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Merge remote-tracking branch 'qatar/master'

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* qatar/master: (24 commits)
  utils: Drop pointless '#if 1' preprocessor directive.
  ac3enc: remove empty ac3_float function that is never called
  ac3enc: split templated float vs. fixed functions into a separate file.
  ac3enc: dynamically allocate AC3EncodeContext fields windowed_samples and mdct
  ac3enc: use function pointer to choose between AC-3 and E-AC-3 header output functions.
  Roll back 4:4:4 H.264 for now Needs some ARM/PPC asm modifications.
  Fix SVQ3 after adding 4:4:4 H.264 support
  H.264: fix CODEC_FLAG_GRAY
  4:4:4 H.264 decoding support
  h264_parser: Fix whitespace after previous change.
  h264_parser: Fix behaviour when PARSER_FLAG_COMPLETE_FRAMES is set.
  wav: remove an invalid free().
  lavf: initialise reference_dts in av_estimate_timings_from_pts.
  h264: don't be so picky on decoding pps in extradata.
  avcodec.h: add or elaborate on some documentation comments.
  h264: change a few comments into error messages
  ac3dec: fix doxy-style for comment ("///>" should be "///<" instead).
  img2: add .dpx to the list of supported file extensions.
  ffv1: fix undefined behavior with insane widths.
  ARM: jrevdct_arm: simplify stack usage
  ...

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2011-06-14 04:55:27 +02:00
commit 173cd695cb
18 changed files with 800 additions and 493 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

8
Changelog
View File

@ -10,6 +10,14 @@ version <next>:
- libxvid aspect pickiness fixed
- Frame multithreaded decoding
- E-AC-3 audio encoder
- ac3enc: add channel coupling support
- floating-point sample format support to the ac3, eac3, dca, aac, and vorbis decoders.
- H264/MPEG frame-level multi-threading
- All av_metadata_* functions renamed to av_dict_* and moved to libavutil
version 0.7_beta2:
- Lots of deprecated API cruft removed
- fft and imdct optimizations for AVX (Sandy Bridge) processors
- showinfo filter added

8
libavcodec/Makefile
View File

@ -63,8 +63,8 @@ OBJS-$(CONFIG_AAC_ENCODER) += aacenc.o aaccoder.o \
OBJS-$(CONFIG_AASC_DECODER) += aasc.o msrledec.o
OBJS-$(CONFIG_AC3_DECODER) += ac3dec.o ac3dec_data.o ac3.o kbdwin.o
OBJS-$(CONFIG_AC3_ENCODER) += ac3enc_combined.o ac3enc_fixed.o ac3enc_float.o ac3tab.o ac3.o kbdwin.o
OBJS-$(CONFIG_AC3_FLOAT_ENCODER) += ac3enc_float.o ac3tab.o ac3.o kbdwin.o
OBJS-$(CONFIG_AC3_FIXED_ENCODER) += ac3enc_fixed.o ac3tab.o ac3.o
OBJS-$(CONFIG_AC3_FLOAT_ENCODER) += ac3enc_float.o ac3tab.o ac3tab.o ac3.o kbdwin.o
OBJS-$(CONFIG_AC3_FIXED_ENCODER) += ac3enc_fixed.o ac3tab.o ac3tab.o ac3.o
OBJS-$(CONFIG_ALAC_DECODER) += alac.o
OBJS-$(CONFIG_ALAC_ENCODER) += alacenc.o
OBJS-$(CONFIG_ALS_DECODER) += alsdec.o bgmc.o mpeg4audio.o
@ -128,8 +128,8 @@ OBJS-$(CONFIG_DVVIDEO_DECODER) += dv.o dvdata.o
OBJS-$(CONFIG_DVVIDEO_ENCODER) += dv.o dvdata.o
OBJS-$(CONFIG_DXA_DECODER) += dxa.o
OBJS-$(CONFIG_EAC3_DECODER) += eac3dec.o eac3dec_data.o
OBJS-$(CONFIG_EAC3_ENCODER) += eac3enc.o ac3enc_float.o ac3tab.o \
ac3.o kbdwin.o
OBJS-$(CONFIG_EAC3_ENCODER) += eac3enc.o ac3enc.o ac3enc_float.o \
ac3tab.o ac3.o kbdwin.o
OBJS-$(CONFIG_EACMV_DECODER) += eacmv.o
OBJS-$(CONFIG_EAMAD_DECODER) += eamad.o eaidct.o mpeg12.o \
mpeg12data.o mpegvideo.o \

2
libavcodec/ac3dec.h
View File

@ -196,7 +196,7 @@ typedef struct {
///@}
///@defgroup arrays aligned arrays
DECLARE_ALIGNED(16, int, fixed_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS]; ///> fixed-point transform coefficients
DECLARE_ALIGNED(16, int, fixed_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS]; ///< fixed-point transform coefficients
DECLARE_ALIGNED(32, float, transform_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS]; ///< transform coefficients
DECLARE_ALIGNED(32, float, delay)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]; ///< delay - added to the next block
DECLARE_ALIGNED(32, float, window)[AC3_BLOCK_SIZE]; ///< window coefficients

456
libavcodec/ac3enc.c
View File

@ -42,7 +42,6 @@
#include "ac3.h"
#include "audioconvert.h"
#include "fft.h"
#include "ac3enc.h"
#include "eac3enc.h"
@ -68,46 +67,6 @@ static const float extmixlev_options[EXTMIXLEV_NUM_OPTIONS] = {
};
#define OFFSET(param) offsetof(AC3EncodeContext, options.param)
#define AC3ENC_PARAM (AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM)
#define AC3ENC_TYPE_AC3_FIXED 0
#define AC3ENC_TYPE_AC3 1
#define AC3ENC_TYPE_EAC3 2
#if CONFIG_AC3ENC_FLOAT
#define AC3ENC_TYPE AC3ENC_TYPE_AC3
#include "ac3enc_opts_template.c"
static AVClass ac3enc_class = { "AC-3 Encoder", av_default_item_name,
ac3_options, LIBAVUTIL_VERSION_INT };
#undef AC3ENC_TYPE
#define AC3ENC_TYPE AC3ENC_TYPE_EAC3
#include "ac3enc_opts_template.c"
static AVClass eac3enc_class = { "E-AC-3 Encoder", av_default_item_name,
eac3_options, LIBAVUTIL_VERSION_INT };
#else
#define AC3ENC_TYPE AC3ENC_TYPE_AC3_FIXED
#include "ac3enc_opts_template.c"
static AVClass ac3enc_class = { "Fixed-Point AC-3 Encoder", av_default_item_name,
ac3fixed_options, LIBAVUTIL_VERSION_INT };
#endif
/* prototypes for functions in ac3enc_fixed.c and ac3enc_float.c */
static av_cold void mdct_end(AC3MDCTContext *mdct);
static av_cold int mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct,
int nbits);
static void apply_window(DSPContext *dsp, SampleType *output, const SampleType *input,
const SampleType *window, unsigned int len);
static int normalize_samples(AC3EncodeContext *s);
static void scale_coefficients(AC3EncodeContext *s);
/**
* LUT for number of exponent groups.
* exponent_group_tab[coupling][exponent strategy-1][number of coefficients]
@ -118,8 +77,7 @@ static uint8_t exponent_group_tab[2][3][256];
/**
* List of supported channel layouts.
*/
#if CONFIG_AC3ENC_FLOAT || !CONFIG_AC3_FLOAT_ENCODER //we need this exactly once compiled in
const int64_t ff_ac3_channel_layouts[] = {
const int64_t ff_ac3_channel_layouts[19] = {
AV_CH_LAYOUT_MONO,
AV_CH_LAYOUT_STEREO,
AV_CH_LAYOUT_2_1,
@ -140,7 +98,6 @@ const int64_t ff_ac3_channel_layouts[] = {
AV_CH_LAYOUT_5POINT1_BACK,
0
};
#endif
/**
@ -233,60 +190,6 @@ static void adjust_frame_size(AC3EncodeContext *s)
}
/**
* Deinterleave input samples.
* Channels are reordered from FFmpeg's default order to AC-3 order.
*/
static void deinterleave_input_samples(AC3EncodeContext *s,
const SampleType *samples)
{
int ch, i;
/* deinterleave and remap input samples */
for (ch = 0; ch < s->channels; ch++) {
const SampleType *sptr;
int sinc;
/* copy last 256 samples of previous frame to the start of the current frame */
memcpy(&s->planar_samples[ch][0], &s->planar_samples[ch][AC3_FRAME_SIZE],
AC3_BLOCK_SIZE * sizeof(s->planar_samples[0][0]));
/* deinterleave */
sinc = s->channels;
sptr = samples + s->channel_map[ch];
for (i = AC3_BLOCK_SIZE; i < AC3_FRAME_SIZE+AC3_BLOCK_SIZE; i++) {
s->planar_samples[ch][i] = *sptr;
sptr += sinc;
}
}
}
/**
* Apply the MDCT to input samples to generate frequency coefficients.
* This applies the KBD window and normalizes the input to reduce precision
* loss due to fixed-point calculations.
*/
static void apply_mdct(AC3EncodeContext *s)
{
int blk, ch;
for (ch = 0; ch < s->channels; ch++) {
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
AC3Block *block = &s->blocks[blk];
const SampleType *input_samples = &s->planar_samples[ch][blk * AC3_BLOCK_SIZE];
apply_window(&s->dsp, s->windowed_samples, input_samples, s->mdct.window, AC3_WINDOW_SIZE);
block->coeff_shift[ch+1] = normalize_samples(s);
s->mdct.fft.mdct_calcw(&s->mdct.fft, block->mdct_coef[ch+1],
s->windowed_samples);
}
}
}
static void compute_coupling_strategy(AC3EncodeContext *s)
{
int blk, ch;
@ -348,296 +251,6 @@ static void compute_coupling_strategy(AC3EncodeContext *s)
}
/**
* Calculate a single coupling coordinate.
*/
static inline float calc_cpl_coord(float energy_ch, float energy_cpl)
{
float coord = 0.125;
if (energy_cpl > 0)
coord *= sqrtf(energy_ch / energy_cpl);
return coord;
}
/**
* Calculate coupling channel and coupling coordinates.
* TODO: Currently this is only used for the floating-point encoder. I was
* able to make it work for the fixed-point encoder, but quality was
* generally lower in most cases than not using coupling. If a more
* adaptive coupling strategy were to be implemented it might be useful
* at that time to use coupling for the fixed-point encoder as well.
*/
static void apply_channel_coupling(AC3EncodeContext *s)
{
#if CONFIG_AC3ENC_FLOAT
LOCAL_ALIGNED_16(float, cpl_coords, [AC3_MAX_BLOCKS], [AC3_MAX_CHANNELS][16]);
LOCAL_ALIGNED_16(int32_t, fixed_cpl_coords, [AC3_MAX_BLOCKS], [AC3_MAX_CHANNELS][16]);
int blk, ch, bnd, i, j;
CoefSumType energy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][16] = {{{0}}};
int num_cpl_coefs = s->num_cpl_subbands * 12;
memset(cpl_coords, 0, AC3_MAX_BLOCKS * sizeof(*cpl_coords));
memset(fixed_cpl_coords, 0, AC3_MAX_BLOCKS * sizeof(*fixed_cpl_coords));
/* calculate coupling channel from fbw channels */
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
AC3Block *block = &s->blocks[blk];
CoefType *cpl_coef = &block->mdct_coef[CPL_CH][s->start_freq[CPL_CH]];
if (!block->cpl_in_use)
continue;
memset(cpl_coef-1, 0, (num_cpl_coefs+4) * sizeof(*cpl_coef));
for (ch = 1; ch <= s->fbw_channels; ch++) {
CoefType *ch_coef = &block->mdct_coef[ch][s->start_freq[CPL_CH]];
if (!block->channel_in_cpl[ch])
continue;
for (i = 0; i < num_cpl_coefs; i++)
cpl_coef[i] += ch_coef[i];
}
/* note: coupling start bin % 4 will always be 1 and num_cpl_coefs
will always be a multiple of 12, so we need to subtract 1 from
the start and add 4 to the length when using optimized
functions which require 16-byte alignment. */
/* coefficients must be clipped to +/- 1.0 in order to be encoded */
s->dsp.vector_clipf(cpl_coef-1, cpl_coef-1, -1.0f, 1.0f, num_cpl_coefs+4);
/* scale coupling coefficients from float to 24-bit fixed-point */
s->ac3dsp.float_to_fixed24(&block->fixed_coef[CPL_CH][s->start_freq[CPL_CH]-1],
cpl_coef-1, num_cpl_coefs+4);
}
/* calculate energy in each band in coupling channel and each fbw channel */
/* TODO: possibly use SIMD to speed up energy calculation */
bnd = 0;
i = s->start_freq[CPL_CH];
while (i < s->cpl_end_freq) {
int band_size = s->cpl_band_sizes[bnd];
for (ch = CPL_CH; ch <= s->fbw_channels; ch++) {
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
AC3Block *block = &s->blocks[blk];
if (!block->cpl_in_use || (ch > CPL_CH && !block->channel_in_cpl[ch]))
continue;
for (j = 0; j < band_size; j++) {
CoefType v = block->mdct_coef[ch][i+j];
MAC_COEF(energy[blk][ch][bnd], v, v);
}
}
}
i += band_size;
bnd++;
}
/* determine which blocks to send new coupling coordinates for */
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
AC3Block *block = &s->blocks[blk];
AC3Block *block0 = blk ? &s->blocks[blk-1] : NULL;
int new_coords = 0;
CoefSumType coord_diff[AC3_MAX_CHANNELS] = {0,};
if (block->cpl_in_use) {
/* calculate coupling coordinates for all blocks and calculate the
average difference between coordinates in successive blocks */
for (ch = 1; ch <= s->fbw_channels; ch++) {
if (!block->channel_in_cpl[ch])
continue;
for (bnd = 0; bnd < s->num_cpl_bands; bnd++) {
cpl_coords[blk][ch][bnd] = calc_cpl_coord(energy[blk][ch][bnd],
energy[blk][CPL_CH][bnd]);
if (blk > 0 && block0->cpl_in_use &&
block0->channel_in_cpl[ch]) {
coord_diff[ch] += fabs(cpl_coords[blk-1][ch][bnd] -
cpl_coords[blk ][ch][bnd]);
}
}
coord_diff[ch] /= s->num_cpl_bands;
}
/* send new coordinates if this is the first block, if previous
* block did not use coupling but this block does, the channels
* using coupling has changed from the previous block, or the
* coordinate difference from the last block for any channel is
* greater than a threshold value. */
if (blk == 0) {
new_coords = 1;
} else if (!block0->cpl_in_use) {
new_coords = 1;
} else {
for (ch = 1; ch <= s->fbw_channels; ch++) {
if (block->channel_in_cpl[ch] && !block0->channel_in_cpl[ch]) {
new_coords = 1;
break;
}
}
if (!new_coords) {
for (ch = 1; ch <= s->fbw_channels; ch++) {
if (block->channel_in_cpl[ch] && coord_diff[ch] > 0.04) {
new_coords = 1;
break;
}
}
}
}
}
block->new_cpl_coords = new_coords;
}
/* calculate final coupling coordinates, taking into account reusing of
coordinates in successive blocks */
for (bnd = 0; bnd < s->num_cpl_bands; bnd++) {
blk = 0;
while (blk < AC3_MAX_BLOCKS) {
int blk1;
CoefSumType energy_cpl;
AC3Block *block = &s->blocks[blk];
if (!block->cpl_in_use) {
blk++;
continue;
}
energy_cpl = energy[blk][CPL_CH][bnd];
blk1 = blk+1;
while (!s->blocks[blk1].new_cpl_coords && blk1 < AC3_MAX_BLOCKS) {
if (s->blocks[blk1].cpl_in_use)
energy_cpl += energy[blk1][CPL_CH][bnd];
blk1++;
}
for (ch = 1; ch <= s->fbw_channels; ch++) {
CoefType energy_ch;
if (!block->channel_in_cpl[ch])
continue;
energy_ch = energy[blk][ch][bnd];
blk1 = blk+1;
while (!s->blocks[blk1].new_cpl_coords && blk1 < AC3_MAX_BLOCKS) {
if (s->blocks[blk1].cpl_in_use)
energy_ch += energy[blk1][ch][bnd];
blk1++;
}
cpl_coords[blk][ch][bnd] = calc_cpl_coord(energy_ch, energy_cpl);
}
blk = blk1;
}
}
/* calculate exponents/mantissas for coupling coordinates */
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
AC3Block *block = &s->blocks[blk];
if (!block->cpl_in_use || !block->new_cpl_coords)
continue;
s->ac3dsp.float_to_fixed24(fixed_cpl_coords[blk][1],
cpl_coords[blk][1],
s->fbw_channels * 16);
s->ac3dsp.extract_exponents(block->cpl_coord_exp[1],
fixed_cpl_coords[blk][1],
s->fbw_channels * 16);
for (ch = 1; ch <= s->fbw_channels; ch++) {
int bnd, min_exp, max_exp, master_exp;
/* determine master exponent */
min_exp = max_exp = block->cpl_coord_exp[ch][0];
for (bnd = 1; bnd < s->num_cpl_bands; bnd++) {
int exp = block->cpl_coord_exp[ch][bnd];
min_exp = FFMIN(exp, min_exp);
max_exp = FFMAX(exp, max_exp);
}
master_exp = ((max_exp - 15) + 2) / 3;
master_exp = FFMAX(master_exp, 0);
while (min_exp < master_exp * 3)
master_exp--;
for (bnd = 0; bnd < s->num_cpl_bands; bnd++) {
block->cpl_coord_exp[ch][bnd] = av_clip(block->cpl_coord_exp[ch][bnd] -
master_exp * 3, 0, 15);
}
block->cpl_master_exp[ch] = master_exp;
/* quantize mantissas */
for (bnd = 0; bnd < s->num_cpl_bands; bnd++) {
int cpl_exp = block->cpl_coord_exp[ch][bnd];
int cpl_mant = (fixed_cpl_coords[blk][ch][bnd] << (5 + cpl_exp + master_exp * 3)) >> 24;
if (cpl_exp == 15)
cpl_mant >>= 1;
else
cpl_mant -= 16;
block->cpl_coord_mant[ch][bnd] = cpl_mant;
}
}
}
if (CONFIG_EAC3_ENCODER && s->eac3)
ff_eac3_set_cpl_states(s);
#endif /* CONFIG_AC3ENC_FLOAT */
}
/**
* Determine rematrixing flags for each block and band.
*/
static void compute_rematrixing_strategy(AC3EncodeContext *s)
{
int nb_coefs;
int blk, bnd, i;
AC3Block *block, *block0;
if (s->channel_mode != AC3_CHMODE_STEREO)
return;
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
block = &s->blocks[blk];
block->new_rematrixing_strategy = !blk;
if (!s->rematrixing_enabled) {
block0 = block;
continue;
}
block->num_rematrixing_bands = 4;
if (block->cpl_in_use) {
block->num_rematrixing_bands -= (s->start_freq[CPL_CH] <= 61);
block->num_rematrixing_bands -= (s->start_freq[CPL_CH] == 37);
if (blk && block->num_rematrixing_bands != block0->num_rematrixing_bands)
block->new_rematrixing_strategy = 1;
}
nb_coefs = FFMIN(block->end_freq[1], block->end_freq[2]);
for (bnd = 0; bnd < block->num_rematrixing_bands; bnd++) {
/* calculate calculate sum of squared coeffs for one band in one block */
int start = ff_ac3_rematrix_band_tab[bnd];
int end = FFMIN(nb_coefs, ff_ac3_rematrix_band_tab[bnd+1]);
CoefSumType sum[4] = {0,};
for (i = start; i < end; i++) {
CoefType lt = block->mdct_coef[1][i];
CoefType rt = block->mdct_coef[2][i];
CoefType md = lt + rt;
CoefType sd = lt - rt;
MAC_COEF(sum[0], lt, lt);
MAC_COEF(sum[1], rt, rt);
MAC_COEF(sum[2], md, md);
MAC_COEF(sum[3], sd, sd);
}
/* compare sums to determine if rematrixing will be used for this band */
if (FFMIN(sum[2], sum[3]) < FFMIN(sum[0], sum[1]))
block->rematrixing_flags[bnd] = 1;
else
block->rematrixing_flags[bnd] = 0;
/* determine if new rematrixing flags will be sent */
if (blk &&
block->rematrixing_flags[bnd] != block0->rematrixing_flags[bnd]) {
block->new_rematrixing_strategy = 1;
}
}
block0 = block;
}
}
/**
* Apply stereo rematrixing to coefficients based on rematrixing flags.
*/
@ -1470,7 +1083,7 @@ static int compute_bit_allocation(AC3EncodeContext *s)
if (s->cpl_on) {
s->cpl_on = 0;
compute_coupling_strategy(s);
compute_rematrixing_strategy(s);
s->compute_rematrixing_strategy(s);
apply_rematrixing(s);
process_exponents(s);
ret = compute_bit_allocation(s);
@ -1990,10 +1603,7 @@ static void output_frame(AC3EncodeContext *s, unsigned char *frame)
init_put_bits(&s->pb, frame, AC3_MAX_CODED_FRAME_SIZE);
if (CONFIG_EAC3_ENCODER && s->eac3)
ff_eac3_output_frame_header(s);
else
ac3_output_frame_header(s);
s->output_frame_header(s);
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++)
output_audio_block(s, blk);
@ -2268,8 +1878,8 @@ static int validate_metadata(AVCodecContext *avctx)
/**
* Encode a single AC-3 frame.
*/
static int ac3_encode_frame(AVCodecContext *avctx, unsigned char *frame,
int buf_size, void *data)
int ff_ac3_encode_frame(AVCodecContext *avctx, unsigned char *frame,
int buf_size, void *data)
{
AC3EncodeContext *s = avctx->priv_data;
const SampleType *samples = data;
@ -2284,19 +1894,19 @@ static int ac3_encode_frame(AVCodecContext *avctx, unsigned char *frame,
if (s->bit_alloc.sr_code == 1 || s->eac3)
adjust_frame_size(s);
deinterleave_input_samples(s, samples);
s->deinterleave_input_samples(s, samples);
apply_mdct(s);
s->apply_mdct(s);
scale_coefficients(s);
s->scale_coefficients(s);
s->cpl_on = s->cpl_enabled;
compute_coupling_strategy(s);
if (s->cpl_on)
apply_channel_coupling(s);
s->apply_channel_coupling(s);
compute_rematrixing_strategy(s);
s->compute_rematrixing_strategy(s);
apply_rematrixing(s);
@ -2319,11 +1929,12 @@ static int ac3_encode_frame(AVCodecContext *avctx, unsigned char *frame,
/**
* Finalize encoding and free any memory allocated by the encoder.
*/
static av_cold int ac3_encode_close(AVCodecContext *avctx)
av_cold int ff_ac3_encode_close(AVCodecContext *avctx)
{
int blk, ch;
AC3EncodeContext *s = avctx->priv_data;
av_freep(&s->windowed_samples);
for (ch = 0; ch < s->channels; ch++)
av_freep(&s->planar_samples[ch]);
av_freep(&s->planar_samples);
@ -2349,7 +1960,8 @@ static av_cold int ac3_encode_close(AVCodecContext *avctx)
av_freep(&block->qmant);
}
mdct_end(&s->mdct);
s->mdct_end(s->mdct);
av_freep(&s->mdct);
av_freep(&avctx->coded_frame);
return 0;
@ -2519,8 +2131,7 @@ static av_cold int validate_options(AVCodecContext *avctx, AC3EncodeContext *s)
(s->channel_mode == AC3_CHMODE_STEREO);
s->cpl_enabled = s->options.channel_coupling &&
s->channel_mode >= AC3_CHMODE_STEREO &&
CONFIG_AC3ENC_FLOAT;
s->channel_mode >= AC3_CHMODE_STEREO && !s->fixed_point;
return 0;
}
@ -2604,6 +2215,8 @@ static av_cold int allocate_buffers(AVCodecContext *avctx)
AC3EncodeContext *s = avctx->priv_data;
int channels = s->channels + 1; /* includes coupling channel */
FF_ALLOC_OR_GOTO(avctx, s->windowed_samples, AC3_WINDOW_SIZE *
sizeof(*s->windowed_samples), alloc_fail);
FF_ALLOC_OR_GOTO(avctx, s->planar_samples, s->channels * sizeof(*s->planar_samples),
alloc_fail);
for (ch = 0; ch < s->channels; ch++) {
@ -2676,7 +2289,7 @@ static av_cold int allocate_buffers(AVCodecContext *avctx)
}
}
if (CONFIG_AC3ENC_FLOAT) {
if (!s->fixed_point) {
FF_ALLOCZ_OR_GOTO(avctx, s->fixed_coef_buffer, AC3_MAX_BLOCKS * channels *
AC3_MAX_COEFS * sizeof(*s->fixed_coef_buffer), alloc_fail);
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
@ -2705,7 +2318,7 @@ alloc_fail:
/**
* Initialize the encoder.
*/
static av_cold int ac3_encode_init(AVCodecContext *avctx)
av_cold int ff_ac3_encode_init(AVCodecContext *avctx)
{
AC3EncodeContext *s = avctx->priv_data;
int ret, frame_size_58;
@ -2735,13 +2348,40 @@ static av_cold int ac3_encode_init(AVCodecContext *avctx)
s->crc_inv[1] = pow_poly((CRC16_POLY >> 1), (8 * frame_size_58) - 16, CRC16_POLY);
}
/* set function pointers */
if (CONFIG_AC3_FIXED_ENCODER && s->fixed_point) {
s->mdct_end = ff_ac3_fixed_mdct_end;
s->mdct_init = ff_ac3_fixed_mdct_init;
s->apply_window = ff_ac3_fixed_apply_window;
s->normalize_samples = ff_ac3_fixed_normalize_samples;
s->scale_coefficients = ff_ac3_fixed_scale_coefficients;
s->deinterleave_input_samples = ff_ac3_fixed_deinterleave_input_samples;
s->apply_mdct = ff_ac3_fixed_apply_mdct;
s->apply_channel_coupling = ff_ac3_fixed_apply_channel_coupling;
s->compute_rematrixing_strategy = ff_ac3_fixed_compute_rematrixing_strategy;
} else if (CONFIG_AC3_ENCODER || CONFIG_EAC3_ENCODER) {
s->mdct_end = ff_ac3_float_mdct_end;
s->mdct_init = ff_ac3_float_mdct_init;
s->apply_window = ff_ac3_float_apply_window;
s->scale_coefficients = ff_ac3_float_scale_coefficients;
s->deinterleave_input_samples = ff_ac3_float_deinterleave_input_samples;
s->apply_mdct = ff_ac3_float_apply_mdct;
s->apply_channel_coupling = ff_ac3_float_apply_channel_coupling;
s->compute_rematrixing_strategy = ff_ac3_float_compute_rematrixing_strategy;
}
if (CONFIG_EAC3_ENCODER && s->eac3)
s->output_frame_header = ff_eac3_output_frame_header;
else
s->output_frame_header = ac3_output_frame_header;
set_bandwidth(s);
exponent_init(s);
bit_alloc_init(s);
ret = mdct_init(avctx, &s->mdct, 9);
FF_ALLOCZ_OR_GOTO(avctx, s->mdct, sizeof(AC3MDCTContext), init_fail);
ret = s->mdct_init(avctx, s->mdct, 9);
if (ret)
goto init_fail;
@ -2758,6 +2398,6 @@ static av_cold int ac3_encode_init(AVCodecContext *avctx)
return 0;
init_fail:
ac3_encode_close(avctx);
ff_ac3_encode_close(avctx);
return ret;
}

83
libavcodec/ac3enc.h
View File

@ -40,18 +40,28 @@
#define CONFIG_AC3ENC_FLOAT 0
#endif
#define OFFSET(param) offsetof(AC3EncodeContext, options.param)
#define AC3ENC_PARAM (AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM)
#define AC3ENC_TYPE_AC3_FIXED 0
#define AC3ENC_TYPE_AC3 1
#define AC3ENC_TYPE_EAC3 2
#if CONFIG_AC3ENC_FLOAT
#define AC3_NAME(x) ff_ac3_float_ ## x
#define MAC_COEF(d,a,b) ((d)+=(a)*(b))
typedef float SampleType;
typedef float CoefType;
typedef float CoefSumType;
#else
#define AC3_NAME(x) ff_ac3_fixed_ ## x
#define MAC_COEF(d,a,b) MAC64(d,a,b)
typedef int16_t SampleType;
typedef int32_t CoefType;
typedef int64_t CoefSumType;
#endif
typedef struct AC3MDCTContext {
const SampleType *window; ///< MDCT window function
FFTContext fft; ///< FFT context for MDCT calculation
@ -128,10 +138,11 @@ typedef struct AC3EncodeContext {
PutBitContext pb; ///< bitstream writer context
DSPContext dsp;
AC3DSPContext ac3dsp; ///< AC-3 optimized functions
AC3MDCTContext mdct; ///< MDCT context
AC3MDCTContext *mdct; ///< MDCT context
AC3Block blocks[AC3_MAX_BLOCKS]; ///< per-block info
int fixed_point; ///< indicates if fixed-point encoder is being used
int eac3; ///< indicates if this is E-AC-3 vs. AC-3
int bitstream_id; ///< bitstream id (bsid)
int bitstream_mode; ///< bitstream mode (bsmod)
@ -189,6 +200,7 @@ typedef struct AC3EncodeContext {
int frame_bits; ///< all frame bits except exponents and mantissas
int exponent_bits; ///< number of bits used for exponents
SampleType *windowed_samples;
SampleType **planar_samples;
uint8_t *bap_buffer;
uint8_t *bap1_buffer;
@ -208,7 +220,74 @@ typedef struct AC3EncodeContext {
uint8_t *ref_bap [AC3_MAX_CHANNELS][AC3_MAX_BLOCKS]; ///< bit allocation pointers (bap)
int ref_bap_set; ///< indicates if ref_bap pointers have been set
DECLARE_ALIGNED(32, SampleType, windowed_samples)[AC3_WINDOW_SIZE];
/* fixed vs. float function pointers */
void (*mdct_end)(AC3MDCTContext *mdct);
int (*mdct_init)(AVCodecContext *avctx, AC3MDCTContext *mdct, int nbits);
void (*apply_window)(DSPContext *dsp, SampleType *output,
const SampleType *input, const SampleType *window,
unsigned int len);
int (*normalize_samples)(struct AC3EncodeContext *s);
void (*scale_coefficients)(struct AC3EncodeContext *s);
/* fixed vs. float templated function pointers */
void (*deinterleave_input_samples)(struct AC3EncodeContext *s,
const SampleType *samples);
void (*apply_mdct)(struct AC3EncodeContext *s);
void (*apply_channel_coupling)(struct AC3EncodeContext *s);
void (*compute_rematrixing_strategy)(struct AC3EncodeContext *s);
/* AC-3 vs. E-AC-3 function pointers */
void (*output_frame_header)(struct AC3EncodeContext *s);
} AC3EncodeContext;
extern const int64_t ff_ac3_channel_layouts[19];
int ff_ac3_encode_init(AVCodecContext *avctx);
int ff_ac3_encode_frame(AVCodecContext *avctx, unsigned char *frame,
int buf_size, void *data);
int ff_ac3_encode_close(AVCodecContext *avctx);
/* prototypes for functions in ac3enc_fixed.c and ac3enc_float.c */
void ff_ac3_fixed_mdct_end(AC3MDCTContext *mdct);
void ff_ac3_float_mdct_end(AC3MDCTContext *mdct);
int ff_ac3_fixed_mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct,
int nbits);
int ff_ac3_float_mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct,
int nbits);
void ff_ac3_fixed_apply_window(DSPContext *dsp, SampleType *output,
const SampleType *input,
const SampleType *window, unsigned int len);
void ff_ac3_float_apply_window(DSPContext *dsp, SampleType *output,
const SampleType *input,
const SampleType *window, unsigned int len);
int ff_ac3_fixed_normalize_samples(AC3EncodeContext *s);
void ff_ac3_fixed_scale_coefficients(AC3EncodeContext *s);
void ff_ac3_float_scale_coefficients(AC3EncodeContext *s);
/* prototypes for functions in ac3enc_template.c */
void ff_ac3_fixed_deinterleave_input_samples(AC3EncodeContext *s,
const SampleType *samples);
void ff_ac3_float_deinterleave_input_samples(AC3EncodeContext *s,
const SampleType *samples);
void ff_ac3_fixed_apply_mdct(AC3EncodeContext *s);
void ff_ac3_float_apply_mdct(AC3EncodeContext *s);
void ff_ac3_fixed_apply_channel_coupling(AC3EncodeContext *s);
void ff_ac3_float_apply_channel_coupling(AC3EncodeContext *s);
void ff_ac3_fixed_compute_rematrixing_strategy(AC3EncodeContext *s);
void ff_ac3_float_compute_rematrixing_strategy(AC3EncodeContext *s);
#endif /* AVCODEC_AC3ENC_H */

38
libavcodec/ac3enc_fixed.c
View File

@ -28,13 +28,20 @@
#define CONFIG_FFT_FLOAT 0
#undef CONFIG_AC3ENC_FLOAT
#include "ac3enc.c"
#include "ac3enc.h"
#define AC3ENC_TYPE AC3ENC_TYPE_AC3_FIXED
#include "ac3enc_opts_template.c"
static AVClass ac3enc_class = { "Fixed-Point AC-3 Encoder", av_default_item_name,
ac3fixed_options, LIBAVUTIL_VERSION_INT };
#include "ac3enc_template.c"
/**
* Finalize MDCT and free allocated memory.
*/
static av_cold void mdct_end(AC3MDCTContext *mdct)
av_cold void AC3_NAME(mdct_end)(AC3MDCTContext *mdct)
{
ff_mdct_end(&mdct->fft);
}
@ -44,8 +51,8 @@ static av_cold void mdct_end(AC3MDCTContext *mdct)
* Initialize MDCT tables.
* @param nbits log2(MDCT size)
*/
static av_cold int mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct,
int nbits)
av_cold int AC3_NAME(mdct_init)(AVCodecContext *avctx, AC3MDCTContext *mdct,
int nbits)
{
int ret = ff_mdct_init(&mdct->fft, nbits, 0, -1.0);
mdct->window = ff_ac3_window;
@ -56,8 +63,9 @@ static av_cold int mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct,
/**
* Apply KBD window to input samples prior to MDCT.
*/
static void apply_window(DSPContext *dsp, int16_t *output, const int16_t *input,
const int16_t *window, unsigned int len)
void AC3_NAME(apply_window)(DSPContext *dsp, int16_t *output,
const int16_t *input, const int16_t *window,
unsigned int len)
{
dsp->apply_window_int16(output, input, window, len);
}
@ -82,7 +90,7 @@ static int log2_tab(AC3EncodeContext *s, int16_t *src, int len)
*
* @return exponent shift
*/
static int normalize_samples(AC3EncodeContext *s)
int AC3_NAME(normalize_samples)(AC3EncodeContext *s)
{
int v = 14 - log2_tab(s, s->windowed_samples, AC3_WINDOW_SIZE);
if (v > 0)
@ -95,7 +103,7 @@ static int normalize_samples(AC3EncodeContext *s)
/**
* Scale MDCT coefficients to 25-bit signed fixed-point.
*/
static void scale_coefficients(AC3EncodeContext *s)
void AC3_NAME(scale_coefficients)(AC3EncodeContext *s)
{
int blk, ch;
@ -109,14 +117,22 @@ static void scale_coefficients(AC3EncodeContext *s)
}
static av_cold int ac3_fixed_encode_init(AVCodecContext *avctx)
{
AC3EncodeContext *s = avctx->priv_data;
s->fixed_point = 1;
return ff_ac3_encode_init(avctx);
}
AVCodec ff_ac3_fixed_encoder = {
"ac3_fixed",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_AC3,
sizeof(AC3EncodeContext),
ac3_encode_init,
ac3_encode_frame,
ac3_encode_close,
ac3_fixed_encode_init,
ff_ac3_encode_frame,
ff_ac3_encode_close,
NULL,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),

58
libavcodec/ac3enc_float.c
View File

@ -27,14 +27,25 @@
*/
#define CONFIG_AC3ENC_FLOAT 1
#include "ac3enc.c"
#include "ac3enc.h"
#include "eac3enc.h"
#include "kbdwin.h"
#if CONFIG_AC3_ENCODER
#define AC3ENC_TYPE AC3ENC_TYPE_AC3
#include "ac3enc_opts_template.c"
static AVClass ac3enc_class = { "AC-3 Encoder", av_default_item_name,
ac3_options, LIBAVUTIL_VERSION_INT };
#endif
#include "ac3enc_template.c"
/**
* Finalize MDCT and free allocated memory.
*/
static av_cold void mdct_end(AC3MDCTContext *mdct)
av_cold void ff_ac3_float_mdct_end(AC3MDCTContext *mdct)
{
ff_mdct_end(&mdct->fft);
av_freep(&mdct->window);
@ -45,8 +56,8 @@ static av_cold void mdct_end(AC3MDCTContext *mdct)
* Initialize MDCT tables.
* @param nbits log2(MDCT size)
*/
static av_cold int mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct,
int nbits)
av_cold int ff_ac3_float_mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct,
int nbits)
{
float *window;
int i, n, n2;
@ -71,27 +82,18 @@ static av_cold int mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct,
/**
* Apply KBD window to input samples prior to MDCT.
*/
static void apply_window(DSPContext *dsp, float *output, const float *input,
const float *window, unsigned int len)
void ff_ac3_float_apply_window(DSPContext *dsp, float *output,
const float *input, const float *window,
unsigned int len)
{
dsp->vector_fmul(output, input, window, len);
}
/**
* Normalize the input samples to use the maximum available precision.
*/
static int normalize_samples(AC3EncodeContext *s)
{
/* Normalization is not needed for floating-point samples, so just return 0 */
return 0;
}
/**
* Scale MDCT coefficients from float to 24-bit fixed-point.
*/
static void scale_coefficients(AC3EncodeContext *s)
void ff_ac3_float_scale_coefficients(AC3EncodeContext *s)
{
int chan_size = AC3_MAX_COEFS * AC3_MAX_BLOCKS;
s->ac3dsp.float_to_fixed24(s->fixed_coef_buffer + chan_size,
@ -106,9 +108,9 @@ AVCodec ff_ac3_float_encoder = {
AVMEDIA_TYPE_AUDIO,
CODEC_ID_AC3,
sizeof(AC3EncodeContext),
ac3_encode_init,
ac3_encode_frame,
ac3_encode_close,
ff_ac3_encode_init,
ff_ac3_encode_frame,
ff_ac3_encode_close,
NULL,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
@ -116,19 +118,3 @@ AVCodec ff_ac3_float_encoder = {
.channel_layouts = ff_ac3_channel_layouts,
};
#endif
#if CONFIG_EAC3_ENCODER
AVCodec ff_eac3_encoder = {
.name = "eac3",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_EAC3,
.priv_data_size = sizeof(AC3EncodeContext),
.init = ac3_encode_init,
.encode = ac3_encode_frame,
.close = ac3_encode_close,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52 E-AC-3"),
.priv_class = &eac3enc_class,
.channel_layouts = ff_ac3_channel_layouts,
};
#endif

3
libavcodec/ac3enc_opts_template.c
View File

@ -19,6 +19,9 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/opt.h"
#include "ac3.h"
#if AC3ENC_TYPE == AC3ENC_TYPE_AC3_FIXED
static const AVOption ac3fixed_options[] = {
#elif AC3ENC_TYPE == AC3ENC_TYPE_AC3

377
libavcodec/ac3enc_template.c Normal file
View File

@ -0,0 +1,377 @@
/*
* AC-3 encoder float/fixed template
* Copyright (c) 2000 Fabrice Bellard
* Copyright (c) 2006-2011 Justin Ruggles <justin.ruggles@gmail.com>
* Copyright (c) 2006-2010 Prakash Punnoor <prakash@punnoor.de>
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* AC-3 encoder float/fixed template
*/
#include <stdint.h>
#include "ac3enc.h"
/**
* Deinterleave input samples.
* Channels are reordered from Libav's default order to AC-3 order.
*/
void AC3_NAME(deinterleave_input_samples)(AC3EncodeContext *s,
const SampleType *samples)
{
int ch, i;
/* deinterleave and remap input samples */
for (ch = 0; ch < s->channels; ch++) {
const SampleType *sptr;
int sinc;
/* copy last 256 samples of previous frame to the start of the current frame */
memcpy(&s->planar_samples[ch][0], &s->planar_samples[ch][AC3_FRAME_SIZE],
AC3_BLOCK_SIZE * sizeof(s->planar_samples[0][0]));
/* deinterleave */
sinc = s->channels;
sptr = samples + s->channel_map[ch];
for (i = AC3_BLOCK_SIZE; i < AC3_FRAME_SIZE+AC3_BLOCK_SIZE; i++) {
s->planar_samples[ch][i] = *sptr;
sptr += sinc;
}
}
}
/**
* Apply the MDCT to input samples to generate frequency coefficients.
* This applies the KBD window and normalizes the input to reduce precision
* loss due to fixed-point calculations.
*/
void AC3_NAME(apply_mdct)(AC3EncodeContext *s)
{
int blk, ch;
for (ch = 0; ch < s->channels; ch++) {
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
AC3Block *block = &s->blocks[blk];
const SampleType *input_samples = &s->planar_samples[ch][blk * AC3_BLOCK_SIZE];
s->apply_window(&s->dsp, s->windowed_samples, input_samples,
s->mdct->window, AC3_WINDOW_SIZE);
if (s->fixed_point)
block->coeff_shift[ch+1] = s->normalize_samples(s);
s->mdct->fft.mdct_calcw(&s->mdct->fft, block->mdct_coef[ch+1],
s->windowed_samples);
}
}
}
/**
* Calculate a single coupling coordinate.
*/
static inline float calc_cpl_coord(float energy_ch, float energy_cpl)
{
float coord = 0.125;
if (energy_cpl > 0)
coord *= sqrtf(energy_ch / energy_cpl);
return coord;
}
/**
* Calculate coupling channel and coupling coordinates.
* TODO: Currently this is only used for the floating-point encoder. I was
* able to make it work for the fixed-point encoder, but quality was
* generally lower in most cases than not using coupling. If a more
* adaptive coupling strategy were to be implemented it might be useful
* at that time to use coupling for the fixed-point encoder as well.
*/
void AC3_NAME(apply_channel_coupling)(AC3EncodeContext *s)
{
#if CONFIG_AC3ENC_FLOAT
LOCAL_ALIGNED_16(float, cpl_coords, [AC3_MAX_BLOCKS], [AC3_MAX_CHANNELS][16]);
LOCAL_ALIGNED_16(int32_t, fixed_cpl_coords, [AC3_MAX_BLOCKS], [AC3_MAX_CHANNELS][16]);
int blk, ch, bnd, i, j;
CoefSumType energy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][16] = {{{0}}};
int num_cpl_coefs = s->num_cpl_subbands * 12;
memset(cpl_coords, 0, AC3_MAX_BLOCKS * sizeof(*cpl_coords));
memset(fixed_cpl_coords, 0, AC3_MAX_BLOCKS * sizeof(*fixed_cpl_coords));
/* calculate coupling channel from fbw channels */
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
AC3Block *block = &s->blocks[blk];
CoefType *cpl_coef = &block->mdct_coef[CPL_CH][s->start_freq[CPL_CH]];
if (!block->cpl_in_use)
continue;
memset(cpl_coef-1, 0, (num_cpl_coefs+4) * sizeof(*cpl_coef));
for (ch = 1; ch <= s->fbw_channels; ch++) {
CoefType *ch_coef = &block->mdct_coef[ch][s->start_freq[CPL_CH]];
if (!block->channel_in_cpl[ch])
continue;
for (i = 0; i < num_cpl_coefs; i++)
cpl_coef[i] += ch_coef[i];
}
/* note: coupling start bin % 4 will always be 1 and num_cpl_coefs
will always be a multiple of 12, so we need to subtract 1 from
the start and add 4 to the length when using optimized
functions which require 16-byte alignment. */
/* coefficients must be clipped to +/- 1.0 in order to be encoded */
s->dsp.vector_clipf(cpl_coef-1, cpl_coef-1, -1.0f, 1.0f, num_cpl_coefs+4);
/* scale coupling coefficients from float to 24-bit fixed-point */
s->ac3dsp.float_to_fixed24(&block->fixed_coef[CPL_CH][s->start_freq[CPL_CH]-1],
cpl_coef-1, num_cpl_coefs+4);
}
/* calculate energy in each band in coupling channel and each fbw channel */
/* TODO: possibly use SIMD to speed up energy calculation */
bnd = 0;
i = s->start_freq[CPL_CH];
while (i < s->cpl_end_freq) {
int band_size = s->cpl_band_sizes[bnd];
for (ch = CPL_CH; ch <= s->fbw_channels; ch++) {
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
AC3Block *block = &s->blocks[blk];
if (!block->cpl_in_use || (ch > CPL_CH && !block->channel_in_cpl[ch]))
continue;
for (j = 0; j < band_size; j++) {
CoefType v = block->mdct_coef[ch][i+j];
MAC_COEF(energy[blk][ch][bnd], v, v);
}
}
}
i += band_size;
bnd++;
}
/* determine which blocks to send new coupling coordinates for */
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
AC3Block *block = &s->blocks[blk];
AC3Block *block0 = blk ? &s->blocks[blk-1] : NULL;
int new_coords = 0;
CoefSumType coord_diff[AC3_MAX_CHANNELS] = {0,};
if (block->cpl_in_use) {
/* calculate coupling coordinates for all blocks and calculate the
average difference between coordinates in successive blocks */
for (ch = 1; ch <= s->fbw_channels; ch++) {
if (!block->channel_in_cpl[ch])
continue;
for (bnd = 0; bnd < s->num_cpl_bands; bnd++) {
cpl_coords[blk][ch][bnd] = calc_cpl_coord(energy[blk][ch][bnd],
energy[blk][CPL_CH][bnd]);
if (blk > 0 && block0->cpl_in_use &&
block0->channel_in_cpl[ch]) {
coord_diff[ch] += fabs(cpl_coords[blk-1][ch][bnd] -
cpl_coords[blk ][ch][bnd]);
}
}
coord_diff[ch] /= s->num_cpl_bands;
}
/* send new coordinates if this is the first block, if previous
* block did not use coupling but this block does, the channels
* using coupling has changed from the previous block, or the
* coordinate difference from the last block for any channel is
* greater than a threshold value. */
if (blk == 0) {
new_coords = 1;
} else if (!block0->cpl_in_use) {
new_coords = 1;
} else {
for (ch = 1; ch <= s->fbw_channels; ch++) {
if (block->channel_in_cpl[ch] && !block0->channel_in_cpl[ch]) {
new_coords = 1;
break;
}
}
if (!new_coords) {
for (ch = 1; ch <= s->fbw_channels; ch++) {
if (block->channel_in_cpl[ch] && coord_diff[ch] > 0.04) {
new_coords = 1;
break;
}
}
}
}
}
block->new_cpl_coords = new_coords;
}
/* calculate final coupling coordinates, taking into account reusing of
coordinates in successive blocks */
for (bnd = 0; bnd < s->num_cpl_bands; bnd++) {
blk = 0;
while (blk < AC3_MAX_BLOCKS) {
int blk1;
CoefSumType energy_cpl;
AC3Block *block = &s->blocks[blk];
if (!block->cpl_in_use) {
blk++;
continue;
}
energy_cpl = energy[blk][CPL_CH][bnd];
blk1 = blk+1;
while (!s->blocks[blk1].new_cpl_coords && blk1 < AC3_MAX_BLOCKS) {
if (s->blocks[blk1].cpl_in_use)
energy_cpl += energy[blk1][CPL_CH][bnd];
blk1++;
}
for (ch = 1; ch <= s->fbw_channels; ch++) {
CoefType energy_ch;
if (!block->channel_in_cpl[ch])
continue;
energy_ch = energy[blk][ch][bnd];
blk1 = blk+1;
while (!s->blocks[blk1].new_cpl_coords && blk1 < AC3_MAX_BLOCKS) {
if (s->blocks[blk1].cpl_in_use)
energy_ch += energy[blk1][ch][bnd];
blk1++;
}
cpl_coords[blk][ch][bnd] = calc_cpl_coord(energy_ch, energy_cpl);
}
blk = blk1;
}
}
/* calculate exponents/mantissas for coupling coordinates */
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
AC3Block *block = &s->blocks[blk];
if (!block->cpl_in_use || !block->new_cpl_coords)
continue;
s->ac3dsp.float_to_fixed24(fixed_cpl_coords[blk][1],
cpl_coords[blk][1],
s->fbw_channels * 16);
s->ac3dsp.extract_exponents(block->cpl_coord_exp[1],
fixed_cpl_coords[blk][1],
s->fbw_channels * 16);
for (ch = 1; ch <= s->fbw_channels; ch++) {
int bnd, min_exp, max_exp, master_exp;
/* determine master exponent */
min_exp = max_exp = block->cpl_coord_exp[ch][0];
for (bnd = 1; bnd < s->num_cpl_bands; bnd++) {
int exp = block->cpl_coord_exp[ch][bnd];
min_exp = FFMIN(exp, min_exp);
max_exp = FFMAX(exp, max_exp);
}
master_exp = ((max_exp - 15) + 2) / 3;
master_exp = FFMAX(master_exp, 0);
while (min_exp < master_exp * 3)
master_exp--;
for (bnd = 0; bnd < s->num_cpl_bands; bnd++) {
block->cpl_coord_exp[ch][bnd] = av_clip(block->cpl_coord_exp[ch][bnd] -
master_exp * 3, 0, 15);
}
block->cpl_master_exp[ch] = master_exp;
/* quantize mantissas */
for (bnd = 0; bnd < s->num_cpl_bands; bnd++) {
int cpl_exp = block->cpl_coord_exp[ch][bnd];
int cpl_mant = (fixed_cpl_coords[blk][ch][bnd] << (5 + cpl_exp + master_exp * 3)) >> 24;
if (cpl_exp == 15)
cpl_mant >>= 1;
else
cpl_mant -= 16;
block->cpl_coord_mant[ch][bnd] = cpl_mant;
}
}
}
if (CONFIG_EAC3_ENCODER && s->eac3)
ff_eac3_set_cpl_states(s);
#endif /* CONFIG_AC3ENC_FLOAT */
}
/**
* Determine rematrixing flags for each block and band.
*/
void AC3_NAME(compute_rematrixing_strategy)(AC3EncodeContext *s)
{
int nb_coefs;
int blk, bnd, i;
AC3Block *block, *av_uninit(block0);
if (s->channel_mode != AC3_CHMODE_STEREO)
return;
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
block = &s->blocks[blk];
block->new_rematrixing_strategy = !blk;
if (!s->rematrixing_enabled) {
block0 = block;
continue;
}
block->num_rematrixing_bands = 4;
if (block->cpl_in_use) {
block->num_rematrixing_bands -= (s->start_freq[CPL_CH] <= 61);
block->num_rematrixing_bands -= (s->start_freq[CPL_CH] == 37);
if (blk && block->num_rematrixing_bands != block0->num_rematrixing_bands)
block->new_rematrixing_strategy = 1;
}
nb_coefs = FFMIN(block->end_freq[1], block->end_freq[2]);
for (bnd = 0; bnd < block->num_rematrixing_bands; bnd++) {
/* calculate calculate sum of squared coeffs for one band in one block */
int start = ff_ac3_rematrix_band_tab[bnd];
int end = FFMIN(nb_coefs, ff_ac3_rematrix_band_tab[bnd+1]);
CoefSumType sum[4] = {0,};
for (i = start; i < end; i++) {
CoefType lt = block->mdct_coef[1][i];
CoefType rt = block->mdct_coef[2][i];
CoefType md = lt + rt;
CoefType sd = lt - rt;
MAC_COEF(sum[0], lt, lt);
MAC_COEF(sum[1], rt, rt);
MAC_COEF(sum[2], md, md);
MAC_COEF(sum[3], sd, sd);
}
/* compare sums to determine if rematrixing will be used for this band */
if (FFMIN(sum[2], sum[3]) < FFMIN(sum[0], sum[1]))
block->rematrixing_flags[bnd] = 1;
else
block->rematrixing_flags[bnd] = 0;
/* determine if new rematrixing flags will be sent */
if (blk &&
block->rematrixing_flags[bnd] != block0->rematrixing_flags[bnd]) {
block->new_rematrixing_strategy = 1;
}
}
block0 = block;
}
}

3
libavcodec/arm/Makefile
View File

@ -5,6 +5,9 @@ OBJS-$(CONFIG_DCA_DECODER) += arm/dcadsp_init_arm.o \
ARMV6-OBJS-$(CONFIG_AC3DSP) += arm/ac3dsp_armv6.o
OBJS-$(CONFIG_MPEGAUDIODSP) += arm/mpegaudiodsp_init_arm.o
ARMV6-OBJS-$(CONFIG_MPEGAUDIODSP) += arm/mpegaudiodsp_fixed_armv6.o
OBJS-$(CONFIG_VP5_DECODER) += arm/vp56dsp_init_arm.o
OBJS-$(CONFIG_VP6_DECODER) += arm/vp56dsp_init_arm.o
OBJS-$(CONFIG_VP8_DECODER) += arm/vp8dsp_init_arm.o

31
libavcodec/arm/jrevdct_arm.S
View File

@ -54,18 +54,13 @@
#define FIX_M_1_961570560_ID 40
#define FIX_M_2_562915447_ID 44
#define FIX_0xFFFF_ID 48
.text
.align
function ff_j_rev_dct_arm, export=1
stmdb sp!, { r4 - r12, lr } @ all callee saved regs
sub sp, sp, #4 @ reserve some space on the stack
str r0, [ sp ] @ save the DCT pointer to the stack
push {r0, r4 - r11, lr}
mov lr, r0 @ lr = pointer to the current row
mov r12, #8 @ r12 = row-counter
adr r11, const_array @ r11 = base pointer to the constants array
movrel r11, const_array @ r11 = base pointer to the constants array
row_loop:
ldrsh r0, [lr, # 0] @ r0 = 'd0'
ldrsh r2, [lr, # 2] @ r2 = 'd2'
@ -102,7 +97,7 @@ row_loop:
add r4, r6, r3, lsl #13 @ r4 = tmp11
rsb r3, r6, r3, lsl #13 @ r3 = tmp12
stmdb sp!, { r0, r2, r3, r4 } @ save on the stack tmp10, tmp13, tmp12, tmp11
push {r0, r2, r3, r4} @ save on the stack tmp10, tmp13, tmp12, tmp11
ldrsh r3, [lr, #10] @ r3 = 'd3'
ldrsh r5, [lr, #12] @ r5 = 'd5'
@ -136,8 +131,8 @@ row_loop:
add r3, r3, r4 @ r3 = tmp2
add r1, r1, r6 @ r1 = tmp3
ldmia sp!, { r0, r2, r4, r6 } @ r0 = tmp10 / r2 = tmp13 / r4 = tmp12 / r6 = tmp11
@ r1 = tmp3 / r3 = tmp2 / r5 = tmp1 / r7 = tmp0
pop {r0, r2, r4, r6} @ r0 = tmp10 / r2 = tmp13 / r4 = tmp12 / r6 = tmp11
@ r1 = tmp3 / r3 = tmp2 / r5 = tmp1 / r7 = tmp0
@ Compute DESCALE(tmp10 + tmp3, CONST_BITS-PASS1_BITS)
add r8, r0, r1
@ -211,7 +206,7 @@ end_of_row_loop:
start_column_loop:
@ Start of column loop
ldr lr, [ sp ]
pop {lr}
mov r12, #8
column_loop:
ldrsh r0, [lr, #( 0*8)] @ r0 = 'd0'
@ -245,7 +240,7 @@ column_loop:
orrs r10, r9, r10
beq empty_odd_column
stmdb sp!, { r0, r2, r4, r6 } @ save on the stack tmp10, tmp13, tmp12, tmp11
push {r0, r2, r4, r6} @ save on the stack tmp10, tmp13, tmp12, tmp11
add r0, r3, r5 @ r0 = 'z2'
add r2, r1, r7 @ r2 = 'z1'
@ -275,8 +270,8 @@ column_loop:
add r3, r3, r4 @ r3 = tmp2
add r1, r1, r6 @ r1 = tmp3
ldmia sp!, { r0, r2, r4, r6 } @ r0 = tmp10 / r2 = tmp13 / r4 = tmp11 / r6 = tmp12
@ r1 = tmp3 / r3 = tmp2 / r5 = tmp1 / r7 = tmp0
pop {r0, r2, r4, r6} @ r0 = tmp10 / r2 = tmp13 / r4 = tmp11 / r6 = tmp12
@ r1 = tmp3 / r3 = tmp2 / r5 = tmp1 / r7 = tmp0
@ Compute DESCALE(tmp10 + tmp3, CONST_BITS+PASS1_BITS+3)
add r8, r0, r1
@ -368,11 +363,10 @@ empty_odd_column:
the_end:
@ The end....
add sp, sp, #4
ldmia sp!, { r4 - r12, pc } @ restore callee saved regs and return
pop {r4 - r11, pc}
endfunc
const_array:
.align
const const_array
.word FIX_0_298631336
.word FIX_0_541196100
.word FIX_0_765366865
@ -386,3 +380,4 @@ const_array:
.word FIX_M_1_961570560
.word FIX_M_2_562915447
.word FIX_0xFFFF
endconst

143
libavcodec/arm/mpegaudiodsp_fixed_armv6.S Normal file
View File

@ -0,0 +1,143 @@
/*
* Copyright (c) 2011 Mans Rullgard <mans@mansr.com>
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "asm.S"
.macro skip args:vararg
.endm
.macro sum8 lo, hi, w, p, t1, t2, t3, t4, rsb=skip, offs=0
ldr \t1, [\w, #4*\offs]
ldr \t2, [\p, #4]!
\rsb \t1, \t1, #0
.irpc i, 135
ldr \t3, [\w, #4*64*\i+4*\offs]
ldr \t4, [\p, #4*64*\i]
smlal \lo, \hi, \t1, \t2
\rsb \t3, \t3, #0
ldr \t1, [\w, #4*64*(\i+1)+4*\offs]
ldr \t2, [\p, #4*64*(\i+1)]
smlal \lo, \hi, \t3, \t4
\rsb \t1, \t1, #0
.endr
ldr \t3, [\w, #4*64*7+4*\offs]
ldr \t4, [\p, #4*64*7]
smlal \lo, \hi, \t1, \t2
\rsb \t3, \t3, #0
smlal \lo, \hi, \t3, \t4
.endm
.macro round rd, lo, hi
lsr \rd, \lo, #24
bic \lo, \lo, #0xff000000
orr \rd, \rd, \hi, lsl #8
mov \hi, #0
ssat \rd, #16, \rd
.endm
function ff_mpadsp_apply_window_fixed_armv6, export=1
push {r2,r4-r11,lr}
add r4, r0, #4*512 @ synth_buf + 512
.rept 4
ldm r0!, {r5-r12}
stm r4!, {r5-r12}
.endr
ldr r4, [sp, #40] @ incr
sub r0, r0, #4*17 @ synth_buf + 16
ldr r8, [r2] @ sum:low
add r2, r0, #4*32 @ synth_buf + 48
rsb r5, r4, r4, lsl #5 @ 31 * incr
lsl r4, r4, #1
asr r9, r8, #31 @ sum:high
add r5, r3, r5, lsl #1 @ samples2
add r6, r1, #4*32 @ w2
str r4, [sp, #40]
sum8 r8, r9, r1, r0, r10, r11, r12, lr
sum8 r8, r9, r1, r2, r10, r11, r12, lr, rsb, 32
round r10, r8, r9
strh r10, [r3], r4
mov lr, #15
1:
ldr r12, [r0, #4]!
ldr r11, [r6, #-4]!
ldr r10, [r1, #4]!
.irpc i, 0246
.if \i
ldr r11, [r6, #4*64*\i]
ldr r10, [r1, #4*64*\i]
.endif
rsb r11, r11, #0
smlal r8, r9, r10, r12
ldr r10, [r0, #4*64*(\i+1)]
.ifeq \i
smull r4, r7, r11, r12
.else
smlal r4, r7, r11, r12
.endif
ldr r11, [r6, #4*64*(\i+1)]
ldr r12, [r1, #4*64*(\i+1)]
rsb r11, r11, #0
smlal r8, r9, r12, r10
.iflt \i-6
ldr r12, [r0, #4*64*(\i+2)]
.else
ldr r12, [r2, #-4]!
.endif
smlal r4, r7, r11, r10
.endr
.irpc i, 0246
ldr r10, [r1, #4*64*\i+4*32]
rsb r12, r12, #0
ldr r11, [r6, #4*64*\i+4*32]
smlal r8, r9, r10, r12
ldr r10, [r2, #4*64*(\i+1)]
smlal r4, r7, r11, r12
ldr r12, [r1, #4*64*(\i+1)+4*32]
rsb r10, r10, #0
ldr r11, [r6, #4*64*(\i+1)+4*32]
smlal r8, r9, r12, r10
.iflt \i-6
ldr r12, [r2, #4*64*(\i+2)]
.else
ldr r12, [sp, #40]
.endif
smlal r4, r7, r11, r10
.endr
round r10, r8, r9
adds r8, r8, r4
adc r9, r9, r7
strh r10, [r3], r12
round r11, r8, r9
subs lr, lr, #1
strh r11, [r5], -r12
bgt 1b
sum8 r8, r9, r1, r0, r10, r11, r12, lr, rsb, 33
pop {r4}
round r10, r8, r9
str r8, [r4]
strh r10, [r3]
pop {r4-r11,pc}
endfunc

33
libavcodec/arm/mpegaudiodsp_init_arm.c Normal file
View File

@ -0,0 +1,33 @@
/*
* Copyright (c) 2011 Mans Rullgard
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <stdint.h>
#include "libavcodec/mpegaudiodsp.h"
#include "config.h"
void ff_mpadsp_apply_window_fixed_armv6(int32_t *synth_buf, int32_t *window,
int *dither, int16_t *out, int incr);
void ff_mpadsp_init_arm(MPADSPContext *s)
{
if (HAVE_ARMV6) {
s->apply_window_fixed = ff_mpadsp_apply_window_fixed_armv6;
}
}

24
libavcodec/eac3enc.c
View File

@ -28,6 +28,13 @@
#include "ac3enc.h"
#include "eac3enc.h"
#define AC3ENC_TYPE AC3ENC_TYPE_EAC3
#include "ac3enc_opts_template.c"
static AVClass eac3enc_class = { "E-AC-3 Encoder", av_default_item_name,
eac3_options, LIBAVUTIL_VERSION_INT };
void ff_eac3_set_cpl_states(AC3EncodeContext *s)
{
int ch, blk;
@ -129,3 +136,20 @@ void ff_eac3_output_frame_header(AC3EncodeContext *s)
/* block start info */
put_bits(&s->pb, 1, 0);
}
#if CONFIG_EAC3_ENCODER
AVCodec ff_eac3_encoder = {
.name = "eac3",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_EAC3,
.priv_data_size = sizeof(AC3EncodeContext),
.init = ff_ac3_encode_init,
.encode = ff_ac3_encode_frame,
.close = ff_ac3_encode_close,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52 E-AC-3"),
.priv_class = &eac3enc_class,
.channel_layouts = ff_ac3_channel_layouts,
};
#endif

22
libavcodec/h264.c
View File

@ -995,7 +995,7 @@ int ff_h264_decode_extradata(H264Context *h)
cnt = *(p++); // Number of pps
for (i = 0; i < cnt; i++) {
nalsize = AV_RB16(p) + 2;
if(decode_nal_units(h, p, nalsize) < 0) {
if (decode_nal_units(h, p, nalsize) < 0) {
av_log(avctx, AV_LOG_ERROR, "Decoding pps %d from avcC failed\n", i);
return -1;
}
@ -2351,8 +2351,8 @@ static int decode_slice_header(H264Context *h, H264Context *h0){
MPV_common_end(s);
}
if (!s->context_initialized) {
if(h != h0){
av_log(h->s.avctx, AV_LOG_ERROR, "we cant (re-)initialize context during parallel decoding\n");
if (h != h0) {
av_log(h->s.avctx, AV_LOG_ERROR, "Cannot (re-)initialize context during parallel decoding.\n");
return -1;
}
@ -2398,8 +2398,8 @@ static int decode_slice_header(H264Context *h, H264Context *h0){
s->avctx->hwaccel = ff_find_hwaccel(s->avctx->codec->id, s->avctx->pix_fmt);
if (MPV_common_init(s) < 0){
av_log(h->s.avctx, AV_LOG_ERROR, "MPV_common_init() failed\n");
if (MPV_common_init(s) < 0) {
av_log(h->s.avctx, AV_LOG_ERROR, "MPV_common_init() failed.\n");
return -1;
}
s->first_field = 0;
@ -2409,8 +2409,8 @@ static int decode_slice_header(H264Context *h, H264Context *h0){
ff_h264_alloc_tables(h);
if (!HAVE_THREADS || !(s->avctx->active_thread_type&FF_THREAD_SLICE)) {
if (context_init(h) < 0){
av_log(h->s.avctx, AV_LOG_ERROR, "context_init() failed\n");
if (context_init(h) < 0) {
av_log(h->s.avctx, AV_LOG_ERROR, "context_init() failed.\n");
return -1;
}
} else {
@ -2428,8 +2428,8 @@ static int decode_slice_header(H264Context *h, H264Context *h0){
}
for(i = 0; i < s->avctx->thread_count; i++)
if(context_init(h->thread_context[i]) < 0){
av_log(h->s.avctx, AV_LOG_ERROR, "context_init() failed\n");
if (context_init(h->thread_context[i]) < 0) {
av_log(h->s.avctx, AV_LOG_ERROR, "context_init() failed.\n");
return -1;
}
}
@ -2737,8 +2737,8 @@ static int decode_slice_header(H264Context *h, H264Context *h0){
av_log(s->avctx, AV_LOG_INFO, "Cannot parallelize deblocking type 1, decoding such frames in sequential order\n");
h0->single_decode_warning = 1;
}
if(h != h0){
av_log(h->s.avctx, AV_LOG_ERROR, "deblocking switched inside frame\n");
if (h != h0) {
av_log(h->s.avctx, AV_LOG_ERROR, "Deblocking switched inside frame.\n");
return 1;
}
}

1
libavcodec/mpegaudiodsp.c
View File

@ -35,6 +35,7 @@ void ff_mpadsp_init(MPADSPContext *s)
s->dct32_float = dct.dct32;
s->dct32_fixed = ff_dct32_fixed;
if (ARCH_ARM) ff_mpadsp_init_arm(s);
if (HAVE_MMX) ff_mpadsp_init_mmx(s);
if (HAVE_ALTIVEC) ff_mpadsp_init_altivec(s);
}

1
libavcodec/mpegaudiodsp.h
View File

@ -47,6 +47,7 @@ void ff_mpa_synth_filter_float(MPADSPContext *s,
float *samples, int incr,
float *sb_samples);
void ff_mpadsp_init_arm(MPADSPContext *s);
void ff_mpadsp_init_mmx(MPADSPContext *s);
void ff_mpadsp_init_altivec(MPADSPContext *s);

2
libavformat/utils.c
View File

@ -1605,14 +1605,12 @@ int64_t av_gen_search(AVFormatContext *s, int stream_index, int64_t target_ts, i
pos = (flags & AVSEEK_FLAG_BACKWARD) ? pos_min : pos_max;
ts = (flags & AVSEEK_FLAG_BACKWARD) ? ts_min : ts_max;
#if 1
pos_min = pos;
ts_min = read_timestamp(s, stream_index, &pos_min, INT64_MAX);
pos_min++;
ts_max = read_timestamp(s, stream_index, &pos_min, INT64_MAX);
av_dlog(s, "pos=0x%"PRIx64" %"PRId64"<=%"PRId64"<=%"PRId64"\n",
pos, ts_min, target_ts, ts_max);
#endif
*ts_ret= ts;
return pos;
}