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 = { _______________________________________________ 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 15:03 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
Reply instructions: You may reply publicly to this message via plain-text email using any one of the following methods: * Save the following mbox file, import it into your mail client, and reply-to-all from there: mbox Avoid top-posting and favor interleaved quoting: https://en.wikipedia.org/wiki/Posting_style#Interleaved_style * Reply using the --to, --cc, and --in-reply-to switches of git-send-email(1): git send-email \ --in-reply-to=AS1PR01MB956477A47F15F92F1C0993438F139@AS1PR01MB9564.eurprd01.prod.exchangelabs.com \ --to=andreas.rheinhardt@outlook.com \ --cc=ffmpeg-devel@ffmpeg.org \ /path/to/YOUR_REPLY https://kernel.org/pub/software/scm/git/docs/git-send-email.html * If your mail client supports setting the In-Reply-To header via mailto: links, try the mailto: link
Git Inbox Mirror of the ffmpeg-devel mailing list - see https://ffmpeg.org/mailman/listinfo/ffmpeg-devel This inbox may be cloned and mirrored by anyone: git clone --mirror https://master.gitmailbox.com/ffmpegdev/0 ffmpegdev/git/0.git # If you have public-inbox 1.1+ installed, you may # initialize and index your mirror using the following commands: public-inbox-init -V2 ffmpegdev ffmpegdev/ https://master.gitmailbox.com/ffmpegdev \ ffmpegdev@gitmailbox.com public-inbox-index ffmpegdev Example config snippet for mirrors. AGPL code for this site: git clone https://public-inbox.org/public-inbox.git