38 #define FLAC_SUBFRAME_CONSTANT 0
39 #define FLAC_SUBFRAME_VERBATIM 1
40 #define FLAC_SUBFRAME_FIXED 8
41 #define FLAC_SUBFRAME_LPC 32
43 #define MAX_FIXED_ORDER 4
44 #define MAX_PARTITION_ORDER 8
45 #define MAX_PARTITIONS (1 << MAX_PARTITION_ORDER)
46 #define MAX_LPC_PRECISION 15
47 #define MIN_LPC_SHIFT 0
48 #define MAX_LPC_SHIFT 15
154 memcpy(&header[18], s->
md5sum, 16);
170 target = (samplerate * block_time_ms) / 1000;
171 for (i = 0; i < 16; i++) {
196 av_log(avctx,
AV_LOG_DEBUG,
" lpc type: Levinson-Durbin recursion with Welch window\n");
264 channels, FLAC_MAX_CHANNELS);
272 for (i = 4; i < 12; i++) {
282 if (freq % 1000 == 0 && freq < 255000) {
285 }
else if (freq % 10 == 0 && freq < 655350) {
288 }
else if (freq < 65535) {
311 s->
options.
block_time_ms = ((
int[]){ 27, 27, 27,105,105,105,105,105,105,105,105,105,105})[level];
318 FF_LPC_TYPE_LEVINSON})[level];
321 s->
options.
min_prediction_order = ((
int[]){ 2, 0, 0, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1})[level];
323 s->
options.
max_prediction_order = ((
int[]){ 3, 4, 4, 6, 8, 8, 8, 8, 12, 12, 12, 32, 32})[level];
330 ORDER_METHOD_SEARCH})[level];
338 s->
options.
min_partition_order = ((
int[]){ 2, 2, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0})[level];
340 s->
options.
max_partition_order = ((
int[]){ 2, 2, 3, 3, 3, 8, 8, 8, 8, 8, 8, 8, 8})[level];
342 #if FF_API_PRIVATE_OPT
348 "invalid min prediction order %d, clamped to %d\n",
364 "invalid max prediction order %d, clamped to %d\n",
384 "invalid min prediction order %d, clamped to %d\n",
390 "invalid max prediction order %d, clamped to %d\n",
448 "output stream will have incorrect "
449 "channel layout.\n");
452 "will use Flac channel layout for "
453 "%d channels.\n", channels);
477 for (i = 0; i < 16; i++) {
496 for (ch = 0; ch < s->
channels; ch++) {
522 #define COPY_SAMPLES(bits) do { \
523 const int ## bits ## _t *samples0 = samples; \
525 for (i = 0, j = 0; i < frame->blocksize; i++) \
526 for (ch = 0; ch < s->channels; ch++, j++) \
527 frame->subframes[ch].samples[i] = samples0[j] >> shift; \
542 for (i = 0; i <
n; i++) {
545 count += (v >> k) + 1 + k;
554 int p, porder, psize;
571 count += pred_order * sub->
obits;
588 for (p = 0; p < 1 << porder; p++) {
601 #define rice_encode_count(sum, n, k) (((n)*((k)+1))+((sum-(n>>1))>>(k)))
613 sum2 = sum - (n >> 1);
614 k =
av_log2(av_clipl_int32(sum2 / n));
615 return FFMIN(k, max_param);
621 int64_t bestbits = INT64_MAX;
624 for (k = 0; k <= max_param; k++) {
625 int64_t
bits = sums[k][i];
626 if (bits < bestbits) {
637 int n,
int pred_order,
int max_param,
int exact)
643 part = (1 << porder);
646 cnt = (n >> porder) - pred_order;
647 for (i = 0; i < part; i++) {
650 all_bits += sums[k][i];
670 const uint32_t *res, *res_end;
675 for (k = 0; k <= kmax; k++) {
676 res = &data[pred_order];
677 res_end = &data[n >> pmax];
678 for (i = 0; i < parts; i++) {
680 uint64_t sum = (1LL + k) * (res_end - res);
681 while (res < res_end)
682 sum += *(res++) >> k;
686 while (res < res_end)
690 res_end += n >> pmax;
698 int parts = (1 <<
level);
699 for (i = 0; i < parts; i++) {
700 for (k=0; k<=kmax; k++)
701 sums[k][i] = sums[k][2*i] + sums[k][2*i+1];
723 for (i = 0; i <
n; i++)
724 udata[i] = (2 * data[i]) ^ (data[i] >> 31);
726 calc_sum_top(pmax, exact ? kmax : 0, udata, n, pred_order, sums);
729 bits[pmin] = UINT32_MAX;
732 if (bits[i] < bits[opt_porder] || pmax == pmin) {
741 return bits[opt_porder];
776 for (i = 0; i < order; i++)
780 for (i = order; i <
n; i++)
782 }
else if (order == 1) {
783 for (i = order; i <
n; i++)
784 res[i] = smp[i] - smp[i-1];
785 }
else if (order == 2) {
786 int a = smp[order-1] - smp[order-2];
787 for (i = order; i <
n; i += 2) {
788 int b = smp[i ] - smp[i-1];
790 a = smp[i+1] - smp[i ];
793 }
else if (order == 3) {
794 int a = smp[order-1] - smp[order-2];
795 int c = smp[order-1] - 2*smp[order-2] + smp[order-3];
796 for (i = order; i <
n; i += 2) {
797 int b = smp[i ] - smp[i-1];
800 a = smp[i+1] - smp[i ];
805 int a = smp[order-1] - smp[order-2];
806 int c = smp[order-1] - 2*smp[order-2] + smp[order-3];
807 int e = smp[order-1] - 3*smp[order-2] + 3*smp[order-3] - smp[order-4];
808 for (i = order; i <
n; i += 2) {
809 int b = smp[i ] - smp[i-1];
813 a = smp[i+1] - smp[i ];
825 int min_order, max_order, opt_order, omethod;
839 for (i = 1; i <
n; i++)
851 memcpy(res, smp, n *
sizeof(
int32_t));
867 bits[0] = UINT32_MAX;
868 for (i = min_order; i <= max_order; i++) {
871 if (bits[i] < bits[opt_order])
874 sub->
order = opt_order;
876 if (sub->
order != max_order) {
893 int levels = 1 << omethod;
896 int opt_index = levels-1;
897 opt_order = max_order-1;
898 bits[opt_index] = UINT32_MAX;
899 for (i = levels-1; i >= 0; i--) {
900 int last_order = order;
901 order = min_order + (((max_order-min_order+1) * (i+1)) / levels)-1;
902 order = av_clip(order, min_order - 1, max_order - 1);
903 if (order == last_order)
913 if (bits[i] < bits[opt_index]) {
923 bits[0] = UINT32_MAX;
924 for (i = min_order-1; i < max_order; i++) {
931 if (bits[i] < bits[opt_order])
939 opt_order = min_order - 1 + (max_order-min_order)/3;
940 memset(bits, -1,
sizeof(bits));
942 for (step = 16; step; step >>= 1) {
943 int last = opt_order;
944 for (i = last-step; i <= last+step; i += step) {
945 if (i < min_order-1 || i >= max_order || bits[i] < UINT32_MAX)
953 if (bits[i] < bits[opt_order])
962 int i, step, improved;
963 int64_t best_score = INT64_MAX;
968 for (i=0; i<opt_order; i++)
973 for (step = 0; step < allsteps; step++) {
979 for (i=0; i<opt_order; i++) {
980 int diff = ((tmp + 1) % 3) - 1;
981 lpc_try[i] = av_clip(coefs[opt_order - 1][i] + diff, -qmax, qmax);
994 if (score < best_score) {
996 memcpy(coefs[opt_order-1], lpc_try,
sizeof(*coefs));
1003 sub->
order = opt_order;
1006 for (i = 0; i < sub->
order; i++)
1064 for (ch = 0; ch < s->
channels; ch++)
1067 count += (8 - (count & 7)) & 7;
1071 if (count > INT_MAX)
1081 for (ch = 0; ch < s->
channels; ch++) {
1091 if (v && !(v & 1)) {
1119 sum[0] = sum[1] = sum[2] = sum[3] = 0;
1120 for (i = 2; i <
n; i++) {
1121 lt = left_ch[i] - 2*left_ch[i-1] + left_ch[i-2];
1122 rt = right_ch[i] - 2*right_ch[i-1] + right_ch[i-2];
1123 sum[2] +=
FFABS((lt + rt) >> 1);
1124 sum[3] +=
FFABS(lt - rt);
1125 sum[0] +=
FFABS(lt);
1126 sum[1] +=
FFABS(rt);
1129 for (i = 0; i < 4; i++) {
1135 score[0] = sum[0] + sum[1];
1136 score[1] = sum[0] + sum[3];
1137 score[2] = sum[1] + sum[3];
1138 score[3] = sum[2] + sum[3];
1142 for (i = 1; i < 4; i++)
1143 if (score[i] < score[best])
1180 for (i = 0; i <
n; i++) {
1182 left[i] = (tmp + right[i]) >> 1;
1183 right[i] = tmp - right[i];
1187 for (i = 0; i <
n; i++)
1188 right[i] = left[i] - right[i];
1191 for (i = 0; i <
n; i++)
1192 left[i] -= right[i];
1227 else if (frame->
bs_code[0] == 7)
1246 for (ch = 0; ch < s->
channels; ch++) {
1248 int i, p, porder, psize;
1264 while (res < frame_end)
1268 for (i = 0; i < sub->
order; i++)
1276 for (i = 0; i < sub->
order; i++)
1290 for (p = 0; p < 1 << porder; p++) {
1293 while (res < part_end)
1295 part_end =
FFMIN(frame_end, part_end + psize);
1336 buf = (
const uint8_t *)samples;
1339 (
const uint16_t *) samples, buf_size / 2);
1344 const int32_t *samples0 = samples;
1363 int frame_bytes, out_bytes, ret;
1373 #if FF_API_SIDEDATA_ONLY_PKT
1388 *got_packet_ptr = 1;
1417 if (frame_bytes < 0) {
1436 if (out_bytes < s->min_framesize)
1441 avpkt->
size = out_bytes;
1445 *got_packet_ptr = 1;
1463 #define FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
1466 {
"lpc_type",
"LPC algorithm", offsetof(
FlacEncodeContext, options.lpc_type),
AV_OPT_TYPE_INT, {.i64 =
FF_LPC_TYPE_DEFAULT },
FF_LPC_TYPE_DEFAULT,
FF_LPC_TYPE_NB-1,
FLAGS,
"lpc_type" },
1471 {
"lpc_passes",
"Number of passes to use for Cholesky factorization during LPC analysis", offsetof(
FlacEncodeContext, options.lpc_passes),
AV_OPT_TYPE_INT, {.i64 = 2 }, 1, INT_MAX,
FLAGS },
1474 {
"prediction_order_method",
"Search method for selecting prediction order", offsetof(
FlacEncodeContext, options.prediction_order_method),
AV_OPT_TYPE_INT, {.i64 = -1 }, -1,
ORDER_METHOD_LOG,
FLAGS,
"predm" },
1481 {
"ch_mode",
"Stereo decorrelation mode", offsetof(
FlacEncodeContext, options.ch_mode),
AV_OPT_TYPE_INT, { .i64 = -1 }, -1,
FLAC_CHMODE_MID_SIDE,
FLAGS,
"ch_mode" },
uint32_t rc_udata[FLAC_MAX_BLOCKSIZE]
#define rice_encode_count(sum, n, k)
const char const char void * val
#define ORDER_METHOD_SEARCH
static int shift(int a, int b)
This structure describes decoded (raw) audio or video data.
#define ORDER_METHOD_8LEVEL
ptrdiff_t const GLvoid * data
static void put_sbits(PutBitContext *pb, int n, int32_t value)
static void put_bits(Jpeg2000EncoderContext *s, int val, int n)
put n times val bit
#define AV_LOG_WARNING
Something somehow does not look correct.
#define LIBAVUTIL_VERSION_INT
av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt, int channels, int bps)
void(* bswap16_buf)(uint16_t *dst, const uint16_t *src, int len)
static av_cold int init(AVCodecContext *avctx)
int ff_flac_get_max_frame_size(int blocksize, int ch, int bps)
Calculate an estimate for the maximum frame size based on verbatim mode.
#define MAX_PARTITION_ORDER
#define PUT_UTF8(val, tmp, PUT_BYTE)
Convert a 32-bit Unicode character to its UTF-8 encoded form (up to 4 bytes long).
#define FLAC_MAX_BLOCKSIZE
#define AV_CH_LAYOUT_STEREO
int bits_per_raw_sample
Bits per sample/pixel of internal libavcodec pixel/sample format.
static int flac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
#define AV_CH_LAYOUT_5POINT0
void(* lpc32_encode)(int32_t *res, const int32_t *smp, int len, int order, const int32_t coefs[32], int shift)
void av_md5_update(AVMD5 *ctx, const uint8_t *src, int len)
Update hash value.
static int select_blocksize(int samplerate, int block_time_ms)
Set blocksize based on samplerate.
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
#define AV_CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
#define av_assert0(cond)
assert() equivalent, that is always enabled.
attribute_deprecated int side_data_only_packets
Encoding only and set by default.
struct AVMD5 * av_md5_alloc(void)
Allocate an AVMD5 context.
enum AVSampleFormat sample_fmt
audio sample format
do not use LPC prediction or use all zero coefficients
int32_t coefs[MAX_LPC_ORDER]
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Public header for CRC hash function implementation.
static uint64_t find_subframe_rice_params(FlacEncodeContext *s, FlacSubframe *sub, int pred_order)
int params[MAX_PARTITIONS]
static const uint8_t header[24]
#define FLAC_MIN_BLOCKSIZE
static void write_subframes(FlacEncodeContext *s)
#define AV_CH_LAYOUT_5POINT1
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
#define ORDER_METHOD_4LEVEL
unsigned int md5_buffer_size
FLAC (Free Lossless Audio Codec) decoder/demuxer common functions.
int exact_rice_parameters
uint64_t rc_sums[32][MAX_PARTITIONS]
void(* lpc16_encode)(int32_t *res, const int32_t *smp, int len, int order, const int32_t coefs[32], int shift)
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
#define FLAC_SUBFRAME_LPC
enum CodingMode coding_mode
simple assert() macros that are a bit more flexible than ISO C assert().
#define AV_CH_LAYOUT_QUAD
const char * name
Name of the codec implementation.
#define COPY_SAMPLES(bits)
#define FLAC_SUBFRAME_VERBATIM
int32_t samples[FLAC_MAX_BLOCKSIZE]
static void remove_wasted_bits(FlacEncodeContext *s)
#define FLAC_SUBFRAME_CONSTANT
uint64_t channel_layout
Audio channel layout.
static int put_bits_count(PutBitContext *s)
#define ORDER_METHOD_2LEVEL
static uint64_t calc_optimal_rice_params(RiceContext *rc, int porder, uint64_t sums[32][MAX_PARTITIONS], int n, int pred_order, int max_param, int exact)
static void frame_end(MpegEncContext *s)
void av_fast_malloc(void *ptr, unsigned int *size, size_t min_size)
Allocate a buffer, reusing the given one if large enough.
#define FLAC_SUBFRAME_FIXED
static int encode_residual_ch(FlacEncodeContext *s, int ch)
av_cold void ff_lpc_end(LPCContext *s)
Uninitialize LPCContext.
#define av_assert1(cond)
assert() equivalent, that does not lie in speed critical code.
static uint64_t rice_count_exact(const int32_t *res, int n, int k)
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
static int encode_frame(FlacEncodeContext *s)
uint32_t av_crc(const AVCRC *ctx, uint32_t crc, const uint8_t *buffer, size_t length)
Calculate the CRC of a block.
#define FLAC_STREAMINFO_SIZE
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
#define AV_CH_FRONT_CENTER
int prediction_order_method
#define AV_CH_LAYOUT_5POINT1_BACK
static int get_max_p_order(int max_porder, int n, int order)
static int write_frame(FlacEncodeContext *s, AVPacket *avpkt)
static int find_optimal_param_exact(uint64_t sums[32][MAX_PARTITIONS], int i, int max_param)
attribute_deprecated int max_prediction_order
static void set_sr_golomb_flac(PutBitContext *pb, int i, int k, int limit, int esc_len)
write signed golomb rice code (flac).
static const AVOption options[]
static void channel_decorrelation(FlacEncodeContext *s)
Perform stereo channel decorrelation.
int frame_size
Number of samples per channel in an audio frame.
The AV_PKT_DATA_NEW_EXTRADATA is used to notify the codec or the format that the extradata buffer was...
const int ff_flac_sample_rate_table[16]
Libavcodec external API header.
static void calc_sum_next(int level, uint64_t sums[32][MAX_PARTITIONS], int kmax)
AVSampleFormat
Audio sample formats.
int sample_rate
samples per second
static void write_frame_header(FlacEncodeContext *s)
static void calc_sum_top(int pmax, int kmax, const uint32_t *data, int n, int pred_order, uint64_t sums[32][MAX_PARTITIONS])
main external API structure.
static int count_frame_header(FlacEncodeContext *s)
Levinson-Durbin recursion.
void av_md5_init(AVMD5 *ctx)
Initialize MD5 hashing.
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
Describe the class of an AVClass context structure.
use the codec default LPC type
#define AV_CH_LAYOUT_5POINT0_BACK
static uint64_t calc_rice_params(RiceContext *rc, uint32_t udata[FLAC_MAX_BLOCKSIZE], uint64_t sums[32][MAX_PARTITIONS], int pmin, int pmax, const int32_t *data, int n, int pred_order, int exact)
static void encode_residual_fixed(int32_t *res, const int32_t *smp, int n, int order)
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
void av_md5_final(AVMD5 *ctx, uint8_t *dst)
Finish hashing and output digest value.
int max_encoded_framesize
int ff_lpc_calc_coefs(LPCContext *s, const int32_t *samples, int blocksize, int min_order, int max_order, int precision, int32_t coefs[][MAX_LPC_ORDER], int *shift, enum FFLPCType lpc_type, int lpc_passes, int omethod, int min_shift, int max_shift, int zero_shift)
Calculate LPC coefficients for multiple orders.
static void write_utf8(PutBitContext *pb, uint32_t val)
av_cold int ff_lpc_init(LPCContext *s, int blocksize, int max_order, enum FFLPCType lpc_type)
Initialize LPCContext.
#define MAX_LPC_PRECISION
static void copy_samples(FlacEncodeContext *s, const void *samples)
Copy channel-interleaved input samples into separate subframes.
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
static void write_frame_footer(FlacEncodeContext *s)
const AVCRC * av_crc_get_table(AVCRCId crc_id)
Get an initialized standard CRC table.
FlacSubframe subframes[FLAC_MAX_CHANNELS]
CompressionOptions options
FFLPCType
LPC analysis type.
#define FF_DISABLE_DEPRECATION_WARNINGS
static int estimate_stereo_mode(const int32_t *left_ch, const int32_t *right_ch, int n, int max_rice_param)
common internal api header.
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
int32_t residual[FLAC_MAX_BLOCKSIZE+11]
const int32_t ff_flac_blocksize_table[16]
#define AV_CODEC_CAP_LOSSLESS
Codec is lossless.
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
static av_cold int flac_encode_close(AVCodecContext *avctx)
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(constuint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(constint16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(constint32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(constint64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(constfloat *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(constdouble *) pi *(INT64_C(1)<< 63)))#defineFMT_PAIR_FUNC(out, in) staticconv_func_type *constfmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};staticvoidcpy1(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, len);}staticvoidcpy2(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 2 *len);}staticvoidcpy4(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 4 *len);}staticvoidcpy8(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, constint *ch_map, intflags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) returnNULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) returnNULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case1:ctx->simd_f=cpy1;break;case2:ctx->simd_f=cpy2;break;case4:ctx->simd_f=cpy4;break;case8:ctx->simd_f=cpy8;break;}}if(HAVE_YASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);returnctx;}voidswri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}intswri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, intlen){intch;intoff=0;constintos=(out->planar?1:out->ch_count)*out->bps;unsignedmisaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){intplanes=in->planar?in->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){intplanes=out->planar?out->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){intplanes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
static av_cold void dprint_compression_options(FlacEncodeContext *s)
av_cold void ff_bswapdsp_init(BswapDSPContext *c)
static int find_optimal_param(uint64_t sum, int n, int max_param)
Solve for d/dk(rice_encode_count) = n-((sum-(n>>1))>>(k+1)) = 0.
static av_always_inline int diff(const uint32_t a, const uint32_t b)
#define FF_ENABLE_DEPRECATION_WARNINGS
attribute_deprecated int min_prediction_order
int channels
number of audio channels
static uint64_t subframe_count_exact(FlacEncodeContext *s, FlacSubframe *sub, int pred_order)
static const AVClass flac_encoder_class
static enum AVSampleFormat sample_fmts[]
Public header for MD5 hash function implementation.
static void init_frame(FlacEncodeContext *s, int nb_samples)
static av_always_inline int64_t ff_samples_to_time_base(AVCodecContext *avctx, int64_t samples)
Rescale from sample rate to AVCodecContext.time_base.
static int update_md5_sum(FlacEncodeContext *s, const void *samples)
uint8_t * av_packet_new_side_data(AVPacket *pkt, enum AVPacketSideDataType type, int size)
Allocate new information of a packet.
This structure stores compressed data.
int nb_samples
number of audio samples (per channel) described by this frame
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
static av_cold int flac_encode_init(AVCodecContext *avctx)
static void write_streaminfo(FlacEncodeContext *s, uint8_t *header)
Write streaminfo metadata block to byte array.
#define FLAC_MAX_CHANNELS