[FFmpeg-devel,v3,1/2] libavcodec: Added DFPWM1a codec

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@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

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@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.
> +@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, \
>