falkTX
/
FFmpeg
mirror of https://github.com/falkTX/FFmpeg.git
1
0
Fork 0
Code Issues 0 Releases 338 Wiki Activity
Browse Source

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>
tags/n1.1
Michael Niedermayer 13 years ago
parent
7d26be63c2 8a58894fc6
commit
6493842900
5 changed files with 277 additions and 161 deletions
Split View
Diff Options
Show Stats Download Patch File Download Diff File
  1. +7
    -2
      libavcodec/flacdsp.c
  2. +2
    -0
      libavcodec/flacdsp.h
  3. +141
    -0
      libavcodec/flacdsp_lpc_template.c
  4. +108
    -147
      libavcodec/flacenc.c
  5. +19
    -12
      tests/fate/flac.mak

+ 7
- 2
libavcodec/flacdsp.c View File

@@ -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:


+ 2
- 0
libavcodec/flacdsp.h View File

@@ -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);


+ 141
- 0
libavcodec/flacdsp_lpc_template.c View File

@@ -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
}

+ 108
- 147
libavcodec/flacenc.c View File

@@ -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)
return -1;
switch (avctx->sample_fmt) {
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;
frame->subframes[ch].obits = 16;
FlacSubframe *sub = &frame->subframes[ch];

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;

frame = &s->frame;
for (i = 0, j = 0; i < frame->blocksize; i++)
for (ch = 0; ch < s->channels; ch++, j++)
frame->subframes[ch].samples[i] = samples[j];
int shift = av_get_bytes_per_sample(s->avctx->sample_fmt) * 8 -
s->avctx->bits_per_raw_sample;

#define COPY_SAMPLES(bits) do { \
const int ## bits ## _t *samples0 = samples; \
frame = &s->frame; \
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)
frame->ch_mode = estimate_stereo_mode(left, right, n);
else
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);
} 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,
(const uint16_t *)samples, buf_size / 2);
buf = s->md5_buffer;
s->dsp.bswap16_buf((uint16_t *)s->md5_buffer,
(const uint16_t *)samples, buf_size / 2);
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,


+ 19
- 12
tests/fate/flac.mak View File

@@ -1,17 +1,24 @@
FATE_FLAC += fate-flac-chmode-indep \
fate-flac-chmode-left_side \
fate-flac-chmode-mid_side \
fate-flac-chmode-right_side \
fate-flac-fixed \
fate-flac-lpc-cholesky \
fate-flac-lpc-levinson \
FATE_FLAC += fate-flac-16-chmode-indep \
fate-flac-16-chmode-left_side \
fate-flac-16-chmode-mid_side \
fate-flac-16-chmode-right_side \
fate-flac-16-fixed \
fate-flac-16-lpc-cholesky \
fate-flac-16-lpc-levinson \
fate-flac-24-comp-8 \

fate-flac-chmode-%: OPTS = -ch_mode $(@:fate-flac-chmode-%=%)
fate-flac-fixed: OPTS = -lpc_type fixed
fate-flac-lpc-%: OPTS = -lpc_type $(@:fate-flac-lpc-%=%)
fate-flac-16-chmode-%: OPTS = -ch_mode $(@:fate-flac-16-chmode-%=%)
fate-flac-16-fixed: OPTS = -lpc_type fixed
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



Write Preview
Loading…
Cancel
Save