diff mbox series

[FFmpeg-devel] FTR decoder

Message ID CAPYw7P713o_j6cim4kuRmdVsQXDafVeWQ5J2gp-sVue5Vsyu7g@mail.gmail.com
State New
Headers show
Series [FFmpeg-devel] FTR decoder | expand

Checks

Context Check Description
yinshiyou/configure_loongarch64 warning Failed to apply patch
andriy/make_x86 fail Make failed

Commit Message

Paul B Mahol Aug. 31, 2022, 4:42 p.m. UTC
Patch attached.

Comments

Andreas Rheinhardt Aug. 31, 2022, 5:55 p.m. UTC | #1
Paul B Mahol:
> Patch attached.

> +    union {
> +        uint64_t u64;
> +        uint8_t  u8[8 + AV_INPUT_BUFFER_PADDING_SIZE];
> +    } tmp;
> +
> +    *poutbuf_size = 0;
> +    *poutbuf = NULL;
> +
> +    if (s->flags & PARSER_FLAG_COMPLETE_FRAMES) {
> +        next = buf_size;
> +    } else {
> +        for (int i = 0; i < buf_size; i++) {
> +            if (ftr->skip > 0) {
> +                ftr->skip--;
> +                if (ftr->skip == 0 && ftr->split) {
> +                    ftr->split = 0;
> +                    next = i;
> +                    break;
> +                } else if (ftr->skip > 0) {
> +                    continue;
> +                }
> +            }
> +
> +            state = (state << 8) | buf[i];
> +            tmp.u64 = av_be2ne64(state);

It is simpler to just use an uint8_t buf[8 +
AV_INPUT_BUFFER_PADDING_SIZE] that is set via AV_RB64(buf, state).

> +            init_get_bits(&bits, tmp.u8 + 8 - AV_AAC_ADTS_HEADER_SIZE,
> +                          AV_AAC_ADTS_HEADER_SIZE * 8);
> +
Andreas Rheinhardt Aug. 31, 2022, 7:15 p.m. UTC | #2
Paul B Mahol:
> diff --git a/libavcodec/ftr.c b/libavcodec/ftr.c
> new file mode 100644
> index 0000000000..03d490a0c9
> --- /dev/null
> +++ b/libavcodec/ftr.c
> @@ -0,0 +1,217 @@
> +/*
> + * 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 "adts_header.h"
> +#include "avcodec.h"
> +#include "codec_internal.h"
> +#include "get_bits.h"
> +#include "internal.h"

You seem to not have rebased your patch upon master: ff_get_buffer() is
now in decode.h and this won't compile; including internal.h seems
superfluous now.

> +
> +typedef struct FTRContext {
> +    AVCodecContext *aac_avctx[64];   // wrapper context for AAC
> +    int nb_context;
> +    AVPacket *packet;
> +} FTRContext;
> +
> +static av_cold int ftr_init(AVCodecContext *avctx)
> +{
> +    FTRContext *s = avctx->priv_data;
> +    const AVCodec *codec;
> +    int ret;
> +
> +    if (avctx->ch_layout.nb_channels > 64 ||
> +        avctx->ch_layout.nb_channels <= 0)
> +        return AVERROR_BUG;

I don't see what is supposed to limit nb_channels to 64. If it isn't
checked somewhere else, you need to return something else then
AVERROR_BUG. EINVAL, ENOSYS or ENOTSUP.

> +
> +    s->packet = av_packet_alloc();
> +    if (!s->packet)
> +        return AVERROR(ENOMEM);
> +
> +    s->nb_context = avctx->ch_layout.nb_channels;
> +
> +    codec = avcodec_find_decoder(AV_CODEC_ID_AAC);

This may return the libfdk-aac decoder if the native ones are disabled.
It uses AV_SAMPLE_FMT_S16, whereas the native ones use a planar format,
namely AV_SAMPLE_FMT_FLTP or . The way you are forwarding the data only
works with planar formats.
IMO you should just add a configure dependency on the native decoder and
force it by using ff_aac_decoder instead of avcodec_find_decoder(). Or
maybe use ff_aac_fixed_decoder to make this codec easily testable?

> +    if (!codec)
> +        return AVERROR_BUG;
> +
> +    for (int i = 0; i < s->nb_context; i++) {
> +        s->aac_avctx[i] = avcodec_alloc_context3(codec);
> +        if (!s->aac_avctx[i])
> +            return AVERROR(ENOMEM);
> +        ret = avcodec_open2(s->aac_avctx[i], codec, NULL);
> +        if (ret < 0)
> +            return ret;
> +    }
> +
> +    avctx->sample_fmt = s->aac_avctx[0]->sample_fmt;
> +
> +    return 0;
> +}
> +
> +static int ftr_decode_frame(AVCodecContext *avctx, AVFrame *frame,
> +                            int *got_frame, AVPacket *avpkt)
> +{
> +    FTRContext *s = avctx->priv_data;
> +    GetBitContext gb;
> +    int ret, ch_offset = 0;
> +
> +    ret = init_get_bits8(&gb, avpkt->data, avpkt->size);
> +    if (ret < 0)
> +        return ret;
> +
> +    frame->nb_samples = 0;
> +
> +    for (int i = 0; i < s->nb_context; i++) {
> +        AVCodecContext *codec_avctx = s->aac_avctx[i];
> +        GetBitContext gb2 = gb;
> +        AACADTSHeaderInfo hdr_info;
> +        AVFrame *iframe = NULL;
> +        int size;
> +
> +        if (get_bits_left(&gb) < 64)
> +            return AVERROR_INVALIDDATA;
> +
> +        memset(&hdr_info, 0, sizeof(hdr_info));
> +
> +        size = ff_adts_header_parse(&gb2, &hdr_info);
> +        if (size <= 0 || size * 8 > get_bits_left(&gb))
> +            return AVERROR_INVALIDDATA;
> +
> +        if (size > s->packet->size) {
> +            if (s->packet->size == 0) {
> +                ret = av_new_packet(s->packet, size);
> +            } else {
> +                ret = av_grow_packet(s->packet, size - s->packet->size);
> +            }

This branch seems superfluous: av_grow_packet() can handle blank packets
just fine.

> +            if (ret < 0)
> +                return ret;
> +        }
> +
> +        ret = av_packet_make_writable(s->packet);
> +        if (ret < 0)
> +            return ret;
> +
> +        memcpy(s->packet->data, avpkt->data + (get_bits_count(&gb) >> 3), size);
> +        s->packet->size = size;
> +
> +        if (size > 12) {
> +            uint8_t *buf = s->packet->data;
> +
> +            if (buf[3] & 0x20) {

Does this happen often? If not, then you can just reuse the given data
(you just need to set pkt->data and size).

> +                int tmp = buf[8];
> +                buf[ 9] = ~buf[9];
> +                buf[11] = ~buf[11];
> +                buf[12] = ~buf[12];
> +                buf[ 8] = ~buf[10];
> +                buf[10] = ~tmp;
> +            }
> +        }
> +
> +        ret = avcodec_send_packet(codec_avctx, s->packet);
> +        if (ret < 0) {
> +            av_log(avctx, AV_LOG_ERROR, "Error submitting a packet for decoding\n");
> +            return ret;
> +        }
> +
> +        iframe = av_frame_alloc();

There is no reason to allocate this temp frame in a loop; it can be
allocated during init just like the temp packet.

> +        if (!iframe)
> +            return AVERROR(ENOMEM);
> +
> +        ret = avcodec_receive_frame(codec_avctx, iframe);
> +        if (ret < 0) {
> +            av_frame_free(&iframe);
> +            return ret;
> +        }
> +
> +        if (!avctx->sample_rate) {
> +            avctx->sample_rate = codec_avctx->sample_rate;
> +        } else {
> +            if (avctx->sample_rate != codec_avctx->sample_rate) {
> +                av_frame_free(&iframe);
> +                return AVERROR_INVALIDDATA;
> +            }
> +        }
> +
> +        if (!frame->nb_samples) {
> +            frame->nb_samples = iframe->nb_samples;
> +            if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
> +                av_frame_free(&iframe);
> +                return ret;
> +            }
> +        } else {
> +            if (frame->nb_samples != iframe->nb_samples) {
> +                av_frame_free(&iframe);
> +                return AVERROR_INVALIDDATA;
> +            }
> +        }
> +
> +        skip_bits_long(&gb, size * 8);
> +
> +        if (ch_offset + iframe->ch_layout.nb_channels > avctx->ch_layout.nb_channels) {
> +            av_frame_free(&iframe);
> +            return AVERROR_INVALIDDATA;
> +        }
> +
> +        for (int ch = 0; ch < iframe->ch_layout.nb_channels; ch++) {
> +            memcpy(frame->extended_data[ch_offset + ch], iframe->extended_data[ch], sizeof(float) * iframe->nb_samples);

One could ref the corresponding buffers; but this would cause problems
with the DR1 flag. I wonder whether we can simply forward get_buffer2 to
the child contexts and keep DR1. (This presumes that the used AAC
decoder has the DR1 flag set, which is true for the native one.)

> +        }
> +
> +        ch_offset += iframe->ch_layout.nb_channels;
> +
> +        av_frame_free(&iframe);
> +
> +        if (ch_offset >= avctx->ch_layout.nb_channels)
> +            break;
> +    }
> +
> +    *got_frame = 1;
> +
> +    return get_bits_count(&gb) >> 3;
> +}
> +
> +static void ftr_flush(AVCodecContext *avctx)
> +{
> +    FTRContext *s = avctx->priv_data;
> +
> +    for (int i = 0; i < s->nb_context; i++)
> +        avcodec_flush_buffers(s->aac_avctx[i]);
> +}
> +
> +static av_cold int ftr_close(AVCodecContext *avctx)
> +{
> +    FTRContext *s = avctx->priv_data;
> +
> +    for (int i = 0; i < s->nb_context; i++)
> +        avcodec_free_context(&s->aac_avctx[i]);
> +    av_packet_free(&s->packet);
> +
> +    return 0;
> +}
> +
> +const FFCodec ff_ftr_decoder = {
> +    .p.name         = "ftr",
> +    .p.long_name    = NULL_IF_CONFIG_SMALL("FTR Voice"),
> +    .p.type         = AVMEDIA_TYPE_AUDIO,
> +    .p.id           = AV_CODEC_ID_FTR,
> +    .init           = ftr_init,
> +    FF_CODEC_DECODE_CB(ftr_decode_frame),
> +    .close          = ftr_close,
> +    .flush          = ftr_flush,
> +    .priv_data_size = sizeof(FTRContext),
> +    .p.capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
> +    .caps_internal  = FF_CODEC_CAP_INIT_CLEANUP,
> +};
Jean-Baptiste Kempf Aug. 31, 2022, 8:25 p.m. UTC | #3
On Wed, 31 Aug 2022, at 18:42, Paul B Mahol wrote:
> Patch attached.

gg
Paul B Mahol Aug. 31, 2022, 9:22 p.m. UTC | #4
New patch updated from feedback received.
Paul B Mahol Aug. 31, 2022, 9:23 p.m. UTC | #5
On Wed, Aug 31, 2022 at 9:15 PM Andreas Rheinhardt <
andreas.rheinhardt@outlook.com> wrote:

> Paul B Mahol:
> > diff --git a/libavcodec/ftr.c b/libavcodec/ftr.c
> > new file mode 100644
> > index 0000000000..03d490a0c9
> > --- /dev/null
> > +++ b/libavcodec/ftr.c
> > @@ -0,0 +1,217 @@
> > +/*
> > + * 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 "adts_header.h"
> > +#include "avcodec.h"
> > +#include "codec_internal.h"
> > +#include "get_bits.h"
> > +#include "internal.h"
>
> You seem to not have rebased your patch upon master: ff_get_buffer() is
> now in decode.h and this won't compile; including internal.h seems
> superfluous now.
>
> > +
> > +typedef struct FTRContext {
> > +    AVCodecContext *aac_avctx[64];   // wrapper context for AAC
> > +    int nb_context;
> > +    AVPacket *packet;
> > +} FTRContext;
> > +
> > +static av_cold int ftr_init(AVCodecContext *avctx)
> > +{
> > +    FTRContext *s = avctx->priv_data;
> > +    const AVCodec *codec;
> > +    int ret;
> > +
> > +    if (avctx->ch_layout.nb_channels > 64 ||
> > +        avctx->ch_layout.nb_channels <= 0)
> > +        return AVERROR_BUG;
>
> I don't see what is supposed to limit nb_channels to 64. If it isn't
> checked somewhere else, you need to return something else then
> AVERROR_BUG. EINVAL, ENOSYS or ENOTSUP.
>
> > +
> > +    s->packet = av_packet_alloc();
> > +    if (!s->packet)
> > +        return AVERROR(ENOMEM);
> > +
> > +    s->nb_context = avctx->ch_layout.nb_channels;
> > +
> > +    codec = avcodec_find_decoder(AV_CODEC_ID_AAC);
>
> This may return the libfdk-aac decoder if the native ones are disabled.
> It uses AV_SAMPLE_FMT_S16, whereas the native ones use a planar format,
> namely AV_SAMPLE_FMT_FLTP or . The way you are forwarding the data only
> works with planar formats.
> IMO you should just add a configure dependency on the native decoder and
> force it by using ff_aac_decoder instead of avcodec_find_decoder(). Or
> maybe use ff_aac_fixed_decoder to make this codec easily testable?
>
> > +    if (!codec)
> > +        return AVERROR_BUG;
> > +
> > +    for (int i = 0; i < s->nb_context; i++) {
> > +        s->aac_avctx[i] = avcodec_alloc_context3(codec);
> > +        if (!s->aac_avctx[i])
> > +            return AVERROR(ENOMEM);
> > +        ret = avcodec_open2(s->aac_avctx[i], codec, NULL);
> > +        if (ret < 0)
> > +            return ret;
> > +    }
> > +
> > +    avctx->sample_fmt = s->aac_avctx[0]->sample_fmt;
> > +
> > +    return 0;
> > +}
> > +
> > +static int ftr_decode_frame(AVCodecContext *avctx, AVFrame *frame,
> > +                            int *got_frame, AVPacket *avpkt)
> > +{
> > +    FTRContext *s = avctx->priv_data;
> > +    GetBitContext gb;
> > +    int ret, ch_offset = 0;
> > +
> > +    ret = init_get_bits8(&gb, avpkt->data, avpkt->size);
> > +    if (ret < 0)
> > +        return ret;
> > +
> > +    frame->nb_samples = 0;
> > +
> > +    for (int i = 0; i < s->nb_context; i++) {
> > +        AVCodecContext *codec_avctx = s->aac_avctx[i];
> > +        GetBitContext gb2 = gb;
> > +        AACADTSHeaderInfo hdr_info;
> > +        AVFrame *iframe = NULL;
> > +        int size;
> > +
> > +        if (get_bits_left(&gb) < 64)
> > +            return AVERROR_INVALIDDATA;
> > +
> > +        memset(&hdr_info, 0, sizeof(hdr_info));
> > +
> > +        size = ff_adts_header_parse(&gb2, &hdr_info);
> > +        if (size <= 0 || size * 8 > get_bits_left(&gb))
> > +            return AVERROR_INVALIDDATA;
> > +
> > +        if (size > s->packet->size) {
> > +            if (s->packet->size == 0) {
> > +                ret = av_new_packet(s->packet, size);
> > +            } else {
> > +                ret = av_grow_packet(s->packet, size - s->packet->size);
> > +            }
>
> This branch seems superfluous: av_grow_packet() can handle blank packets
> just fine.
>
> > +            if (ret < 0)
> > +                return ret;
> > +        }
> > +
> > +        ret = av_packet_make_writable(s->packet);
> > +        if (ret < 0)
> > +            return ret;
> > +
> > +        memcpy(s->packet->data, avpkt->data + (get_bits_count(&gb) >>
> 3), size);
> > +        s->packet->size = size;
> > +
> > +        if (size > 12) {
> > +            uint8_t *buf = s->packet->data;
> > +
> > +            if (buf[3] & 0x20) {
>
> Does this happen often? If not, then you can just reuse the given data
> (you just need to set pkt->data and size).
>

It happens almost always.


>
> > +                int tmp = buf[8];
> > +                buf[ 9] = ~buf[9];
> > +                buf[11] = ~buf[11];
> > +                buf[12] = ~buf[12];
> > +                buf[ 8] = ~buf[10];
> > +                buf[10] = ~tmp;
> > +            }
> > +        }
> > +
> > +        ret = avcodec_send_packet(codec_avctx, s->packet);
> > +        if (ret < 0) {
> > +            av_log(avctx, AV_LOG_ERROR, "Error submitting a packet for
> decoding\n");
> > +            return ret;
> > +        }
> > +
> > +        iframe = av_frame_alloc();
>
> There is no reason to allocate this temp frame in a loop; it can be
> allocated during init just like the temp packet.
>
> > +        if (!iframe)
> > +            return AVERROR(ENOMEM);
> > +
> > +        ret = avcodec_receive_frame(codec_avctx, iframe);
> > +        if (ret < 0) {
> > +            av_frame_free(&iframe);
> > +            return ret;
> > +        }
> > +
> > +        if (!avctx->sample_rate) {
> > +            avctx->sample_rate = codec_avctx->sample_rate;
> > +        } else {
> > +            if (avctx->sample_rate != codec_avctx->sample_rate) {
> > +                av_frame_free(&iframe);
> > +                return AVERROR_INVALIDDATA;
> > +            }
> > +        }
> > +
> > +        if (!frame->nb_samples) {
> > +            frame->nb_samples = iframe->nb_samples;
> > +            if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
> > +                av_frame_free(&iframe);
> > +                return ret;
> > +            }
> > +        } else {
> > +            if (frame->nb_samples != iframe->nb_samples) {
> > +                av_frame_free(&iframe);
> > +                return AVERROR_INVALIDDATA;
> > +            }
> > +        }
> > +
> > +        skip_bits_long(&gb, size * 8);
> > +
> > +        if (ch_offset + iframe->ch_layout.nb_channels >
> avctx->ch_layout.nb_channels) {
> > +            av_frame_free(&iframe);
> > +            return AVERROR_INVALIDDATA;
> > +        }
> > +
> > +        for (int ch = 0; ch < iframe->ch_layout.nb_channels; ch++) {
> > +            memcpy(frame->extended_data[ch_offset + ch],
> iframe->extended_data[ch], sizeof(float) * iframe->nb_samples);
>
> One could ref the corresponding buffers; but this would cause problems
> with the DR1 flag. I wonder whether we can simply forward get_buffer2 to
> the child contexts and keep DR1. (This presumes that the used AAC
> decoder has the DR1 flag set, which is true for the native one.)
>
> > +        }
> > +
> > +        ch_offset += iframe->ch_layout.nb_channels;
> > +
> > +        av_frame_free(&iframe);
> > +
> > +        if (ch_offset >= avctx->ch_layout.nb_channels)
> > +            break;
> > +    }
> > +
> > +    *got_frame = 1;
> > +
> > +    return get_bits_count(&gb) >> 3;
> > +}
> > +
> > +static void ftr_flush(AVCodecContext *avctx)
> > +{
> > +    FTRContext *s = avctx->priv_data;
> > +
> > +    for (int i = 0; i < s->nb_context; i++)
> > +        avcodec_flush_buffers(s->aac_avctx[i]);
> > +}
> > +
> > +static av_cold int ftr_close(AVCodecContext *avctx)
> > +{
> > +    FTRContext *s = avctx->priv_data;
> > +
> > +    for (int i = 0; i < s->nb_context; i++)
> > +        avcodec_free_context(&s->aac_avctx[i]);
> > +    av_packet_free(&s->packet);
> > +
> > +    return 0;
> > +}
> > +
> > +const FFCodec ff_ftr_decoder = {
> > +    .p.name         = "ftr",
> > +    .p.long_name    = NULL_IF_CONFIG_SMALL("FTR Voice"),
> > +    .p.type         = AVMEDIA_TYPE_AUDIO,
> > +    .p.id           = AV_CODEC_ID_FTR,
> > +    .init           = ftr_init,
> > +    FF_CODEC_DECODE_CB(ftr_decode_frame),
> > +    .close          = ftr_close,
> > +    .flush          = ftr_flush,
> > +    .priv_data_size = sizeof(FTRContext),
> > +    .p.capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
> > +    .caps_internal  = FF_CODEC_CAP_INIT_CLEANUP,
> > +};
>
> _______________________________________________
> ffmpeg-devel mailing list
> ffmpeg-devel@ffmpeg.org
> https://ffmpeg.org/mailman/listinfo/ffmpeg-devel
>
> To unsubscribe, visit link above, or email
> ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe".
>
Paul B Mahol Sept. 22, 2022, 9:09 a.m. UTC | #6
On 8/31/22, Paul B Mahol <onemda@gmail.com> wrote:
> New patch updated from feedback received.
>

Will apply soon.
Anton Khirnov Sept. 22, 2022, 1:21 p.m. UTC | #7
Quoting Paul B Mahol (2022-09-22 11:09:03)
> On 8/31/22, Paul B Mahol <onemda@gmail.com> wrote:
> > New patch updated from feedback received.
> >
> 
> Will apply soon.

This needs tests.
diff mbox series

Patch

From d56c7b938f5a1a291f7a46b0d06ed45f6e723b82 Mon Sep 17 00:00:00 2001
From: Paul B Mahol <onemda@gmail.com>
Date: Tue, 30 Aug 2022 17:14:46 +0200
Subject: [PATCH] avcodec: add FTR audio decoder

Signed-off-by: Paul B Mahol <onemda@gmail.com>
---
 libavcodec/Makefile     |   2 +
 libavcodec/allcodecs.c  |   1 +
 libavcodec/codec_desc.c |   7 ++
 libavcodec/codec_id.h   |   1 +
 libavcodec/ftr.c        | 217 ++++++++++++++++++++++++++++++++++++++++
 libavcodec/ftr_parser.c | 108 ++++++++++++++++++++
 libavcodec/parsers.c    |   1 +
 libavcodec/utils.c      |   1 +
 libavformat/avidec.c    |   5 +-
 libavformat/riff.c      |   3 +
 10 files changed, 345 insertions(+), 1 deletion(-)
 create mode 100644 libavcodec/ftr.c
 create mode 100644 libavcodec/ftr_parser.c

diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index cb80f73d99..8ff9588013 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -359,6 +359,7 @@  OBJS-$(CONFIG_FMVC_DECODER)            += fmvc.o
 OBJS-$(CONFIG_FOURXM_DECODER)          += 4xm.o
 OBJS-$(CONFIG_FRAPS_DECODER)           += fraps.o
 OBJS-$(CONFIG_FRWU_DECODER)            += frwu.o
+OBJS-$(CONFIG_FTR_DECODER)             += ftr.o
 OBJS-$(CONFIG_G2M_DECODER)             += g2meet.o elsdec.o mjpegdec_common.o
 OBJS-$(CONFIG_G723_1_DECODER)          += g723_1dec.o g723_1.o \
                                           acelp_vectors.o celp_filters.o celp_math.o
@@ -1130,6 +1131,7 @@  OBJS-$(CONFIG_DVBSUB_PARSER)           += dvbsub_parser.o
 OBJS-$(CONFIG_DVD_NAV_PARSER)          += dvd_nav_parser.o
 OBJS-$(CONFIG_DVDSUB_PARSER)           += dvdsub_parser.o
 OBJS-$(CONFIG_FLAC_PARSER)             += flac_parser.o flacdata.o flac.o
+OBJS-$(CONFIG_FTR_PARSER)              += ftr_parser.o
 OBJS-$(CONFIG_G723_1_PARSER)           += g723_1_parser.o
 OBJS-$(CONFIG_G729_PARSER)             += g729_parser.o
 OBJS-$(CONFIG_GIF_PARSER)              += gif_parser.o
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 6939a4e25f..f7631cd497 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -466,6 +466,7 @@  extern const FFCodec ff_fastaudio_decoder;
 extern const FFCodec ff_ffwavesynth_decoder;
 extern const FFCodec ff_flac_encoder;
 extern const FFCodec ff_flac_decoder;
+extern const FFCodec ff_ftr_decoder;
 extern const FFCodec ff_g723_1_encoder;
 extern const FFCodec ff_g723_1_decoder;
 extern const FFCodec ff_g729_decoder;
diff --git a/libavcodec/codec_desc.c b/libavcodec/codec_desc.c
index 06dfe55d0f..d6523845ea 100644
--- a/libavcodec/codec_desc.c
+++ b/libavcodec/codec_desc.c
@@ -3290,6 +3290,13 @@  static const AVCodecDescriptor codec_descriptors[] = {
         .long_name = NULL_IF_CONFIG_SMALL("DFPWM (Dynamic Filter Pulse Width Modulation)"),
         .props     = AV_CODEC_PROP_LOSSY,
     },
+    {
+        .id        = AV_CODEC_ID_FTR,
+        .type      = AVMEDIA_TYPE_AUDIO,
+        .name      = "ftr",
+        .long_name = NULL_IF_CONFIG_SMALL("FTR Voice"),
+        .props     = AV_CODEC_PROP_INTRA_ONLY | AV_CODEC_PROP_LOSSY,
+    },
 
     /* subtitle codecs */
     {
diff --git a/libavcodec/codec_id.h b/libavcodec/codec_id.h
index 2247bc0309..dc8b30eb93 100644
--- a/libavcodec/codec_id.h
+++ b/libavcodec/codec_id.h
@@ -527,6 +527,7 @@  enum AVCodecID {
     AV_CODEC_ID_FASTAUDIO,
     AV_CODEC_ID_MSNSIREN,
     AV_CODEC_ID_DFPWM,
+    AV_CODEC_ID_FTR,
 
     /* subtitle codecs */
     AV_CODEC_ID_FIRST_SUBTITLE = 0x17000,          ///< A dummy ID pointing at the start of subtitle codecs.
diff --git a/libavcodec/ftr.c b/libavcodec/ftr.c
new file mode 100644
index 0000000000..03d490a0c9
--- /dev/null
+++ b/libavcodec/ftr.c
@@ -0,0 +1,217 @@ 
+/*
+ * 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 "adts_header.h"
+#include "avcodec.h"
+#include "codec_internal.h"
+#include "get_bits.h"
+#include "internal.h"
+
+typedef struct FTRContext {
+    AVCodecContext *aac_avctx[64];   // wrapper context for AAC
+    int nb_context;
+    AVPacket *packet;
+} FTRContext;
+
+static av_cold int ftr_init(AVCodecContext *avctx)
+{
+    FTRContext *s = avctx->priv_data;
+    const AVCodec *codec;
+    int ret;
+
+    if (avctx->ch_layout.nb_channels > 64 ||
+        avctx->ch_layout.nb_channels <= 0)
+        return AVERROR_BUG;
+
+    s->packet = av_packet_alloc();
+    if (!s->packet)
+        return AVERROR(ENOMEM);
+
+    s->nb_context = avctx->ch_layout.nb_channels;
+
+    codec = avcodec_find_decoder(AV_CODEC_ID_AAC);
+    if (!codec)
+        return AVERROR_BUG;
+
+    for (int i = 0; i < s->nb_context; i++) {
+        s->aac_avctx[i] = avcodec_alloc_context3(codec);
+        if (!s->aac_avctx[i])
+            return AVERROR(ENOMEM);
+        ret = avcodec_open2(s->aac_avctx[i], codec, NULL);
+        if (ret < 0)
+            return ret;
+    }
+
+    avctx->sample_fmt = s->aac_avctx[0]->sample_fmt;
+
+    return 0;
+}
+
+static int ftr_decode_frame(AVCodecContext *avctx, AVFrame *frame,
+                            int *got_frame, AVPacket *avpkt)
+{
+    FTRContext *s = avctx->priv_data;
+    GetBitContext gb;
+    int ret, ch_offset = 0;
+
+    ret = init_get_bits8(&gb, avpkt->data, avpkt->size);
+    if (ret < 0)
+        return ret;
+
+    frame->nb_samples = 0;
+
+    for (int i = 0; i < s->nb_context; i++) {
+        AVCodecContext *codec_avctx = s->aac_avctx[i];
+        GetBitContext gb2 = gb;
+        AACADTSHeaderInfo hdr_info;
+        AVFrame *iframe = NULL;
+        int size;
+
+        if (get_bits_left(&gb) < 64)
+            return AVERROR_INVALIDDATA;
+
+        memset(&hdr_info, 0, sizeof(hdr_info));
+
+        size = ff_adts_header_parse(&gb2, &hdr_info);
+        if (size <= 0 || size * 8 > get_bits_left(&gb))
+            return AVERROR_INVALIDDATA;
+
+        if (size > s->packet->size) {
+            if (s->packet->size == 0) {
+                ret = av_new_packet(s->packet, size);
+            } else {
+                ret = av_grow_packet(s->packet, size - s->packet->size);
+            }
+            if (ret < 0)
+                return ret;
+        }
+
+        ret = av_packet_make_writable(s->packet);
+        if (ret < 0)
+            return ret;
+
+        memcpy(s->packet->data, avpkt->data + (get_bits_count(&gb) >> 3), size);
+        s->packet->size = size;
+
+        if (size > 12) {
+            uint8_t *buf = s->packet->data;
+
+            if (buf[3] & 0x20) {
+                int tmp = buf[8];
+                buf[ 9] = ~buf[9];
+                buf[11] = ~buf[11];
+                buf[12] = ~buf[12];
+                buf[ 8] = ~buf[10];
+                buf[10] = ~tmp;
+            }
+        }
+
+        ret = avcodec_send_packet(codec_avctx, s->packet);
+        if (ret < 0) {
+            av_log(avctx, AV_LOG_ERROR, "Error submitting a packet for decoding\n");
+            return ret;
+        }
+
+        iframe = av_frame_alloc();
+        if (!iframe)
+            return AVERROR(ENOMEM);
+
+        ret = avcodec_receive_frame(codec_avctx, iframe);
+        if (ret < 0) {
+            av_frame_free(&iframe);
+            return ret;
+        }
+
+        if (!avctx->sample_rate) {
+            avctx->sample_rate = codec_avctx->sample_rate;
+        } else {
+            if (avctx->sample_rate != codec_avctx->sample_rate) {
+                av_frame_free(&iframe);
+                return AVERROR_INVALIDDATA;
+            }
+        }
+
+        if (!frame->nb_samples) {
+            frame->nb_samples = iframe->nb_samples;
+            if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
+                av_frame_free(&iframe);
+                return ret;
+            }
+        } else {
+            if (frame->nb_samples != iframe->nb_samples) {
+                av_frame_free(&iframe);
+                return AVERROR_INVALIDDATA;
+            }
+        }
+
+        skip_bits_long(&gb, size * 8);
+
+        if (ch_offset + iframe->ch_layout.nb_channels > avctx->ch_layout.nb_channels) {
+            av_frame_free(&iframe);
+            return AVERROR_INVALIDDATA;
+        }
+
+        for (int ch = 0; ch < iframe->ch_layout.nb_channels; ch++) {
+            memcpy(frame->extended_data[ch_offset + ch], iframe->extended_data[ch], sizeof(float) * iframe->nb_samples);
+        }
+
+        ch_offset += iframe->ch_layout.nb_channels;
+
+        av_frame_free(&iframe);
+
+        if (ch_offset >= avctx->ch_layout.nb_channels)
+            break;
+    }
+
+    *got_frame = 1;
+
+    return get_bits_count(&gb) >> 3;
+}
+
+static void ftr_flush(AVCodecContext *avctx)
+{
+    FTRContext *s = avctx->priv_data;
+
+    for (int i = 0; i < s->nb_context; i++)
+        avcodec_flush_buffers(s->aac_avctx[i]);
+}
+
+static av_cold int ftr_close(AVCodecContext *avctx)
+{
+    FTRContext *s = avctx->priv_data;
+
+    for (int i = 0; i < s->nb_context; i++)
+        avcodec_free_context(&s->aac_avctx[i]);
+    av_packet_free(&s->packet);
+
+    return 0;
+}
+
+const FFCodec ff_ftr_decoder = {
+    .p.name         = "ftr",
+    .p.long_name    = NULL_IF_CONFIG_SMALL("FTR Voice"),
+    .p.type         = AVMEDIA_TYPE_AUDIO,
+    .p.id           = AV_CODEC_ID_FTR,
+    .init           = ftr_init,
+    FF_CODEC_DECODE_CB(ftr_decode_frame),
+    .close          = ftr_close,
+    .flush          = ftr_flush,
+    .priv_data_size = sizeof(FTRContext),
+    .p.capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
+    .caps_internal  = FF_CODEC_CAP_INIT_CLEANUP,
+};
diff --git a/libavcodec/ftr_parser.c b/libavcodec/ftr_parser.c
new file mode 100644
index 0000000000..58cc7d6421
--- /dev/null
+++ b/libavcodec/ftr_parser.c
@@ -0,0 +1,108 @@ 
+/*
+ * FTR parser
+ * Copyright (c) 2022 Paul B Mahol
+ *
+ * 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
+ * FTR parser
+ */
+
+#include "parser.h"
+#include "get_bits.h"
+#include "adts_header.h"
+#include "adts_parser.h"
+#include "mpeg4audio.h"
+
+typedef struct FTRParseContext {
+    ParseContext pc;
+    int skip;
+    int split;
+    int frame_index;
+} FTRParseContext;
+
+static int ftr_parse(AVCodecParserContext *s, AVCodecContext *avctx,
+                     const uint8_t **poutbuf, int *poutbuf_size,
+                     const uint8_t *buf, int buf_size)
+{
+    FTRParseContext *ftr = s->priv_data;
+    uint64_t state = ftr->pc.state64;
+    int next = END_NOT_FOUND;
+    GetBitContext bits;
+    AACADTSHeaderInfo hdr;
+    int size;
+    union {
+        uint64_t u64;
+        uint8_t  u8[8 + AV_INPUT_BUFFER_PADDING_SIZE];
+    } tmp;
+
+    *poutbuf_size = 0;
+    *poutbuf = NULL;
+
+    if (s->flags & PARSER_FLAG_COMPLETE_FRAMES) {
+        next = buf_size;
+    } else {
+        for (int i = 0; i < buf_size; i++) {
+            if (ftr->skip > 0) {
+                ftr->skip--;
+                if (ftr->skip == 0 && ftr->split) {
+                    ftr->split = 0;
+                    next = i;
+                    break;
+                } else if (ftr->skip > 0) {
+                    continue;
+                }
+            }
+
+            state = (state << 8) | buf[i];
+            tmp.u64 = av_be2ne64(state);
+            init_get_bits(&bits, tmp.u8 + 8 - AV_AAC_ADTS_HEADER_SIZE,
+                          AV_AAC_ADTS_HEADER_SIZE * 8);
+
+            if ((size = ff_adts_header_parse(&bits, &hdr)) > 0) {
+                ftr->skip = size - 6;
+                ftr->frame_index += ff_mpeg4audio_channels[hdr.chan_config];
+                if (ftr->frame_index >= avctx->ch_layout.nb_channels) {
+                    ftr->frame_index = 0;
+                    ftr->split = 1;
+                }
+            }
+        }
+
+        ftr->pc.state64 = state;
+
+        if (ff_combine_frame(&ftr->pc, next, &buf, &buf_size) < 0) {
+            *poutbuf = NULL;
+            *poutbuf_size = 0;
+            return buf_size;
+        }
+    }
+
+    *poutbuf      = buf;
+    *poutbuf_size = buf_size;
+
+    return next;
+}
+
+const AVCodecParser ff_ftr_parser = {
+    .codec_ids      = { AV_CODEC_ID_FTR },
+    .priv_data_size = sizeof(FTRParseContext),
+    .parser_parse   = ftr_parse,
+    .parser_close   = ff_parse_close,
+};
diff --git a/libavcodec/parsers.c b/libavcodec/parsers.c
index a8d52af6cb..ad72e147fd 100644
--- a/libavcodec/parsers.c
+++ b/libavcodec/parsers.c
@@ -42,6 +42,7 @@  extern const AVCodecParser ff_dvbsub_parser;
 extern const AVCodecParser ff_dvdsub_parser;
 extern const AVCodecParser ff_dvd_nav_parser;
 extern const AVCodecParser ff_flac_parser;
+extern const AVCodecParser ff_ftr_parser;
 extern const AVCodecParser ff_g723_1_parser;
 extern const AVCodecParser ff_g729_parser;
 extern const AVCodecParser ff_gif_parser;
diff --git a/libavcodec/utils.c b/libavcodec/utils.c
index e73e3a7d08..0536174f1e 100644
--- a/libavcodec/utils.c
+++ b/libavcodec/utils.c
@@ -638,6 +638,7 @@  static int get_audio_frame_duration(enum AVCodecID id, int sr, int ch, int ba,
     case AV_CODEC_ID_MP2:
     case AV_CODEC_ID_MUSEPACK7:    return 1152;
     case AV_CODEC_ID_AC3:          return 1536;
+    case AV_CODEC_ID_FTR:          return 1024;
     }
 
     if (sr > 0) {
diff --git a/libavformat/avidec.c b/libavformat/avidec.c
index 937d9e6ffb..212b154d80 100644
--- a/libavformat/avidec.c
+++ b/libavformat/avidec.c
@@ -1555,7 +1555,10 @@  resync:
             } else {
                 pkt->flags |= AV_PKT_FLAG_KEY;
             }
-            ast->frame_offset += get_duration(ast, pkt->size);
+            if (st->codecpar->codec_id == AV_CODEC_ID_FTR)
+                ast->frame_offset++;
+            else
+                ast->frame_offset += get_duration(ast, pkt->size);
         }
         ast->remaining -= err;
         if (!ast->remaining) {
diff --git a/libavformat/riff.c b/libavformat/riff.c
index 6c06ad2d60..59fc7abcbd 100644
--- a/libavformat/riff.c
+++ b/libavformat/riff.c
@@ -558,6 +558,7 @@  const AVCodecTag ff_codec_wav_tags[] = {
     { AV_CODEC_ID_WMALOSSLESS,     0x0163 },
     { AV_CODEC_ID_XMA1,            0x0165 },
     { AV_CODEC_ID_XMA2,            0x0166 },
+    { AV_CODEC_ID_FTR,             0x0180 },
     { AV_CODEC_ID_ADPCM_CT,        0x0200 },
     { AV_CODEC_ID_DVAUDIO,         0x0215 },
     { AV_CODEC_ID_DVAUDIO,         0x0216 },
@@ -583,8 +584,10 @@  const AVCodecTag ff_codec_wav_tags[] = {
     { AV_CODEC_ID_PCM_MULAW,       0x6c75 },
     { AV_CODEC_ID_AAC,             0x706d },
     { AV_CODEC_ID_AAC,             0x4143 },
+    { AV_CODEC_ID_FTR,             0x4180 },
     { AV_CODEC_ID_XAN_DPCM,        0x594a },
     { AV_CODEC_ID_G729,            0x729A },
+    { AV_CODEC_ID_FTR,             0x8180 },
     { AV_CODEC_ID_G723_1,          0xA100 }, /* Comverse Infosys Ltd. G723 1 */
     { AV_CODEC_ID_AAC,             0xA106 },
     { AV_CODEC_ID_SPEEX,           0xA109 },
-- 
2.37.2