From: Andreas Rheinhardt <andreas.rheinhardt@outlook.com> To: ffmpeg-devel@ffmpeg.org Subject: Re: [FFmpeg-devel] [PATCH v4 1/2] libavcodec/flacdec: Implement decoding of 32 bit-per-sample PCM Date: Wed, 14 Sep 2022 15:03:08 +0200 Message-ID: <GV1P250MB073709D6FFD01F5CFF122A518F469@GV1P250MB0737.EURP250.PROD.OUTLOOK.COM> (raw) In-Reply-To: <20220903092144.569716-2-mvanb1@gmail.com> Martijn van Beurden: > Add decoding of FLAC files coding for 32 bit-per-sample PCM to libavcodec. > --- > libavcodec/flac.c | 4 +- > libavcodec/flacdec.c | 248 ++++++++++++++++++++++++++++++++++++++---- > libavcodec/get_bits.h | 12 ++ > libavcodec/mathops.h | 9 ++ > 4 files changed, 250 insertions(+), 23 deletions(-) > > diff --git a/libavcodec/flac.c b/libavcodec/flac.c > index 352d663c67..174b4801be 100644 > --- a/libavcodec/flac.c > +++ b/libavcodec/flac.c > @@ -28,7 +28,7 @@ > #include "flacdata.h" > #include "flac_parse.h" > > -static const int8_t sample_size_table[] = { 0, 8, 12, 0, 16, 20, 24, 0 }; > +static const int8_t sample_size_table[] = { 0, 8, 12, 0, 16, 20, 24, 32 }; > > static const AVChannelLayout flac_channel_layouts[8] = { > AV_CHANNEL_LAYOUT_MONO, > @@ -82,7 +82,7 @@ int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb, > > /* bits per sample */ > bps_code = get_bits(gb, 3); > - if (bps_code == 3 || bps_code == 7) { > + if (bps_code == 3) { > av_log(avctx, AV_LOG_ERROR + log_level_offset, > "invalid sample size code (%d)\n", > bps_code); > diff --git a/libavcodec/flacdec.c b/libavcodec/flacdec.c > index c5d9e95168..14a9202569 100644 > --- a/libavcodec/flacdec.c > +++ b/libavcodec/flacdec.c > @@ -64,6 +64,9 @@ typedef struct FLACContext { > int32_t *decoded[FLAC_MAX_CHANNELS]; ///< decoded samples > uint8_t *decoded_buffer; > unsigned int decoded_buffer_size; > + int64_t *decoded_33bps; ///< decoded samples for a 33 bps subframe > + uint8_t *decoded_buffer_33bps; Why do you need a new buffer here instead of just reusing decoded_buffer? > + unsigned int decoded_buffer_size_33bps; > int buggy_lpc; ///< use workaround for old lavc encoded files > > FLACDSPContext dsp; > @@ -154,6 +157,24 @@ static int allocate_buffers(FLACContext *s) > s->stream_info.channels, > s->stream_info.max_blocksize, > AV_SAMPLE_FMT_S32P, 0); > + if (ret >= 0 && s->stream_info.bps == 32 && s->stream_info.channels == 2) { > + buf_size = av_samples_get_buffer_size(NULL, 1, > + s->stream_info.max_blocksize, > + AV_SAMPLE_FMT_S64P, 0); > + if (buf_size < 0) > + return buf_size; > + > + av_fast_malloc(&s->decoded_buffer_33bps, &s->decoded_buffer_size_33bps, buf_size); > + if (!s->decoded_buffer) You are checking the wrong variable. > + return AVERROR(ENOMEM); > + > + ret = av_samples_fill_arrays((uint8_t **)&s->decoded_33bps, NULL, > + s->decoded_buffer_33bps, > + 1, > + s->stream_info.max_blocksize, > + AV_SAMPLE_FMT_S64P, 0); > + > + } > return ret < 0 ? ret : 0; > } > > @@ -331,6 +352,94 @@ static int decode_subframe_fixed(FLACContext *s, int32_t *decoded, > return 0; > } > > +static int decode_subframe_fixed_wide(FLACContext *s, int32_t *decoded, > + int pred_order, int bps) > +{ > + const int blocksize = s->blocksize; > + int i; > + int ret; > + > + /* warm up samples */ > + for (i = 0; i < pred_order; i++) { > + decoded[i] = get_sbits_long(&s->gb, bps); > + } > + > + if ((ret = decode_residuals(s, decoded, pred_order)) < 0) > + return ret; > + > + switch (pred_order) { > + case 0: > + break; > + case 1: > + for (i = pred_order; i < blocksize; i++) > + decoded[i] += decoded[i-1]; > + break; > + case 2: > + for (i = pred_order; i < blocksize; i++) > + decoded[i] = (int64_t)decoded[i] + 2*(int64_t)decoded[i-1] - (int64_t)decoded[i-2]; > + break; > + case 3: > + for (i = pred_order; i < blocksize; i++) > + decoded[i] = (int64_t)decoded[i] + 3*(int64_t)decoded[i-1] - 3*(int64_t)decoded[i-2] + (int64_t)decoded[i-3]; > + break; > + case 4: > + for (i = pred_order; i < blocksize; i++) > + decoded[i] = (int64_t)decoded[i] + 4*(int64_t)decoded[i-1] - 6*(int64_t)decoded[i-2] + 4*(int64_t)decoded[i-3] - (int64_t)decoded[i-4]; > + break; > + default: > + av_log(s->avctx, AV_LOG_ERROR, "illegal pred order %d\n", pred_order); > + return AVERROR_INVALIDDATA; > + } > + > + return 0; > +} > + > + > +static int decode_subframe_fixed_33bps(FLACContext *s, int64_t *decoded, > + int32_t *residual, int pred_order) > +{ > + const int blocksize = s->blocksize; > + int i; > + int ret; > + > + /* warm up samples */ > + for (i = 0; i < pred_order; i++) { > + decoded[i] = get_sbits64(&s->gb, 33); > + av_log(s->avctx, AV_LOG_DEBUG, "warm-up %d = %" PRId64 "\n", i, decoded[i]); > + } > + > + if ((ret = decode_residuals(s, residual, pred_order)) < 0) > + return ret; > + > + switch (pred_order) { > + case 0: > + for (i = pred_order; i < blocksize; i++) > + decoded[i] = residual[i]; > + break; > + case 1: > + for (i = pred_order; i < blocksize; i++) > + decoded[i] = residual[i] + decoded[i-1]; > + break; > + case 2: > + for (i = pred_order; i < blocksize; i++) > + decoded[i] = residual[i] + 2*decoded[i-1] - decoded[i-2]; > + break; > + case 3: > + for (i = pred_order; i < blocksize; i++) > + decoded[i] = residual[i] + 3*decoded[i-1] - 3*decoded[i-2] + decoded[i-3]; > + break; > + case 4: > + for (i = pred_order; i < blocksize; i++) > + decoded[i] = residual[i] + 4*decoded[i-1] - 6*decoded[i-2] + 4*decoded[i-3] - decoded[i-4]; > + break; > + default: > + av_log(s->avctx, AV_LOG_ERROR, "illegal pred order %d\n", pred_order); > + return AVERROR_INVALIDDATA; > + } > + > + return 0; > +} > + > static void lpc_analyze_remodulate(SUINT32 *decoded, const int coeffs[32], > int order, int qlevel, int len, int bps) > { > @@ -402,12 +511,53 @@ static int decode_subframe_lpc(FLACContext *s, int32_t *decoded, int pred_order, > return 0; > } > > +static int decode_subframe_lpc_33bps(FLACContext *s, int64_t *decoded, > + int32_t *residual, int pred_order) > +{ > + int i, j, ret; > + int coeff_prec, qlevel; > + int coeffs[32]; > + > + /* warm up samples */ > + for (i = 0; i < pred_order; i++) { > + decoded[i] = get_sbits64(&s->gb, 33); > + } > + > + coeff_prec = get_bits(&s->gb, 4) + 1; > + if (coeff_prec == 16) { > + av_log(s->avctx, AV_LOG_ERROR, "invalid coeff precision\n"); > + return AVERROR_INVALIDDATA; > + } > + qlevel = get_sbits(&s->gb, 5); > + if (qlevel < 0) { > + av_log(s->avctx, AV_LOG_ERROR, "qlevel %d not supported, maybe buggy stream\n", > + qlevel); > + return AVERROR_INVALIDDATA; > + } > + > + for (i = 0; i < pred_order; i++) { > + coeffs[pred_order - i - 1] = get_sbits(&s->gb, coeff_prec); > + } > + > + if ((ret = decode_residuals(s, residual, pred_order)) < 0) > + return ret; > + > + for (i = pred_order; i < s->blocksize; i++, decoded++) { > + int64_t sum = 0; > + for (j = 0; j < pred_order; j++) > + sum += (int64_t)coeffs[j] * decoded[j]; > + decoded[j] = residual[i] + (sum >> qlevel); > + } > + > + return 0; > +} > + > static inline int decode_subframe(FLACContext *s, int channel) > { > int32_t *decoded = s->decoded[channel]; > int type, wasted = 0; > int bps = s->stream_info.bps; > - int i, tmp, ret; > + int i, ret; > > if (channel == 0) { > if (s->ch_mode == FLAC_CHMODE_RIGHT_SIDE) > @@ -436,34 +586,63 @@ static inline int decode_subframe(FLACContext *s, int channel) > wasted = 1 + get_unary(&s->gb, 1, get_bits_left(&s->gb)); > bps -= wasted; > } > - if (bps > 32) { > - avpriv_report_missing_feature(s->avctx, "Decorrelated bit depth > 32"); > - return AVERROR_PATCHWELCOME; > - } > > //FIXME use av_log2 for types > if (type == 0) { > - tmp = get_sbits_long(&s->gb, bps); > - for (i = 0; i < s->blocksize; i++) > - decoded[i] = tmp; > + if (bps < 33) { > + int32_t tmp = get_sbits_long(&s->gb, bps); > + for (i = 0; i < s->blocksize; i++) > + decoded[i] = tmp; > + } else { > + int64_t tmp = get_sbits64(&s->gb, 33); > + for (i = 0; i < s->blocksize; i++) > + s->decoded_33bps[i] = tmp; > + } > } else if (type == 1) { > - for (i = 0; i < s->blocksize; i++) > - decoded[i] = get_sbits_long(&s->gb, bps); > + if (bps < 33) { > + for (i = 0; i < s->blocksize; i++) > + decoded[i] = get_sbits_long(&s->gb, bps); > + } else { > + for (i = 0; i < s->blocksize; i++) > + s->decoded_33bps[i] = get_sbits64(&s->gb, 33); > + } > } else if ((type >= 8) && (type <= 12)) { > - if ((ret = decode_subframe_fixed(s, decoded, type & ~0x8, bps)) < 0) > - return ret; > + int order = type & ~0x8; > + if (bps < 33) { > + if (bps + order <= 32) { > + if ((ret = decode_subframe_fixed(s, decoded, order, bps)) < 0) > + return ret; > + } else { > + if ((ret = decode_subframe_fixed_wide(s, decoded, order, bps)) < 0) > + return ret; > + } > + } else { > + if ((ret = decode_subframe_fixed_33bps(s, s->decoded_33bps, decoded, order)) < 0) > + return ret; > + } > } else if (type >= 32) { > - if ((ret = decode_subframe_lpc(s, decoded, (type & ~0x20)+1, bps)) < 0) > - return ret; > + if (bps < 33) { > + if ((ret = decode_subframe_lpc(s, decoded, (type & ~0x20)+1, bps)) < 0) > + return ret; > + } else { > + if ((ret = decode_subframe_lpc_33bps(s, s->decoded_33bps, decoded, (type & ~0x20)+1)) < 0) > + return ret; > + } > } else { > av_log(s->avctx, AV_LOG_ERROR, "invalid coding type\n"); > return AVERROR_INVALIDDATA; > } > > - if (wasted && wasted < 32) { > - int i; > - for (i = 0; i < s->blocksize; i++) > - decoded[i] = (unsigned)decoded[i] << wasted; > + if (wasted) { > + if (wasted+bps == 33) { > + int i; > + for (i = 0; i < s->blocksize; i++) > + s->decoded_33bps[i] = (uint64_t)decoded[i] << wasted; > + } else if (wasted < 32) { > + int i; > + for (i = 0; i < s->blocksize; i++) > + decoded[i] = (unsigned)decoded[i] << wasted; > + } > } > > return 0; > @@ -554,6 +733,26 @@ static int decode_frame(FLACContext *s) > return 0; > } > > +static void decorrelate_33bps(int ch_mode, int32_t **decoded, int64_t *decoded_33bps, int len) > +{ > + int i; > + if (ch_mode == FLAC_CHMODE_LEFT_SIDE ) { > + for (i = 0; i < len; i++) > + decoded[1][i] = decoded[0][i] - decoded_33bps[i]; > + } else if (ch_mode == FLAC_CHMODE_RIGHT_SIDE ) { > + for (i = 0; i < len; i++) > + decoded[0][i] = decoded[1][i] + decoded_33bps[i]; > + } else if (ch_mode == FLAC_CHMODE_MID_SIDE ) { > + for (i = 0; i < len; i++) { > + uint64_t a = decoded[0][i]; > + int64_t b = decoded_33bps[i]; > + a -= b >> 1; > + decoded[0][i] = (a + b); > + decoded[1][i] = a; > + } > + } > +} > + > static int flac_decode_frame(AVCodecContext *avctx, AVFrame *frame, > int *got_frame_ptr, AVPacket *avpkt) > { > @@ -612,9 +811,15 @@ static int flac_decode_frame(AVCodecContext *avctx, AVFrame *frame, > if ((ret = ff_thread_get_buffer(avctx, frame, 0)) < 0) > return ret; > > - s->dsp.decorrelate[s->ch_mode](frame->data, s->decoded, > - s->stream_info.channels, > - s->blocksize, s->sample_shift); > + if (s->stream_info.bps == 32 && s->ch_mode > 0) { > + decorrelate_33bps(s->ch_mode, s->decoded, s->decoded_33bps, s->blocksize); > + s->dsp.decorrelate[0](frame->data, s->decoded, s->stream_info.channels, > + s->blocksize, s->sample_shift); > + } else { > + s->dsp.decorrelate[s->ch_mode](frame->data, s->decoded, > + s->stream_info.channels, > + s->blocksize, s->sample_shift); > + } > > if (bytes_read > buf_size) { > av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", bytes_read - buf_size); > @@ -635,6 +840,7 @@ static av_cold int flac_decode_close(AVCodecContext *avctx) > FLACContext *s = avctx->priv_data; > > av_freep(&s->decoded_buffer); > + av_freep(&s->decoded_buffer_33bps); > > return 0; > } > diff --git a/libavcodec/get_bits.h b/libavcodec/get_bits.h > index 992765dc92..52d13b8242 100644 > --- a/libavcodec/get_bits.h > +++ b/libavcodec/get_bits.h > @@ -596,6 +596,18 @@ static inline int get_sbits_long(GetBitContext *s, int n) > return sign_extend(get_bits_long(s, n), n); > } > > +/** > + * Read 0-64 bits as a signed integer. > + */ > +static inline int64_t get_sbits64(GetBitContext *s, int n) > +{ > + // sign_extend(x, 0) is undefined > + if (!n) > + return 0; > + > + return sign_extend64(get_bits64(s, n), n); > +} > + > /** > * Show 0-32 bits. > */ > diff --git a/libavcodec/mathops.h b/libavcodec/mathops.h > index f81d21f9c4..8a82d9d086 100644 > --- a/libavcodec/mathops.h > +++ b/libavcodec/mathops.h > @@ -137,6 +137,15 @@ static inline av_const int sign_extend(int val, unsigned bits) > } > #endif > > +#ifndef sign_extend64 > +static inline av_const int64_t sign_extend64(int64_t val, unsigned bits) > +{ > + unsigned shift = 8 * sizeof(int64_t) - bits; > + union { uint64_t u; int64_t s; } v = { (uint64_t) val << shift }; > + return v.s >> shift; > +} > +#endif > + > #ifndef zero_extend > static inline av_const unsigned zero_extend(unsigned val, unsigned bits) > { _______________________________________________ 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:[~2022-09-14 13:03 UTC|newest] Thread overview: 9+ messages / expand[flat|nested] mbox.gz Atom feed top 2022-09-03 9:21 [FFmpeg-devel] [PATCH v4 0/2] 32bps FLAC patches Martijn van Beurden 2022-09-03 9:21 ` [FFmpeg-devel] [PATCH v4 1/2] libavcodec/flacdec: Implement decoding of 32 bit-per-sample PCM Martijn van Beurden 2022-09-14 13:03 ` Andreas Rheinhardt [this message] 2022-09-14 15:21 ` Martijn van Beurden 2022-09-14 15:48 ` Paul B Mahol 2022-09-14 17:10 ` Andreas Rheinhardt 2022-09-03 9:21 ` [FFmpeg-devel] [PATCH v4 2/2] libavcodec/flacenc: Implement encoding " Martijn van Beurden 2022-09-10 10:32 ` [FFmpeg-devel] [PATCH v4 0/2] 32bps FLAC patches Martijn van Beurden 2022-09-10 13:22 ` Paul B Mahol
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