From patchwork Sat Jan 9 19:19:17 2021 Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: 8bit X-Patchwork-Submitter: Lynne X-Patchwork-Id: 24867 Return-Path: X-Original-To: patchwork@ffaux-bg.ffmpeg.org Delivered-To: patchwork@ffaux-bg.ffmpeg.org Received: from ffbox0-bg.mplayerhq.hu (ffbox0-bg.ffmpeg.org [79.124.17.100]) by ffaux.localdomain (Postfix) with ESMTP id DF4A044ABA6 for ; Sat, 9 Jan 2021 21:19:24 +0200 (EET) Received: from [127.0.1.1] (localhost [127.0.0.1]) by ffbox0-bg.mplayerhq.hu (Postfix) with ESMTP id B368A68A779; Sat, 9 Jan 2021 21:19:24 +0200 (EET) X-Original-To: ffmpeg-devel@ffmpeg.org Delivered-To: ffmpeg-devel@ffmpeg.org Received: from w4.tutanota.de (w4.tutanota.de [81.3.6.165]) by ffbox0-bg.mplayerhq.hu (Postfix) with ESMTPS id DAAC6687F93 for ; Sat, 9 Jan 2021 21:19:18 +0200 (EET) Received: from w3.tutanota.de (unknown [192.168.1.164]) by w4.tutanota.de (Postfix) with ESMTP id ED65310600E8 for ; Sat, 9 Jan 2021 19:19:17 +0000 (UTC) DKIM-Signature: v=1; a=rsa-sha256; q=dns/txt; c=relaxed/relaxed; t=1610219957; s=s1; d=lynne.ee; h=From:From:To:To:Subject:Subject:Content-Description:Content-ID:Content-Type:Content-Type:Content-Transfer-Encoding:Cc:Date:Date:In-Reply-To:MIME-Version:MIME-Version:Message-ID:Message-ID:Reply-To:References:Sender; bh=jqYsLAyRHdsmyUeZ+5JVp1K5C3gonmVDkiTR8jqViVM=; b=PeiwV2Tu2X4oa46y+jMZduX7GRt6oG630GnUHmEVUzSr/6NmH4a0McbbTjo6S9zT 96bWv4/xOF7EY6lOXIcZlYsoH1Wd/DSqFmsfkDjUzeDL5Au/+DJndWZvsh/0NOIl65X 8LYAxrb91y+M+g4+hXUep9RJMt/Pfl/Y63WuPj/znH0SGJyITU3C/xciPsfjlAAast4 AdebReMc7v2BG1M7466m16uxVwAJE4J1e5EZDp72W2Ip6j64qpHUOBpP0572TIslCiA mCAr5ZVQM70djpdqmWZSlB41NAcNhyBpInB/+aD9CDDVuASZXHchxrKSwda0uJPFUV0 zKxAzzP0Xg== Date: Sat, 9 Jan 2021 20:19:17 +0100 (CET) From: Lynne To: Ffmpeg Devel Message-ID: MIME-Version: 1.0 Subject: [FFmpeg-devel] [PATCH 1/6] ac3enc_fixed: convert to 32-bit sample format X-BeenThere: ffmpeg-devel@ffmpeg.org X-Mailman-Version: 2.1.20 Precedence: list List-Id: FFmpeg development discussions and patches List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Reply-To: FFmpeg development discussions and patches Errors-To: ffmpeg-devel-bounces@ffmpeg.org Sender: "ffmpeg-devel" 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% Patch attached. Subject: [PATCH 1/6] 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% --- doc/encoders.texi | 7 ++-- libavcodec/Makefile | 2 +- libavcodec/ac3enc.h | 11 ++++--- libavcodec/ac3enc_fixed.c | 63 ++++++++++++++++-------------------- libavcodec/ac3enc_template.c | 21 ++++-------- libavcodec/version.h | 4 +-- tests/fate/ac3.mak | 2 +- 7 files changed, 48 insertions(+), 62 deletions(-) diff --git a/doc/encoders.texi b/doc/encoders.texi index 0b1c69e982..60e763a704 100644 --- a/doc/encoders.texi +++ b/doc/encoders.texi @@ -151,10 +151,9 @@ the undocumented RealAudio 3 (a.k.a. dnet). The @var{ac3} encoder uses floating-point math, while the @var{ac3_fixed} encoder only uses fixed-point integer math. This does not mean that one is always faster, just that one or the other may be better suited to a -particular system. The floating-point encoder will generally produce better -quality audio for a given bitrate. The @var{ac3_fixed} encoder is not the -default codec for any of the output formats, so it must be specified explicitly -using the option @code{-acodec ac3_fixed} in order to use it. +particular system. The @var{ac3_fixed} encoder is not the default codec for +any of the output formats, so it must be specified explicitly using the option +@code{-acodec ac3_fixed} in order to use it. @subsection AC-3 Metadata diff --git a/libavcodec/Makefile b/libavcodec/Makefile index 35318f4f4d..0546e6f6c5 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -181,7 +181,7 @@ OBJS-$(CONFIG_AC3_DECODER) += ac3dec_float.o ac3dec_data.o ac3.o kbd OBJS-$(CONFIG_AC3_FIXED_DECODER) += ac3dec_fixed.o ac3dec_data.o ac3.o kbdwin.o ac3tab.o OBJS-$(CONFIG_AC3_ENCODER) += ac3enc_float.o ac3enc.o ac3tab.o \ ac3.o kbdwin.o -OBJS-$(CONFIG_AC3_FIXED_ENCODER) += ac3enc_fixed.o ac3enc.o ac3tab.o ac3.o +OBJS-$(CONFIG_AC3_FIXED_ENCODER) += ac3enc_fixed.o ac3enc.o ac3tab.o ac3.o kbdwin.o OBJS-$(CONFIG_AC3_MF_ENCODER) += mfenc.o mf_utils.o OBJS-$(CONFIG_ACELP_KELVIN_DECODER) += g729dec.o lsp.o celp_math.o celp_filters.o acelp_filters.o acelp_pitch_delay.o acelp_vectors.o g729postfilter.o OBJS-$(CONFIG_AGM_DECODER) += agm.o diff --git a/libavcodec/ac3enc.h b/libavcodec/ac3enc.h index 044564ecb4..ba62891371 100644 --- a/libavcodec/ac3enc.h +++ b/libavcodec/ac3enc.h @@ -30,8 +30,6 @@ #include -#include "libavutil/float_dsp.h" - #include "ac3.h" #include "ac3dsp.h" #include "avcodec.h" @@ -53,6 +51,7 @@ #define AC3ENC_TYPE_EAC3 2 #if AC3ENC_FLOAT +#include "libavutil/float_dsp.h" #define AC3_NAME(x) ff_ac3_float_ ## x #define MAC_COEF(d,a,b) ((d)+=(a)*(b)) #define COEF_MIN (-16777215.0/16777216.0) @@ -62,12 +61,13 @@ typedef float SampleType; typedef float CoefType; typedef float CoefSumType; #else +#include "libavutil/fixed_dsp.h" #define AC3_NAME(x) ff_ac3_fixed_ ## x #define MAC_COEF(d,a,b) MAC64(d,a,b) #define COEF_MIN -16777215 #define COEF_MAX 16777215 #define NEW_CPL_COORD_THRESHOLD 503317 -typedef int16_t SampleType; +typedef int32_t SampleType; typedef int32_t CoefType; typedef int64_t CoefSumType; #endif @@ -141,7 +141,6 @@ typedef struct AC3Block { uint16_t **qmant; ///< quantized mantissas uint8_t **cpl_coord_exp; ///< coupling coord exponents (cplcoexp) uint8_t **cpl_coord_mant; ///< coupling coord mantissas (cplcomant) - uint8_t coeff_shift[AC3_MAX_CHANNELS]; ///< fixed-point coefficient shift values uint8_t new_rematrixing_strategy; ///< send new rematrixing flags in this block int num_rematrixing_bands; ///< number of rematrixing bands uint8_t rematrixing_flags[4]; ///< rematrixing flags @@ -165,7 +164,11 @@ typedef struct AC3EncodeContext { AVCodecContext *avctx; ///< parent AVCodecContext PutBitContext pb; ///< bitstream writer context AudioDSPContext adsp; +#if AC3ENC_FLOAT AVFloatDSPContext *fdsp; +#else + AVFixedDSPContext *fdsp; +#endif MECmpContext mecc; AC3DSPContext ac3dsp; ///< AC-3 optimized functions FFTContext mdct; ///< FFT context for MDCT calculation diff --git a/libavcodec/ac3enc_fixed.c b/libavcodec/ac3enc_fixed.c index 7818dd8c35..7aaa55f2e7 100644 --- a/libavcodec/ac3enc_fixed.c +++ b/libavcodec/ac3enc_fixed.c @@ -26,12 +26,14 @@ * fixed-point AC-3 encoder. */ -#define FFT_FLOAT 0 #define AC3ENC_FLOAT 0 +#define FFT_FLOAT 0 +#define FFT_FIXED_32 1 #include "internal.h" #include "audiodsp.h" #include "ac3enc.h" #include "eac3enc.h" +#include "kbdwin.h" #define AC3ENC_TYPE AC3ENC_TYPE_AC3_FIXED #include "ac3enc_opts_template.c" @@ -43,37 +45,6 @@ static const AVClass ac3enc_class = { .version = LIBAVUTIL_VERSION_INT, }; -/* - * Normalize the input samples to use the maximum available precision. - * This assumes signed 16-bit input samples. - */ -static int normalize_samples(AC3EncodeContext *s) -{ - int v = s->ac3dsp.ac3_max_msb_abs_int16(s->windowed_samples, AC3_WINDOW_SIZE); - v = 14 - av_log2(v); - if (v > 0) - s->ac3dsp.ac3_lshift_int16(s->windowed_samples, AC3_WINDOW_SIZE, v); - /* +6 to right-shift from 31-bit to 25-bit */ - return v + 6; -} - - -/* - * Scale MDCT coefficients to 25-bit signed fixed-point. - */ -static void scale_coefficients(AC3EncodeContext *s) -{ - int blk, ch; - - for (blk = 0; blk < s->num_blocks; blk++) { - AC3Block *block = &s->blocks[blk]; - for (ch = 1; ch <= s->channels; ch++) { - s->ac3dsp.ac3_rshift_int32(block->mdct_coef[ch], AC3_MAX_COEFS, - block->coeff_shift[ch]); - } - } -} - static void sum_square_butterfly(AC3EncodeContext *s, int64_t sum[4], const int32_t *coef0, const int32_t *coef1, int len) @@ -118,9 +89,10 @@ static CoefType calc_cpl_coord(CoefSumType energy_ch, CoefSumType energy_cpl) static av_cold void ac3_fixed_mdct_end(AC3EncodeContext *s) { ff_mdct_end(&s->mdct); + av_freep(&s->fdsp); + av_freep(&s->mdct_window); } - /** * Initialize MDCT tables. * @@ -130,7 +102,28 @@ static av_cold void ac3_fixed_mdct_end(AC3EncodeContext *s) static av_cold int ac3_fixed_mdct_init(AC3EncodeContext *s) { int ret = ff_mdct_init(&s->mdct, 9, 0, -1.0); - s->mdct_window = ff_ac3_window; + if (ret < 0) + return ret; + + int32_t *iwin = av_malloc_array(AC3_WINDOW_SIZE, sizeof(*iwin)); + if (!iwin) + return AVERROR(ENOMEM); + + float fwin[AC3_WINDOW_SIZE/2]; + ff_kbd_window_init(fwin, 5.0, AC3_WINDOW_SIZE/2); + + for (int i = 0; i < AC3_WINDOW_SIZE/2; i++) + iwin[i] = lrintf(fwin[i] * (1 << 22)); + + for (int i = 0; i < AC3_WINDOW_SIZE/2; i++) + iwin[AC3_WINDOW_SIZE-1-i] = iwin[i]; + + s->mdct_window = iwin; + + s->fdsp = avpriv_alloc_fixed_dsp(s->avctx->flags & AV_CODEC_FLAG_BITEXACT); + if (!s->fdsp) + return AVERROR(ENOMEM); + return ret; } @@ -155,7 +148,7 @@ AVCodec ff_ac3_fixed_encoder = { .init = ac3_fixed_encode_init, .encode2 = ff_ac3_fixed_encode_frame, .close = ff_ac3_encode_close, - .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16P, + .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE }, .priv_class = &ac3enc_class, .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP, diff --git a/libavcodec/ac3enc_template.c b/libavcodec/ac3enc_template.c index 0fdc95b968..de6eba71d8 100644 --- a/libavcodec/ac3enc_template.c +++ b/libavcodec/ac3enc_template.c @@ -91,18 +91,11 @@ static void apply_mdct(AC3EncodeContext *s) AC3Block *block = &s->blocks[blk]; const SampleType *input_samples = &s->planar_samples[ch][blk * AC3_BLOCK_SIZE]; -#if AC3ENC_FLOAT s->fdsp->vector_fmul(s->windowed_samples, input_samples, - s->mdct_window, AC3_WINDOW_SIZE); -#else - s->ac3dsp.apply_window_int16(s->windowed_samples, input_samples, - s->mdct_window, AC3_WINDOW_SIZE); - - block->coeff_shift[ch + 1] = normalize_samples(s); -#endif + s->mdct_window, AC3_WINDOW_SIZE); - s->mdct.mdct_calcw(&s->mdct, block->mdct_coef[ch+1], - s->windowed_samples); + s->mdct.mdct_calc(&s->mdct, block->mdct_coef[ch+1], + s->windowed_samples); } } } @@ -390,9 +383,6 @@ int AC3_NAME(encode_frame)(AVCodecContext *avctx, AVPacket *avpkt, apply_mdct(s); - if (!AC3ENC_FLOAT) - scale_coefficients(s); - clip_coefficients(&s->adsp, s->blocks[0].mdct_coef[1], AC3_MAX_COEFS * s->num_blocks * s->channels); @@ -404,8 +394,9 @@ int AC3_NAME(encode_frame)(AVCodecContext *avctx, AVPacket *avpkt, compute_rematrixing_strategy(s); - if (AC3ENC_FLOAT) - scale_coefficients(s); +#if AC3ENC_FLOAT + scale_coefficients(s); +#endif return ff_ac3_encode_frame_common_end(avctx, avpkt, frame, got_packet_ptr); } diff --git a/libavcodec/version.h b/libavcodec/version.h index 5b92afe60a..1420439044 100644 --- a/libavcodec/version.h +++ b/libavcodec/version.h @@ -28,8 +28,8 @@ #include "libavutil/version.h" #define LIBAVCODEC_VERSION_MAJOR 58 -#define LIBAVCODEC_VERSION_MINOR 115 -#define LIBAVCODEC_VERSION_MICRO 102 +#define LIBAVCODEC_VERSION_MINOR 116 +#define LIBAVCODEC_VERSION_MICRO 100 #define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \ LIBAVCODEC_VERSION_MINOR, \ diff --git a/tests/fate/ac3.mak b/tests/fate/ac3.mak index 757cd51cf2..d76e22bade 100644 --- a/tests/fate/ac3.mak +++ b/tests/fate/ac3.mak @@ -90,7 +90,7 @@ fate-ac3-fixed-encode: tests/data/asynth-44100-2.wav fate-ac3-fixed-encode: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav fate-ac3-fixed-encode: CMD = md5 -i $(SRC) -c ac3_fixed -ab 128k -f ac3 -flags +bitexact -af aresample fate-ac3-fixed-encode: CMP = oneline -fate-ac3-fixed-encode: REF = a1d1fc116463b771abf5aef7ed37d7b1 +fate-ac3-fixed-encode: REF = 1f548175e11a95e62ce20e442fcc8d08 FATE_EAC3-$(call ALLYES, EAC3_DEMUXER EAC3_MUXER EAC3_CORE_BSF) += fate-eac3-core-bsf 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