//#define DEBUG
//#define ASSERT_LEVEL 2
+#include <stdint.h>
+
#include "libavutil/audioconvert.h"
#include "libavutil/avassert.h"
#include "libavutil/crc.h"
#include "ac3dsp.h"
#include "ac3.h"
#include "audioconvert.h"
+#include "fft.h"
#ifndef CONFIG_AC3ENC_FLOAT
#define AC3_REMATRIXING_NONE 1
#define AC3_REMATRIXING_ALWAYS 3
-/** Scale a float value by 2^bits and convert to an integer. */
-#define SCALE_FLOAT(a, bits) lrintf((a) * (float)(1 << (bits)))
-
-
#if CONFIG_AC3ENC_FLOAT
-#include "ac3enc_float.h"
+#define MAC_COEF(d,a,b) ((d)+=(a)*(b))
+typedef float SampleType;
+typedef float CoefType;
+typedef float CoefSumType;
#else
-#include "ac3enc_fixed.h"
+#define MAC_COEF(d,a,b) MAC64(d,a,b)
+typedef int16_t SampleType;
+typedef int32_t CoefType;
+typedef int64_t CoefSumType;
#endif
-
-/**
- * Encoding Options used by AVOption.
- */
-typedef struct AC3EncOptions {
- /* AC-3 metadata options*/
- int dialogue_level;
- int bitstream_mode;
- float center_mix_level;
- float surround_mix_level;
- int dolby_surround_mode;
- int audio_production_info;
- int mixing_level;
- int room_type;
- int copyright;
- int original;
- int extended_bsi_1;
- int preferred_stereo_downmix;
- float ltrt_center_mix_level;
- float ltrt_surround_mix_level;
- float loro_center_mix_level;
- float loro_surround_mix_level;
- int extended_bsi_2;
- int dolby_surround_ex_mode;
- int dolby_headphone_mode;
- int ad_converter_type;
-
- /* other encoding options */
- int allow_per_frame_metadata;
-} AC3EncOptions;
+typedef struct AC3MDCTContext {
+ const SampleType *window; ///< MDCT window function
+ FFTContext fft; ///< FFT context for MDCT calculation
+} AC3MDCTContext;
/**
* Data for a single audio block.
uint8_t exp_strategy[AC3_MAX_CHANNELS][AC3_MAX_BLOCKS]; ///< exponent strategies
- DECLARE_ALIGNED(16, SampleType, windowed_samples)[AC3_WINDOW_SIZE];
+ DECLARE_ALIGNED(32, SampleType, windowed_samples)[AC3_WINDOW_SIZE];
} AC3EncodeContext;
typedef struct AC3Mant {
#define OFFSET(param) offsetof(AC3EncodeContext, options.param)
#define AC3ENC_PARAM (AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM)
-static const AVOption options[] = {
+#if CONFIG_AC3ENC_FLOAT || !CONFIG_AC3_FLOAT_ENCODER //we need this exactly once compiled in
+const AVOption ff_ac3_options[] = {
/* Metadata Options */
{"per_frame_metadata", "Allow Changing Metadata Per-Frame", OFFSET(allow_per_frame_metadata), FF_OPT_TYPE_INT, 0, 0, 1, AC3ENC_PARAM},
/* downmix levels */
{"hdcd", "HDCD", 0, FF_OPT_TYPE_CONST, 1, INT_MIN, INT_MAX, AC3ENC_PARAM, "ad_conv_type"},
{NULL}
};
+#endif
#if CONFIG_AC3ENC_FLOAT
static AVClass ac3enc_class = { "AC-3 Encoder", av_default_item_name,
- options, LIBAVUTIL_VERSION_INT };
+ ff_ac3_options, LIBAVUTIL_VERSION_INT };
#else
static AVClass ac3enc_class = { "Fixed-Point AC-3 Encoder", av_default_item_name,
- options, LIBAVUTIL_VERSION_INT };
+ ff_ac3_options, LIBAVUTIL_VERSION_INT };
#endif
static av_cold int mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct,
int nbits);
-static void mdct512(AC3MDCTContext *mdct, CoefType *out, SampleType *in);
-
static void apply_window(DSPContext *dsp, SampleType *output, const SampleType *input,
const SampleType *window, unsigned int len);
/**
* List of supported channel layouts.
*/
-static const int64_t ac3_channel_layouts[] = {
+#if CONFIG_AC3ENC_FLOAT || !CONFIG_AC3_FLOAT_ENCODER //we need this exactly once compiled in
+const int64_t ff_ac3_channel_layouts[] = {
AV_CH_LAYOUT_MONO,
AV_CH_LAYOUT_STEREO,
AV_CH_LAYOUT_2_1,
AV_CH_LAYOUT_5POINT1_BACK,
0
};
+#endif
+
+
+/**
+ * LUT to select the bandwidth code based on the bit rate, sample rate, and
+ * number of full-bandwidth channels.
+ * bandwidth_tab[fbw_channels-1][sample rate code][bit rate code]
+ */
+static const uint8_t ac3_bandwidth_tab[5][3][19] = {
+// 32 40 48 56 64 80 96 112 128 160 192 224 256 320 384 448 512 576 640
+
+ { { 0, 0, 0, 12, 16, 32, 48, 48, 48, 48, 48, 48, 48, 48, 48, 48, 48, 48, 48 },
+ { 0, 0, 0, 16, 20, 36, 56, 56, 56, 56, 56, 56, 56, 56, 56, 56, 56, 56, 56 },
+ { 0, 0, 0, 32, 40, 60, 60, 60, 60, 60, 60, 60, 60, 60, 60, 60, 60, 60, 60 } },
+
+ { { 0, 0, 0, 0, 0, 0, 0, 20, 24, 32, 48, 48, 48, 48, 48, 48, 48, 48, 48 },
+ { 0, 0, 0, 0, 0, 0, 4, 24, 28, 36, 56, 56, 56, 56, 56, 56, 56, 56, 56 },
+ { 0, 0, 0, 0, 0, 0, 20, 44, 52, 60, 60, 60, 60, 60, 60, 60, 60, 60, 60 } },
+
+ { { 0, 0, 0, 0, 0, 0, 0, 0, 0, 16, 24, 32, 40, 48, 48, 48, 48, 48, 48 },
+ { 0, 0, 0, 0, 0, 0, 0, 0, 4, 20, 28, 36, 44, 56, 56, 56, 56, 56, 56 },
+ { 0, 0, 0, 0, 0, 0, 0, 0, 20, 40, 48, 60, 60, 60, 60, 60, 60, 60, 60 } },
+
+ { { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 12, 24, 32, 48, 48, 48, 48, 48, 48 },
+ { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 16, 28, 36, 56, 56, 56, 56, 56, 56 },
+ { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 32, 48, 60, 60, 60, 60, 60, 60, 60 } },
+
+ { { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 8, 20, 32, 40, 48, 48, 48, 48 },
+ { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 12, 24, 36, 44, 56, 56, 56, 56 },
+ { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 28, 44, 60, 60, 60, 60, 60, 60 } }
+};
/**
block->coeff_shift[ch] = normalize_samples(s);
- mdct512(&s->mdct, block->mdct_coef[ch], s->windowed_samples);
+ s->mdct.fft.mdct_calcw(&s->mdct.fft, block->mdct_coef[ch],
+ s->windowed_samples);
}
}
}
*/
static void extract_exponents(AC3EncodeContext *s)
{
- int blk, ch, i;
+ int blk, ch;
for (ch = 0; ch < s->channels; ch++) {
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
AC3Block *block = &s->blocks[blk];
- uint8_t *exp = block->exp[ch];
- int32_t *coef = block->fixed_coef[ch];
- for (i = 0; i < AC3_MAX_COEFS; i++) {
- int e;
- int v = abs(coef[i]);
- if (v == 0)
- e = 24;
- else {
- e = 23 - av_log2(v);
- if (e >= 24) {
- e = 24;
- coef[i] = 0;
- }
- av_assert2(e >= 0);
- }
- exp[i] = e;
- }
+ s->ac3dsp.extract_exponents(block->exp[ch], block->fixed_coef[ch],
+ AC3_MAX_COEFS);
}
}
}
bw_code = av_clip((fbw_coeffs - 73) / 3, 0, 60);
} else {
/* use default bandwidth setting */
- /* XXX: should compute the bandwidth according to the frame
- size, so that we avoid annoying high frequency artifacts */
- bw_code = 50;
+ bw_code = ac3_bandwidth_tab[s->fbw_channels-1][s->bit_alloc.sr_code][s->frame_size_code/2];
}
/* set number of coefficients for each channel */
projects / ffmpeg / blobdiff