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ac3enc_fixed: convert to 32-bit sample format
The AC3 encoder used to be a separate library called "Aften", which got merged into libavcodec (literally, SVN commits and all). The merge preserved as much features from the library as possible. The code had two versions - a fixed point version and a floating point version. FFmpeg had floating point DSP code used by other codecs, the AC3 decoder including, so the floating-point DSP was simply replaced with FFmpeg's own functions. However, FFmpeg had no fixed-point audio code at that point. So the encoder brought along its own fixed-point DSP functions, including a fixed-point MDCT. The fixed-point MDCT itself is trivially just a float MDCT with a different type and each multiply being a fixed-point multiply. So over time, it got refactored, and the FFT used for all other codecs was templated. Due to design decisions at the time, the fixed-point version of the encoder operates at 16-bits of precision. Although convenient, this, even at the time, was inadequate and inefficient. The encoder is noisy, does not produce output comparable to the float encoder, and even rings at higher frequencies due to the badly approximated winow function. Enter MIPS (owned by Imagination Technologies at the time). They wanted quick fixed-point decoding on their FPUless cores. So they contributed patches to template the AC3 decoder so it had both a fixed-point and a floating-point version. They also did the same for the AAC decoder. They however, used 32-bit samples. Not 16-bits. And we did not have 32-bit fixed-point DSP functions, including an MDCT. But instead of templating our MDCT to output 3 versions (float, 32-bit fixed and 16-bit fixed), they simply copy-pasted their own MDCT into ours, and completely ifdeffed our own MDCT code out if a 32-bit fixed point MDCT was selected. This is also the status quo nowadays - 2 separate MDCTs, one which produces floating point and 16-bit fixed point versions, and one sort-of integrated which produces 32-bit MDCT. MIPS weren't all that interested in encoding, so they left the encoder as-is, and they didn't care much about the ifdeffery, mess or quality - it's not their problem. So the MDCT/FFT code has always been a thorn in anyone looking to clean up code's eye. Backstory over. Internally AC3 operates on 25-bit fixed-point coefficients. So for the floating point version, the encoder simply runs the float MDCT, and converts the resulting coefficients to 25-bit fixed-point, as AC3 is inherently a fixed-point codec. For the fixed-point version, the input is 16-bit samples, so to maximize precision the frame samples are analyzed and the highest set bit is detected via ac3_max_msb_abs_int16(), and the coefficients are then scaled up via ac3_lshift_int16(), so the input for the FFT is always at least 14 bits, computed in normalize_samples(). After FFT, the coefficients are scaled up to 25 bits. This patch simply changes the encoder to accept 32-bit samples, reusing the already well-optimized 32-bit MDCT code, allowing us to clean up and drop a large part of a very messy code of ours, as well as prepare for the future lavu/tx conversion. The coefficients are simply scaled down to 25 bits during windowing, skipping 2 separate scalings, as the hacks to extend precision are simply no longer necessary. There's no point in running the MDCT always at 32 bits when you're going to drop 6 bits off anyway, the headroom is plenty, and the MDCT rounds properly. This also makes the encoder even slightly more accurate over the float version, as there's no coefficient conversion step necessary. SIZE SAVINGS: ARM32: HARDCODED TABLES: BASE - 10709590 DROP DSP - 10702872 - diff: -6.56KiB DROP MDCT - 10667932 - diff: -34.12KiB - both: -40.68KiB DROP FFT - 10336652 - diff: -323.52KiB - all: -364.20KiB SOFTCODED TABLES: BASE - 9685096 DROP DSP - 9678378 - diff: -6.56KiB DROP MDCT - 9643466 - diff: -34.09KiB - both: -40.65KiB DROP FFT - 9573918 - diff: -67.92KiB - all: -108.57KiB ARM64: HARDCODED TABLES: BASE - 14641112 DROP DSP - 14633806 - diff: -7.13KiB DROP MDCT - 14604812 - diff: -28.31KiB - both: -35.45KiB DROP FFT - 14286826 - diff: -310.53KiB - all: -345.98KiB SOFTCODED TABLES: BASE - 13636238 DROP DSP - 13628932 - diff: -7.13KiB DROP MDCT - 13599866 - diff: -28.38KiB - both: -35.52KiB DROP FFT - 13542080 - diff: -56.43KiB - all: -91.95KiB x86: HARDCODED TABLES: BASE - 12367336 DROP DSP - 12354698 - diff: -12.34KiB DROP MDCT - 12331024 - diff: -23.12KiB - both: -35.46KiB DROP FFT - 12029788 - diff: -294.18KiB - all: -329.64KiB SOFTCODED TABLES: BASE - 11358094 DROP DSP - 11345456 - diff: -12.34KiB DROP MDCT - 11321742 - diff: -23.16KiB - both: -35.50KiB DROP FFT - 11276946 - diff: -43.75KiB - all: -79.25KiB PERFORMANCE (10min random s32le): ARM32 - before - 39.9x - 0m15.046s ARM32 - after - 28.2x - 0m21.525s Speed: -30% ARM64 - before - 36.1x - 0m16.637s ARM64 - after - 36.0x - 0m16.727s Speed: -0.5% x86 - before - 184x - 0m3.277s x86 - after - 190x - 0m3.187s Speed: +3%
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@ -151,10 +151,9 @@ the undocumented RealAudio 3 (a.k.a. dnet).
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The @var{ac3} encoder uses floating-point math, while the @var{ac3_fixed}
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encoder only uses fixed-point integer math. This does not mean that one is
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always faster, just that one or the other may be better suited to a
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particular system. The floating-point encoder will generally produce better
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quality audio for a given bitrate. The @var{ac3_fixed} encoder is not the
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default codec for any of the output formats, so it must be specified explicitly
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using the option @code{-acodec ac3_fixed} in order to use it.
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particular system. The @var{ac3_fixed} encoder is not the default codec for
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any of the output formats, so it must be specified explicitly using the option
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@code{-acodec ac3_fixed} in order to use it.
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@subsection AC-3 Metadata
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@ -181,7 +181,7 @@ OBJS-$(CONFIG_AC3_DECODER) += ac3dec_float.o ac3dec_data.o ac3.o kbd
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OBJS-$(CONFIG_AC3_FIXED_DECODER) += ac3dec_fixed.o ac3dec_data.o ac3.o kbdwin.o ac3tab.o
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OBJS-$(CONFIG_AC3_ENCODER) += ac3enc_float.o ac3enc.o ac3tab.o \
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ac3.o kbdwin.o
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OBJS-$(CONFIG_AC3_FIXED_ENCODER) += ac3enc_fixed.o ac3enc.o ac3tab.o ac3.o
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OBJS-$(CONFIG_AC3_FIXED_ENCODER) += ac3enc_fixed.o ac3enc.o ac3tab.o ac3.o kbdwin.o
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OBJS-$(CONFIG_AC3_MF_ENCODER) += mfenc.o mf_utils.o
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OBJS-$(CONFIG_ACELP_KELVIN_DECODER) += g729dec.o lsp.o celp_math.o celp_filters.o acelp_filters.o acelp_pitch_delay.o acelp_vectors.o g729postfilter.o
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OBJS-$(CONFIG_AGM_DECODER) += agm.o
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@ -2047,6 +2047,7 @@ av_cold int ff_ac3_encode_close(AVCodecContext *avctx)
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int blk, ch;
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AC3EncodeContext *s = avctx->priv_data;
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av_freep(&s->mdct_window);
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av_freep(&s->windowed_samples);
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if (s->planar_samples)
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for (ch = 0; ch < s->channels; ch++)
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@ -30,8 +30,6 @@
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#include <stdint.h>
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#include "libavutil/float_dsp.h"
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#include "ac3.h"
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#include "ac3dsp.h"
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#include "avcodec.h"
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@ -53,6 +51,7 @@
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#define AC3ENC_TYPE_EAC3 2
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#if AC3ENC_FLOAT
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#include "libavutil/float_dsp.h"
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#define AC3_NAME(x) ff_ac3_float_ ## x
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#define MAC_COEF(d,a,b) ((d)+=(a)*(b))
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#define COEF_MIN (-16777215.0/16777216.0)
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@ -62,12 +61,13 @@ typedef float SampleType;
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typedef float CoefType;
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typedef float CoefSumType;
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#else
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#include "libavutil/fixed_dsp.h"
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#define AC3_NAME(x) ff_ac3_fixed_ ## x
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#define MAC_COEF(d,a,b) MAC64(d,a,b)
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#define COEF_MIN -16777215
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#define COEF_MAX 16777215
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#define NEW_CPL_COORD_THRESHOLD 503317
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typedef int16_t SampleType;
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typedef int32_t SampleType;
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typedef int32_t CoefType;
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typedef int64_t CoefSumType;
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#endif
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@ -141,7 +141,6 @@ typedef struct AC3Block {
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uint16_t **qmant; ///< quantized mantissas
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uint8_t **cpl_coord_exp; ///< coupling coord exponents (cplcoexp)
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uint8_t **cpl_coord_mant; ///< coupling coord mantissas (cplcomant)
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uint8_t coeff_shift[AC3_MAX_CHANNELS]; ///< fixed-point coefficient shift values
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uint8_t new_rematrixing_strategy; ///< send new rematrixing flags in this block
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int num_rematrixing_bands; ///< number of rematrixing bands
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uint8_t rematrixing_flags[4]; ///< rematrixing flags
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@ -165,7 +164,11 @@ typedef struct AC3EncodeContext {
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AVCodecContext *avctx; ///< parent AVCodecContext
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PutBitContext pb; ///< bitstream writer context
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AudioDSPContext adsp;
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#if AC3ENC_FLOAT
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AVFloatDSPContext *fdsp;
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#else
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AVFixedDSPContext *fdsp;
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#endif
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MECmpContext mecc;
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AC3DSPContext ac3dsp; ///< AC-3 optimized functions
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FFTContext mdct; ///< FFT context for MDCT calculation
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@ -26,12 +26,14 @@
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* fixed-point AC-3 encoder.
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*/
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#define FFT_FLOAT 0
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#define AC3ENC_FLOAT 0
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#define FFT_FLOAT 0
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#define FFT_FIXED_32 1
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#include "internal.h"
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#include "audiodsp.h"
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#include "ac3enc.h"
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#include "eac3enc.h"
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#include "kbdwin.h"
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#define AC3ENC_TYPE AC3ENC_TYPE_AC3_FIXED
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#include "ac3enc_opts_template.c"
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@ -43,37 +45,6 @@ static const AVClass ac3enc_class = {
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.version = LIBAVUTIL_VERSION_INT,
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};
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/*
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* Normalize the input samples to use the maximum available precision.
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* This assumes signed 16-bit input samples.
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*/
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static int normalize_samples(AC3EncodeContext *s)
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{
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int v = s->ac3dsp.ac3_max_msb_abs_int16(s->windowed_samples, AC3_WINDOW_SIZE);
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v = 14 - av_log2(v);
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if (v > 0)
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s->ac3dsp.ac3_lshift_int16(s->windowed_samples, AC3_WINDOW_SIZE, v);
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/* +6 to right-shift from 31-bit to 25-bit */
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return v + 6;
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}
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/*
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* Scale MDCT coefficients to 25-bit signed fixed-point.
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*/
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static void scale_coefficients(AC3EncodeContext *s)
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{
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int blk, ch;
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for (blk = 0; blk < s->num_blocks; blk++) {
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AC3Block *block = &s->blocks[blk];
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for (ch = 1; ch <= s->channels; ch++) {
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s->ac3dsp.ac3_rshift_int32(block->mdct_coef[ch], AC3_MAX_COEFS,
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block->coeff_shift[ch]);
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}
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}
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}
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static void sum_square_butterfly(AC3EncodeContext *s, int64_t sum[4],
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const int32_t *coef0, const int32_t *coef1,
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int len)
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@ -120,7 +91,6 @@ static av_cold void ac3_fixed_mdct_end(AC3EncodeContext *s)
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ff_mdct_end(&s->mdct);
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}
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/**
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* Initialize MDCT tables.
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*
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@ -129,9 +99,25 @@ static av_cold void ac3_fixed_mdct_end(AC3EncodeContext *s)
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*/
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static av_cold int ac3_fixed_mdct_init(AC3EncodeContext *s)
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{
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int ret = ff_mdct_init(&s->mdct, 9, 0, -1.0);
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s->mdct_window = ff_ac3_window;
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return ret;
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float fwin[AC3_BLOCK_SIZE];
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int32_t *iwin = av_malloc_array(AC3_WINDOW_SIZE, sizeof(*iwin));
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if (!iwin)
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return AVERROR(ENOMEM);
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ff_kbd_window_init(fwin, 5.0, AC3_WINDOW_SIZE/2);
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for (int i = 0; i < AC3_WINDOW_SIZE/2; i++) {
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iwin[i] = lrintf(fwin[i] * (1 << 22));
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iwin[AC3_WINDOW_SIZE-1-i] = lrintf(fwin[i] * (1 << 22));
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}
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s->mdct_window = iwin;
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s->fdsp = avpriv_alloc_fixed_dsp(s->avctx->flags & AV_CODEC_FLAG_BITEXACT);
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if (!s->fdsp)
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return AVERROR(ENOMEM);
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return ff_mdct_init(&s->mdct, 9, 0, -1.0);
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}
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@ -155,7 +141,7 @@ AVCodec ff_ac3_fixed_encoder = {
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.init = ac3_fixed_encode_init,
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.encode2 = ff_ac3_fixed_encode_frame,
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.close = ff_ac3_encode_close,
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.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16P,
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.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32P,
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AV_SAMPLE_FMT_NONE },
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.priv_class = &ac3enc_class,
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.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
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static av_cold void ac3_float_mdct_end(AC3EncodeContext *s)
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{
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ff_mdct_end(&s->mdct);
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av_freep(&s->mdct_window);
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}
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AC3Block *block = &s->blocks[blk];
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const SampleType *input_samples = &s->planar_samples[ch][blk * AC3_BLOCK_SIZE];
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#if AC3ENC_FLOAT
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s->fdsp->vector_fmul(s->windowed_samples, input_samples,
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s->mdct_window, AC3_WINDOW_SIZE);
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#else
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s->ac3dsp.apply_window_int16(s->windowed_samples, input_samples,
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s->mdct_window, AC3_WINDOW_SIZE);
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s->mdct_window, AC3_WINDOW_SIZE);
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block->coeff_shift[ch + 1] = normalize_samples(s);
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#endif
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s->mdct.mdct_calcw(&s->mdct, block->mdct_coef[ch+1],
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s->windowed_samples);
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s->mdct.mdct_calc(&s->mdct, block->mdct_coef[ch+1],
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s->windowed_samples);
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}
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}
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}
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@ -390,9 +383,6 @@ int AC3_NAME(encode_frame)(AVCodecContext *avctx, AVPacket *avpkt,
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apply_mdct(s);
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if (!AC3ENC_FLOAT)
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scale_coefficients(s);
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clip_coefficients(&s->adsp, s->blocks[0].mdct_coef[1],
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AC3_MAX_COEFS * s->num_blocks * s->channels);
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@ -404,8 +394,9 @@ int AC3_NAME(encode_frame)(AVCodecContext *avctx, AVPacket *avpkt,
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compute_rematrixing_strategy(s);
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if (AC3ENC_FLOAT)
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scale_coefficients(s);
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#if AC3ENC_FLOAT
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scale_coefficients(s);
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#endif
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return ff_ac3_encode_frame_common_end(avctx, avpkt, frame, got_packet_ptr);
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}
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#include "libavutil/version.h"
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#define LIBAVCODEC_VERSION_MAJOR 58
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#define LIBAVCODEC_VERSION_MINOR 116
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#define LIBAVCODEC_VERSION_MINOR 117
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#define LIBAVCODEC_VERSION_MICRO 100
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#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
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fate-ac3-fixed-encode: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
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fate-ac3-fixed-encode: CMD = md5 -i $(SRC) -c ac3_fixed -ab 128k -f ac3 -flags +bitexact -af aresample
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fate-ac3-fixed-encode: CMP = oneline
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fate-ac3-fixed-encode: REF = a1d1fc116463b771abf5aef7ed37d7b1
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fate-ac3-fixed-encode: REF = 1f548175e11a95e62ce20e442fcc8d08
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FATE_EAC3-$(call ALLYES, EAC3_DEMUXER EAC3_MUXER EAC3_CORE_BSF) += fate-eac3-core-bsf
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fate-eac3-core-bsf: CMD = md5pipe -i $(TARGET_SAMPLES)/eac3/the_great_wall_7.1.eac3 -c:a copy -bsf:a eac3_core -fflags +bitexact -f eac3
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FATE_FFMPEG-$(call ALLYES, PCM_S16LE_DEMUXER AC3_MUXER PCM_S16LE_DECODER AC3_FIXED_ENCODER) += fate-unknown_layout-ac3
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fate-unknown_layout-ac3: $(AREF)
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fate-unknown_layout-ac3: CMD = md5 -auto_conversion_filters \
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-guess_layout_max 0 -f s16le -ac 1 -ar 44100 -i $(TARGET_PATH)/$(AREF) \
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-guess_layout_max 0 -f s32le -ac 1 -ar 44100 -i $(TARGET_PATH)/$(AREF) \
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-f ac3 -flags +bitexact -c ac3_fixed
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@ -1 +1 @@
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bbb7550d6d93973c10f4ee13c87cf799
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febdb165cfd6cba375aa086195e61213
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@ -1,2 +1,2 @@
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e30681d05d6f3d24108d3614600bf116 *tests/data/lavf/lavf.rm
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8dfb8d4556d61d3615e0d0012ffe540c *tests/data/lavf/lavf.rm
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346424 tests/data/lavf/lavf.rm
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