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authorMichael Niedermayer <michaelni@gmx.at>2011-06-14 04:55:27 +0200
committerMichael Niedermayer <michaelni@gmx.at>2011-06-14 04:56:26 +0200
commit173cd695cbb79a50a0738ce7bcc966cb40f4a28a (patch)
tree1a8025d98e71b5950eb12ed24c2c8787a5c185e3 /libavcodec
parentfdb5e02901111a6a53f8386d82afae0aa2d746a7 (diff)
parent35bdaf3d427b6856df01d41ee826bd515440ec46 (diff)
downloadffmpeg-173cd695cbb79a50a0738ce7bcc966cb40f4a28a.tar.gz
Merge remote-tracking branch 'qatar/master'
* 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>
Diffstat (limited to 'libavcodec')
-rw-r--r--libavcodec/Makefile8
-rw-r--r--libavcodec/ac3dec.h2
-rw-r--r--libavcodec/ac3enc.c456
-rw-r--r--libavcodec/ac3enc.h83
-rw-r--r--libavcodec/ac3enc_fixed.c38
-rw-r--r--libavcodec/ac3enc_float.c58
-rw-r--r--libavcodec/ac3enc_opts_template.c3
-rw-r--r--libavcodec/ac3enc_template.c377
-rw-r--r--libavcodec/arm/Makefile3
-rw-r--r--libavcodec/arm/jrevdct_arm.S31
-rw-r--r--libavcodec/arm/mpegaudiodsp_fixed_armv6.S143
-rw-r--r--libavcodec/arm/mpegaudiodsp_init_arm.c33
-rw-r--r--libavcodec/eac3enc.c24
-rw-r--r--libavcodec/h264.c22
-rw-r--r--libavcodec/mpegaudiodsp.c1
-rw-r--r--libavcodec/mpegaudiodsp.h1
16 files changed, 792 insertions, 491 deletions
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index aa091bf2e5..f4613749a0 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -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 \
diff --git a/libavcodec/ac3dec.h b/libavcodec/ac3dec.h
index 10695b7b20..377e5154d7 100644
--- a/libavcodec/ac3dec.h
+++ b/libavcodec/ac3dec.h
@@ -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
diff --git a/libavcodec/ac3enc.c b/libavcodec/ac3enc.c
index 4f153191a0..f61e7f8c4d 100644
--- a/libavcodec/ac3enc.c
+++ b/libavcodec/ac3enc.c
@@ -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;
@@ -349,296 +252,6 @@ 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.
*/
static void apply_rematrixing(AC3EncodeContext *s)
@@ -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;
}
diff --git a/libavcodec/ac3enc.h b/libavcodec/ac3enc.h
index d1f5548297..d5f662b4aa 100644
--- a/libavcodec/ac3enc.h
+++ b/libavcodec/ac3enc.h
@@ -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 */
diff --git a/libavcodec/ac3enc_fixed.c b/libavcodec/ac3enc_fixed.c
index 6b1ee88c9f..f4d447e3b2 100644
--- a/libavcodec/ac3enc_fixed.c
+++ b/libavcodec/ac3enc_fixed.c
@@ -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)"),
diff --git a/libavcodec/ac3enc_float.c b/libavcodec/ac3enc_float.c
index 2ab24db561..9e798106f3 100644
--- a/libavcodec/ac3enc_float.c
+++ b/libavcodec/ac3enc_float.c
@@ -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
diff --git a/libavcodec/ac3enc_opts_template.c b/libavcodec/ac3enc_opts_template.c
index e16e0d0878..39138a1083 100644
--- a/libavcodec/ac3enc_opts_template.c
+++ b/libavcodec/ac3enc_opts_template.c
@@ -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
diff --git a/libavcodec/ac3enc_template.c b/libavcodec/ac3enc_template.c
new file mode 100644
index 0000000000..d88fa225a1
--- /dev/null
+++ b/libavcodec/ac3enc_template.c
@@ -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;
+ }
+}
diff --git a/libavcodec/arm/Makefile b/libavcodec/arm/Makefile
index a5abfdd128..3374f0e2bd 100644
--- a/libavcodec/arm/Makefile
+++ b/libavcodec/arm/Makefile
@@ -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
diff --git a/libavcodec/arm/jrevdct_arm.S b/libavcodec/arm/jrevdct_arm.S
index 4fcf35101d..93cbbbe8eb 100644
--- a/libavcodec/arm/jrevdct_arm.S
+++ b/libavcodec/arm/jrevdct_arm.S
@@ -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
diff --git a/libavcodec/arm/mpegaudiodsp_fixed_armv6.S b/libavcodec/arm/mpegaudiodsp_fixed_armv6.S
new file mode 100644
index 0000000000..9ec731480b
--- /dev/null
+++ b/libavcodec/arm/mpegaudiodsp_fixed_armv6.S
@@ -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
diff --git a/libavcodec/arm/mpegaudiodsp_init_arm.c b/libavcodec/arm/mpegaudiodsp_init_arm.c
new file mode 100644
index 0000000000..94a55787ad
--- /dev/null
+++ b/libavcodec/arm/mpegaudiodsp_init_arm.c
@@ -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;
+ }
+}
diff --git a/libavcodec/eac3enc.c b/libavcodec/eac3enc.c
index 20f4b879c6..d37acaf20b 100644
--- a/libavcodec/eac3enc.c
+++ b/libavcodec/eac3enc.c
@@ -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
diff --git a/libavcodec/h264.c b/libavcodec/h264.c
index 4da2807663..5c812144c9 100644
--- a/libavcodec/h264.c
+++ b/libavcodec/h264.c
@@ -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;
}
}
diff --git a/libavcodec/mpegaudiodsp.c b/libavcodec/mpegaudiodsp.c
index 064acd1e74..d98d25bb21 100644
--- a/libavcodec/mpegaudiodsp.c
+++ b/libavcodec/mpegaudiodsp.c
@@ -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);
}
diff --git a/libavcodec/mpegaudiodsp.h b/libavcodec/mpegaudiodsp.h
index a47019cc4b..8a18db8325 100644
--- a/libavcodec/mpegaudiodsp.h
+++ b/libavcodec/mpegaudiodsp.h
@@ -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);