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[92.64.99.37]) by smtp.googlemail.com with ESMTPSA id 3sm4637096ejq.159.2021.12.19.12.53.40 (version=TLS1_3 cipher=TLS_AES_256_GCM_SHA384 bits=256/256); Sun, 19 Dec 2021 12:53:40 -0800 (PST) From: Martijn van Beurden To: ffmpeg-devel@ffmpeg.org Date: Sun, 19 Dec 2021 21:53:18 +0100 Message-Id: <20211219205318.9362-1-mvanb1@gmail.com> X-Mailer: git-send-email 2.30.2 MIME-Version: 1.0 Subject: [FFmpeg-devel] [PATCH] libavcodec/flacenc: add backward-compatible 32 bit-per-sample capability X-BeenThere: ffmpeg-devel@ffmpeg.org X-Mailman-Version: 2.1.29 Precedence: list List-Id: FFmpeg development discussions and patches List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Reply-To: FFmpeg development discussions and patches Cc: Martijn van Beurden Errors-To: ffmpeg-devel-bounces@ffmpeg.org Sender: "ffmpeg-devel" X-TUID: INc2+29P9yY2 Enables creation of FLAC files with up to 32 bits-per-sample, up from the previous limit of 24 bit. This is a feature requested for RAWcooked, the archiving community has a need for storing files with 32-bit integer audio samples. See https://github.com/MediaArea/RAWcooked/issues/356 Restrictions to the encoder are added so created files are compatible with existing decoders. Stereo decorrelation is disabled on 32 bit-per-sample, because a side channel would need 33 bit-per-sample, causing problems in existing 32 bit datapaths. Also only LPC encoding is enabled, because decoders capable of processing 24-bit files already use 64-bit processing for LPC, but not for fixed subframes. Furthermore, predictions and residuals are checked for causing integer overflow, reverting to a verbatim (store) subframe in case no LPC coeffs can be found that do not cause overflow. ffmpeg's FLAC decoder has been forward-compatible with this change since commit c720b9ce98 (May 2015). libFLAC is forward-compatible since release 1.2.1 (September 2007), the flac command line tool however blocks 32-bit files out of caution, it having been untested until now. --- libavcodec/flacdsp.c | 47 ++++++++++++++++++++++ libavcodec/flacdsp.h | 3 ++ libavcodec/flacenc.c | 95 +++++++++++++++++++++++++++++++++++++------- 3 files changed, 130 insertions(+), 15 deletions(-) diff --git a/libavcodec/flacdsp.c b/libavcodec/flacdsp.c index bc9a5dbed9..422f6578ba 100644 --- a/libavcodec/flacdsp.c +++ b/libavcodec/flacdsp.c @@ -43,6 +43,53 @@ #define PLANAR 1 #include "flacdsp_template.c" +#define ZIGZAG_32BIT_MAX 0x3FFFFFFF +#define ZIGZAG_32BIT_MIN -0x3FFFFFFF + +int ff_flacdsp_lpc_encode_c_32_overflow_detect(int32_t *res, const int32_t *smp, int len, + int order, int32_t *coefs, int shift) +{ + /* This function checks for every prediction and every residual + * whether they cause integer overflow in existing decoders. In + * case the prediction exceeds int32_t limits, prediction + * coefficients are lowered accordingly. If the residual exceeds + * ZIGZAG_32BIT_MAX and _MIN or coefficients have been lowered + * twice but the prediction still overflows, give up */ + int lpc_reduction_tries = 0; + int64_t pmax; + for (int i = 0; i < order; i++) + res[i] = smp[i]; + do { + pmax = 0; + for (int i = order; i < len; i++) { + int64_t p = 0, tmp; + for (int j = 0; j < order; j++) + p += (int64_t)coefs[j]*smp[(i-1)-j]; + p >>= shift; + tmp = smp[i] - p; + if (p > INT32_MAX && p > pmax) + pmax = p; + else if (p < INT32_MIN && (p * -1) > pmax) + pmax = p * -1; + if (tmp > ZIGZAG_32BIT_MAX || tmp < ZIGZAG_32BIT_MIN) + return 0; + res[i] = tmp; + } + + if (pmax > 0) { + double scale; + if(lpc_reduction_tries >= 2) + return 0; + lpc_reduction_tries++; + scale = (double)INT32_MAX/pmax; + for (int i = 0; i < order; i++) + coefs[i] *= scale; + } + } while (pmax > 0); + return 1; +} + + static void flac_lpc_16_c(int32_t *decoded, const int coeffs[32], int pred_order, int qlevel, int len) { diff --git a/libavcodec/flacdsp.h b/libavcodec/flacdsp.h index 7bb0dd0e9a..5978a4722a 100644 --- a/libavcodec/flacdsp.h +++ b/libavcodec/flacdsp.h @@ -40,4 +40,7 @@ void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt, int channels, i void ff_flacdsp_init_arm(FLACDSPContext *c, enum AVSampleFormat fmt, int channels, int bps); void ff_flacdsp_init_x86(FLACDSPContext *c, enum AVSampleFormat fmt, int channels, int bps); +int ff_flacdsp_lpc_encode_c_32_overflow_detect(int32_t *res, const int32_t *smp, int len, + int order, int32_t *coefs, int shift); + #endif /* AVCODEC_FLACDSP_H */ diff --git a/libavcodec/flacenc.c b/libavcodec/flacenc.c index 595928927d..64d4b8a71d 100644 --- a/libavcodec/flacenc.c +++ b/libavcodec/flacenc.c @@ -254,10 +254,30 @@ static av_cold int flac_encode_init(AVCodecContext *avctx) s->bps_code = 4; break; case AV_SAMPLE_FMT_S32: - if (avctx->bits_per_raw_sample != 24) - av_log(avctx, AV_LOG_WARNING, "encoding as 24 bits-per-sample\n"); - avctx->bits_per_raw_sample = 24; - s->bps_code = 6; + if (avctx->bits_per_raw_sample > 0 && avctx->bits_per_raw_sample <= 24){ + if(avctx->bits_per_raw_sample < 24) + av_log(avctx, AV_LOG_WARNING, "encoding as 24 bits-per-sample\n"); + avctx->bits_per_raw_sample = 24; + s->bps_code = 6; + } else { + av_log(avctx, AV_LOG_WARNING, "non-streamable bits-per-sample\n"); + s->bps_code = 0; + if (avctx->bits_per_raw_sample == 0) + avctx->bits_per_raw_sample = 32; + if(s->options.lpc_type != FF_LPC_TYPE_LEVINSON && + s->options.lpc_type != FF_LPC_TYPE_CHOLESKY){ + av_log(avctx, AV_LOG_WARNING, "forcing lpc_type levinson, lpc_type fixed or none not supported with >24 bits-per-sample FLAC\n"); + s->options.lpc_type = FF_LPC_TYPE_LEVINSON; + } + if (avctx->bits_per_raw_sample == 32){ + /* Because stereo decorrelation can raise the bitdepth of + * a subframe to 33 bits, we disable it */ + if(s->options.ch_mode != FLAC_CHMODE_INDEPENDENT){ + av_log(avctx, AV_LOG_WARNING, "disabling stereo decorrelation, not supported with 32 bits-per-sample FLAC\n"); + s->options.ch_mode = FLAC_CHMODE_INDEPENDENT; + } + } + } break; } @@ -686,7 +706,7 @@ static uint64_t calc_rice_params(RiceContext *rc, tmp_rc.coding_mode = rc->coding_mode; - for (i = 0; i < n; i++) + for (i = pred_order; i < n; i++) udata[i] = (2 * data[i]) ^ (data[i] >> 31); calc_sum_top(pmax, exact ? kmax : 0, udata, n, pred_order, sums); @@ -868,7 +888,11 @@ static int encode_residual_ch(FlacEncodeContext *s, int ch) order = av_clip(order, min_order - 1, max_order - 1); if (order == last_order) continue; - if (s->bps_code * 4 + s->options.lpc_coeff_precision + av_log2(order) <= 32) { + if (s->avctx->bits_per_raw_sample > 24) { + if(!ff_flacdsp_lpc_encode_c_32_overflow_detect(res, smp, n, order+1, + coefs[order], shift[order])) + continue; + } else if (s->bps_code * 4 + s->options.lpc_coeff_precision + av_log2(order) <= 32) { s->flac_dsp.lpc16_encode(res, smp, n, order+1, coefs[order], shift[order]); } else { @@ -888,7 +912,11 @@ static int encode_residual_ch(FlacEncodeContext *s, int ch) opt_order = 0; bits[0] = UINT32_MAX; for (i = min_order-1; i < max_order; i++) { - if (s->bps_code * 4 + s->options.lpc_coeff_precision + av_log2(i) <= 32) { + if (s->avctx->bits_per_raw_sample > 24) { + if(!ff_flacdsp_lpc_encode_c_32_overflow_detect(res, smp, n, i+1, + coefs[i], shift[i])) + continue; + } else if (s->bps_code * 4 + s->options.lpc_coeff_precision + av_log2(i) <= 32) { s->flac_dsp.lpc16_encode(res, smp, n, i+1, coefs[i], shift[i]); } else { s->flac_dsp.lpc32_encode(res, smp, n, i+1, coefs[i], shift[i]); @@ -910,7 +938,11 @@ static int encode_residual_ch(FlacEncodeContext *s, int ch) for (i = last-step; i <= last+step; i += step) { if (i < min_order-1 || i >= max_order || bits[i] < UINT32_MAX) continue; - if (s->bps_code * 4 + s->options.lpc_coeff_precision + av_log2(i) <= 32) { + if (s->avctx->bits_per_raw_sample > 24) { + if(!ff_flacdsp_lpc_encode_c_32_overflow_detect(res, smp, n, i+1, + coefs[i], shift[i])) + continue; + } else if (s->bps_code * 4 + s->options.lpc_coeff_precision + av_log2(i) <= 32) { s->flac_dsp.lpc32_encode(res, smp, n, i+1, coefs[i], shift[i]); } else { s->flac_dsp.lpc16_encode(res, smp, n, i+1, coefs[i], shift[i]); @@ -951,7 +983,11 @@ static int encode_residual_ch(FlacEncodeContext *s, int ch) if (diffsum >8) continue; - if (s->bps_code * 4 + s->options.lpc_coeff_precision + av_log2(opt_order - 1) <= 32) { + if (s->avctx->bits_per_raw_sample > 24) { + if(!ff_flacdsp_lpc_encode_c_32_overflow_detect(res, smp, n, opt_order, + lpc_try, shift[opt_order-1])) + continue; + } else if (s->bps_code * 4 + s->options.lpc_coeff_precision + av_log2(opt_order-1) <= 32) { s->flac_dsp.lpc16_encode(res, smp, n, opt_order, lpc_try, shift[opt_order-1]); } else { s->flac_dsp.lpc32_encode(res, smp, n, opt_order, lpc_try, shift[opt_order-1]); @@ -972,7 +1008,16 @@ static int encode_residual_ch(FlacEncodeContext *s, int ch) for (i = 0; i < sub->order; i++) sub->coefs[i] = coefs[sub->order-1][i]; - if (s->bps_code * 4 + s->options.lpc_coeff_precision + av_log2(opt_order) <= 32) { + if (s->avctx->bits_per_raw_sample > 24) { + if (!ff_flacdsp_lpc_encode_c_32_overflow_detect(res, smp, n, sub->order, + sub->coefs, sub->shift)) { + /* No coefs found that do not cause integer overflow, + * so return a verbatim subframe instead */ + sub->type = sub->type_code = FLAC_SUBFRAME_VERBATIM; + memcpy(res, smp, n * sizeof(int32_t)); + return subframe_count_exact(s, sub, 0); + } + } else if (s->bps_code * 4 + s->options.lpc_coeff_precision + av_log2(sub->order) <= 32) { s->flac_dsp.lpc16_encode(res, smp, n, sub->order, sub->coefs, sub->shift); } else { s->flac_dsp.lpc32_encode(res, smp, n, sub->order, sub->coefs, sub->shift); @@ -1227,12 +1272,22 @@ static void write_subframes(FlacEncodeContext *s) if (sub->type == FLAC_SUBFRAME_CONSTANT) { put_sbits(&s->pb, sub->obits, res[0]); } else if (sub->type == FLAC_SUBFRAME_VERBATIM) { - while (res < frame_end) - put_sbits(&s->pb, sub->obits, *res++); + if (sub->obits < 32) { + while (res < frame_end) + put_sbits(&s->pb, sub->obits, *res++); + } else { + while (res < frame_end) + put_bits32(&s->pb, *res++); + } } else { /* warm-up samples */ - for (i = 0; i < sub->order; i++) - put_sbits(&s->pb, sub->obits, *res++); + if(sub->obits < 32){ + for (i = 0; i < sub->order; i++) + put_sbits(&s->pb, sub->obits, *res++); + }else{ + for (i = 0; i < sub->order; i++) + put_bits32(&s->pb, *res++); + } /* LPC coefficients */ if (sub->type == FLAC_SUBFRAME_LPC) { @@ -1305,7 +1360,7 @@ static int update_md5_sum(FlacEncodeContext *s, const void *samples) (const uint16_t *) samples, buf_size / 2); buf = s->md5_buffer; #endif - } else { + } else if (s->avctx->bits_per_raw_sample <= 24) { int i; const int32_t *samples0 = samples; uint8_t *tmp = s->md5_buffer; @@ -1315,6 +1370,16 @@ static int update_md5_sum(FlacEncodeContext *s, const void *samples) AV_WL24(tmp + 3*i, v); } buf = s->md5_buffer; + } else { + /* s->avctx->bits_per_raw_sample <= 32 */ + int i; + const int32_t *samples0 = samples; + uint8_t *tmp = s->md5_buffer; + + for (i = 0; i < s->frame.blocksize * s->channels; i++) { + AV_WL32(tmp + 4*i, samples0[i]); + } + buf = s->md5_buffer; } av_md5_update(s->md5ctx, buf, buf_size);