Merge remote-tracking branch 'qatar/master'

* qatar/master:
  FATE: add a 24-bit FLAC encoding test
  FATE: rename FLAC tests from flac-* to flac-16-*
  flacenc: use RICE2 entropy coding mode for 24-bit
  flacenc: add 24-bit encoding
  flacdsp: move lpc encoding from FLAC encoder to FLACDSPContext

Merged-by: Michael Niedermayer <michaelni@gmx.at>

libavcodec/flacdsp.c

@@ -26,6 +26,7 @@
 #define SAMPLE_SIZE 16
 #define PLANAR 0
 #include "flacdsp_template.c"
+#include "flacdsp_lpc_template.c"
 
 #undef  PLANAR
 #define PLANAR 1
@@ -36,6 +37,7 @@
 #define SAMPLE_SIZE 32
 #define PLANAR 0
 #include "flacdsp_template.c"
+#include "flacdsp_lpc_template.c"
 
 #undef  PLANAR
 #define PLANAR 1
@@ -86,10 +88,13 @@ static void flac_lpc_32_c(int32_t *decoded, const int coeffs[32],
 av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt,
                              int bps)
 {
-    if (bps > 16)
+    if (bps > 16) {
         c->lpc            = flac_lpc_32_c;
-    else
+        c->lpc_encode     = flac_lpc_encode_c_32;
+    } else {
         c->lpc            = flac_lpc_16_c;
+        c->lpc_encode     = flac_lpc_encode_c_16;
+    }
 
     switch (fmt) {
     case AV_SAMPLE_FMT_S32:

libavcodec/flacdsp.h

@@ -27,6 +27,8 @@ typedef struct FLACDSPContext {
                            int len, int shift);
     void (*lpc)(int32_t *samples, const int coeffs[32], int order,
                 int qlevel, int len);
+    void (*lpc_encode)(int32_t *res, const int32_t *smp, int len, int order,
+                       const int32_t *coefs, int shift);
 } FLACDSPContext;
 
 void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt, int bps);

libavcodec/flacdsp_lpc_template.c

@@ -0,0 +1,141 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <stdint.h>
+#include "libavutil/avutil.h"
+#include "mathops.h"
+
+#undef FUNC
+#undef sum_type
+#undef MUL
+#undef CLIP
+#undef FSUF
+
+#define FUNC(n) AV_JOIN(n ## _, SAMPLE_SIZE)
+
+#if SAMPLE_SIZE == 32
+#   define sum_type  int64_t
+#   define MUL(a, b) MUL64(a, b)
+#   define CLIP(x) av_clipl_int32(x)
+#else
+#   define sum_type  int32_t
+#   define MUL(a, b) ((a) * (b))
+#   define CLIP(x) (x)
+#endif
+
+#define LPC1(x) {           \
+    int c = coefs[(x)-1];   \
+    p0   += MUL(c, s);      \
+    s     = smp[i-(x)+1];   \
+    p1   += MUL(c, s);      \
+}
+
+static av_always_inline void FUNC(lpc_encode_unrolled)(int32_t *res,
+                                  const int32_t *smp, int len, int order,
+                                  const int32_t *coefs, int shift, int big)
+{
+    int i;
+    for (i = order; i < len; i += 2) {
+        int s  = smp[i-order];
+        sum_type p0 = 0, p1 = 0;
+        if (big) {
+            switch (order) {
+            case 32: LPC1(32)
+            case 31: LPC1(31)
+            case 30: LPC1(30)
+            case 29: LPC1(29)
+            case 28: LPC1(28)
+            case 27: LPC1(27)
+            case 26: LPC1(26)
+            case 25: LPC1(25)
+            case 24: LPC1(24)
+            case 23: LPC1(23)
+            case 22: LPC1(22)
+            case 21: LPC1(21)
+            case 20: LPC1(20)
+            case 19: LPC1(19)
+            case 18: LPC1(18)
+            case 17: LPC1(17)
+            case 16: LPC1(16)
+            case 15: LPC1(15)
+            case 14: LPC1(14)
+            case 13: LPC1(13)
+            case 12: LPC1(12)
+            case 11: LPC1(11)
+            case 10: LPC1(10)
+            case  9: LPC1( 9)
+                     LPC1( 8)
+                     LPC1( 7)
+                     LPC1( 6)
+                     LPC1( 5)
+                     LPC1( 4)
+                     LPC1( 3)
+                     LPC1( 2)
+                     LPC1( 1)
+            }
+        } else {
+            switch (order) {
+            case  8: LPC1( 8)
+            case  7: LPC1( 7)
+            case  6: LPC1( 6)
+            case  5: LPC1( 5)
+            case  4: LPC1( 4)
+            case  3: LPC1( 3)
+            case  2: LPC1( 2)
+            case  1: LPC1( 1)
+            }
+        }
+        res[i  ] = smp[i  ] - CLIP(p0 >> shift);
+        res[i+1] = smp[i+1] - CLIP(p1 >> shift);
+    }
+}
+
+static void FUNC(flac_lpc_encode_c)(int32_t *res, const int32_t *smp, int len,
+                                    int order, const int32_t *coefs, int shift)
+{
+    int i;
+    for (i = 0; i < order; i++)
+        res[i] = smp[i];
+#if CONFIG_SMALL
+    for (i = order; i < len; i += 2) {
+        int j;
+        int s  = smp[i];
+        sum_type p0 = 0, p1 = 0;
+        for (j = 0; j < order; j++) {
+            int c = coefs[j];
+            p1   += MUL(c, s);
+            s     = smp[i-j-1];
+            p0   += MUL(c, s);
+        }
+        res[i  ] = smp[i  ] - CLIP(p0 >> shift);
+        res[i+1] = smp[i+1] - CLIP(p1 >> shift);
+    }
+#else
+    switch (order) {
+    case  1: FUNC(lpc_encode_unrolled)(res, smp, len,     1, coefs, shift, 0); break;
+    case  2: FUNC(lpc_encode_unrolled)(res, smp, len,     2, coefs, shift, 0); break;
+    case  3: FUNC(lpc_encode_unrolled)(res, smp, len,     3, coefs, shift, 0); break;
+    case  4: FUNC(lpc_encode_unrolled)(res, smp, len,     4, coefs, shift, 0); break;
+    case  5: FUNC(lpc_encode_unrolled)(res, smp, len,     5, coefs, shift, 0); break;
+    case  6: FUNC(lpc_encode_unrolled)(res, smp, len,     6, coefs, shift, 0); break;
+    case  7: FUNC(lpc_encode_unrolled)(res, smp, len,     7, coefs, shift, 0); break;
+    case  8: FUNC(lpc_encode_unrolled)(res, smp, len,     8, coefs, shift, 0); break;
+    default: FUNC(lpc_encode_unrolled)(res, smp, len, order, coefs, shift, 1); break;
+    }
+#endif
+}

libavcodec/flacenc.c

@@ -32,6 +32,7 @@
 #include "lpc.h"
 #include "flac.h"
 #include "flacdata.h"
+#include "flacdsp.h"
 
 #define FLAC_SUBFRAME_CONSTANT  0
 #define FLAC_SUBFRAME_VERBATIM  1
@@ -43,7 +44,11 @@
 #define MAX_PARTITIONS     (1 << MAX_PARTITION_ORDER)
 #define MAX_LPC_PRECISION  15
 #define MAX_LPC_SHIFT      15
-#define MAX_RICE_PARAM     14
+
+enum CodingMode {
+    CODING_MODE_RICE  = 4,
+    CODING_MODE_RICE2 = 5,
+};
 
 typedef struct CompressionOptions {
     int compression_level;
@@ -60,6 +65,7 @@ typedef struct CompressionOptions {
 } CompressionOptions;
 
 typedef struct RiceContext {
+    enum CodingMode coding_mode;
     int porder;
     int params[MAX_PARTITIONS];
 } RiceContext;
@@ -92,6 +98,7 @@ typedef struct FlacEncodeContext {
     int channels;
     int samplerate;
     int sr_code[2];
+    int bps_code;
     int max_blocksize;
     int min_framesize;
     int max_framesize;
@@ -107,6 +114,7 @@ typedef struct FlacEncodeContext {
     uint8_t *md5_buffer;
     unsigned int md5_buffer_size;
     DSPContext dsp;
+    FLACDSPContext flac_dsp;
 } FlacEncodeContext;
 
 
@@ -127,7 +135,7 @@ static void write_streaminfo(FlacEncodeContext *s, uint8_t *header)
     put_bits(&pb, 24, s->max_framesize);
     put_bits(&pb, 20, s->samplerate);
     put_bits(&pb, 3, s->channels-1);
-    put_bits(&pb, 5, 15);       /* bits per sample - 1 */
+    put_bits(&pb,  5, s->avctx->bits_per_raw_sample - 1);
     /* write 36-bit sample count in 2 put_bits() calls */
     put_bits(&pb, 24, (s->sample_count & 0xFFFFFF000LL) >> 12);
     put_bits(&pb, 12,  s->sample_count & 0x000000FFFLL);
@@ -227,8 +235,18 @@ static av_cold int flac_encode_init(AVCodecContext *avctx)
 
     s->avctx = avctx;
 
-    if (avctx->sample_fmt != AV_SAMPLE_FMT_S16)
+    switch (avctx->sample_fmt) {
-        return -1;
+    case AV_SAMPLE_FMT_S16:
+        avctx->bits_per_raw_sample = 16;
+        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;
+        break;
+    }
 
     if (channels < 1 || channels > FLAC_MAX_CHANNELS)
         return -1;
@@ -358,7 +376,8 @@ static av_cold int flac_encode_init(AVCodecContext *avctx)
 
     /* set maximum encoded frame size in verbatim mode */
     s->max_framesize = ff_flac_get_max_frame_size(s->avctx->frame_size,
-                                                  s->channels, 16);
+                                                  s->channels,
+                                                  s->avctx->bits_per_raw_sample);
 
     /* initialize MD5 context */
     s->md5ctx = av_md5_alloc();
@@ -408,6 +427,8 @@ static av_cold int flac_encode_init(AVCodecContext *avctx)
                       s->options.max_prediction_order, FF_LPC_TYPE_LEVINSON);
 
     ff_dsputil_init(&s->dsp, avctx);
+    ff_flacdsp_init(&s->flac_dsp, avctx->sample_fmt,
+                    avctx->bits_per_raw_sample);
 
     dprint_compression_options(s);
 
@@ -442,8 +463,15 @@ static void init_frame(FlacEncodeContext *s, int nb_samples)
     }
 
     for (ch = 0; ch < s->channels; ch++) {
-        frame->subframes[ch].wasted = 0;
+        FlacSubframe *sub = &frame->subframes[ch];
-        frame->subframes[ch].obits = 16;
+
+        sub->wasted = 0;
+        sub->obits  = s->avctx->bits_per_raw_sample;
+
+        if (sub->obits > 16)
+            sub->rc.coding_mode = CODING_MODE_RICE2;
+        else
+            sub->rc.coding_mode = CODING_MODE_RICE;
     }
 
     frame->verbatim_only = 0;
@@ -453,15 +481,25 @@ static void init_frame(FlacEncodeContext *s, int nb_samples)
 /**
  * Copy channel-interleaved input samples into separate subframes.
  */
-static void copy_samples(FlacEncodeContext *s, const int16_t *samples)
+static void copy_samples(FlacEncodeContext *s, const void *samples)
 {
     int i, j, ch;
     FlacFrame *frame;
+    int shift = av_get_bytes_per_sample(s->avctx->sample_fmt) * 8 -
+                s->avctx->bits_per_raw_sample;
 
-    frame = &s->frame;
+#define COPY_SAMPLES(bits) do {                                     \
-    for (i = 0, j = 0; i < frame->blocksize; i++)
+    const int ## bits ## _t *samples0 = samples;                    \
-        for (ch = 0; ch < s->channels; ch++, j++)
+    frame = &s->frame;                                              \
-            frame->subframes[ch].samples[i] = samples[j];
+    for (i = 0, j = 0; i < frame->blocksize; i++)                   \
+        for (ch = 0; ch < s->channels; ch++, j++)                   \
+            frame->subframes[ch].samples[i] = samples0[j] >> shift; \
+} while (0)
+
+    if (s->avctx->sample_fmt == AV_SAMPLE_FMT_S16)
+        COPY_SAMPLES(16);
+    else
+        COPY_SAMPLES(32);
 }
 
 
@@ -515,7 +553,7 @@ static uint64_t subframe_count_exact(FlacEncodeContext *s, FlacSubframe *sub,
         part_end = psize;
         for (p = 0; p < 1 << porder; p++) {
             int k = sub->rc.params[p];
-            count += 4;
+            count += sub->rc.coding_mode;
             count += rice_count_exact(&sub->residual[i], part_end - i, k);
             i = part_end;
             part_end = FFMIN(s->frame.blocksize, part_end + psize);
@@ -531,7 +569,7 @@ static uint64_t subframe_count_exact(FlacEncodeContext *s, FlacSubframe *sub,
 /**
  * Solve for d/dk(rice_encode_count) = n-((sum-(n>>1))>>(k+1)) = 0.
  */
-static int find_optimal_param(uint64_t sum, int n)
+static int find_optimal_param(uint64_t sum, int n, int max_param)
 {
     int k;
     uint64_t sum2;
@@ -540,7 +578,7 @@ static int find_optimal_param(uint64_t sum, int n)
         return 0;
     sum2 = sum - (n >> 1);
     k    = av_log2(av_clipl_int32(sum2 / n));
-    return FFMIN(k, MAX_RICE_PARAM);
+    return FFMIN(k, max_param);
 }
 
 
@@ -548,15 +586,17 @@ static uint64_t calc_optimal_rice_params(RiceContext *rc, int porder,
                                          uint64_t *sums, int n, int pred_order)
 {
     int i;
-    int k, cnt, part;
+    int k, cnt, part, max_param;
     uint64_t all_bits;
 
+    max_param = (1 << rc->coding_mode) - 2;
+
     part     = (1 << porder);
     all_bits = 4 * part;
 
     cnt = (n >> porder) - pred_order;
     for (i = 0; i < part; i++) {
-        k = find_optimal_param(sums[i], cnt);
+        k = find_optimal_param(sums[i], cnt, max_param);
         rc->params[i] = k;
         all_bits += rice_encode_count(sums[i], cnt, k);
         cnt = n >> porder;
@@ -609,6 +649,8 @@ static uint64_t calc_rice_params(RiceContext *rc, int pmin, int pmax,
     av_assert1(pmax >= 0 && pmax <= MAX_PARTITION_ORDER);
     av_assert1(pmin <= pmax);
 
+    tmp_rc.coding_mode = rc->coding_mode;
+
     udata = av_malloc(n * sizeof(uint32_t));
     for (i = 0; i < n; i++)
         udata[i] = (2*data[i]) ^ (data[i]>>31);
@@ -647,7 +689,7 @@ static uint64_t find_subframe_rice_params(FlacEncodeContext *s,
     int pmax = get_max_p_order(s->options.max_partition_order,
                                s->frame.blocksize, pred_order);
 
-    uint64_t bits = 8 + pred_order * sub->obits + 2 + 4;
+    uint64_t bits = 8 + pred_order * sub->obits + 2 + sub->rc.coding_mode;
     if (sub->type == FLAC_SUBFRAME_LPC)
         bits += 4 + 5 + pred_order * s->options.lpc_coeff_precision;
     bits += calc_rice_params(&sub->rc, pmin, pmax, sub->residual,
@@ -707,110 +749,6 @@ static void encode_residual_fixed(int32_t *res, const int32_t *smp, int n,
 }
 
 
-#define LPC1(x) {\
-    int c = coefs[(x)-1];\
-    p0   += c * s;\
-    s     = smp[i-(x)+1];\
-    p1   += c * s;\
-}
-
-static av_always_inline void encode_residual_lpc_unrolled(int32_t *res,
-                                    const int32_t *smp, int n, int order,
-                                    const int32_t *coefs, int shift, int big)
-{
-    int i;
-    for (i = order; i < n; i += 2) {
-        int s  = smp[i-order];
-        int p0 = 0, p1 = 0;
-        if (big) {
-            switch (order) {
-            case 32: LPC1(32)
-            case 31: LPC1(31)
-            case 30: LPC1(30)
-            case 29: LPC1(29)
-            case 28: LPC1(28)
-            case 27: LPC1(27)
-            case 26: LPC1(26)
-            case 25: LPC1(25)
-            case 24: LPC1(24)
-            case 23: LPC1(23)
-            case 22: LPC1(22)
-            case 21: LPC1(21)
-            case 20: LPC1(20)
-            case 19: LPC1(19)
-            case 18: LPC1(18)
-            case 17: LPC1(17)
-            case 16: LPC1(16)
-            case 15: LPC1(15)
-            case 14: LPC1(14)
-            case 13: LPC1(13)
-            case 12: LPC1(12)
-            case 11: LPC1(11)
-            case 10: LPC1(10)
-            case  9: LPC1( 9)
-                     LPC1( 8)
-                     LPC1( 7)
-                     LPC1( 6)
-                     LPC1( 5)
-                     LPC1( 4)
-                     LPC1( 3)
-                     LPC1( 2)
-                     LPC1( 1)
-            }
-        } else {
-            switch (order) {
-            case  8: LPC1( 8)
-            case  7: LPC1( 7)
-            case  6: LPC1( 6)
-            case  5: LPC1( 5)
-            case  4: LPC1( 4)
-            case  3: LPC1( 3)
-            case  2: LPC1( 2)
-            case  1: LPC1( 1)
-            }
-        }
-        res[i  ] = smp[i  ] - (p0 >> shift);
-        res[i+1] = smp[i+1] - (p1 >> shift);
-    }
-}
-
-
-static void encode_residual_lpc(int32_t *res, const int32_t *smp, int n,
-                                int order, const int32_t *coefs, int shift)
-{
-    int i;
-    for (i = 0; i < order; i++)
-        res[i] = smp[i];
-#if CONFIG_SMALL
-    for (i = order; i < n; i += 2) {
-        int j;
-        int s  = smp[i];
-        int p0 = 0, p1 = 0;
-        for (j = 0; j < order; j++) {
-            int c = coefs[j];
-            p1   += c * s;
-            s     = smp[i-j-1];
-            p0   += c * s;
-        }
-        res[i  ] = smp[i  ] - (p0 >> shift);
-        res[i+1] = smp[i+1] - (p1 >> shift);
-    }
-#else
-    switch (order) {
-    case  1: encode_residual_lpc_unrolled(res, smp, n, 1, coefs, shift, 0); break;
-    case  2: encode_residual_lpc_unrolled(res, smp, n, 2, coefs, shift, 0); break;
-    case  3: encode_residual_lpc_unrolled(res, smp, n, 3, coefs, shift, 0); break;
-    case  4: encode_residual_lpc_unrolled(res, smp, n, 4, coefs, shift, 0); break;
-    case  5: encode_residual_lpc_unrolled(res, smp, n, 5, coefs, shift, 0); break;
-    case  6: encode_residual_lpc_unrolled(res, smp, n, 6, coefs, shift, 0); break;
-    case  7: encode_residual_lpc_unrolled(res, smp, n, 7, coefs, shift, 0); break;
-    case  8: encode_residual_lpc_unrolled(res, smp, n, 8, coefs, shift, 0); break;
-    default: encode_residual_lpc_unrolled(res, smp, n, order, coefs, shift, 1); break;
-    }
-#endif
-}
-
-
 static int encode_residual_ch(FlacEncodeContext *s, int ch)
 {
     int i, n;
@@ -892,7 +830,8 @@ static int encode_residual_ch(FlacEncodeContext *s, int ch)
             order = min_order + (((max_order-min_order+1) * (i+1)) / levels)-1;
             if (order < 0)
                 order = 0;
-            encode_residual_lpc(res, smp, n, order+1, coefs[order], shift[order]);
+            s->flac_dsp.lpc_encode(res, smp, n, order+1, coefs[order],
+                                   shift[order]);
             bits[i] = find_subframe_rice_params(s, sub, order+1);
             if (bits[i] < bits[opt_index]) {
                 opt_index = i;
@@ -906,7 +845,7 @@ 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++) {
-            encode_residual_lpc(res, smp, n, i+1, coefs[i], shift[i]);
+            s->flac_dsp.lpc_encode(res, smp, n, i+1, coefs[i], shift[i]);
             bits[i] = find_subframe_rice_params(s, sub, i+1);
             if (bits[i] < bits[opt_order])
                 opt_order = i;
@@ -924,7 +863,7 @@ 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;
-                encode_residual_lpc(res, smp, n, i+1, coefs[i], shift[i]);
+                s->flac_dsp.lpc_encode(res, smp, n, i+1, coefs[i], shift[i]);
                 bits[i] = find_subframe_rice_params(s, sub, i+1);
                 if (bits[i] < bits[opt_order])
                     opt_order = i;
@@ -939,7 +878,7 @@ static int encode_residual_ch(FlacEncodeContext *s, int ch)
     for (i = 0; i < sub->order; i++)
         sub->coefs[i] = coefs[sub->order-1][i];
 
-    encode_residual_lpc(res, smp, n, sub->order, sub->coefs, sub->shift);
+    s->flac_dsp.lpc_encode(res, smp, n, sub->order, sub->coefs, sub->shift);
 
     find_subframe_rice_params(s, sub, sub->order);
 
@@ -1025,12 +964,18 @@ static void remove_wasted_bits(FlacEncodeContext *s)
 
             sub->wasted = v;
             sub->obits -= 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;
         }
     }
 }
 
 
-static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n)
+static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n,
+                                int max_rice_param)
 {
     int i, best;
     int32_t lt, rt;
@@ -1050,7 +995,7 @@ static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n)
     }
     /* estimate bit counts */
     for (i = 0; i < 4; i++) {
-        k      = find_optimal_param(2 * sum[i], n);
+        k      = find_optimal_param(2 * sum[i], n, max_rice_param);
         sum[i] = rice_encode_count( 2 * sum[i], n, k);
     }
 
@@ -1089,9 +1034,10 @@ static void channel_decorrelation(FlacEncodeContext *s)
         return;
     }
 
-    if (s->options.ch_mode < 0)
+    if (s->options.ch_mode < 0) {
-        frame->ch_mode = estimate_stereo_mode(left, right, n);
+        int max_rice_param = (1 << frame->subframes[0].rc.coding_mode) - 2;
-    else
+        frame->ch_mode = estimate_stereo_mode(left, right, n, max_rice_param);
+    } else
         frame->ch_mode = s->options.ch_mode;
 
     /* perform decorrelation and adjust bits-per-sample */
@@ -1140,7 +1086,7 @@ static void write_frame_header(FlacEncodeContext *s)
     else
         put_bits(&s->pb, 4, frame->ch_mode + FLAC_MAX_CHANNELS - 1);
 
-    put_bits(&s->pb, 3, 4); /* bits-per-sample code */
+    put_bits(&s->pb, 3, s->bps_code);
     put_bits(&s->pb, 1, 0);
     write_utf8(&s->pb, s->frame_count);
 
@@ -1200,7 +1146,7 @@ static void write_subframes(FlacEncodeContext *s)
             }
 
             /* rice-encoded block */
-            put_bits(&s->pb, 2, 0);
+            put_bits(&s->pb, 2, sub->rc.coding_mode - 4);
 
             /* partition order */
             porder  = sub->rc.porder;
@@ -1211,7 +1157,7 @@ static void write_subframes(FlacEncodeContext *s)
             part_end  = &sub->residual[psize];
             for (p = 0; p < 1 << porder; p++) {
                 int k = sub->rc.params[p];
-                put_bits(&s->pb, 4, k);
+                put_bits(&s->pb, sub->rc.coding_mode, k);
                 while (res < part_end)
                     set_sr_golomb_flac(&s->pb, *res++, k, INT32_MAX, 0);
                 part_end = FFMIN(frame_end, part_end + psize);
@@ -1242,23 +1188,38 @@ static int write_frame(FlacEncodeContext *s, AVPacket *avpkt)
 }
 
 
-static int update_md5_sum(FlacEncodeContext *s, const int16_t *samples)
+static int update_md5_sum(FlacEncodeContext *s, const void *samples)
 {
     const uint8_t *buf;
-    int buf_size = s->frame.blocksize * s->channels * 2;
+    int buf_size = s->frame.blocksize * s->channels *
+                   ((s->avctx->bits_per_raw_sample + 7) / 8);
 
-    if (HAVE_BIGENDIAN) {
+    if (s->avctx->bits_per_raw_sample > 16 || HAVE_BIGENDIAN) {
         av_fast_malloc(&s->md5_buffer, &s->md5_buffer_size, buf_size);
         if (!s->md5_buffer)
             return AVERROR(ENOMEM);
     }
 
-    buf = (const uint8_t *)samples;
+    if (s->avctx->bits_per_raw_sample <= 16) {
+        buf = (const uint8_t *)samples;
 #if HAVE_BIGENDIAN
-    s->dsp.bswap16_buf((uint16_t *)s->md5_buffer,
+        s->dsp.bswap16_buf((uint16_t *)s->md5_buffer,
-                       (const uint16_t *)samples, buf_size / 2);
+                           (const uint16_t *)samples, buf_size / 2);
-    buf = s->md5_buffer;
+        buf = s->md5_buffer;
 #endif
+    } else {
+        int i;
+        const int32_t *samples0 = samples;
+        uint8_t *tmp            = s->md5_buffer;
+
+        for (i = 0; i < s->frame.blocksize * s->channels; i++) {
+            int32_t v = samples0[i] >> 8;
+            *tmp++    = (v      ) & 0xFF;
+            *tmp++    = (v >>  8) & 0xFF;
+            *tmp++    = (v >> 16) & 0xFF;
+        }
+        buf = s->md5_buffer;
+    }
     av_md5_update(s->md5ctx, buf, buf_size);
 
     return 0;
@@ -1269,7 +1230,6 @@ static int flac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                              const AVFrame *frame, int *got_packet_ptr)
 {
     FlacEncodeContext *s;
-    const int16_t *samples;
     int frame_bytes, out_bytes, ret;
 
     s = avctx->priv_data;
@@ -1281,17 +1241,17 @@ static int flac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
         write_streaminfo(s, avctx->extradata);
         return 0;
     }
-    samples = (const int16_t *)frame->data[0];
 
     /* change max_framesize for small final frame */
     if (frame->nb_samples < s->frame.blocksize) {
         s->max_framesize = ff_flac_get_max_frame_size(frame->nb_samples,
-                                                      s->channels, 16);
+                                                      s->channels,
+                                                      avctx->bits_per_raw_sample);
     }
 
     init_frame(s, frame->nb_samples);
 
-    copy_samples(s, samples);
+    copy_samples(s, frame->data[0]);
 
     channel_decorrelation(s);
 
@@ -1317,7 +1277,7 @@ static int flac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
 
     s->frame_count++;
     s->sample_count += frame->nb_samples;
-    if ((ret = update_md5_sum(s, samples)) < 0) {
+    if ((ret = update_md5_sum(s, frame->data[0])) < 0) {
         av_log(avctx, AV_LOG_ERROR, "Error updating MD5 checksum\n");
         return ret;
     }
@@ -1394,6 +1354,7 @@ AVCodec ff_flac_encoder = {
     .close          = flac_encode_close,
     .capabilities   = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_LOSSLESS,
     .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
+                                                     AV_SAMPLE_FMT_S32,
                                                      AV_SAMPLE_FMT_NONE },
     .long_name      = NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),
     .priv_class     = &flac_encoder_class,

tests/fate/flac.mak

@@ -1,17 +1,24 @@
-FATE_FLAC += fate-flac-chmode-indep                                     \
+FATE_FLAC += fate-flac-16-chmode-indep                                  \
-             fate-flac-chmode-left_side                                 \
+             fate-flac-16-chmode-left_side                              \
-             fate-flac-chmode-mid_side                                  \
+             fate-flac-16-chmode-mid_side                               \
-             fate-flac-chmode-right_side                                \
+             fate-flac-16-chmode-right_side                             \
-             fate-flac-fixed                                            \
+             fate-flac-16-fixed                                         \
-             fate-flac-lpc-cholesky                                     \
+             fate-flac-16-lpc-cholesky                                  \
-             fate-flac-lpc-levinson                                     \
+             fate-flac-16-lpc-levinson                                  \
+             fate-flac-24-comp-8                                        \
 
-fate-flac-chmode-%: OPTS = -ch_mode $(@:fate-flac-chmode-%=%)
+fate-flac-16-chmode-%: OPTS = -ch_mode $(@:fate-flac-16-chmode-%=%)
-fate-flac-fixed: OPTS = -lpc_type fixed
+fate-flac-16-fixed:    OPTS = -lpc_type fixed
-fate-flac-lpc-%: OPTS = -lpc_type $(@:fate-flac-lpc-%=%)
+fate-flac-16-lpc-%:    OPTS = -lpc_type $(@:fate-flac-16-lpc-%=%)
+
+fate-flac-16-%: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
+fate-flac-16-%: CMD = enc_dec_pcm flac wav s16le $(REF) -c flac $(OPTS)
+
+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 $(REF) -c flac $(OPTS)
 
-fate-flac-%: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
-fate-flac-%: CMD = enc_dec_pcm flac wav s16le $(REF) -c flac $(OPTS)
 fate-flac-%: CMP = oneoff
 fate-flac-%: FUZZ = 0