[FFmpeg-devel] [PATCH v3 1/2] libavcodec: Added DFPWM1a codec
James Almer
jamrial at gmail.com
Sun Feb 27 05:02:57 EET 2022
On 2/26/2022 8:26 PM, Jack Bruienne wrote:
>
> From the wiki page (https://wiki.vexatos.com/dfpwm):
>> DFPWM (Dynamic Filter Pulse Width Modulation) is an audio codec
>> created by Ben “GreaseMonkey” Russell in 2012, originally to be used
>> as a voice codec for asiekierka's pixmess, a C remake of 64pixels.
>> It is a 1-bit-per-sample codec which uses a dynamic-strength one-pole
>> low-pass filter as a predictor. Due to the fact that a raw DPFWM decoding
>> creates a high-pitched whine, it is often followed by some
>> post-processing
>> filters to make the stream more listenable.
>
> It has recently gained popularity through the ComputerCraft mod for
> Minecraft, which added support for audio through this codec, as well as
> the Computronics expansion which preceeded the official support. These
> both implement the slightly adjusted 1a version of the codec, which is
> the version I have chosen for this patch.
>
> This patch adds a new codec (with encoding and decoding) for DFPWM1a.
>
> The codec sources are pretty simple: they use the reference codec with
> a basic wrapper to connect it to the FFmpeg AVCodec system.
>
> This patch will be highly useful to ComputerCraft developers who are
> working with audio, as it is the standard format for audio, and there
> are few user-friendly encoders out there. It will streamline the process
> for importing audio, replacing the need to write code or use tools that
> require very specific input formats.
>
> You may use the CraftOS-PC program (https://www.craftos-pc.cc) to test
> out DFPWM playback. To use it, run the program and type this command:
> "attach left speaker" Then run "speaker play <file.dfpwm>" for each file.
> The app runs in a sandbox, so files have to be transferred in first;
> the easiest way to do this is to simply drag the file on the window.
> (Or copy files to the folder at https://www.craftos-pc.cc/docs/saves.)
>
> Sample DFPWM files can be generated with an online tool at
> https://music.madefor.cc. This is the current best way to encode DFPWM
> files. Simply drag an audio file onto the page, and it will encode it,
> giving a download link on the page.
>
> I've made sure to update all of the docs as per Developer§7, and I've
> tested it as per section 8. Test files encoded to DFPWM play correctly
> in ComputerCraft, and other files that work in CC are correctly decoded.
> I have also verified that corrupt files do not crash the decoder - this
> should theoretically not be an issue as the result size is constant with
> respect to the input size.
>
> Changes since the prior v2 patch:
> I've found that the reference encoder has a few errors, and sounds
> worse than the Java-based implementation that is used most often. I got
> in contact with someone who knows DFPWM much better than I do, and I
> worked with them to make a few adjustments that should improve the
> audio quality. I also made sure that the output matches the Java
> codec exactly, so it should have the exact same quality as other codecs.
>
> Signed-off-by: Jack Bruienne <jackbruienne at gmail.com>
> ---
> Changelog | 1 +
> MAINTAINERS | 1 +
> doc/general_contents.texi | 1 +
> libavcodec/Makefile | 2 +
> libavcodec/allcodecs.c | 2 +
> libavcodec/codec_desc.c | 7 +++
> libavcodec/codec_id.h | 1 +
> libavcodec/dfpwmdec.c | 129 ++++++++++++++++++++++++++++++++++++++
> libavcodec/dfpwmenc.c | 123 ++++++++++++++++++++++++++++++++++++
> libavcodec/utils.c | 2 +
> libavcodec/version.h | 2 +-
> 11 files changed, 270 insertions(+), 1 deletion(-)
> create mode 100644 libavcodec/dfpwmdec.c
> create mode 100644 libavcodec/dfpwmenc.c
> diff --git a/Changelog b/Changelog
> index 5ad2cef..5170a6a 100644
> --- a/Changelog
> +++ b/Changelog
> @@ -4,6 +4,7 @@ releases are sorted from youngest to oldest.
> version 5.1:
> - dialogue enhance audio filter
> - dropped obsolete XvMC hwaccel
> +- DFPWM audio encoder/decoder
>
>
> version 5.0:
> diff --git a/MAINTAINERS b/MAINTAINERS
> index f33ccbd..57b6f33 100644
> --- a/MAINTAINERS
> +++ b/MAINTAINERS
> @@ -161,6 +161,7 @@ Codecs:
> cscd.c Reimar Doeffinger
> cuviddec.c Timo Rothenpieler
> dca* foo86
> + dfpwm* Jack Bruienne
> dirac* Rostislav Pehlivanov
> dnxhd* Baptiste Coudurier
> dolby_e* foo86
> diff --git a/doc/general_contents.texi b/doc/general_contents.texi
> index df1692c..14aeaed 100644
> --- a/doc/general_contents.texi
> +++ b/doc/general_contents.texi
> @@ -1194,6 +1194,7 @@ following image formats are supported:
> @item CRI HCA @tab @tab X
> @item Delphine Software International CIN audio @tab @tab X
> @tab Codec used in Delphine Software International games.
> + at item DFPWM @tab X @tab X
> @item Digital Speech Standard - Standard Play mode (DSS SP) @tab @tab X
> @item Discworld II BMV Audio @tab @tab X
> @item COOK @tab @tab X
> diff --git a/libavcodec/Makefile b/libavcodec/Makefile
> index 6076b4a..7474220 100644
> --- a/libavcodec/Makefile
> +++ b/libavcodec/Makefile
> @@ -289,6 +289,8 @@ OBJS-$(CONFIG_DERF_DPCM_DECODER) += dpcm.o
> OBJS-$(CONFIG_DIRAC_DECODER) += diracdec.o dirac.o diracdsp.o diractab.o \
> dirac_arith.o dirac_dwt.o dirac_vlc.o
> OBJS-$(CONFIG_DFA_DECODER) += dfa.o
> +OBJS-$(CONFIG_DFPWM_DECODER) += dfpwmdec.o
> +OBJS-$(CONFIG_DFPWM_ENCODER) += dfpwmenc.o
> OBJS-$(CONFIG_DNXHD_DECODER) += dnxhddec.o dnxhddata.o
> OBJS-$(CONFIG_DNXHD_ENCODER) += dnxhdenc.o dnxhddata.o
> OBJS-$(CONFIG_DOLBY_E_DECODER) += dolby_e.o dolby_e_parse.o kbdwin.o
> diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
> index d1e1019..c3a0c26 100644
> --- a/libavcodec/allcodecs.c
> +++ b/libavcodec/allcodecs.c
> @@ -437,6 +437,8 @@ extern const AVCodec ff_bmv_audio_decoder;
> extern const AVCodec ff_cook_decoder;
> extern const AVCodec ff_dca_encoder;
> extern const AVCodec ff_dca_decoder;
> +extern const AVCodec ff_dfpwm_encoder;
> +extern const AVCodec ff_dfpwm_decoder;
> extern const AVCodec ff_dolby_e_decoder;
> extern const AVCodec ff_dsd_lsbf_decoder;
> extern const AVCodec ff_dsd_msbf_decoder;
> diff --git a/libavcodec/codec_desc.c b/libavcodec/codec_desc.c
> index 725c687..81f3b3c 100644
> --- a/libavcodec/codec_desc.c
> +++ b/libavcodec/codec_desc.c
> @@ -3237,6 +3237,13 @@ static const AVCodecDescriptor codec_descriptors[] = {
> .long_name = NULL_IF_CONFIG_SMALL("MSN Siren"),
> .props = AV_CODEC_PROP_INTRA_ONLY | AV_CODEC_PROP_LOSSY,
> },
> + {
> + .id = AV_CODEC_ID_DFPWM,
> + .type = AVMEDIA_TYPE_AUDIO,
> + .name = "dfpwm",
> + .long_name = NULL_IF_CONFIG_SMALL("DFPWM (Dynamic Filter Pulse Width Modulation)"),
> + .props = AV_CODEC_PROP_LOSSY,
> + },
>
> /* subtitle codecs */
> {
> diff --git a/libavcodec/codec_id.h b/libavcodec/codec_id.h
> index ab265ec..3ffb9bd 100644
> --- a/libavcodec/codec_id.h
> +++ b/libavcodec/codec_id.h
> @@ -516,6 +516,7 @@ enum AVCodecID {
> AV_CODEC_ID_HCA,
> AV_CODEC_ID_FASTAUDIO,
> AV_CODEC_ID_MSNSIREN,
> + AV_CODEC_ID_DFPWM,
>
> /* subtitle codecs */
> AV_CODEC_ID_FIRST_SUBTITLE = 0x17000, ///< A dummy ID pointing at the start of subtitle codecs.
> diff --git a/libavcodec/dfpwmdec.c b/libavcodec/dfpwmdec.c
> new file mode 100644
> index 0000000..b783aad
> --- /dev/null
> +++ b/libavcodec/dfpwmdec.c
> @@ -0,0 +1,129 @@
> +/*
> + * DFPWM decoder
> + * Copyright (c) 2022 Jack Bruienne
> + * Copyright (c) 2012, 2016 Ben "GreaseMonkey" Russell
> + *
> + * This file is part of FFmpeg.
> + *
> + * FFmpeg 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.
> + *
> + * FFmpeg 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 FFmpeg; if not, write to the Free Software
> + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
> + */
> +
> +/**
> + * @file
> + * DFPWM1a decoder
> + */
> +
> +#include "libavutil/internal.h"
> +#include "avcodec.h"
> +#include "codec_id.h"
> +#include "internal.h"
> +
> +typedef struct {
> + int fq, q, s, lt;
> +} DFPWMState;
> +
> +// DFPWM codec from https://github.com/ChenThread/dfpwm/blob/master/1a/
> +// Licensed in the public domain
> +
> +#ifndef CONST_PREC
Remove this check. Nothing should have defined it at this point.
> +#define CONST_PREC 10
> +#endif
> +
> +static void au_decompress(DFPWMState *state, int fs, int len, uint8_t *outbuf, uint8_t *inbuf)
> +{
> + int i, j;
> + uint8_t d;
Unsigned is better than fixed size types for a scalar like this.
> + for (i = 0; i < len; i++) {
for (int i...
> + // get bits
> + d = *(inbuf++);
> + for (j = 0; j < 8; j++) {
for (int j...
> + int nq, lq, st, ns, ov;
> + // set target
> + int t = ((d&1) ? 127 : -128);
> + d >>= 1;
> +
> + // adjust charge
> + nq = state->q + ((state->s * (t-state->q) + (1<<(CONST_PREC-1)))>>CONST_PREC);
> + if(nq == state->q && nq != t)
> + nq += (t == 127 ? 1 : -1);
> + lq = state->q;
> + state->q = nq;
> +
> + // adjust strength
> + st = (t != state->lt ? 0 : (1<<CONST_PREC)-1);
> + ns = state->s;
> + if(ns != st)
> + ns += (st != 0 ? 1 : -1);
> +#if CONST_PREC > 8
Same, remove this check.
> + if(ns < (2<<(CONST_PREC-8))) ns = (2<<(CONST_PREC-8));
> +#endif
> + state->s = ns;
> +
> + // FILTER: perform antijerk
> + ov = (t != state->lt ? (nq+lq+1)>>1 : nq);
> +
> + // FILTER: perform LPF
> + state->fq += ((fs*(ov-state->fq) + 0x80)>>8);
> + ov = state->fq;
> +
> + // output sample
> + *(outbuf++) = ov + 128;
> +
> + state->lt = t;
> + }
> + }
> +}
> +
> +static av_cold int dfpwm_dec_init(struct AVCodecContext *ctx)
> +{
> + DFPWMState *state = ctx->priv_data;
> +
> + state->fq = 0;
> + state->q = 0;
> + state->s = 0;
> + state->lt = -128;
> +
> + return 0;
> +}
> +
> +static int dfpwm_dec_frame(struct AVCodecContext *ctx, void *data,
> + int *got_frame, struct AVPacket *packet)
> +{
> + DFPWMState *state = ctx->priv_data;
> + AVFrame *frame = data;
> +
> + frame->format = AV_SAMPLE_FMT_U8;
> + frame->nb_samples = packet->size * 8;
> + frame->channel_layout = AV_CH_LAYOUT_MONO;
These should be set in the AVCodecContext in dfpwm_dec_init() above.
ff_get_buffer() will then copy them to the frame here.
Also, missing channels and sample_rate.
> +
> + ff_get_buffer(ctx, frame, 0);
> +
> + au_decompress(state, 140, packet->size, frame->data[0], packet->data);
> +
> + if (got_frame) *got_frame = 1;
got_frame is never NULL.
> + return packet->size;
> +}
> +
> +const AVCodec ff_dfpwm_decoder = {
> + .name = "dfpwm",
> + .long_name = NULL_IF_CONFIG_SMALL("DFPWM1a audio"),
> + .type = AVMEDIA_TYPE_AUDIO,
> + .id = AV_CODEC_ID_DFPWM,
> + .priv_data_size = sizeof(DFPWMState),
> + .init = dfpwm_dec_init,
> + .decode = dfpwm_dec_frame,
> + .capabilities = AV_CODEC_CAP_DR1,
Also needs AV_CODEC_CAP_CHANNEL_CONF.
> + .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
> +};
> \ No newline at end of file
Fix this please.
> diff --git a/libavcodec/dfpwmenc.c b/libavcodec/dfpwmenc.c
> new file mode 100644
> index 0000000..c973cc8
> --- /dev/null
> +++ b/libavcodec/dfpwmenc.c
> @@ -0,0 +1,123 @@
> +/*
> + * DFPWM encoder
> + * Copyright (c) 2022 Jack Bruienne
> + * Copyright (c) 2012, 2016 Ben "GreaseMonkey" Russell
> + *
> + * This file is part of FFmpeg.
> + *
> + * FFmpeg 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.
> + *
> + * FFmpeg 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 FFmpeg; if not, write to the Free Software
> + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
> + */
> +
> +/**
> + * @file
> + * DFPWM1a encoder
> + */
> +
> +#include "libavutil/internal.h"
> +#include "avcodec.h"
> +#include "codec_id.h"
> +#include "internal.h"
> +
> +typedef struct {
> + int fq, q, s, lt;
> +} DFPWMState;
> +
> +// DFPWM codec from https://github.com/ChenThread/dfpwm/blob/master/1a/
> +// Licensed in the public domain
> +
> +#ifndef CONST_PREC
Same as above.
> +#define CONST_PREC 10
> +#endif
> +
> +// note, len denotes how many compressed bytes there are (uncompressed bytes / 8).
> +static void au_compress(DFPWMState *state, int len, uint8_t *outbuf, uint8_t *inbuf)
> +{
> + int i, j;
> + uint8_t d = 0;
> + for (i = 0; i < len; i++) {
> + for (j = 0; j < 8; j++) {
Same.
> + int nq, st, ns;
> + // get sample
> + int v = *(inbuf++) - 128;
> + // set bit / target
> + int t = (v > state->q || (v == state->q && v == 127) ? 127 : -128);
> + d >>= 1;
> + if(t > 0)
> + d |= 0x80;
> +
> + // adjust charge
> + nq = state->q + ((state->s * (t-state->q) + (1<<(CONST_PREC-1)))>>CONST_PREC);
> + if(nq == state->q && nq != t)
> + nq += (t == 127 ? 1 : -1);
> + state->q = nq;
> +
> + // adjust strength
> + st = (t != state->lt ? 0 : (1<<CONST_PREC)-1);
> + ns = state->s;
> + if(ns != st)
> + ns += (st != 0 ? 1 : -1);
> +#if CONST_PREC > 8
Same
> + if(ns < (2<<(CONST_PREC-8))) ns = (2<<(CONST_PREC-8));
> +#endif
> + state->s = ns;
> +
> + state->lt = t;
> + }
> +
> + // output bits
> + *(outbuf++) = d;
> + }
> +}
> +
> +static av_cold int dfpwm_enc_init(struct AVCodecContext *ctx)
> +{
> + DFPWMState *state = ctx->priv_data;
> +
> + state->fq = 0;
> + state->q = 0;
> + state->s = 0;
> + state->lt = -128;
Sample rate?
> +
> + return 0;
> +}
> +
> +static int dfpwm_enc_frame(struct AVCodecContext *ctx, struct AVPacket *packet,
> + const struct AVFrame *frame, int *got_packet)
> +{
> + DFPWMState *state = ctx->priv_data;
> + int size = frame->nb_samples / 8 + (frame->nb_samples % 8 > 0 ? 1 : 0);
> +
> + if (packet->size < size) av_grow_packet(packet, size - packet->size);
> + else if (packet->size > size) av_shrink_packet(packet, size);
The packet is always "clean" at this point. These checks are
unnecessary, and the else path will never be taken.
You should for that matter use ff_get_encode_buffer() to allocate the
packet buffer, and set the AV_CODEC_CAP_DR1 capability for it.
> +
> + au_compress(state, size, packet->data, frame->data[0]);
> +
> + if (got_packet) *got_packet = 1;
Same, got_packet is never NULL.
> + return 0;
> +}
> +
> +const AVCodec ff_dfpwm_encoder = {
> + .name = "dfpwm",
> + .long_name = NULL_IF_CONFIG_SMALL("DFPWM1a audio"),
> + .type = AVMEDIA_TYPE_AUDIO,
> + .id = AV_CODEC_ID_DFPWM,
> + .priv_data_size = sizeof(DFPWMState),
> + .init = dfpwm_enc_init,
> + .encode2 = dfpwm_enc_frame,
> + .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_NONE},
> + .channel_layouts = (const uint64_t[]){AV_CH_LAYOUT_MONO, 0},
> + .capabilities = AV_CODEC_CAP_VARIABLE_FRAME_SIZE,
> + .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
> +};
> diff --git a/libavcodec/utils.c b/libavcodec/utils.c
> index 6f9d90a..066da76 100644
> --- a/libavcodec/utils.c
> +++ b/libavcodec/utils.c
> @@ -577,6 +577,8 @@ enum AVCodecID av_get_pcm_codec(enum AVSampleFormat fmt, int be)
> int av_get_bits_per_sample(enum AVCodecID codec_id)
> {
> switch (codec_id) {
> + case AV_CODEC_ID_DFPWM:
> + return 1;
> case AV_CODEC_ID_ADPCM_SBPRO_2:
> return 2;
> case AV_CODEC_ID_ADPCM_SBPRO_3:
> diff --git a/libavcodec/version.h b/libavcodec/version.h
> index d900503..84f3979 100644
> --- a/libavcodec/version.h
> +++ b/libavcodec/version.h
> @@ -28,7 +28,7 @@
> #include "libavutil/version.h"
>
> #define LIBAVCODEC_VERSION_MAJOR 59
> -#define LIBAVCODEC_VERSION_MINOR 21
> +#define LIBAVCODEC_VERSION_MINOR 22
> #define LIBAVCODEC_VERSION_MICRO 100
>
> #define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
>
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