[FFmpeg-devel] [PATCH] libavcodec, libavformat: Added DFPWM1a codec and raw format
Paul B Mahol
onemda at gmail.com
Fri Feb 25 10:15:42 EET 2022
On Fri, Feb 25, 2022 at 02:54:35AM -0500, 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 both a new codec (with encoding and decoding), as well as
> a raw audio format to be able to read/write the raw files that are most
> commonly used (as no other container format supports it yet).
>
> The codec sources are pretty simple: they use the reference codec with
> a basic wrapper to connect it to the FFmpeg AVCodec system. There's a
> bit of extra code to convert from unsigned to signed 8-bit audio, as the
> codec implementation operates on signed data, which FFmpeg doesn't support.
>
> The muxers are mostly copied from the PCM demuxer and the raw muxers, as
> DFPWM is typically stored as raw data.
>
> 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.
>
> One thing I noticed is that this sample file fails to decode to raw:
> https://samples.ffmpeg.org/ogg/virginradio-three-consecutive-chains.ogg
> It reports "Application provided invalid, non monotonically increasing
> dts to muxer in stream 0", which appears to be because the initial
> timestamp is not 0:00. This affects all raw muxers, including PCM.
>
> Signed-off-by: Jack Bruienne <jackbruienne at gmail.com>
> ---
> Changelog | 1 +
> MAINTAINERS | 2 +
> doc/general_contents.texi | 2 +
> libavcodec/Makefile | 2 +
> libavcodec/allcodecs.c | 2 +
> libavcodec/codec_desc.c | 7 ++
> libavcodec/codec_id.h | 1 +
> libavcodec/dfpwmdec.c | 138 +++++++++++++++++++++++++++++++++++++
> libavcodec/dfpwmenc.c | 140 ++++++++++++++++++++++++++++++++++++++
> libavcodec/utils.c | 2 +
> libavcodec/version.h | 2 +-
> libavformat/Makefile | 2 +
> libavformat/allformats.c | 2 +
> libavformat/dfpwmdec.c | 107 +++++++++++++++++++++++++++++
> libavformat/rawenc.c | 13 ++++
> libavformat/version.h | 4 +-
> 16 files changed, 424 insertions(+), 3 deletions(-)
> create mode 100644 libavcodec/dfpwmdec.c
> create mode 100644 libavcodec/dfpwmenc.c
> create mode 100644 libavformat/dfpwmdec.c
>
> diff --git a/Changelog b/Changelog
> index 5ad2cef..ec688da 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 and raw muxer/demuxer
Keep empty line here above and everywhere else you had removed them.
Is patch corrupted somehow? Just attach file.
And split demuxer and muxer addition from decoder and encoder.
> version 5.0:
> diff --git a/MAINTAINERS b/MAINTAINERS
> index f33ccbd..931cf4b 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
> @@ -415,6 +416,7 @@ Muxers/Demuxers:
> dashdec.c Steven Liu
> dashenc.c Karthick Jeyapal
> daud.c Reimar Doeffinger
> + dfpwmdec.c Jack Bruienne
> dss.c Oleksij Rempel
> dtsdec.c foo86
> dtshddec.c Paul B Mahol
> diff --git a/doc/general_contents.texi b/doc/general_contents.texi
> index df1692c..fcd9da1 100644
> --- a/doc/general_contents.texi
> +++ b/doc/general_contents.texi
> @@ -578,6 +578,7 @@ library:
> @item raw aptX @tab X @tab X
> @item raw aptX HD @tab X @tab X
> @item raw Chinese AVS video @tab X @tab X
> + at item raw DFPWM @tab X @tab X
> @item raw Dirac @tab X @tab X
> @item raw DNxHD @tab X @tab X
> @item raw DTS @tab X @tab X
> @@ -1194,6 +1195,7 @@ 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..87ca591 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("DFPWM1a audio"),
You could use full description here instead of just DFPWM1a.
> + .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,
Keep empty line here.
> /* 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..9f12841
> --- /dev/null
> +++ b/libavcodec/dfpwmdec.c
> @@ -0,0 +1,138 @@
> +/*
> + * 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
> +#define CONST_PREC 10
> +#endif
> +
> +static void au_decompress(DFPWMState *state, int fs, int len, int8_t
> *outbuf, uint8_t *inbuf)
> +{
> + int i, j;
> + uint8_t d;
> + for (i = 0; i < len; i++) {
> + // get bits
> + d = *(inbuf++);
> + for (j = 0; j < 8; 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)
> + state->q += (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
> + if(ns < 1+(1<<(CONST_PREC-8))) ns = 1+(1<<(CONST_PREC-8));
> +#endif
> + state->s = ns;
> +
> + // FILTER: perform antijerk
> + ov = (t != state->lt ? (nq+lq)>>1 : nq);
> +
> + // FILTER: perform LPF
> + state->fq += ((fs*(ov-state->fq) + 0x80)>>8);
> + ov = state->fq;
> +
> + // output sample
> + *(outbuf++) = ov;
> +
> + 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 av_cold int dfpwm_dec_end(struct AVCodecContext *ctx)
> +{
> + return 0;
> +}
Remove if not gonna be used and is empty.
> +
> +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;
> +
> + av_frame_get_buffer(frame, 0);
Use ff_get_buffer()
See how libavcodec/pcm.c codecs handles that.
> +
> + au_decompress(state, 100, packet->size, frame->data[0], packet->data);
> +
> + // convert from signed to unsigned 8-bit, as DFPWM outputs S8 but
> FFmpeg needs U8
> + for (int i = 0; i < packet->size * 8; i++) frame->data[0][i] =
> ((int8_t*)frame->data[0])[i] + 128;
Cant you remove this wrapper, for such trivial code is not helping here.
Than you can do conversion from S8 to U8 on the fly without need for extra buffer.
> +
> + if (got_frame) *got_frame = 1;
> + 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,
> + .close = dfpwm_dec_end,
> + .decode = dfpwm_dec_frame,
> + .capabilities = 0,
This flags are not correct, or optimal in any way.
> + .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
> +};
> \ No newline at end of file
> diff --git a/libavcodec/dfpwmenc.c b/libavcodec/dfpwmenc.c
> new file mode 100644
> index 0000000..97fc42e
> --- /dev/null
> +++ b/libavcodec/dfpwmenc.c
> @@ -0,0 +1,140 @@
> +/*
> + * 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
> +#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, int8_t
> *inbuf)
> +{
> + int i, j;
> + uint8_t d = 0;
> + for (i = 0; i < len; i++) {
> + for (j = 0; j < 8; j++) {
> + int nq, st, ns;
> + // get sample
> + int v = *(inbuf++);
> + // set bit / target
> + int t = (v < state->q || v == -128 ? -128 : 127);
> + 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
> + if(ns < 1+(1<<(CONST_PREC-8))) ns = 1+(1<<(CONST_PREC-8));
> +#endif
> + state->s = ns;
> +
> + state->lt = t;
> +
> + //fprintf(stderr, "%4i %4i %4i %4i\n", v, *q, *s, t);
> + //usleep(10000);
> + }
> +
> + // 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;
> +
> + return 0;
> +}
> +
> +static av_cold int dfpwm_enc_end(struct AVCodecContext *ctx)
> +{
> + return 0;
> +}
Please remove functions that do nothing.
> +
> +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);
> + int8_t *data = av_malloc(size * 8);
> +
> + if (!data) return AVERROR(ENOMEM);
> +
> + if (packet->size < size) av_grow_packet(packet, size - packet->size);
> + else if (packet->size > size) av_shrink_packet(packet, size);
> +
> + // make a temporary S8 buffer as DFPWM needs S8 but FFmpeg uses U8
> + for (int i = 0; i < frame->nb_samples; i++) data[i] = frame->data[0][i]
> - 128;
> + for (int i = frame->nb_samples; i < size * 8; i++) data[i] = 0;
> +
> + au_compress(state, size, packet->data, data);
> + av_free(data);
Using extra buffer for such trivial code is not optimal solution.
> +
> + if (got_packet) *got_packet = 1;
> + 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,
> + .close = dfpwm_enc_end,
> + .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, \
> diff --git a/libavformat/Makefile b/libavformat/Makefile
> index 6566e40..b89073a 100644
> --- a/libavformat/Makefile
> +++ b/libavformat/Makefile
> @@ -165,6 +165,8 @@ OBJS-$(CONFIG_DAUD_MUXER) += daudenc.o
> OBJS-$(CONFIG_DCSTR_DEMUXER) += dcstr.o
> OBJS-$(CONFIG_DERF_DEMUXER) += derf.o pcm.o
> OBJS-$(CONFIG_DFA_DEMUXER) += dfa.o
> +OBJS-$(CONFIG_DFPWM_DEMUXER) += dfpwmdec.o pcm.o
> +OBJS-$(CONFIG_DFPWM_MUXER) += rawenc.o
> OBJS-$(CONFIG_DHAV_DEMUXER) += dhav.o
> OBJS-$(CONFIG_DIRAC_DEMUXER) += diracdec.o rawdec.o
> OBJS-$(CONFIG_DIRAC_MUXER) += rawenc.o
> diff --git a/libavformat/allformats.c b/libavformat/allformats.c
> index d066a77..587ad59 100644
> --- a/libavformat/allformats.c
> +++ b/libavformat/allformats.c
> @@ -124,6 +124,8 @@ extern const AVOutputFormat ff_daud_muxer;
> extern const AVInputFormat ff_dcstr_demuxer;
> extern const AVInputFormat ff_derf_demuxer;
> extern const AVInputFormat ff_dfa_demuxer;
> +extern const AVInputFormat ff_dfpwm_demuxer;
> +extern const AVOutputFormat ff_dfpwm_muxer;
> extern const AVInputFormat ff_dhav_demuxer;
> extern const AVInputFormat ff_dirac_demuxer;
> extern const AVOutputFormat ff_dirac_muxer;
> diff --git a/libavformat/dfpwmdec.c b/libavformat/dfpwmdec.c
> new file mode 100644
> index 0000000..ad5bfa5
> --- /dev/null
> +++ b/libavformat/dfpwmdec.c
> @@ -0,0 +1,107 @@
> +/*
> + * RAW PCM demuxers
> + * Copyright (c) 2002 Fabrice Bellard
> + * Copyright (c) 2022 Jack Bruienne
> + *
> + * 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
> + */
> +
> +#include "libavutil/avstring.h"
> +#include "avformat.h"
> +#include "internal.h"
> +#include "pcm.h"
> +#include "libavutil/log.h"
> +#include "libavutil/opt.h"
> +#include "libavutil/avassert.h"
> +
> +typedef struct DFPWMAudioDemuxerContext {
> + AVClass *class;
> + int sample_rate;
> +} DFPWMAudioDemuxerContext;
> +
> +static int dfpwm_read_header(AVFormatContext *s)
> +{
> + DFPWMAudioDemuxerContext *s1 = s->priv_data;
> + AVCodecParameters *par;
> + AVStream *st;
> + uint8_t *mime_type = NULL;
> +
> + st = avformat_new_stream(s, NULL);
> + if (!st)
> + return AVERROR(ENOMEM);
> + par = st->codecpar;
> +
> + par->codec_type = AVMEDIA_TYPE_AUDIO;
> + par->codec_id = s->iformat->raw_codec_id;
> + par->sample_rate = s1->sample_rate;
> + par->channels = 1;
> +
> + av_opt_get(s->pb, "mime_type", AV_OPT_SEARCH_CHILDREN, &mime_type);
> + if (mime_type && s->iformat->mime_type) {
> + int rate = 0;
> + const char *options;
> + if (av_stristart(mime_type, s->iformat->mime_type, &options)) { /*
> audio/L16 */
> + while (options = strchr(options, ';')) {
> + options++;
> + if (!rate)
> + sscanf(options, " rate=%d", &rate);
> + }
> + if (rate <= 0) {
> + av_log(s, AV_LOG_ERROR,
> + "Invalid sample_rate found in mime_type \"%s\"\n",
> + mime_type);
> + av_freep(&mime_type);
> + return AVERROR_INVALIDDATA;
> + }
> + par->sample_rate = rate;
> + }
> + }
> + av_freep(&mime_type);
> +
> + par->bits_per_coded_sample = av_get_bits_per_sample(par->codec_id);
> +
> + av_assert0(par->bits_per_coded_sample > 0);
> +
> + par->block_align = 1;
> +
> + avpriv_set_pts_info(st, 64, 1, par->sample_rate);
> + return 0;
> +}
> +
> +static const AVOption dfpwm_options[] = {
> + { "sample_rate", "", offsetof(DFPWMAudioDemuxerContext, sample_rate),
> AV_OPT_TYPE_INT, {.i64 = 44100}, 0, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
> + { NULL },
> +};
> +static const AVClass dfpwm_demuxer_class = {
> + .class_name = "dfpwm demuxer",
> + .item_name = av_default_item_name,
> + .option = dfpwm_options,
> + .version = LIBAVUTIL_VERSION_INT,
> +};
> +
> +const AVInputFormat ff_dfpwm_demuxer = {
> + .name = "dfpwm",
> + .long_name = NULL_IF_CONFIG_SMALL("raw DFPWM1a data"),
> + .priv_data_size = sizeof(DFPWMAudioDemuxerContext),
> + .read_header = dfpwm_read_header,
> + .read_packet = ff_pcm_read_packet,
> + .read_seek = ff_pcm_read_seek,
> + .flags = AVFMT_GENERIC_INDEX,
> + .extensions = "dfpwm",
> + .raw_codec_id = AV_CODEC_ID_DFPWM,
> + .priv_class = &dfpwm_demuxer_class,
> +};
> \ No newline at end of file
> diff --git a/libavformat/rawenc.c b/libavformat/rawenc.c
> index 4bbae77..17b627b 100644
> --- a/libavformat/rawenc.c
> +++ b/libavformat/rawenc.c
> @@ -192,6 +192,19 @@ const AVOutputFormat ff_data_muxer = {
> };
> #endif
> +#if CONFIG_DFPWM_MUXER
> +const AVOutputFormat ff_dfpwm_muxer = {
> + .name = "dfpwm",
> + .long_name = NULL_IF_CONFIG_SMALL("raw DFPWM1a audio"),
> + .extensions = "dfpwm",
> + .audio_codec = AV_CODEC_ID_DFPWM,
> + .video_codec = AV_CODEC_ID_NONE,
> + .init = force_one_stream,
> + .write_packet = ff_raw_write_packet,
> + .flags = AVFMT_NOTIMESTAMPS,
> +};
> +#endif
> +
> #if CONFIG_DIRAC_MUXER
> const AVOutputFormat ff_dirac_muxer = {
> .name = "dirac",
> diff --git a/libavformat/version.h b/libavformat/version.h
> index 2623457..0f89af4 100644
> --- a/libavformat/version.h
> +++ b/libavformat/version.h
> @@ -32,8 +32,8 @@
> // Major bumping may affect Ticket5467, 5421, 5451(compatibility with
> Chromium)
> // Also please add any ticket numbers that you believe might be affected
> here
> #define LIBAVFORMAT_VERSION_MAJOR 59
> -#define LIBAVFORMAT_VERSION_MINOR 17
> -#define LIBAVFORMAT_VERSION_MICRO 102
> +#define LIBAVFORMAT_VERSION_MINOR 18
> +#define LIBAVFORMAT_VERSION_MICRO 100
> #define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR,
> \
>
> LIBAVFORMAT_VERSION_MINOR, \
> --
> 2.35.1
>
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