37 #define FLAC_SUBFRAME_CONSTANT 0
38 #define FLAC_SUBFRAME_VERBATIM 1
39 #define FLAC_SUBFRAME_FIXED 8
40 #define FLAC_SUBFRAME_LPC 32
42 #define MAX_FIXED_ORDER 4
43 #define MAX_PARTITION_ORDER 8
44 #define MAX_PARTITIONS (1 << MAX_PARTITION_ORDER)
45 #define MAX_LPC_PRECISION 15
46 #define MAX_LPC_SHIFT 15
152 memcpy(&header[18], s->
md5sum, 16);
168 target = (samplerate * block_time_ms) / 1000;
169 for (i = 0; i < 16; i++) {
194 av_log(avctx,
AV_LOG_DEBUG,
" lpc type: Levinson-Durbin recursion with Welch window\n");
262 channels, FLAC_MAX_CHANNELS);
270 for (i = 4; i < 12; i++) {
280 if (freq % 1000 == 0 && freq < 255000) {
283 }
else if (freq % 10 == 0 && freq < 655350) {
286 }
else if (freq < 65535) {
309 s->
options.
block_time_ms = ((
int[]){ 27, 27, 27,105,105,105,105,105,105,105,105,105,105})[level];
316 FF_LPC_TYPE_LEVINSON})[level];
318 s->
options.
min_prediction_order = ((
int[]){ 2, 0, 0, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1})[level];
319 s->
options.
max_prediction_order = ((
int[]){ 3, 4, 4, 6, 8, 8, 8, 8, 12, 12, 12, 32, 32})[level];
326 ORDER_METHOD_SEARCH})[level];
334 s->
options.
min_partition_order = ((
int[]){ 2, 2, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0})[level];
336 s->
options.
max_partition_order = ((
int[]){ 2, 2, 3, 3, 3, 8, 8, 8, 8, 8, 8, 8, 8})[level];
424 "output stream will have incorrect "
425 "channel layout.\n");
428 "will use Flac channel layout for "
429 "%d channels.\n", channels);
453 for (i = 0; i < 16; i++) {
472 for (ch = 0; ch < s->
channels; ch++) {
498 #define COPY_SAMPLES(bits) do { \
499 const int ## bits ## _t *samples0 = samples; \
501 for (i = 0, j = 0; i < frame->blocksize; i++) \
502 for (ch = 0; ch < s->channels; ch++, j++) \
503 frame->subframes[ch].samples[i] = samples0[j] >> shift; \
518 for (i = 0; i <
n; i++) {
521 count += (v >> k) + 1 + k;
530 int p, porder, psize;
547 count += pred_order * sub->
obits;
564 for (p = 0; p < 1 << porder; p++) {
577 #define rice_encode_count(sum, n, k) (((n)*((k)+1))+((sum-(n>>1))>>(k)))
589 sum2 = sum - (n >> 1);
590 k =
av_log2(av_clipl_int32(sum2 / n));
591 return FFMIN(k, max_param);
597 int64_t bestbits = INT64_MAX;
600 for (k = 0; k <= max_param; k++) {
601 int64_t
bits = sums[k][i];
602 if (bits < bestbits) {
613 int n,
int pred_order,
int max_param,
int exact)
619 part = (1 << porder);
622 cnt = (n >> porder) - pred_order;
623 for (i = 0; i < part; i++) {
626 all_bits += sums[k][i];
646 const uint32_t *res, *res_end;
651 for (k = 0; k <= kmax; k++) {
652 res = &data[pred_order];
653 res_end = &data[n >> pmax];
654 for (i = 0; i < parts; i++) {
656 uint64_t sum = (1LL + k) * (res_end - res);
657 while (res < res_end)
658 sum += *(res++) >> k;
662 while (res < res_end)
666 res_end += n >> pmax;
674 int parts = (1 <<
level);
675 for (i = 0; i < parts; i++) {
676 for (k=0; k<=kmax; k++)
677 sums[k][i] = sums[k][2*i] + sums[k][2*i+1];
699 for (i = 0; i <
n; i++)
700 udata[i] = (2 * data[i]) ^ (data[i] >> 31);
702 calc_sum_top(pmax, exact ? kmax : 0, udata, n, pred_order, sums);
705 bits[pmin] = UINT32_MAX;
708 if (bits[i] < bits[opt_porder]) {
717 return bits[opt_porder];
752 for (i = 0; i < order; i++)
756 for (i = order; i <
n; i++)
758 }
else if (order == 1) {
759 for (i = order; i <
n; i++)
760 res[i] = smp[i] - smp[i-1];
761 }
else if (order == 2) {
762 int a = smp[order-1] - smp[order-2];
763 for (i = order; i <
n; i += 2) {
764 int b = smp[i ] - smp[i-1];
766 a = smp[i+1] - smp[i ];
769 }
else if (order == 3) {
770 int a = smp[order-1] - smp[order-2];
771 int c = smp[order-1] - 2*smp[order-2] + smp[order-3];
772 for (i = order; i <
n; i += 2) {
773 int b = smp[i ] - smp[i-1];
776 a = smp[i+1] - smp[i ];
781 int a = smp[order-1] - smp[order-2];
782 int c = smp[order-1] - 2*smp[order-2] + smp[order-3];
783 int e = smp[order-1] - 3*smp[order-2] + 3*smp[order-3] - smp[order-4];
784 for (i = order; i <
n; i += 2) {
785 int b = smp[i ] - smp[i-1];
789 a = smp[i+1] - smp[i ];
801 int min_order, max_order, opt_order, omethod;
815 for (i = 1; i <
n; i++)
827 memcpy(res, smp, n *
sizeof(
int32_t));
843 bits[0] = UINT32_MAX;
844 for (i = min_order; i <= max_order; i++) {
847 if (bits[i] < bits[opt_order])
850 sub->
order = opt_order;
852 if (sub->
order != max_order) {
869 int levels = 1 << omethod;
872 int opt_index = levels-1;
873 opt_order = max_order-1;
874 bits[opt_index] = UINT32_MAX;
875 for (i = levels-1; i >= 0; i--) {
876 int last_order = order;
877 order = min_order + (((max_order-min_order+1) * (i+1)) / levels)-1;
878 order = av_clip(order, min_order - 1, max_order - 1);
879 if (order == last_order)
889 if (bits[i] < bits[opt_index]) {
899 bits[0] = UINT32_MAX;
900 for (i = min_order-1; i < max_order; i++) {
907 if (bits[i] < bits[opt_order])
915 opt_order = min_order - 1 + (max_order-min_order)/3;
916 memset(bits, -1,
sizeof(bits));
918 for (step = 16; step; step >>= 1) {
919 int last = opt_order;
920 for (i = last-step; i <= last+step; i += step) {
921 if (i < min_order-1 || i >= max_order || bits[i] < UINT32_MAX)
929 if (bits[i] < bits[opt_order])
938 int i, step, improved;
939 int64_t best_score = INT64_MAX;
944 for (i=0; i<opt_order; i++)
949 for (step = 0; step < allsteps; step++) {
955 for (i=0; i<opt_order; i++) {
956 int diff = ((tmp + 1) % 3) - 1;
957 lpc_try[i] = av_clip(coefs[opt_order - 1][i] + diff, -qmax, qmax);
970 if (score < best_score) {
972 memcpy(coefs[opt_order-1], lpc_try,
sizeof(*coefs));
979 sub->
order = opt_order;
982 for (i = 0; i < sub->
order; i++)
1040 for (ch = 0; ch < s->
channels; ch++)
1043 count += (8 - (count & 7)) & 7;
1047 if (count > INT_MAX)
1057 for (ch = 0; ch < s->
channels; ch++) {
1067 if (v && !(v & 1)) {
1095 sum[0] = sum[1] = sum[2] = sum[3] = 0;
1096 for (i = 2; i <
n; i++) {
1097 lt = left_ch[i] - 2*left_ch[i-1] + left_ch[i-2];
1098 rt = right_ch[i] - 2*right_ch[i-1] + right_ch[i-2];
1099 sum[2] +=
FFABS((lt + rt) >> 1);
1100 sum[3] +=
FFABS(lt - rt);
1101 sum[0] +=
FFABS(lt);
1102 sum[1] +=
FFABS(rt);
1105 for (i = 0; i < 4; i++) {
1111 score[0] = sum[0] + sum[1];
1112 score[1] = sum[0] + sum[3];
1113 score[2] = sum[1] + sum[3];
1114 score[3] = sum[2] + sum[3];
1118 for (i = 1; i < 4; i++)
1119 if (score[i] < score[best])
1156 for (i = 0; i <
n; i++) {
1158 left[i] = (tmp + right[i]) >> 1;
1159 right[i] = tmp - right[i];
1163 for (i = 0; i <
n; i++)
1164 right[i] = left[i] - right[i];
1167 for (i = 0; i <
n; i++)
1168 left[i] -= right[i];
1203 else if (frame->
bs_code[0] == 7)
1222 for (ch = 0; ch < s->
channels; ch++) {
1224 int i, p, porder, psize;
1240 while (res < frame_end)
1244 for (i = 0; i < sub->
order; i++)
1252 for (i = 0; i < sub->
order; i++)
1266 for (p = 0; p < 1 << porder; p++) {
1269 while (res < part_end)
1271 part_end =
FFMIN(frame_end, part_end + psize);
1312 buf = (
const uint8_t *)samples;
1315 (
const uint16_t *) samples, buf_size / 2);
1320 const int32_t *samples0 = samples;
1339 int frame_bytes, out_bytes,
ret;
1358 *got_packet_ptr = 1;
1387 if (frame_bytes < 0) {
1406 if (out_bytes < s->min_framesize)
1411 avpkt->
size = out_bytes;
1415 *got_packet_ptr = 1;
1433 #define FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
1436 {
"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" },
1441 {
"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 },
1444 {
"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" },
1451 {
"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]
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size)
Check AVPacket size and/or allocate data.
#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
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 max_shift, int zero_shift)
Calculate LPC coefficients for multiple orders.
void(* bswap16_buf)(uint16_t *dst, const uint16_t *src, int len)
static av_cold int init(AVCodecContext *avctx)
int av_ctz(int v)
Trailing zero bit count.
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 CODEC_CAP_LOSSLESS
Codec is lossless.
#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.
#define av_assert0(cond)
assert() equivalent, that is always enabled.
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 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.
static uint64_t find_subframe_rice_params(FlacEncodeContext *s, FlacSubframe *sub, int pred_order)
int params[MAX_PARTITIONS]
static const uint8_t header[24]
int duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
#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.
#define CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
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 CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
#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]
Libavcodec external API header.
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)
av_cold void ff_bswapdsp_init(BswapDSPContext *c)
#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)
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.
int side_data_only_packets
Encoding only.
#define FLAC_STREAMINFO_SIZE
#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)
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.
const int ff_flac_sample_rate_table[16]
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.
av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt, int channels, int bps)
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)
void av_md5_final(AVMD5 *ctx, uint8_t *dst)
Finish hashing and output digest value.
int max_encoded_framesize
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.
void av_fast_malloc(void *ptr, unsigned int *size, size_t min_size)
Allocate a buffer, reusing the given one if large enough.
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]
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)
static av_cold void dprint_compression_options(FlacEncodeContext *s)
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)
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[]
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