContext *avctx, AVFrame *frame)
>      GetBitContext *gb = &s->gb;
>      int ret;
>  
> +again:
>      if (!s->pkt->data) {
>          ret = ff_decode_get_packet(avctx, s->pkt);
>          if (ret < 0)
> @@ -313,22 +317,31 @@ static int binkaudio_receive_frame(AVCodecContext *avctx, AVFrame *frame)
>      }
>  
>      /* get output buffer */
> -    frame->nb_samples = s->frame_len;
> -    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
> -        return ret;
> +    if (s->ch_offset == 0) {
> +        frame->nb_samples = s->frame_len;
> +        if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
> +            return ret;
> +    }
>  
>      if (decode_block(s, (float **)frame->extended_data,
> -                     avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT)) {
> +                     avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT,
> +                     FFMIN(MAX_CHANNELS, s->channels), s->ch_offset)) {
>          av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n");
>          return AVERROR_INVALIDDATA;
>      }
> +    s->ch_offset += MAX_CHANNELS;
>      get_bits_align32(gb);
>      if (!get_bits_left(gb)) {
>          memset(gb, 0, sizeof(*gb));
>          av_packet_unref(s->pkt);
>      }
> +    if (s->ch_offset >= s->channels) {
> +        s->ch_offset = 0;
> +    } else {
> +        goto again;
> +    }

Is it really intended that the data for one multi-channel frame is
divided into several input packets?

>  
> -    frame->nb_samples = s->block_size / avctx->ch_layout.nb_channels;
> +    frame->nb_samples = s->block_size / FFMIN(avctx->ch_layout.nb_channels, MAX_CHANNELS);
>  
>      return 0;
>  fail:
> @@ -343,6 +356,7 @@ static void decode_flush(AVCodecContext *avctx)
>      /* s->pkt coincides with avctx->internal->in_pkt
>       * and is unreferenced generically when flushing. */
>      s->first = 1;
> +    s->ch_offset = 0;
>  }
>  
>  const AVCodec ff_binkaudio_rdft_decoder = {

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  reply	other threads:[~2022-03-18 15:03 UTC|newest]

Thread overview: 4+ messages / expand[flat|nested]  mbox.gz  Atom feed  top
2022-03-18 13:04 Paul B Mahol
2022-03-18 15:03 ` Andreas Rheinhardt [this message]
2022-03-18 15:21   ` Paul B Mahol
2022-03-20  4:37     ` Peter Ross

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