From: Paul B Mahol <onemda@gmail.com> To: FFmpeg development discussions and patches <ffmpeg-devel@ffmpeg.org> Subject: Re: [FFmpeg-devel] [PATCH] avcodec/binkaudio: add support for >2 channels dct codec Date: Fri, 18 Mar 2022 16:21:44 +0100 Message-ID: <CAPYw7P50hz16JXtUYTnrM9=xmYbXRFon1Z-pWZb_xbrKrGmL8w@mail.gmail.com> (raw) In-Reply-To: <AS1PR01MB956477A47F15F92F1C0993438F139@AS1PR01MB9564.eurprd01.prod.exchangelabs.com> On 3/18/22, Andreas Rheinhardt <andreas.rheinhardt@outlook.com> wrote: > 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? You are missing big picture here, >2 files have channels in different packets interleaved. Something like in XMA. (And nothing signals how are they interleaved. so its worse than in XMA) So it is working fine. I just need another look for possible regressions and security implications. Renaming MAX_CHANNELS is not useful as that is not property of both codecs. > >> >> - 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 = { > > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > https://ffmpeg.org/mailman/listinfo/ffmpeg-devel > > To unsubscribe, visit link above, or email > ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". > _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe".
next prev parent reply other threads:[~2022-03-18 15:21 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 2022-03-18 15:21 ` Paul B Mahol [this message] 2022-03-20 4:37 ` Peter Ross
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