* [FFmpeg-devel] [PATCH v3 0/2] 32bps FLAC patches @ 2022-09-02 14:20 Martijn van Beurden 2022-09-02 14:20 ` [FFmpeg-devel] [PATCH v3 1/2] libavcodec/flacdec: Implement decoding of 32 bit-per-sample PCM Martijn van Beurden 2022-09-02 14:20 ` [FFmpeg-devel] [PATCH v3 2/2] libavcodec/flacenc: Implement encoding " Martijn van Beurden 0 siblings, 2 replies; 6+ messages in thread From: Martijn van Beurden @ 2022-09-02 14:20 UTC (permalink / raw) To: ffmpeg-devel; +Cc: Martijn van Beurden Recently libFLAC gained the ability (not in any released version yet though) to create FLAC files containing 32-bit int PCM samples. To keep complexity reasonable, the choice was made to limit residuals to 32-bit integers, which the encoder must make sure of. In case the encoder cannot find any predictor of which the residuals fit this limit, it must default to using a verbatim subframe. Tests have shown that this does not happen often (<0.1% of subframes on a music corpus of various styles). See also discussion here: https://github.com/ietf-wg-cellar/flac-specification/pull/148 These two patches implement decoding and encoding following this format. The decoding patch has been send in for review earlier, but didn't get any feedback yet, so I'm sending it in again, unchanged, this time with a patch for encoding too. Changes since v1: fix copy-paste error in encoder, several invocations of lpc_encode_choose_datapath used wrong parameters, making FATE fail as compression was less than it should be Changes since v2: Rebased decoder part as it didn't apply anymore Martijn van Beurden (2): libavcodec/flacdec: Implement decoding of 32 bit-per-sample PCM libavcodec/flacenc: Implement encoding of 32 bit-per-sample PCM libavcodec/flac.c | 4 +- libavcodec/flacdec.c | 248 ++++++++++++++++++-- libavcodec/flacenc.c | 499 ++++++++++++++++++++++++++++++++-------- libavcodec/get_bits.h | 12 + libavcodec/mathops.h | 9 + libavcodec/put_bits.h | 7 + libavcodec/put_golomb.h | 19 +- 7 files changed, 668 insertions(+), 130 deletions(-) -- 2.30.2 _______________________________________________ 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". ^ permalink raw reply [flat|nested] 6+ messages in thread
* [FFmpeg-devel] [PATCH v3 1/2] libavcodec/flacdec: Implement decoding of 32 bit-per-sample PCM 2022-09-02 14:20 [FFmpeg-devel] [PATCH v3 0/2] 32bps FLAC patches Martijn van Beurden @ 2022-09-02 14:20 ` Martijn van Beurden 2022-09-02 14:20 ` [FFmpeg-devel] [PATCH v3 2/2] libavcodec/flacenc: Implement encoding " Martijn van Beurden 1 sibling, 0 replies; 6+ messages in thread From: Martijn van Beurden @ 2022-09-02 14:20 UTC (permalink / raw) To: ffmpeg-devel; +Cc: 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; + 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) + 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) { -- 2.30.2 _______________________________________________ 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". ^ permalink raw reply [flat|nested] 6+ messages in thread
* [FFmpeg-devel] [PATCH v3 2/2] libavcodec/flacenc: Implement encoding of 32 bit-per-sample PCM 2022-09-02 14:20 [FFmpeg-devel] [PATCH v3 0/2] 32bps FLAC patches Martijn van Beurden 2022-09-02 14:20 ` [FFmpeg-devel] [PATCH v3 1/2] libavcodec/flacdec: Implement decoding of 32 bit-per-sample PCM Martijn van Beurden @ 2022-09-02 14:20 ` Martijn van Beurden 2022-09-02 15:10 ` Andreas Rheinhardt 1 sibling, 1 reply; 6+ messages in thread From: Martijn van Beurden @ 2022-09-02 14:20 UTC (permalink / raw) To: ffmpeg-devel; +Cc: Martijn van Beurden Add encoding of 32 bit-per-sample PCM to FLAC files to libavcodec. Coding to this format is at this point considered experimental and -strict -2 is needed to get ffmpeg to encode such files. --- libavcodec/flacenc.c | 499 ++++++++++++++++++++++++++++++++-------- libavcodec/put_bits.h | 7 + libavcodec/put_golomb.h | 19 +- 3 files changed, 418 insertions(+), 107 deletions(-) diff --git a/libavcodec/flacenc.c b/libavcodec/flacenc.c index 0170e02ae8..84973bb2a5 100644 --- a/libavcodec/flacenc.c +++ b/libavcodec/flacenc.c @@ -95,6 +95,7 @@ typedef struct FlacSubframe { typedef struct FlacFrame { FlacSubframe subframes[FLAC_MAX_CHANNELS]; + int64_t samples_33bps[FLAC_MAX_BLOCKSIZE]; int blocksize; int bs_code[2]; uint8_t crc8; @@ -282,10 +283,23 @@ 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 <= 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 if (avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) { + av_log(avctx, AV_LOG_WARNING, + "encoding as 24 bits-per-sample, more is considered " + "experimental. Add -strict %d if you want to encode " + "more than 24 bits-per-sample\n", + FF_COMPLIANCE_EXPERIMENTAL); + avctx->bits_per_raw_sample = 24; + s->bps_code = 6; + } else { + avctx->bits_per_raw_sample = 32; + s->bps_code = 7; + } break; } @@ -533,8 +547,7 @@ static uint64_t rice_count_exact(const int32_t *res, int n, int k) uint64_t count = 0; for (i = 0; i < n; i++) { - int32_t v = -2 * res[i] - 1; - v ^= v >> 31; + unsigned v = ((unsigned)(res[i]) << 1) ^ (res[i] >> 31); count += (v >> k) + 1 + k; } return count; @@ -713,8 +726,8 @@ static uint64_t calc_rice_params(RiceContext *rc, tmp_rc.coding_mode = rc->coding_mode; - for (i = 0; i < n; i++) - udata[i] = (2 * data[i]) ^ (data[i] >> 31); + for (i = pred_order; i < n; i++) + udata[i] = ((unsigned)(data[i]) << 1) ^ (data[i] >> 31); calc_sum_top(pmax, exact ? kmax : 0, udata, n, pred_order, sums); @@ -812,6 +825,180 @@ static void encode_residual_fixed(int32_t *res, const int32_t *smp, int n, } +static int encode_residual_fixed_with_residual_limit(int32_t *res, const int32_t *smp, + int n, int order) +{ + /* This function checks for every residual whether it can be + * contained in <INT32_MIN,INT32_MAX]. In case it doesn't, the + * function that called this function has to try something else */ + int i; + int64_t res64; + + for (i = 0; i < order; i++) + res[i] = smp[i]; + + if (order == 0) { + for (i = order; i < n; i++) { + if (smp[i] == INT32_MIN) + return 1; + res[i] = smp[i]; + } + } else if (order == 1) { + for (i = order; i < n; i++) { + res64 = (int64_t)smp[i] - smp[i-1]; + if (res64 <= INT32_MIN || res64 > INT32_MAX) + return 1; + res[i] = res64; + } + } else if (order == 2) { + for (i = order; i < n; i++) { + res64 = (int64_t)smp[i] - 2*(int64_t)smp[i-1] + smp[i-2]; + if (res64 <= INT32_MIN || res64 > INT32_MAX) + return 1; + res[i] = res64; + } + } else if (order == 3) { + for (i = order; i < n; i++) { + res64 = (int64_t)smp[i] - 3*(int64_t)smp[i-1] + 3*(int64_t)smp[i-2] - smp[i-3]; + if (res64 <= INT32_MIN || res64 > INT32_MAX) + return 1; + res[i] = res64; + } + } else { + for (i = order; i < n; i++) { + res64 = (int64_t)smp[i] - 4*(int64_t)smp[i-1] + 6*(int64_t)smp[i-2] - 4*(int64_t)smp[i-3] + smp[i-4]; + if (res64 <= INT32_MIN || res64 > INT32_MAX) + return 1; + res[i] = res64; + } + } + return 0; +} + + +static int encode_residual_fixed_with_residual_limit_33bps(int32_t *res, const int64_t *smp, + int n, int order) +{ + /* This function checks for every residual whether it can be + * contained in <INT32_MIN,INT32_MAX]. In case it doesn't, the + * function that called this function has to try something else */ + int i; + int64_t res64; + + if (order == 0) { + for (i = order; i < n; i++) { + if (smp[i] <= INT32_MIN || smp[i] > INT32_MAX) + return 1; + res[i] = smp[i]; + } + } else if (order == 1) { + for (i = order; i < n; i++) { + res64 = smp[i] - smp[i-1]; + if (res64 <= INT32_MIN || res64 > INT32_MAX) + return 1; + res[i] = res64; + } + } else if (order == 2) { + for (i = order; i < n; i++) { + res64 = smp[i] - 2*smp[i-1] + smp[i-2]; + if (res64 <= INT32_MIN || res64 > INT32_MAX) + return 1; + res[i] = res64; + } + } else if (order == 3) { + for (i = order; i < n; i++) { + res64 = smp[i] - 3*smp[i-1] + 3*smp[i-2] - smp[i-3]; + if (res64 <= INT32_MIN || res64 > INT32_MAX) + return 1; + res[i] = res64; + } + } else { + for (i = order; i < n; i++) { + res64 = smp[i] - 4*smp[i-1] + 6*smp[i-2] - 4*smp[i-3] + smp[i-4]; + if (res64 <= INT32_MIN || res64 > INT32_MAX) + return 1; + res[i] = res64; + } + } + return 0; +} + + +static int lpc_encode_with_residual_limit(int32_t *res, const int32_t *smp, int len, + int order, int32_t *coefs, int shift) +{ + /* This function checks for every residual whether it can be + * contained in <INT32_MIN,INT32_MAX]. In case it doesn't, the + * function that called this function has to try something else */ + for (int i = 0; i < order; i++) + res[i] = smp[i]; + 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 (tmp <= INT32_MIN || tmp > INT32_MAX) + return 1; + res[i] = tmp; + } + return 0; +} + +static int lpc_encode_with_residual_limit_33bps(int32_t *res, const int64_t *smp, int len, + int order, int32_t *coefs, int shift) +{ + /* This function checks for every residual whether it can be + * contained in <INT32_MIN,INT32_MAX]. In case it doesn't, the + * function that called this function has to try something else */ + 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 (tmp <= INT32_MIN || tmp > INT32_MAX) + return 1; + res[i] = tmp; + } + return 0; +} + +static int lpc_encode_choose_datapath(FlacEncodeContext *s, int32_t bps, + int32_t *res, const int32_t *smp, + const int64_t *smp_33bps, int len, + int order, int32_t *coefs, int shift) +{ + uint64_t max_residual_value = 0; + int64_t max_sample_value = ((int64_t)(1) << (bps-1)); + /* This calculates the max size of any residual with the current + * predictor, so we know whether we need to check the residual */ + for (int i = 0; i < order; i++) + max_residual_value += FFABS(max_sample_value * coefs[i]); + max_residual_value >>= shift; + max_residual_value += max_sample_value; + if (bps > 32) { + if (lpc_encode_with_residual_limit_33bps(res, smp_33bps, len, order, coefs, shift)) + return 1; + } else if (max_residual_value > INT32_MAX) { + if (lpc_encode_with_residual_limit(res, smp, len, order, coefs, shift)) + return 1; + } else if (bps + s->options.lpc_coeff_precision + av_log2(order) <= 32) { + s->flac_dsp.lpc16_encode(res, smp, len, order, coefs, shift); + } else { + s->flac_dsp.lpc32_encode(res, smp, len, order, coefs, shift); + } + return 0; +} + +#define DEFAULT_TO_VERBATIM() \ +{ \ + sub->type = sub->type_code = FLAC_SUBFRAME_VERBATIM; \ + if (sub->obits <= 32) \ + memcpy(res, smp, n * sizeof(int32_t)); \ + return subframe_count_exact(s, sub, 0); \ +} + static int encode_residual_ch(FlacEncodeContext *s, int ch) { int i, n; @@ -821,28 +1008,38 @@ static int encode_residual_ch(FlacEncodeContext *s, int ch) int32_t coefs[MAX_LPC_ORDER][MAX_LPC_ORDER]; int shift[MAX_LPC_ORDER]; int32_t *res, *smp; + int64_t *smp_33bps; - frame = &s->frame; - sub = &frame->subframes[ch]; - res = sub->residual; - smp = sub->samples; - n = frame->blocksize; + frame = &s->frame; + sub = &frame->subframes[ch]; + res = sub->residual; + smp = sub->samples; + smp_33bps = frame->samples_33bps; + n = frame->blocksize; /* CONSTANT */ - for (i = 1; i < n; i++) - if(smp[i] != smp[0]) - break; - if (i == n) { - sub->type = sub->type_code = FLAC_SUBFRAME_CONSTANT; - res[0] = smp[0]; - return subframe_count_exact(s, sub, 0); + if (sub->obits > 32) { + for (i = 1; i < n; i++) + if(smp_33bps[i] != smp_33bps[0]) + break; + if (i == n) { + sub->type = sub->type_code = FLAC_SUBFRAME_CONSTANT; + return subframe_count_exact(s, sub, 0); + } + } else { + for (i = 1; i < n; i++) + if(smp[i] != smp[0]) + break; + if (i == n) { + sub->type = sub->type_code = FLAC_SUBFRAME_CONSTANT; + res[0] = smp[0]; + return subframe_count_exact(s, sub, 0); + } } /* VERBATIM */ if (frame->verbatim_only || n < 5) { - sub->type = sub->type_code = FLAC_SUBFRAME_VERBATIM; - memcpy(res, smp, n * sizeof(int32_t)); - return subframe_count_exact(s, sub, 0); + DEFAULT_TO_VERBATIM(); } min_order = s->options.min_prediction_order; @@ -859,15 +1056,32 @@ static int encode_residual_ch(FlacEncodeContext *s, int ch) opt_order = 0; bits[0] = UINT32_MAX; for (i = min_order; i <= max_order; i++) { - encode_residual_fixed(res, smp, n, i); + if (sub->obits == 33) { + if (encode_residual_fixed_with_residual_limit_33bps(res, smp_33bps, n, i)) + continue; + } else if (sub->obits + i >= 32) { + if (encode_residual_fixed_with_residual_limit(res, smp, n, i)) + continue; + } else + encode_residual_fixed(res, smp, n, i); bits[i] = find_subframe_rice_params(s, sub, i); if (bits[i] < bits[opt_order]) opt_order = i; } + if (opt_order == 0 && bits[0] == UINT32_MAX) { + /* No predictor found with residuals within <INT32_MIN,INT32_MAX], + * so encode a verbatim subframe instead */ + DEFAULT_TO_VERBATIM(); + } sub->order = opt_order; sub->type_code = sub->type | sub->order; if (sub->order != max_order) { - encode_residual_fixed(res, smp, n, sub->order); + if (sub->obits == 33) + encode_residual_fixed_with_residual_limit_33bps(res, smp_33bps, n, sub->order); + else if (sub->obits + i >= 32) + encode_residual_fixed_with_residual_limit(res, smp, n, sub->order); + else + encode_residual_fixed(res, smp, n, sub->order); find_subframe_rice_params(s, sub, sub->order); } return subframe_count_exact(s, sub, sub->order); @@ -875,6 +1089,14 @@ static int encode_residual_ch(FlacEncodeContext *s, int ch) /* LPC */ sub->type = FLAC_SUBFRAME_LPC; + if (sub->obits == 33) + /* As ff_lpc_calc_coefs is shared with other codecs and the LSB + * probably isn't predictable anyway, throw away LSB for analysis + * so it fits 32 bit int and existing function can be used + * unmodified */ + for (i = 0; i < n; i++) + smp[i] = smp_33bps[i] >> 1; + opt_order = ff_lpc_calc_coefs(&s->lpc_ctx, smp, n, min_order, max_order, s->options.lpc_coeff_precision, coefs, shift, s->options.lpc_type, s->options.lpc_passes, omethod, @@ -895,13 +1117,8 @@ 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) { - s->flac_dsp.lpc16_encode(res, smp, n, order+1, coefs[order], - shift[order]); - } else { - s->flac_dsp.lpc32_encode(res, smp, n, order+1, coefs[order], - shift[order]); - } + if(lpc_encode_choose_datapath(s, sub->obits, res, smp, smp_33bps, n, order+1, coefs[order], shift[order])) + continue; bits[i] = find_subframe_rice_params(s, sub, order+1); if (bits[i] < bits[opt_index]) { opt_index = i; @@ -915,11 +1132,8 @@ 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) { - 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]); - } + if(lpc_encode_choose_datapath(s, sub->obits, res, smp, smp_33bps, n, i+1, coefs[i], shift[i])) + continue; bits[i] = find_subframe_rice_params(s, sub, i+1); if (bits[i] < bits[opt_order]) opt_order = i; @@ -937,11 +1151,8 @@ 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) { - 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]); - } + if(lpc_encode_choose_datapath(s, sub->obits, res, smp, smp_33bps, n, i+1, coefs[i], shift[i])) + continue; bits[i] = find_subframe_rice_params(s, sub, i+1); if (bits[i] < bits[opt_order]) opt_order = i; @@ -978,11 +1189,8 @@ 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) { - 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]); - } + if(lpc_encode_choose_datapath(s, sub->obits, res, smp, smp_33bps, n, opt_order, lpc_try, shift[opt_order-1])) + continue; score = find_subframe_rice_params(s, sub, opt_order); if (score < best_score) { best_score = score; @@ -999,10 +1207,10 @@ 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) { - 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); + if(lpc_encode_choose_datapath(s, sub->obits, res, smp, smp_33bps, n, sub->order, sub->coefs, sub->shift)) { + /* No predictor found with residuals within <INT32_MIN,INT32_MAX], + * so encode a verbatim subframe instead */ + DEFAULT_TO_VERBATIM(); } find_subframe_rice_params(s, sub, sub->order); @@ -1069,54 +1277,88 @@ static int encode_frame(FlacEncodeContext *s) static void remove_wasted_bits(FlacEncodeContext *s) { - int ch, i; + int ch, i, wasted_bits; for (ch = 0; ch < s->channels; ch++) { FlacSubframe *sub = &s->frame.subframes[ch]; - int32_t v = 0; - for (i = 0; i < s->frame.blocksize; i++) { - v |= sub->samples[i]; - if (v & 1) - break; - } + if (sub->obits > 32) { + int64_t v = 0; + for (i = 0; i < s->frame.blocksize; i++) { + v |= s->frame.samples_33bps[i]; + if (v & 1) + break; + } + + if (!v || (v & 1)) + return; - if (v && !(v & 1)) { v = ff_ctz(v); + /* If any wasted bits are found, samples are moved + * from frame.samples_33bps to frame.subframes[ch] */ for (i = 0; i < s->frame.blocksize; i++) - sub->samples[i] >>= v; + sub->samples[i] = s->frame.samples_33bps[i] >> v; + wasted_bits = v; + } else { + int32_t v = 0; + for (i = 0; i < s->frame.blocksize; i++) { + v |= sub->samples[i]; + if (v & 1) + break; + } + + if (!v || (v & 1)) + return; - sub->wasted = v; - sub->obits -= v; + v = ff_ctz(v); - /* for 24-bit, check if removing wasted bits makes the range better - suited for using RICE instead of RICE2 for entropy coding */ - if (sub->obits <= 17) - sub->rc.coding_mode = CODING_MODE_RICE; + for (i = 0; i < s->frame.blocksize; i++) + sub->samples[i] >>= v; + wasted_bits = v; } + + sub->wasted = wasted_bits; + sub->obits -= wasted_bits; + + /* for 24-bit, check if removing wasted bits makes the range better + * suited for using RICE instead of RICE2 for entropy coding */ + if (sub->obits <= 17) + sub->rc.coding_mode = CODING_MODE_RICE; } } static int estimate_stereo_mode(const int32_t *left_ch, const int32_t *right_ch, int n, - int max_rice_param) + int max_rice_param, int bps) { int i, best; - int32_t lt, rt; uint64_t sum[4]; uint64_t score[4]; int k; /* calculate sum of 2nd order residual for each channel */ sum[0] = sum[1] = sum[2] = sum[3] = 0; - for (i = 2; i < n; i++) { - lt = left_ch[i] - 2*left_ch[i-1] + left_ch[i-2]; - rt = right_ch[i] - 2*right_ch[i-1] + right_ch[i-2]; - sum[2] += FFABS((lt + rt) >> 1); - sum[3] += FFABS(lt - rt); - sum[0] += FFABS(lt); - sum[1] += FFABS(rt); + if(bps < 30) { + int32_t lt, rt; + for (i = 2; i < n; i++) { + lt = left_ch[i] - 2*left_ch[i-1] + left_ch[i-2]; + rt = right_ch[i] - 2*right_ch[i-1] + right_ch[i-2]; + sum[2] += FFABS((lt + rt) >> 1); + sum[3] += FFABS(lt - rt); + sum[0] += FFABS(lt); + sum[1] += FFABS(rt); + } + } else { + int64_t lt, rt; + for (i = 2; i < n; i++) { + lt = (int64_t)left_ch[i] - 2*(int64_t)left_ch[i-1] + left_ch[i-2]; + rt = (int64_t)right_ch[i] - 2*(int64_t)right_ch[i-1] + right_ch[i-2]; + sum[2] += FFABS((lt + rt) >> 1); + sum[3] += FFABS(lt - rt); + sum[0] += FFABS(lt); + sum[1] += FFABS(rt); + } } /* estimate bit counts */ for (i = 0; i < 4; i++) { @@ -1147,12 +1389,14 @@ static void channel_decorrelation(FlacEncodeContext *s) { FlacFrame *frame; int32_t *left, *right; + int64_t *side_33bps; int i, n; - frame = &s->frame; - n = frame->blocksize; - left = frame->subframes[0].samples; - right = frame->subframes[1].samples; + frame = &s->frame; + n = frame->blocksize; + left = frame->subframes[0].samples; + right = frame->subframes[1].samples; + side_33bps = frame->samples_33bps; if (s->channels != 2) { frame->ch_mode = FLAC_CHMODE_INDEPENDENT; @@ -1161,29 +1405,49 @@ static void channel_decorrelation(FlacEncodeContext *s) if (s->options.ch_mode < 0) { int max_rice_param = (1 << frame->subframes[0].rc.coding_mode) - 2; - frame->ch_mode = estimate_stereo_mode(left, right, n, max_rice_param); + frame->ch_mode = estimate_stereo_mode(left, right, n, max_rice_param, s->avctx->bits_per_raw_sample); } else frame->ch_mode = s->options.ch_mode; /* perform decorrelation and adjust bits-per-sample */ if (frame->ch_mode == FLAC_CHMODE_INDEPENDENT) return; - if (frame->ch_mode == FLAC_CHMODE_MID_SIDE) { - int32_t tmp; - for (i = 0; i < n; i++) { - tmp = left[i]; - left[i] = (tmp + right[i]) >> 1; - right[i] = tmp - right[i]; + if(s->avctx->bits_per_raw_sample == 32) { + if (frame->ch_mode == FLAC_CHMODE_MID_SIDE) { + int64_t tmp; + for (i = 0; i < n; i++) { + tmp = left[i]; + left[i] = (tmp + right[i]) >> 1; + side_33bps[i] = tmp - right[i]; + } + frame->subframes[1].obits++; + } else if (frame->ch_mode == FLAC_CHMODE_LEFT_SIDE) { + for (i = 0; i < n; i++) + side_33bps[i] = (int64_t)left[i] - right[i]; + frame->subframes[1].obits++; + } else { + for (i = 0; i < n; i++) + side_33bps[i] = (int64_t)left[i] - right[i]; + frame->subframes[0].obits++; } - frame->subframes[1].obits++; - } else if (frame->ch_mode == FLAC_CHMODE_LEFT_SIDE) { - for (i = 0; i < n; i++) - right[i] = left[i] - right[i]; - frame->subframes[1].obits++; } else { - for (i = 0; i < n; i++) - left[i] -= right[i]; - frame->subframes[0].obits++; + if (frame->ch_mode == FLAC_CHMODE_MID_SIDE) { + int32_t tmp; + for (i = 0; i < n; i++) { + tmp = left[i]; + left[i] = (tmp + right[i]) >> 1; + right[i] = tmp - right[i]; + } + frame->subframes[1].obits++; + } else if (frame->ch_mode == FLAC_CHMODE_LEFT_SIDE) { + for (i = 0; i < n; i++) + right[i] = left[i] - right[i]; + frame->subframes[1].obits++; + } else { + for (i = 0; i < n; i++) + left[i] -= right[i]; + frame->subframes[0].obits++; + } } } @@ -1252,14 +1516,38 @@ static void write_subframes(FlacEncodeContext *s) /* subframe */ if (sub->type == FLAC_SUBFRAME_CONSTANT) { - put_sbits(&s->pb, sub->obits, res[0]); + if(sub->obits == 33) + put_sbits64(&s->pb, 33, s->frame.samples_33bps[0]); + else if(sub->obits == 32) + put_bits32(&s->pb, res[0]); + else + 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 == 33) { + int64_t *res64 = s->frame.samples_33bps; + int64_t *frame_end64 = &s->frame.samples_33bps[s->frame.blocksize]; + while (res64 < frame_end64) + put_sbits64(&s->pb, 33, (*res64++)); + } else if (sub->obits == 32) { + while (res < frame_end) + put_bits32(&s->pb, *res++); + } else { + while (res < frame_end) + put_sbits(&s->pb, sub->obits, *res++); + } } else { /* warm-up samples */ - for (i = 0; i < sub->order; i++) - put_sbits(&s->pb, sub->obits, *res++); + if (sub->obits == 33) { + for (i = 0; i < sub->order; i++) + put_sbits64(&s->pb, 33, s->frame.samples_33bps[i]); + res += sub->order; + } else if (sub->obits == 32) { + for (i = 0; i < sub->order; i++) + put_bits32(&s->pb, *res++); + } else { + for (i = 0; i < sub->order; i++) + put_sbits(&s->pb, sub->obits, *res++); + } /* LPC coefficients */ if (sub->type == FLAC_SUBFRAME_LPC) { @@ -1284,7 +1572,7 @@ static void write_subframes(FlacEncodeContext *s) int k = sub->rc.params[p]; put_bits(&s->pb, sub->rc.coding_mode, k); while (res < part_end) - set_sr_golomb_flac(&s->pb, *res++, k, INT32_MAX, 0); + set_sr_golomb_flac(&s->pb, *res++, k); part_end = FFMIN(frame_end, part_end + psize); } } @@ -1332,7 +1620,7 @@ 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; @@ -1342,6 +1630,15 @@ 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); diff --git a/libavcodec/put_bits.h b/libavcodec/put_bits.h index 4b4f977ad5..11360fe7a4 100644 --- a/libavcodec/put_bits.h +++ b/libavcodec/put_bits.h @@ -363,6 +363,13 @@ static inline void put_bits64(PutBitContext *s, int n, uint64_t value) } } +static inline void put_sbits64(PutBitContext *pb, int n, int64_t value) +{ + av_assert2(n >= 0 && n <= 64); + + put_bits64(pb, n, (uint64_t)(value) & (~(UINT64_MAX << n))); +} + /** * Return the pointer to the byte where the bitstream writer will put * the next bit. diff --git a/libavcodec/put_golomb.h b/libavcodec/put_golomb.h index 9ca911fc3c..ca4b182ae8 100644 --- a/libavcodec/put_golomb.h +++ b/libavcodec/put_golomb.h @@ -154,15 +154,22 @@ static inline void set_sr_golomb(PutBitContext *pb, int i, int k, int limit, /** * write signed golomb rice code (flac). */ -static inline void set_sr_golomb_flac(PutBitContext *pb, int i, int k, - int limit, int esc_len) +static inline void set_sr_golomb_flac(PutBitContext *pb, int i, int k) { - int v; + unsigned v, e; - v = -2 * i - 1; - v ^= (v >> 31); + v = ((unsigned)(i) << 1) ^ (i >> 31); - set_ur_golomb_jpegls(pb, v, k, limit, esc_len); + e = (v >> k) + 1; + while (e > 31) { + put_bits(pb, 31, 0); + e -= 31; + } + put_bits(pb, e, 1); + if (k) { + unsigned mask = UINT32_MAX >> (32-k); + put_bits(pb, k, v & mask); + } } #endif /* AVCODEC_PUT_GOLOMB_H */ -- 2.30.2 _______________________________________________ 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". ^ permalink raw reply [flat|nested] 6+ messages in thread
* Re: [FFmpeg-devel] [PATCH v3 2/2] libavcodec/flacenc: Implement encoding of 32 bit-per-sample PCM 2022-09-02 14:20 ` [FFmpeg-devel] [PATCH v3 2/2] libavcodec/flacenc: Implement encoding " Martijn van Beurden @ 2022-09-02 15:10 ` Andreas Rheinhardt 2022-09-02 18:43 ` Martijn van Beurden 0 siblings, 1 reply; 6+ messages in thread From: Andreas Rheinhardt @ 2022-09-02 15:10 UTC (permalink / raw) To: ffmpeg-devel Martijn van Beurden: > Add encoding of 32 bit-per-sample PCM to FLAC files to libavcodec. > Coding to this format is at this point considered experimental and > -strict -2 is needed to get ffmpeg to encode such files. > --- > libavcodec/flacenc.c | 499 ++++++++++++++++++++++++++++++++-------- > libavcodec/put_bits.h | 7 + > libavcodec/put_golomb.h | 19 +- > 3 files changed, 418 insertions(+), 107 deletions(-) > > diff --git a/libavcodec/flacenc.c b/libavcodec/flacenc.c > index 0170e02ae8..84973bb2a5 100644 > --- a/libavcodec/flacenc.c > +++ b/libavcodec/flacenc.c > @@ -95,6 +95,7 @@ typedef struct FlacSubframe { > > typedef struct FlacFrame { > FlacSubframe subframes[FLAC_MAX_CHANNELS]; > + int64_t samples_33bps[FLAC_MAX_BLOCKSIZE]; > int blocksize; > int bs_code[2]; > uint8_t crc8; > @@ -282,10 +283,23 @@ 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 <= 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 if (avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) { > + av_log(avctx, AV_LOG_WARNING, > + "encoding as 24 bits-per-sample, more is considered " > + "experimental. Add -strict %d if you want to encode " > + "more than 24 bits-per-sample\n", > + FF_COMPLIANCE_EXPERIMENTAL); > + avctx->bits_per_raw_sample = 24; > + s->bps_code = 6; > + } else { > + avctx->bits_per_raw_sample = 32; > + s->bps_code = 7; > + } > break; > } > > @@ -533,8 +547,7 @@ static uint64_t rice_count_exact(const int32_t *res, int n, int k) > uint64_t count = 0; > > for (i = 0; i < n; i++) { > - int32_t v = -2 * res[i] - 1; > - v ^= v >> 31; > + unsigned v = ((unsigned)(res[i]) << 1) ^ (res[i] >> 31); > count += (v >> k) + 1 + k; > } > return count; > @@ -713,8 +726,8 @@ static uint64_t calc_rice_params(RiceContext *rc, > > tmp_rc.coding_mode = rc->coding_mode; > > - for (i = 0; i < n; i++) > - udata[i] = (2 * data[i]) ^ (data[i] >> 31); > + for (i = pred_order; i < n; i++) > + udata[i] = ((unsigned)(data[i]) << 1) ^ (data[i] >> 31); > > calc_sum_top(pmax, exact ? kmax : 0, udata, n, pred_order, sums); > > @@ -812,6 +825,180 @@ static void encode_residual_fixed(int32_t *res, const int32_t *smp, int n, > } > > > +static int encode_residual_fixed_with_residual_limit(int32_t *res, const int32_t *smp, > + int n, int order) > +{ > + /* This function checks for every residual whether it can be > + * contained in <INT32_MIN,INT32_MAX]. In case it doesn't, the > + * function that called this function has to try something else */ > + int i; > + int64_t res64; > + > + for (i = 0; i < order; i++) > + res[i] = smp[i]; > + > + if (order == 0) { > + for (i = order; i < n; i++) { > + if (smp[i] == INT32_MIN) > + return 1; > + res[i] = smp[i]; > + } > + } else if (order == 1) { > + for (i = order; i < n; i++) { > + res64 = (int64_t)smp[i] - smp[i-1]; > + if (res64 <= INT32_MIN || res64 > INT32_MAX) > + return 1; > + res[i] = res64; > + } > + } else if (order == 2) { > + for (i = order; i < n; i++) { > + res64 = (int64_t)smp[i] - 2*(int64_t)smp[i-1] + smp[i-2]; > + if (res64 <= INT32_MIN || res64 > INT32_MAX) > + return 1; > + res[i] = res64; > + } > + } else if (order == 3) { > + for (i = order; i < n; i++) { > + res64 = (int64_t)smp[i] - 3*(int64_t)smp[i-1] + 3*(int64_t)smp[i-2] - smp[i-3]; > + if (res64 <= INT32_MIN || res64 > INT32_MAX) > + return 1; > + res[i] = res64; > + } > + } else { > + for (i = order; i < n; i++) { > + res64 = (int64_t)smp[i] - 4*(int64_t)smp[i-1] + 6*(int64_t)smp[i-2] - 4*(int64_t)smp[i-3] + smp[i-4]; > + if (res64 <= INT32_MIN || res64 > INT32_MAX) > + return 1; > + res[i] = res64; > + } > + } > + return 0; > +} > + > + > +static int encode_residual_fixed_with_residual_limit_33bps(int32_t *res, const int64_t *smp, > + int n, int order) > +{ > + /* This function checks for every residual whether it can be > + * contained in <INT32_MIN,INT32_MAX]. In case it doesn't, the > + * function that called this function has to try something else */ > + int i; > + int64_t res64; > + > + if (order == 0) { > + for (i = order; i < n; i++) { > + if (smp[i] <= INT32_MIN || smp[i] > INT32_MAX) > + return 1; > + res[i] = smp[i]; > + } > + } else if (order == 1) { > + for (i = order; i < n; i++) { > + res64 = smp[i] - smp[i-1]; > + if (res64 <= INT32_MIN || res64 > INT32_MAX) > + return 1; > + res[i] = res64; > + } > + } else if (order == 2) { > + for (i = order; i < n; i++) { > + res64 = smp[i] - 2*smp[i-1] + smp[i-2]; > + if (res64 <= INT32_MIN || res64 > INT32_MAX) > + return 1; > + res[i] = res64; > + } > + } else if (order == 3) { > + for (i = order; i < n; i++) { > + res64 = smp[i] - 3*smp[i-1] + 3*smp[i-2] - smp[i-3]; > + if (res64 <= INT32_MIN || res64 > INT32_MAX) > + return 1; > + res[i] = res64; > + } > + } else { > + for (i = order; i < n; i++) { > + res64 = smp[i] - 4*smp[i-1] + 6*smp[i-2] - 4*smp[i-3] + smp[i-4]; > + if (res64 <= INT32_MIN || res64 > INT32_MAX) > + return 1; > + res[i] = res64; > + } > + } > + return 0; > +} > + > + > +static int lpc_encode_with_residual_limit(int32_t *res, const int32_t *smp, int len, > + int order, int32_t *coefs, int shift) > +{ > + /* This function checks for every residual whether it can be > + * contained in <INT32_MIN,INT32_MAX]. In case it doesn't, the > + * function that called this function has to try something else */ > + for (int i = 0; i < order; i++) > + res[i] = smp[i]; > + 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 (tmp <= INT32_MIN || tmp > INT32_MAX) > + return 1; > + res[i] = tmp; > + } > + return 0; > +} > + > +static int lpc_encode_with_residual_limit_33bps(int32_t *res, const int64_t *smp, int len, > + int order, int32_t *coefs, int shift) > +{ > + /* This function checks for every residual whether it can be > + * contained in <INT32_MIN,INT32_MAX]. In case it doesn't, the > + * function that called this function has to try something else */ > + 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 (tmp <= INT32_MIN || tmp > INT32_MAX) > + return 1; > + res[i] = tmp; > + } > + return 0; > +} > + > +static int lpc_encode_choose_datapath(FlacEncodeContext *s, int32_t bps, > + int32_t *res, const int32_t *smp, > + const int64_t *smp_33bps, int len, > + int order, int32_t *coefs, int shift) > +{ > + uint64_t max_residual_value = 0; > + int64_t max_sample_value = ((int64_t)(1) << (bps-1)); > + /* This calculates the max size of any residual with the current > + * predictor, so we know whether we need to check the residual */ > + for (int i = 0; i < order; i++) > + max_residual_value += FFABS(max_sample_value * coefs[i]); > + max_residual_value >>= shift; > + max_residual_value += max_sample_value; > + if (bps > 32) { > + if (lpc_encode_with_residual_limit_33bps(res, smp_33bps, len, order, coefs, shift)) > + return 1; > + } else if (max_residual_value > INT32_MAX) { > + if (lpc_encode_with_residual_limit(res, smp, len, order, coefs, shift)) > + return 1; > + } else if (bps + s->options.lpc_coeff_precision + av_log2(order) <= 32) { > + s->flac_dsp.lpc16_encode(res, smp, len, order, coefs, shift); > + } else { > + s->flac_dsp.lpc32_encode(res, smp, len, order, coefs, shift); > + } > + return 0; > +} > + > +#define DEFAULT_TO_VERBATIM() \ > +{ \ > + sub->type = sub->type_code = FLAC_SUBFRAME_VERBATIM; \ > + if (sub->obits <= 32) \ > + memcpy(res, smp, n * sizeof(int32_t)); \ > + return subframe_count_exact(s, sub, 0); \ > +} > + > static int encode_residual_ch(FlacEncodeContext *s, int ch) > { > int i, n; > @@ -821,28 +1008,38 @@ static int encode_residual_ch(FlacEncodeContext *s, int ch) > int32_t coefs[MAX_LPC_ORDER][MAX_LPC_ORDER]; > int shift[MAX_LPC_ORDER]; > int32_t *res, *smp; > + int64_t *smp_33bps; > > - frame = &s->frame; > - sub = &frame->subframes[ch]; > - res = sub->residual; > - smp = sub->samples; > - n = frame->blocksize; > + frame = &s->frame; > + sub = &frame->subframes[ch]; > + res = sub->residual; > + smp = sub->samples; > + smp_33bps = frame->samples_33bps; > + n = frame->blocksize; > > /* CONSTANT */ > - for (i = 1; i < n; i++) > - if(smp[i] != smp[0]) > - break; > - if (i == n) { > - sub->type = sub->type_code = FLAC_SUBFRAME_CONSTANT; > - res[0] = smp[0]; > - return subframe_count_exact(s, sub, 0); > + if (sub->obits > 32) { > + for (i = 1; i < n; i++) > + if(smp_33bps[i] != smp_33bps[0]) > + break; > + if (i == n) { > + sub->type = sub->type_code = FLAC_SUBFRAME_CONSTANT; > + return subframe_count_exact(s, sub, 0); > + } > + } else { > + for (i = 1; i < n; i++) > + if(smp[i] != smp[0]) > + break; > + if (i == n) { > + sub->type = sub->type_code = FLAC_SUBFRAME_CONSTANT; > + res[0] = smp[0]; > + return subframe_count_exact(s, sub, 0); > + } > } > > /* VERBATIM */ > if (frame->verbatim_only || n < 5) { > - sub->type = sub->type_code = FLAC_SUBFRAME_VERBATIM; > - memcpy(res, smp, n * sizeof(int32_t)); > - return subframe_count_exact(s, sub, 0); > + DEFAULT_TO_VERBATIM(); > } > > min_order = s->options.min_prediction_order; > @@ -859,15 +1056,32 @@ static int encode_residual_ch(FlacEncodeContext *s, int ch) > opt_order = 0; > bits[0] = UINT32_MAX; > for (i = min_order; i <= max_order; i++) { > - encode_residual_fixed(res, smp, n, i); > + if (sub->obits == 33) { > + if (encode_residual_fixed_with_residual_limit_33bps(res, smp_33bps, n, i)) > + continue; > + } else if (sub->obits + i >= 32) { > + if (encode_residual_fixed_with_residual_limit(res, smp, n, i)) > + continue; > + } else > + encode_residual_fixed(res, smp, n, i); > bits[i] = find_subframe_rice_params(s, sub, i); > if (bits[i] < bits[opt_order]) > opt_order = i; > } > + if (opt_order == 0 && bits[0] == UINT32_MAX) { > + /* No predictor found with residuals within <INT32_MIN,INT32_MAX], > + * so encode a verbatim subframe instead */ > + DEFAULT_TO_VERBATIM(); > + } > sub->order = opt_order; > sub->type_code = sub->type | sub->order; > if (sub->order != max_order) { > - encode_residual_fixed(res, smp, n, sub->order); > + if (sub->obits == 33) > + encode_residual_fixed_with_residual_limit_33bps(res, smp_33bps, n, sub->order); > + else if (sub->obits + i >= 32) > + encode_residual_fixed_with_residual_limit(res, smp, n, sub->order); > + else > + encode_residual_fixed(res, smp, n, sub->order); > find_subframe_rice_params(s, sub, sub->order); > } > return subframe_count_exact(s, sub, sub->order); > @@ -875,6 +1089,14 @@ static int encode_residual_ch(FlacEncodeContext *s, int ch) > > /* LPC */ > sub->type = FLAC_SUBFRAME_LPC; > + if (sub->obits == 33) > + /* As ff_lpc_calc_coefs is shared with other codecs and the LSB > + * probably isn't predictable anyway, throw away LSB for analysis > + * so it fits 32 bit int and existing function can be used > + * unmodified */ > + for (i = 0; i < n; i++) > + smp[i] = smp_33bps[i] >> 1; > + > opt_order = ff_lpc_calc_coefs(&s->lpc_ctx, smp, n, min_order, max_order, > s->options.lpc_coeff_precision, coefs, shift, s->options.lpc_type, > s->options.lpc_passes, omethod, > @@ -895,13 +1117,8 @@ 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) { > - s->flac_dsp.lpc16_encode(res, smp, n, order+1, coefs[order], > - shift[order]); > - } else { > - s->flac_dsp.lpc32_encode(res, smp, n, order+1, coefs[order], > - shift[order]); > - } > + if(lpc_encode_choose_datapath(s, sub->obits, res, smp, smp_33bps, n, order+1, coefs[order], shift[order])) > + continue; > bits[i] = find_subframe_rice_params(s, sub, order+1); > if (bits[i] < bits[opt_index]) { > opt_index = i; > @@ -915,11 +1132,8 @@ 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) { > - 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]); > - } > + if(lpc_encode_choose_datapath(s, sub->obits, res, smp, smp_33bps, n, i+1, coefs[i], shift[i])) > + continue; > bits[i] = find_subframe_rice_params(s, sub, i+1); > if (bits[i] < bits[opt_order]) > opt_order = i; > @@ -937,11 +1151,8 @@ 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) { > - 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]); > - } > + if(lpc_encode_choose_datapath(s, sub->obits, res, smp, smp_33bps, n, i+1, coefs[i], shift[i])) > + continue; > bits[i] = find_subframe_rice_params(s, sub, i+1); > if (bits[i] < bits[opt_order]) > opt_order = i; > @@ -978,11 +1189,8 @@ 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) { > - 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]); > - } > + if(lpc_encode_choose_datapath(s, sub->obits, res, smp, smp_33bps, n, opt_order, lpc_try, shift[opt_order-1])) > + continue; > score = find_subframe_rice_params(s, sub, opt_order); > if (score < best_score) { > best_score = score; > @@ -999,10 +1207,10 @@ 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) { > - 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); > + if(lpc_encode_choose_datapath(s, sub->obits, res, smp, smp_33bps, n, sub->order, sub->coefs, sub->shift)) { > + /* No predictor found with residuals within <INT32_MIN,INT32_MAX], > + * so encode a verbatim subframe instead */ > + DEFAULT_TO_VERBATIM(); > } > > find_subframe_rice_params(s, sub, sub->order); > @@ -1069,54 +1277,88 @@ static int encode_frame(FlacEncodeContext *s) > > static void remove_wasted_bits(FlacEncodeContext *s) > { > - int ch, i; > + int ch, i, wasted_bits; > > for (ch = 0; ch < s->channels; ch++) { > FlacSubframe *sub = &s->frame.subframes[ch]; > - int32_t v = 0; > > - for (i = 0; i < s->frame.blocksize; i++) { > - v |= sub->samples[i]; > - if (v & 1) > - break; > - } > + if (sub->obits > 32) { > + int64_t v = 0; > + for (i = 0; i < s->frame.blocksize; i++) { > + v |= s->frame.samples_33bps[i]; > + if (v & 1) > + break; > + } > + > + if (!v || (v & 1)) > + return; > > - if (v && !(v & 1)) { > v = ff_ctz(v); > > + /* If any wasted bits are found, samples are moved > + * from frame.samples_33bps to frame.subframes[ch] */ > for (i = 0; i < s->frame.blocksize; i++) > - sub->samples[i] >>= v; > + sub->samples[i] = s->frame.samples_33bps[i] >> v; > + wasted_bits = v; > + } else { > + int32_t v = 0; > + for (i = 0; i < s->frame.blocksize; i++) { > + v |= sub->samples[i]; > + if (v & 1) > + break; > + } > + > + if (!v || (v & 1)) > + return; > > - sub->wasted = v; > - sub->obits -= v; > + v = ff_ctz(v); > > - /* for 24-bit, check if removing wasted bits makes the range better > - suited for using RICE instead of RICE2 for entropy coding */ > - if (sub->obits <= 17) > - sub->rc.coding_mode = CODING_MODE_RICE; > + for (i = 0; i < s->frame.blocksize; i++) > + sub->samples[i] >>= v; > + wasted_bits = v; > } > + > + sub->wasted = wasted_bits; > + sub->obits -= wasted_bits; > + > + /* for 24-bit, check if removing wasted bits makes the range better > + * suited for using RICE instead of RICE2 for entropy coding */ > + if (sub->obits <= 17) > + sub->rc.coding_mode = CODING_MODE_RICE; > } > } > > > static int estimate_stereo_mode(const int32_t *left_ch, const int32_t *right_ch, int n, > - int max_rice_param) > + int max_rice_param, int bps) > { > int i, best; > - int32_t lt, rt; > uint64_t sum[4]; > uint64_t score[4]; > int k; > > /* calculate sum of 2nd order residual for each channel */ > sum[0] = sum[1] = sum[2] = sum[3] = 0; > - for (i = 2; i < n; i++) { > - lt = left_ch[i] - 2*left_ch[i-1] + left_ch[i-2]; > - rt = right_ch[i] - 2*right_ch[i-1] + right_ch[i-2]; > - sum[2] += FFABS((lt + rt) >> 1); > - sum[3] += FFABS(lt - rt); > - sum[0] += FFABS(lt); > - sum[1] += FFABS(rt); > + if(bps < 30) { > + int32_t lt, rt; > + for (i = 2; i < n; i++) { > + lt = left_ch[i] - 2*left_ch[i-1] + left_ch[i-2]; > + rt = right_ch[i] - 2*right_ch[i-1] + right_ch[i-2]; > + sum[2] += FFABS((lt + rt) >> 1); > + sum[3] += FFABS(lt - rt); > + sum[0] += FFABS(lt); > + sum[1] += FFABS(rt); > + } > + } else { > + int64_t lt, rt; > + for (i = 2; i < n; i++) { > + lt = (int64_t)left_ch[i] - 2*(int64_t)left_ch[i-1] + left_ch[i-2]; > + rt = (int64_t)right_ch[i] - 2*(int64_t)right_ch[i-1] + right_ch[i-2]; > + sum[2] += FFABS((lt + rt) >> 1); > + sum[3] += FFABS(lt - rt); > + sum[0] += FFABS(lt); > + sum[1] += FFABS(rt); > + } > } > /* estimate bit counts */ > for (i = 0; i < 4; i++) { > @@ -1147,12 +1389,14 @@ static void channel_decorrelation(FlacEncodeContext *s) > { > FlacFrame *frame; > int32_t *left, *right; > + int64_t *side_33bps; > int i, n; > > - frame = &s->frame; > - n = frame->blocksize; > - left = frame->subframes[0].samples; > - right = frame->subframes[1].samples; > + frame = &s->frame; > + n = frame->blocksize; > + left = frame->subframes[0].samples; > + right = frame->subframes[1].samples; > + side_33bps = frame->samples_33bps; > > if (s->channels != 2) { > frame->ch_mode = FLAC_CHMODE_INDEPENDENT; > @@ -1161,29 +1405,49 @@ static void channel_decorrelation(FlacEncodeContext *s) > > if (s->options.ch_mode < 0) { > int max_rice_param = (1 << frame->subframes[0].rc.coding_mode) - 2; > - frame->ch_mode = estimate_stereo_mode(left, right, n, max_rice_param); > + frame->ch_mode = estimate_stereo_mode(left, right, n, max_rice_param, s->avctx->bits_per_raw_sample); > } else > frame->ch_mode = s->options.ch_mode; > > /* perform decorrelation and adjust bits-per-sample */ > if (frame->ch_mode == FLAC_CHMODE_INDEPENDENT) > return; > - if (frame->ch_mode == FLAC_CHMODE_MID_SIDE) { > - int32_t tmp; > - for (i = 0; i < n; i++) { > - tmp = left[i]; > - left[i] = (tmp + right[i]) >> 1; > - right[i] = tmp - right[i]; > + if(s->avctx->bits_per_raw_sample == 32) { > + if (frame->ch_mode == FLAC_CHMODE_MID_SIDE) { > + int64_t tmp; > + for (i = 0; i < n; i++) { > + tmp = left[i]; > + left[i] = (tmp + right[i]) >> 1; > + side_33bps[i] = tmp - right[i]; > + } > + frame->subframes[1].obits++; > + } else if (frame->ch_mode == FLAC_CHMODE_LEFT_SIDE) { > + for (i = 0; i < n; i++) > + side_33bps[i] = (int64_t)left[i] - right[i]; > + frame->subframes[1].obits++; > + } else { > + for (i = 0; i < n; i++) > + side_33bps[i] = (int64_t)left[i] - right[i]; > + frame->subframes[0].obits++; > } > - frame->subframes[1].obits++; > - } else if (frame->ch_mode == FLAC_CHMODE_LEFT_SIDE) { > - for (i = 0; i < n; i++) > - right[i] = left[i] - right[i]; > - frame->subframes[1].obits++; > } else { > - for (i = 0; i < n; i++) > - left[i] -= right[i]; > - frame->subframes[0].obits++; > + if (frame->ch_mode == FLAC_CHMODE_MID_SIDE) { > + int32_t tmp; > + for (i = 0; i < n; i++) { > + tmp = left[i]; > + left[i] = (tmp + right[i]) >> 1; > + right[i] = tmp - right[i]; > + } > + frame->subframes[1].obits++; > + } else if (frame->ch_mode == FLAC_CHMODE_LEFT_SIDE) { > + for (i = 0; i < n; i++) > + right[i] = left[i] - right[i]; > + frame->subframes[1].obits++; > + } else { > + for (i = 0; i < n; i++) > + left[i] -= right[i]; > + frame->subframes[0].obits++; > + } > } > } > > @@ -1252,14 +1516,38 @@ static void write_subframes(FlacEncodeContext *s) > > /* subframe */ > if (sub->type == FLAC_SUBFRAME_CONSTANT) { > - put_sbits(&s->pb, sub->obits, res[0]); > + if(sub->obits == 33) > + put_sbits64(&s->pb, 33, s->frame.samples_33bps[0]); > + else if(sub->obits == 32) > + put_bits32(&s->pb, res[0]); > + else > + 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 == 33) { > + int64_t *res64 = s->frame.samples_33bps; > + int64_t *frame_end64 = &s->frame.samples_33bps[s->frame.blocksize]; > + while (res64 < frame_end64) > + put_sbits64(&s->pb, 33, (*res64++)); > + } else if (sub->obits == 32) { > + while (res < frame_end) > + put_bits32(&s->pb, *res++); > + } else { > + while (res < frame_end) > + put_sbits(&s->pb, sub->obits, *res++); > + } > } else { > /* warm-up samples */ > - for (i = 0; i < sub->order; i++) > - put_sbits(&s->pb, sub->obits, *res++); > + if (sub->obits == 33) { > + for (i = 0; i < sub->order; i++) > + put_sbits64(&s->pb, 33, s->frame.samples_33bps[i]); > + res += sub->order; > + } else if (sub->obits == 32) { > + for (i = 0; i < sub->order; i++) > + put_bits32(&s->pb, *res++); > + } else { > + for (i = 0; i < sub->order; i++) > + put_sbits(&s->pb, sub->obits, *res++); > + } > > /* LPC coefficients */ > if (sub->type == FLAC_SUBFRAME_LPC) { > @@ -1284,7 +1572,7 @@ static void write_subframes(FlacEncodeContext *s) > int k = sub->rc.params[p]; > put_bits(&s->pb, sub->rc.coding_mode, k); > while (res < part_end) > - set_sr_golomb_flac(&s->pb, *res++, k, INT32_MAX, 0); > + set_sr_golomb_flac(&s->pb, *res++, k); > part_end = FFMIN(frame_end, part_end + psize); > } > } > @@ -1332,7 +1620,7 @@ 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; > @@ -1342,6 +1630,15 @@ 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); > > diff --git a/libavcodec/put_bits.h b/libavcodec/put_bits.h > index 4b4f977ad5..11360fe7a4 100644 > --- a/libavcodec/put_bits.h > +++ b/libavcodec/put_bits.h > @@ -363,6 +363,13 @@ static inline void put_bits64(PutBitContext *s, int n, uint64_t value) > } > } > > +static inline void put_sbits64(PutBitContext *pb, int n, int64_t value) > +{ > + av_assert2(n >= 0 && n <= 64); > + > + put_bits64(pb, n, (uint64_t)(value) & (~(UINT64_MAX << n))); Shifting by 64 bits here is UB, so better modify the assert to disallow it. And rename the function to put_sbits63(). > +} > + > /** > * Return the pointer to the byte where the bitstream writer will put > * the next bit. > diff --git a/libavcodec/put_golomb.h b/libavcodec/put_golomb.h > index 9ca911fc3c..ca4b182ae8 100644 > --- a/libavcodec/put_golomb.h > +++ b/libavcodec/put_golomb.h > @@ -154,15 +154,22 @@ static inline void set_sr_golomb(PutBitContext *pb, int i, int k, int limit, > /** > * write signed golomb rice code (flac). > */ > -static inline void set_sr_golomb_flac(PutBitContext *pb, int i, int k, > - int limit, int esc_len) > +static inline void set_sr_golomb_flac(PutBitContext *pb, int i, int k) This seems to be only used by flacenc.c, so IMO it would be better to move it there. > { > - int v; > + unsigned v, e; > > - v = -2 * i - 1; > - v ^= (v >> 31); > + v = ((unsigned)(i) << 1) ^ (i >> 31); > > - set_ur_golomb_jpegls(pb, v, k, limit, esc_len); > + e = (v >> k) + 1; > + while (e > 31) { > + put_bits(pb, 31, 0); > + e -= 31; > + } > + put_bits(pb, e, 1); > + if (k) { > + unsigned mask = UINT32_MAX >> (32-k); > + put_bits(pb, k, v & mask); > + } > } > > #endif /* AVCODEC_PUT_GOLOMB_H */ _______________________________________________ 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". ^ permalink raw reply [flat|nested] 6+ messages in thread
* Re: [FFmpeg-devel] [PATCH v3 2/2] libavcodec/flacenc: Implement encoding of 32 bit-per-sample PCM 2022-09-02 15:10 ` Andreas Rheinhardt @ 2022-09-02 18:43 ` Martijn van Beurden 2022-09-02 19:04 ` Andreas Rheinhardt 0 siblings, 1 reply; 6+ messages in thread From: Martijn van Beurden @ 2022-09-02 18:43 UTC (permalink / raw) To: FFmpeg development discussions and patches First of all, thanks for reviewing. Op vr 2 sep. 2022 om 17:11 schreef Andreas Rheinhardt < andreas.rheinhardt@outlook.com>: > > +static inline void put_sbits64(PutBitContext *pb, int n, int64_t value) > > +{ > > + av_assert2(n >= 0 && n <= 64); > > + > > + put_bits64(pb, n, (uint64_t)(value) & (~(UINT64_MAX << n))); > > Shifting by 64 bits here is UB, so better modify the assert to disallow > it. And rename the function to put_sbits63(). > > I could also add specific handling for the 64-bit case. Perhaps something like put_bits64(pb, n, (uint64_t)(value) & (~((n < 64)?(UINT64_MAX << n):0))); or should I leave that to whoever needs that functionality? > > -static inline void set_sr_golomb_flac(PutBitContext *pb, int i, int k, > > - int limit, int esc_len) > > +static inline void set_sr_golomb_flac(PutBitContext *pb, int i, int k) > > This seems to be only used by flacenc.c, so IMO it would be better to > move it there. > > Yes, I was wondering what to do with this, similarly with the functions in get_bits, mathops and put_bits. I suppose the additions to get_bits, mathops and put_bits might be useful to others in the future. The golomb code is probably not useful for other codecs indeed. Is that reasoning valid? _______________________________________________ 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". ^ permalink raw reply [flat|nested] 6+ messages in thread
* Re: [FFmpeg-devel] [PATCH v3 2/2] libavcodec/flacenc: Implement encoding of 32 bit-per-sample PCM 2022-09-02 18:43 ` Martijn van Beurden @ 2022-09-02 19:04 ` Andreas Rheinhardt 0 siblings, 0 replies; 6+ messages in thread From: Andreas Rheinhardt @ 2022-09-02 19:04 UTC (permalink / raw) To: ffmpeg-devel Martijn van Beurden: > First of all, thanks for reviewing. > > Op vr 2 sep. 2022 om 17:11 schreef Andreas Rheinhardt < > andreas.rheinhardt@outlook.com>: > >>> +static inline void put_sbits64(PutBitContext *pb, int n, int64_t value) >>> +{ >>> + av_assert2(n >= 0 && n <= 64); >>> + >>> + put_bits64(pb, n, (uint64_t)(value) & (~(UINT64_MAX << n))); >> >> Shifting by 64 bits here is UB, so better modify the assert to disallow >> it. And rename the function to put_sbits63(). >> >> > I could also add specific handling for the 64-bit case. Perhaps something > like > > put_bits64(pb, n, (uint64_t)(value) & (~((n < 64)?(UINT64_MAX << n):0))); > > or should I leave that to whoever needs that functionality? > Leave it to whoever needs this. > >>> -static inline void set_sr_golomb_flac(PutBitContext *pb, int i, int k, >>> - int limit, int esc_len) >>> +static inline void set_sr_golomb_flac(PutBitContext *pb, int i, int k) >> >> This seems to be only used by flacenc.c, so IMO it would be better to >> move it there. >> >> > Yes, I was wondering what to do with this, similarly with the functions in > get_bits, mathops and put_bits. I suppose the additions to get_bits, > mathops and put_bits might be useful to others in the future. The golomb > code is probably not useful for other codecs indeed. Is that reasoning > valid? Yes. _______________________________________________ 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". ^ permalink raw reply [flat|nested] 6+ messages in thread
end of thread, other threads:[~2022-09-02 19:05 UTC | newest] Thread overview: 6+ messages (download: mbox.gz / follow: Atom feed) -- links below jump to the message on this page -- 2022-09-02 14:20 [FFmpeg-devel] [PATCH v3 0/2] 32bps FLAC patches Martijn van Beurden 2022-09-02 14:20 ` [FFmpeg-devel] [PATCH v3 1/2] libavcodec/flacdec: Implement decoding of 32 bit-per-sample PCM Martijn van Beurden 2022-09-02 14:20 ` [FFmpeg-devel] [PATCH v3 2/2] libavcodec/flacenc: Implement encoding " Martijn van Beurden 2022-09-02 15:10 ` Andreas Rheinhardt 2022-09-02 18:43 ` Martijn van Beurden 2022-09-02 19:04 ` Andreas Rheinhardt
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