[FFmpeg-devel] [PATCH v7 0/3] 32bps FLAC patches

[FFmpeg-devel] [PATCH v7 0/3] 32bps FLAC patches

[Martijn van Beurden]

Recently libFLAC gained the ability (first released in FLAC 1.4.0)
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.

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

Changes since v3:
Moved put_golomb part to flacenc.c (as it is now quite specific to FLAC)
and renamed put_sbits64 to put_sbits63 (and changed assert accordingly)

Changes since v4:
Fix check of decoded_buffer_33bps malloc. Fix reading of wasted bits
for 33bps subframe. Add fate test

Changes since v5:
Slimmed down 32 bps fate flac test from 2 to 1 input file. Replaced
-strict -2 with -strict experimental in fate and encoder warning.

Changes since v6:
Deduplicated function bodies with preprocessor macros in both
encoder and decoder, declared variables in loop body where possible
and changed ff_ctz to ff_ctzll where necessary

Martijn van Beurden (3):
  libavcodec/flacdec: Implement decoding of 32 bit-per-sample PCM
  libavcodec/flacenc: Implement encoding of 32 bit-per-sample PCM
  fate/flac: Add test of 32 bps encoding/decoding

 libavcodec/flac.c       |   4 +-
 libavcodec/flacdec.c    | 218 ++++++++++++++++--
 libavcodec/flacenc.c    | 480 +++++++++++++++++++++++++++++++---------
 libavcodec/get_bits.h   |  12 +
 libavcodec/mathops.h    |   9 +
 libavcodec/put_bits.h   |   7 +
 libavcodec/put_golomb.h |  14 --
 tests/fate/flac.mak     |   4 +
 8 files changed, 602 insertions(+), 146 deletions(-)

-- 
2.30.2

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[FFmpeg-devel] [PATCH v7 1/3] libavcodec/flacdec: Implement decoding of 32 bit-per-sample PCM

[Martijn van Beurden]

Add decoding of FLAC files coding for 32 bit-per-sample PCM to libavcodec.
---
 libavcodec/flac.c     |   4 +-
 libavcodec/flacdec.c  | 218 +++++++++++++++++++++++++++++++++++++-----
 libavcodec/get_bits.h |  12 +++
 libavcodec/mathops.h  |   9 ++
 4 files changed, 219 insertions(+), 24 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 5b8547a98f..56ef033a42 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_33bps)
+            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 +352,62 @@ static int decode_subframe_fixed(FLACContext *s, int32_t *decoded,
     return 0;
 }
 
+#define DECODER_SUBFRAME_FIXED_WIDE(residual) {                       \
+    const int blocksize = s->blocksize;                               \
+    int ret;                                                          \
+                                                                      \
+    if ((ret = decode_residuals(s, residual, pred_order)) < 0)        \
+        return ret;                                                   \
+                                                                      \
+    switch (pred_order) {                                             \
+    case 0:                                                           \
+	for (int i = pred_order; i < blocksize; i++)                  \
+            decoded[i] = residual[i];                                 \
+        break;                                                        \
+    case 1:                                                           \
+        for (int i = pred_order; i < blocksize; i++)                  \
+            decoded[i] = (int64_t)residual[i] + (int64_t)decoded[i-1];\
+        break;                                                        \
+    case 2:                                                           \
+        for (int i = pred_order; i < blocksize; i++)                  \
+            decoded[i] = (int64_t)residual[i] + 2*(int64_t)decoded[i-1] - (int64_t)decoded[i-2];  \
+        break;                                                        \
+    case 3:                                                           \
+        for (int i = pred_order; i < blocksize; i++)                  \
+            decoded[i] = (int64_t)residual[i] + 3*(int64_t)decoded[i-1] - 3*(int64_t)decoded[i-2] + (int64_t)decoded[i-3];   \
+        break;                                                        \
+    case 4:                                                           \
+        for (int i = pred_order; i < blocksize; i++)                  \
+            decoded[i] = (int64_t)residual[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_wide(FLACContext *s, int32_t *decoded,
+                                      int pred_order, int bps)
+{
+    /* warm up samples */
+    for (int i = 0; i < pred_order; i++) {
+        decoded[i] = get_sbits_long(&s->gb, bps);
+    }
+    DECODER_SUBFRAME_FIXED_WIDE(decoded);
+}
+
+
+static int decode_subframe_fixed_33bps(FLACContext *s, int64_t *decoded,
+                                       int32_t *residual, int pred_order)
+{
+    /* warm up samples */                                             \
+    for (int i = 0; i < pred_order; i++) {                            \
+        decoded[i] = get_sbits64(&s->gb, 33);                         \
+    }                                                                 \
+    DECODER_SUBFRAME_FIXED_WIDE(residual);
+}
+
 static void lpc_analyze_remodulate(SUINT32 *decoded, const int coeffs[32],
                                    int order, int qlevel, int len, int bps)
 {
@@ -402,12 +479,53 @@ 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)
@@ -427,7 +545,7 @@ static inline int decode_subframe(FLACContext *s, int channel)
         int left = get_bits_left(&s->gb);
         if ( left <= 0 ||
             (left < bps && !show_bits_long(&s->gb, left)) ||
-                           !show_bits_long(&s->gb, bps)) {
+                           !show_bits_long(&s->gb, bps-1)) {
             av_log(s->avctx, AV_LOG_ERROR,
                    "Invalid number of wasted bits > available bits (%d) - left=%d\n",
                    bps, left);
@@ -436,34 +554,63 @@ 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 +701,26 @@ 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 +779,15 @@ 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 +808,7 @@ 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 c89054d6ed..a1dc323304 100644
--- a/libavcodec/mathops.h
+++ b/libavcodec/mathops.h
@@ -138,6 +138,15 @@ 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

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[FFmpeg-devel] [PATCH v7 2/3] libavcodec/flacenc: Implement encoding of 32 bit-per-sample PCM

[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 experimental is needed to get ffmpeg to encode such files.
---
 libavcodec/flacenc.c    | 480 +++++++++++++++++++++++++++++++---------
 libavcodec/put_bits.h   |   7 +
 libavcodec/put_golomb.h |  14 --
 3 files changed, 379 insertions(+), 122 deletions(-)

diff --git a/libavcodec/flacenc.c b/libavcodec/flacenc.c
index 5d8c3f82be..1e4eb0d319 100644
--- a/libavcodec/flacenc.c
+++ b/libavcodec/flacenc.c
@@ -31,7 +31,6 @@
 #include "codec_internal.h"
 #include "encode.h"
 #include "put_bits.h"
-#include "put_golomb.h"
 #include "lpc.h"
 #include "flac.h"
 #include "flacdata.h"
@@ -95,6 +94,7 @@ 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 +282,22 @@ 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 experimental if you want "
+                   "to encode more than 24 bits-per-sample\n");
+            avctx->bits_per_raw_sample = 24;
+            s->bps_code                = 6;
+        } else {
+            avctx->bits_per_raw_sample = 32;
+            s->bps_code = 7;
+        }
         break;
     }
 
@@ -533,8 +545,7 @@ 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 +724,8 @@ 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 +823,130 @@ static void encode_residual_fixed(int32_t *res, const int32_t *smp, int n,
 }
 
 
+/* These four functions check 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.
+ * Each function is duplicated, once for int32_t input, once for
+ * int64_t input */
+#define ENCODE_RESIDUAL_FIXED_WITH_RESIDUAL_LIMIT()                   \
+{                                                                     \
+    for (int i = 0; i < order; i++)                                   \
+        res[i] = smp[i];                                              \
+    if (order == 0) {                                                 \
+        for (int i = order; i < n; i++) {                             \
+            if (smp[i] == INT32_MIN)                                  \
+                return 1;                                             \
+            res[i] = smp[i];                                          \
+        }                                                             \
+    } else if (order == 1) {                                          \
+        for (int i = order; i < n; i++) {                             \
+            int64_t 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 (int i = order; i < n; i++) {                             \
+            int64_t 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 (int i = order; i < n; i++) {                             \
+            int64_t 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 (int i = order; i < n; i++) {                             \
+            int64_t 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(int32_t *res, const int32_t *smp,
+                                                      int n, int order)
+{
+    ENCODE_RESIDUAL_FIXED_WITH_RESIDUAL_LIMIT();
+}
+
+
+static int encode_residual_fixed_with_residual_limit_33bps(int32_t *res, const int64_t *smp,
+                                                           int n, int order)
+{
+    ENCODE_RESIDUAL_FIXED_WITH_RESIDUAL_LIMIT();
+}
+
+#define LPC_ENCODE_WITH_RESIDUAL_LIMIT()                 \
+{                                                        \
+   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(int32_t *res, const int32_t *smp, int len,
+                                               int order, int32_t *coefs, int shift)
+{
+    LPC_ENCODE_WITH_RESIDUAL_LIMIT();
+}
+
+static int lpc_encode_with_residual_limit_33bps(int32_t *res, const int64_t *smp, int len,
+                                               int order, int32_t *coefs, int shift)
+{
+    LPC_ENCODE_WITH_RESIDUAL_LIMIT();
+}
+
+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 +956,38 @@ 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 +1004,32 @@ 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 +1037,14 @@ 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 +1065,8 @@ 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 +1080,8 @@ 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 +1099,8 @@ 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 +1137,8 @@ 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 +1155,10 @@ 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);
@@ -1069,57 +1225,91 @@ 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;
+
+            v = ff_ctzll(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] = 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;
 
-        if (v && !(v & 1)) {
             v = ff_ctz(v);
 
             for (i = 0; i < s->frame.blocksize; i++)
                 sub->samples[i] >>= v;
+            wasted_bits = v;
+        }
 
-            sub->wasted = v;
-            sub->obits -= 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;
-        }
+        /* 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;
+    int best;
     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 (int 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 (int 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++) {
+    for (int i = 0; i < 4; i++) {
         k      = find_optimal_param(2 * sum[i], n, max_rice_param);
         sum[i] = rice_encode_count( 2 * sum[i], n, k);
     }
@@ -1132,7 +1322,7 @@ static int estimate_stereo_mode(const int32_t *left_ch, const int32_t *right_ch,
 
     /* return mode with lowest score */
     best = 0;
-    for (i = 1; i < 4; i++)
+    for (int i = 1; i < 4; i++)
         if (score[i] < score[best])
             best = i;
 
@@ -1147,12 +1337,14 @@ static void channel_decorrelation(FlacEncodeContext *s)
 {
     FlacFrame *frame;
     int32_t *left, *right;
-    int i, n;
+    int64_t *side_33bps;
+    int 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 +1353,49 @@ 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 (int 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 (int i = 0; i < n; i++)
+                side_33bps[i] = (int64_t)left[i] - right[i];
+            frame->subframes[1].obits++;
+        } else {
+            for (int 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 (int 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 (int i = 0; i < n; i++)
+                right[i] = left[i] - right[i];
+            frame->subframes[1].obits++;
+        } else {
+            for (int i = 0; i < n; i++)
+                left[i] -= right[i];
+            frame->subframes[0].obits++;
+        }
     }
 }
 
@@ -1232,13 +1444,32 @@ static void write_frame_header(FlacEncodeContext *s)
 }
 
 
+static inline void set_sr_golomb_flac(PutBitContext *pb, int i, int k)
+{
+    unsigned v, e;
+
+    v = ((unsigned)(i) << 1) ^ (i >> 31);
+
+    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);
+    }
+}
+
+
 static void write_subframes(FlacEncodeContext *s)
 {
     int ch;
 
     for (ch = 0; ch < s->channels; ch++) {
         FlacSubframe *sub = &s->frame.subframes[ch];
-        int i, p, porder, psize;
+        int p, porder, psize;
         int32_t *part_end;
         int32_t *res       =  sub->residual;
         int32_t *frame_end = &sub->residual[s->frame.blocksize];
@@ -1252,21 +1483,45 @@ 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_sbits63(&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_sbits63(&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 (int i = 0; i < sub->order; i++)
+                    put_sbits63(&s->pb, 33, s->frame.samples_33bps[i]);
+                res += sub->order;
+            } else if (sub->obits == 32) {
+                for (int i = 0; i < sub->order; i++)
+                    put_bits32(&s->pb, *res++);
+            } else {
+                for (int i = 0; i < sub->order; i++)
+                    put_sbits(&s->pb, sub->obits, *res++);
+            }
 
             /* LPC coefficients */
             if (sub->type == FLAC_SUBFRAME_LPC) {
                 int cbits = s->options.lpc_coeff_precision;
                 put_bits( &s->pb, 4, cbits-1);
                 put_sbits(&s->pb, 5, sub->shift);
-                for (i = 0; i < sub->order; i++)
+                for (int i = 0; i < sub->order; i++)
                     put_sbits(&s->pb, cbits, sub->coefs[i]);
             }
 
@@ -1284,7 +1539,7 @@ 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 +1587,7 @@ 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 +1597,15 @@ 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..4561dc131a 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_sbits63(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..df47fd2c0a 100644
--- a/libavcodec/put_golomb.h
+++ b/libavcodec/put_golomb.h
@@ -151,18 +151,4 @@ static inline void set_sr_golomb(PutBitContext *pb, int i, int k, int limit,
     set_ur_golomb(pb, v, k, limit, esc_len);
 }
 
-/**
- * write signed golomb rice code (flac).
- */
-static inline void set_sr_golomb_flac(PutBitContext *pb, int i, int k,
-                                      int limit, int esc_len)
-{
-    int v;
-
-    v  = -2 * i - 1;
-    v ^= (v >> 31);
-
-    set_ur_golomb_jpegls(pb, v, k, limit, esc_len);
-}
-
 #endif /* AVCODEC_PUT_GOLOMB_H */
-- 
2.30.2

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[FFmpeg-devel] [PATCH v7 3/3] fate/flac: Add test of 32 bps encoding/decoding

[Martijn van Beurden]

---
 tests/fate/flac.mak | 4 ++++
 1 file changed, 4 insertions(+)

diff --git a/tests/fate/flac.mak b/tests/fate/flac.mak
index 115cc965e1..3424b2bf82 100644
--- a/tests/fate/flac.mak
+++ b/tests/fate/flac.mak
@@ -6,6 +6,7 @@ FATE_FLAC += fate-flac-16-chmode-indep                                  \
              fate-flac-16-lpc-cholesky                                  \
              fate-flac-16-lpc-levinson                                  \
              fate-flac-24-comp-8                                        \
+             fate-flac-32-wasted-bits                                   \
              fate-flac-rice-params                                      \
 
 fate-flac-16-chmode-%: OPTS = -ch_mode $(@:fate-flac-16-chmode-%=%)
@@ -20,6 +21,9 @@ fate-flac-24-comp-%: OPTS = -compression_level $(@:fate-flac-24-comp-%=%)
 fate-flac-24-%: REF = $(SAMPLES)/audio-reference/divertimenti_2ch_96kHz_s24.wav
 fate-flac-24-%: CMD = enc_dec_pcm flac wav s24le $(subst $(SAMPLES),$(TARGET_SAMPLES),$(REF)) -c flac $(OPTS)
 
+fate-flac-32-wasted-bits: REF = $(SAMPLES)/audio-reference/drums_2ch_44kHz_s32_wastedbits.wav
+fate-flac-32-wasted-bits: CMD = enc_dec_pcm flac wav s32le $(subst $(SAMPLES),$(TARGET_SAMPLES),$(REF)) -c flac -strict experimental
+
 fate-flac-rice-params: REF = $(SAMPLES)/audio-reference/chorusnoise_2ch_44kHz_s16.wav
 fate-flac-rice-params: CMD = enc_dec_pcm flac wav s16le $(subst $(SAMPLES),$(TARGET_SAMPLES),$(REF)) -c flac
 
-- 
2.30.2

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Re: [FFmpeg-devel] [PATCH v7 0/3] 32bps FLAC patches

[Martijn van Beurden]

Op di 11 okt. 2022 om 19:24 schreef Martijn van Beurden <mvanb1@gmail.com>:
>
> Recently libFLAC gained the ability (first released in FLAC 1.4.0)
> 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.
>
> 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
>
> Changes since v3:
> Moved put_golomb part to flacenc.c (as it is now quite specific to FLAC)
> and renamed put_sbits64 to put_sbits63 (and changed assert accordingly)
>
> Changes since v4:
> Fix check of decoded_buffer_33bps malloc. Fix reading of wasted bits
> for 33bps subframe. Add fate test
>
> Changes since v5:
> Slimmed down 32 bps fate flac test from 2 to 1 input file. Replaced
> -strict -2 with -strict experimental in fate and encoder warning.
>
> Changes since v6:
> Deduplicated function bodies with preprocessor macros in both
> encoder and decoder, declared variables in loop body where possible
> and changed ff_ctz to ff_ctzll where necessary
>
> Martijn van Beurden (3):
>   libavcodec/flacdec: Implement decoding of 32 bit-per-sample PCM
>   libavcodec/flacenc: Implement encoding of 32 bit-per-sample PCM
>   fate/flac: Add test of 32 bps encoding/decoding
>
>  libavcodec/flac.c       |   4 +-
>  libavcodec/flacdec.c    | 218 ++++++++++++++++--
>  libavcodec/flacenc.c    | 480 +++++++++++++++++++++++++++++++---------
>  libavcodec/get_bits.h   |  12 +
>  libavcodec/mathops.h    |   9 +
>  libavcodec/put_bits.h   |   7 +
>  libavcodec/put_golomb.h |  14 --
>  tests/fate/flac.mak     |   4 +
>  8 files changed, 602 insertions(+), 146 deletions(-)
>

Hi all,

With this email, I would like to renew the attention of the mailing
list for these patches.

Kind regards, Martijn van Beurden
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Re: [FFmpeg-devel] [PATCH v7 0/3] 32bps FLAC patches

[Martijn van Beurden]

Op vr 28 okt. 2022 om 22:13 schreef Martijn van Beurden <mvanb1@gmail.com>:
>
> Op di 11 okt. 2022 om 19:24 schreef Martijn van Beurden <mvanb1@gmail.com>:
> >
> > Recently libFLAC gained the ability (first released in FLAC 1.4.0)
> > 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.
> >
> > 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
> >
> > Changes since v3:
> > Moved put_golomb part to flacenc.c (as it is now quite specific to FLAC)
> > and renamed put_sbits64 to put_sbits63 (and changed assert accordingly)
> >
> > Changes since v4:
> > Fix check of decoded_buffer_33bps malloc. Fix reading of wasted bits
> > for 33bps subframe. Add fate test
> >
> > Changes since v5:
> > Slimmed down 32 bps fate flac test from 2 to 1 input file. Replaced
> > -strict -2 with -strict experimental in fate and encoder warning.
> >
> > Changes since v6:
> > Deduplicated function bodies with preprocessor macros in both
> > encoder and decoder, declared variables in loop body where possible
> > and changed ff_ctz to ff_ctzll where necessary
> >
> > Martijn van Beurden (3):
> >   libavcodec/flacdec: Implement decoding of 32 bit-per-sample PCM
> >   libavcodec/flacenc: Implement encoding of 32 bit-per-sample PCM
> >   fate/flac: Add test of 32 bps encoding/decoding
> >
> >  libavcodec/flac.c       |   4 +-
> >  libavcodec/flacdec.c    | 218 ++++++++++++++++--
> >  libavcodec/flacenc.c    | 480 +++++++++++++++++++++++++++++++---------
> >  libavcodec/get_bits.h   |  12 +
> >  libavcodec/mathops.h    |   9 +
> >  libavcodec/put_bits.h   |   7 +
> >  libavcodec/put_golomb.h |  14 --
> >  tests/fate/flac.mak     |   4 +
> >  8 files changed, 602 insertions(+), 146 deletions(-)
> >
>
> Hi all,
>
> With this email, I would like to renew the attention of the mailing
> list for these patches.
>
> Kind regards, Martijn van Beurden

Ping
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Re: [FFmpeg-devel] [PATCH v7 0/3] 32bps FLAC patches

[Martijn van Beurden]

Op vr 11 nov. 2022 om 22:11 schreef Martijn van Beurden <mvanb1@gmail.com>:
>
> Op vr 28 okt. 2022 om 22:13 schreef Martijn van Beurden <mvanb1@gmail.com>:
> >
> > Op di 11 okt. 2022 om 19:24 schreef Martijn van Beurden <mvanb1@gmail.com>:
> > >
> > > Recently libFLAC gained the ability (first released in FLAC 1.4.0)
> > > 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.
> > >
> > > 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
> > >
> > > Changes since v3:
> > > Moved put_golomb part to flacenc.c (as it is now quite specific to FLAC)
> > > and renamed put_sbits64 to put_sbits63 (and changed assert accordingly)
> > >
> > > Changes since v4:
> > > Fix check of decoded_buffer_33bps malloc. Fix reading of wasted bits
> > > for 33bps subframe. Add fate test
> > >
> > > Changes since v5:
> > > Slimmed down 32 bps fate flac test from 2 to 1 input file. Replaced
> > > -strict -2 with -strict experimental in fate and encoder warning.
> > >
> > > Changes since v6:
> > > Deduplicated function bodies with preprocessor macros in both
> > > encoder and decoder, declared variables in loop body where possible
> > > and changed ff_ctz to ff_ctzll where necessary
> > >
> > > Martijn van Beurden (3):
> > >   libavcodec/flacdec: Implement decoding of 32 bit-per-sample PCM
> > >   libavcodec/flacenc: Implement encoding of 32 bit-per-sample PCM
> > >   fate/flac: Add test of 32 bps encoding/decoding
> > >
> > >  libavcodec/flac.c       |   4 +-
> > >  libavcodec/flacdec.c    | 218 ++++++++++++++++--
> > >  libavcodec/flacenc.c    | 480 +++++++++++++++++++++++++++++++---------
> > >  libavcodec/get_bits.h   |  12 +
> > >  libavcodec/mathops.h    |   9 +
> > >  libavcodec/put_bits.h   |   7 +
> > >  libavcodec/put_golomb.h |  14 --
> > >  tests/fate/flac.mak     |   4 +
> > >  8 files changed, 602 insertions(+), 146 deletions(-)
> > >
> >
> > Hi all,
> >
> > With this email, I would like to renew the attention of the mailing
> > list for these patches.
> >
> > Kind regards, Martijn van Beurden
>
> Ping

Ping
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Re: [FFmpeg-devel] [PATCH v7 0/3] 32bps FLAC patches

[Paul B Mahol]

On 10/11/22, Martijn van Beurden <mvanb1@gmail.com> wrote:
> Recently libFLAC gained the ability (first released in FLAC 1.4.0)
> 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.
>
> 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
>
> Changes since v3:
> Moved put_golomb part to flacenc.c (as it is now quite specific to FLAC)
> and renamed put_sbits64 to put_sbits63 (and changed assert accordingly)
>
> Changes since v4:
> Fix check of decoded_buffer_33bps malloc. Fix reading of wasted bits
> for 33bps subframe. Add fate test
>
> Changes since v5:
> Slimmed down 32 bps fate flac test from 2 to 1 input file. Replaced
> -strict -2 with -strict experimental in fate and encoder warning.
>
> Changes since v6:
> Deduplicated function bodies with preprocessor macros in both
> encoder and decoder, declared variables in loop body where possible
> and changed ff_ctz to ff_ctzll where necessary
>
> Martijn van Beurden (3):
>   libavcodec/flacdec: Implement decoding of 32 bit-per-sample PCM
>   libavcodec/flacenc: Implement encoding of 32 bit-per-sample PCM
>   fate/flac: Add test of 32 bps encoding/decoding
>
>  libavcodec/flac.c       |   4 +-
>  libavcodec/flacdec.c    | 218 ++++++++++++++++--
>  libavcodec/flacenc.c    | 480 +++++++++++++++++++++++++++++++---------
>  libavcodec/get_bits.h   |  12 +
>  libavcodec/mathops.h    |   9 +
>  libavcodec/put_bits.h   |   7 +
>  libavcodec/put_golomb.h |  14 --
>  tests/fate/flac.mak     |   4 +
>  8 files changed, 602 insertions(+), 146 deletions(-)
>
> --
> 2.30.2
>



If this is OK, will apply soon.
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Re: [FFmpeg-devel] [PATCH v7 0/3] 32bps FLAC patches

[Paul B Mahol]

On 11/25/22, Paul B Mahol <onemda@gmail.com> wrote:
> On 10/11/22, Martijn van Beurden <mvanb1@gmail.com> wrote:
>> Recently libFLAC gained the ability (first released in FLAC 1.4.0)
>> 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.
>>
>> 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
>>
>> Changes since v3:
>> Moved put_golomb part to flacenc.c (as it is now quite specific to FLAC)
>> and renamed put_sbits64 to put_sbits63 (and changed assert accordingly)
>>
>> Changes since v4:
>> Fix check of decoded_buffer_33bps malloc. Fix reading of wasted bits
>> for 33bps subframe. Add fate test
>>
>> Changes since v5:
>> Slimmed down 32 bps fate flac test from 2 to 1 input file. Replaced
>> -strict -2 with -strict experimental in fate and encoder warning.
>>
>> Changes since v6:
>> Deduplicated function bodies with preprocessor macros in both
>> encoder and decoder, declared variables in loop body where possible
>> and changed ff_ctz to ff_ctzll where necessary
>>
>> Martijn van Beurden (3):
>>   libavcodec/flacdec: Implement decoding of 32 bit-per-sample PCM
>>   libavcodec/flacenc: Implement encoding of 32 bit-per-sample PCM
>>   fate/flac: Add test of 32 bps encoding/decoding
>>
>>  libavcodec/flac.c       |   4 +-
>>  libavcodec/flacdec.c    | 218 ++++++++++++++++--
>>  libavcodec/flacenc.c    | 480 +++++++++++++++++++++++++++++++---------
>>  libavcodec/get_bits.h   |  12 +
>>  libavcodec/mathops.h    |   9 +
>>  libavcodec/put_bits.h   |   7 +
>>  libavcodec/put_golomb.h |  14 --
>>  tests/fate/flac.mak     |   4 +
>>  8 files changed, 602 insertions(+), 146 deletions(-)
>>
>> --
>> 2.30.2
>>
>
>
>
> If this is OK, will apply soon.


Last time to say no to this patchset.
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Re: [FFmpeg-devel] [PATCH v7 0/3] 32bps FLAC patches

[Martijn van Beurden]

Op do 1 dec. 2022 om 13:31 schreef Paul B Mahol <onemda@gmail.com>:
>
> On 11/25/22, Paul B Mahol <onemda@gmail.com> wrote:
> > On 10/11/22, Martijn van Beurden <mvanb1@gmail.com> wrote:
> >> Recently libFLAC gained the ability (first released in FLAC 1.4.0)
> >> 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.
> >>
> >> 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
> >>
> >> Changes since v3:
> >> Moved put_golomb part to flacenc.c (as it is now quite specific to FLAC)
> >> and renamed put_sbits64 to put_sbits63 (and changed assert accordingly)
> >>
> >> Changes since v4:
> >> Fix check of decoded_buffer_33bps malloc. Fix reading of wasted bits
> >> for 33bps subframe. Add fate test
> >>
> >> Changes since v5:
> >> Slimmed down 32 bps fate flac test from 2 to 1 input file. Replaced
> >> -strict -2 with -strict experimental in fate and encoder warning.
> >>
> >> Changes since v6:
> >> Deduplicated function bodies with preprocessor macros in both
> >> encoder and decoder, declared variables in loop body where possible
> >> and changed ff_ctz to ff_ctzll where necessary
> >>
> >> Martijn van Beurden (3):
> >>   libavcodec/flacdec: Implement decoding of 32 bit-per-sample PCM
> >>   libavcodec/flacenc: Implement encoding of 32 bit-per-sample PCM
> >>   fate/flac: Add test of 32 bps encoding/decoding
> >>
> >>  libavcodec/flac.c       |   4 +-
> >>  libavcodec/flacdec.c    | 218 ++++++++++++++++--
> >>  libavcodec/flacenc.c    | 480 +++++++++++++++++++++++++++++++---------
> >>  libavcodec/get_bits.h   |  12 +
> >>  libavcodec/mathops.h    |   9 +
> >>  libavcodec/put_bits.h   |   7 +
> >>  libavcodec/put_golomb.h |  14 --
> >>  tests/fate/flac.mak     |   4 +
> >>  8 files changed, 602 insertions(+), 146 deletions(-)
> >>
> >> --
> >> 2.30.2
> >>
> >
> >
> >
> > If this is OK, will apply soon.
>
>
> Last time to say no to this patchset.

I haven't had any comments the last few weeks, it still applies
against current git and passed FATE. If you could apply it I would be
much obliged.

Kind regards,

Martijn van Beurden
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Re: [FFmpeg-devel] [PATCH v7 0/3] 32bps FLAC patches

[Paul B Mahol]

On 12/11/22, Martijn van Beurden <mvanb1@gmail.com> wrote:
> Op do 1 dec. 2022 om 13:31 schreef Paul B Mahol <onemda@gmail.com>:
>>
>> On 11/25/22, Paul B Mahol <onemda@gmail.com> wrote:
>> > On 10/11/22, Martijn van Beurden <mvanb1@gmail.com> wrote:
>> >> Recently libFLAC gained the ability (first released in FLAC 1.4.0)
>> >> 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.
>> >>
>> >> 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
>> >>
>> >> Changes since v3:
>> >> Moved put_golomb part to flacenc.c (as it is now quite specific to
>> >> FLAC)
>> >> and renamed put_sbits64 to put_sbits63 (and changed assert
>> >> accordingly)
>> >>
>> >> Changes since v4:
>> >> Fix check of decoded_buffer_33bps malloc. Fix reading of wasted bits
>> >> for 33bps subframe. Add fate test
>> >>
>> >> Changes since v5:
>> >> Slimmed down 32 bps fate flac test from 2 to 1 input file. Replaced
>> >> -strict -2 with -strict experimental in fate and encoder warning.
>> >>
>> >> Changes since v6:
>> >> Deduplicated function bodies with preprocessor macros in both
>> >> encoder and decoder, declared variables in loop body where possible
>> >> and changed ff_ctz to ff_ctzll where necessary
>> >>
>> >> Martijn van Beurden (3):
>> >>   libavcodec/flacdec: Implement decoding of 32 bit-per-sample PCM
>> >>   libavcodec/flacenc: Implement encoding of 32 bit-per-sample PCM
>> >>   fate/flac: Add test of 32 bps encoding/decoding
>> >>
>> >>  libavcodec/flac.c       |   4 +-
>> >>  libavcodec/flacdec.c    | 218 ++++++++++++++++--
>> >>  libavcodec/flacenc.c    | 480
>> >> +++++++++++++++++++++++++++++++---------
>> >>  libavcodec/get_bits.h   |  12 +
>> >>  libavcodec/mathops.h    |   9 +
>> >>  libavcodec/put_bits.h   |   7 +
>> >>  libavcodec/put_golomb.h |  14 --
>> >>  tests/fate/flac.mak     |   4 +
>> >>  8 files changed, 602 insertions(+), 146 deletions(-)
>> >>
>> >> --
>> >> 2.30.2
>> >>
>> >
>> >
>> >
>> > If this is OK, will apply soon.
>>
>>
>> Last time to say no to this patchset.
>
> I haven't had any comments the last few weeks, it still applies
> against current git and passed FATE. If you could apply it I would be
> much obliged.
>
> Kind regards,
>
> Martijn van Beurden
>

Hi, if this still pastes FATE and compiles and applies I gonna apply
it in next 48h.
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Re: [FFmpeg-devel] [PATCH v7 0/3] 32bps FLAC patches

[Paul B Mahol]

On 12/19/22, Paul B Mahol <onemda@gmail.com> wrote:
> On 12/11/22, Martijn van Beurden <mvanb1@gmail.com> wrote:
>> Op do 1 dec. 2022 om 13:31 schreef Paul B Mahol <onemda@gmail.com>:
>>>
>>> On 11/25/22, Paul B Mahol <onemda@gmail.com> wrote:
>>> > On 10/11/22, Martijn van Beurden <mvanb1@gmail.com> wrote:
>>> >> Recently libFLAC gained the ability (first released in FLAC 1.4.0)
>>> >> 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.
>>> >>
>>> >> 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
>>> >>
>>> >> Changes since v3:
>>> >> Moved put_golomb part to flacenc.c (as it is now quite specific to
>>> >> FLAC)
>>> >> and renamed put_sbits64 to put_sbits63 (and changed assert
>>> >> accordingly)
>>> >>
>>> >> Changes since v4:
>>> >> Fix check of decoded_buffer_33bps malloc. Fix reading of wasted bits
>>> >> for 33bps subframe. Add fate test
>>> >>
>>> >> Changes since v5:
>>> >> Slimmed down 32 bps fate flac test from 2 to 1 input file. Replaced
>>> >> -strict -2 with -strict experimental in fate and encoder warning.
>>> >>
>>> >> Changes since v6:
>>> >> Deduplicated function bodies with preprocessor macros in both
>>> >> encoder and decoder, declared variables in loop body where possible
>>> >> and changed ff_ctz to ff_ctzll where necessary
>>> >>
>>> >> Martijn van Beurden (3):
>>> >>   libavcodec/flacdec: Implement decoding of 32 bit-per-sample PCM
>>> >>   libavcodec/flacenc: Implement encoding of 32 bit-per-sample PCM
>>> >>   fate/flac: Add test of 32 bps encoding/decoding
>>> >>
>>> >>  libavcodec/flac.c       |   4 +-
>>> >>  libavcodec/flacdec.c    | 218 ++++++++++++++++--
>>> >>  libavcodec/flacenc.c    | 480
>>> >> +++++++++++++++++++++++++++++++---------
>>> >>  libavcodec/get_bits.h   |  12 +
>>> >>  libavcodec/mathops.h    |   9 +
>>> >>  libavcodec/put_bits.h   |   7 +
>>> >>  libavcodec/put_golomb.h |  14 --
>>> >>  tests/fate/flac.mak     |   4 +
>>> >>  8 files changed, 602 insertions(+), 146 deletions(-)
>>> >>
>>> >> --
>>> >> 2.30.2
>>> >>
>>> >
>>> >
>>> >
>>> > If this is OK, will apply soon.
>>>
>>>
>>> Last time to say no to this patchset.
>>
>> I haven't had any comments the last few weeks, it still applies
>> against current git and passed FATE. If you could apply it I would be
>> much obliged.
>>
>> Kind regards,
>>
>> Martijn van Beurden
>>
>
> Hi, if this still pastes FATE and compiles and applies I gonna apply
> it in next 48h.
>

Pushed.

Next time do not put tabs in non Makefile parts in patch or I will
reject pushing it.
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Re: [FFmpeg-devel] [PATCH v7 0/3] 32bps FLAC patches

[Martijn van Beurden]

Op ma 26 dec. 2022 21:14 schreef Paul B Mahol <onemda@gmail.com>:

> Pushed.
>
> Next time do not put tabs in non Makefile parts in patch or I will
> reject pushing it.
>

Thanks. Sorry about the tabs, I forgot to check, I will try to be more
careful next time.

>
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