Re: [FFmpeg-devel] [PATCH] Whisper audio filter

[Marvin Scholz <epirat07-at-gmail.com@ffmpeg.org>]

On 9 Jul 2025, at 9:23, Vittorio Palmisano wrote:

> It adds a new audio filter for running audio transcriptions with the whisper model.
> Documentation and examples are included into the patch.
>
> Signed-off-by: Vittorio Palmisano <vpalmisano@gmail.com>
> ---
>  configure                |   5 +
>  doc/filters.texi         | 101 ++++++++
>  libavfilter/Makefile     |   2 +
>  libavfilter/af_whisper.c | 494 +++++++++++++++++++++++++++++++++++++++
>  libavfilter/allfilters.c |   2 +
>  5 files changed, 604 insertions(+)
>  create mode 100644 libavfilter/af_whisper.c
>
> diff --git a/configure b/configure
> index 2ccafe7c20..573dfc67dc 100755
> --- a/configure
> +++ b/configure
> @@ -337,6 +337,7 @@ External library support:
>    --enable-vapoursynth     enable VapourSynth demuxer [no]
>    --disable-xlib           disable xlib [autodetect]
>    --disable-zlib           disable zlib [autodetect]
> +  --enable-whisper         enable whisper filter [no]
>
>    The following libraries provide various hardware acceleration features:
>    --disable-amf            disable AMF video encoding code [autodetect]
> @@ -2003,6 +2004,7 @@ EXTERNAL_LIBRARY_LIST="
>      pocketsphinx
>      vapoursynth
>      vulkan_static
> +    whisper
>  "
>
>  HWACCEL_AUTODETECT_LIBRARY_LIST="
> @@ -4059,6 +4061,7 @@ xstack_qsv_filter_deps="libmfx"
>  xstack_qsv_filter_select="qsvvpp"
>  pad_vaapi_filter_deps="vaapi_1"
>  drawbox_vaapi_filter_deps="vaapi_1"
> +whisper_filter_deps="whisper"
>
>  # examples
>  avio_http_serve_files_deps="avformat avutil fork"
> @@ -7108,6 +7111,8 @@ enabled libvo_amrwbenc    && require libvo_amrwbenc vo-amrwbenc/enc_if.h E_IF_in
>  enabled libvorbis         && require_pkg_config libvorbis vorbis vorbis/codec.h vorbis_info_init &&
>                               require_pkg_config libvorbisenc vorbisenc vorbis/vorbisenc.h vorbis_encode_init
>
> +enabled whisper           && require_pkg_config whisper "whisper >= 1.7.5" whisper.h whisper_init_from_file_with_params
> +
>  enabled libvpx            && {
>      enabled libvpx_vp8_decoder && {
>          check_pkg_config libvpx_vp8_decoder "vpx >= 1.4.0" "vpx/vpx_decoder.h vpx/vp8dx.h" vpx_codec_vp8_dx ||
> diff --git a/doc/filters.texi b/doc/filters.texi
> index ed2956fe75..c00e73478f 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -7682,6 +7682,107 @@ There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6 dB, etc.
>  In other words, raising the volume by +4 dB does not cause any clipping,
>  raising it by +5 dB causes clipping for 6 samples, etc.
>
> +@anchor{whisper}
> +@section whisper
> +
> +It runs a automatic speech recognition using the OpenAI’s Whisper model.
> +
> +It requires the whisper.cpp library (https://github.com/ggml-org/whisper.cpp)
> +as a pre-requisite. After installing the library it can be enabled using:
> +@code{./configure --enable-whisper}.
> +
> +The filter has following options:
> +
> +@table @option
> +@item model
> +The file path of the downloaded whisper.cpp model (mandatory).
> +
> +@item language
> +The language to use for transcription ('auto' for auto-detect).
> +Default value: @code{"auto"}
> +
> +@item queue
> +The maximum size in milliseconds that will be queued into the filter before
> +processing the audio with whisper
> +Default value: @code{"3000"}
> +
> +@item use_gpu
> +If the GPU support should be enabled.
> +Default value: @code{"true"}
> +
> +@item gpu_device
> +The GPU device to use.
> +Default value: @code{"0"}
> +
> +@item destination
> +If set, the transcription output will be sent to the specified file or URL
> +(use one of the FFmpeg AVIO protocols); otherwise, the output will be logged as
> +info messages.
> +The output will also be set in the "lavfi.whisper.text" frame metadata.
> +
> +@item format
> +The destination format string; it could be "text" (only the transcribed text
> +will be sent to the destination), "srt" (subtitle format) or "json".
> +Default value: @code{"text"}
> +
> +@item vad_model
> +Path to the VAD model file. If set, the filter will load an additional voice
> +activity detection module (https://github.com/snakers4/silero-vad) that will be
> +used to fragment the audio queue; use this option setting a valid path obtained
> +from the whisper.cpp repository (e.g. "../whisper.cpp/models/ggml-silero-v5.1.2.bin")
> +and increase the queue parameter to an higher value (e.g. 10000)
> +
> +@item vad_threshold
> +The VAD threshold to use.
> +Default value: @code{"0.5"}
> +
> +@item vad_min_speech_duration
> +The minimum VAD speaking duration in milliseconds.
> +Default value: @code{"50"}
> +
> +@item vad_min_silence_duration
> +The minimum VAD silence duration in milliseconds.
> +Default value: @code{"500"}
> +
> +@end table
> +
> +@subsection Examples
> +@itemize
> +
> +@item
> +Run a transcription with srt file generation:
> +@example
> +ffmpeg -i input.mp4 -vn -af "aformat=sample_rates=16000:channel_layouts=mono,whisper=model=../whisper.cpp/models/ggml-base.en.bin\
> +:language=en\
> +:queue=3000\
> +:destination=output.srt\
> +:format=srt" -f null -
> +@end example
> +
> +@item
> +Run a transcription and send the output in JSON format to an HTTP service:
> +@example
> +ffmpeg -i input.mp4 -vn -af "aformat=sample_rates=16000:channel_layouts=mono,whisper=model=../whisper.cpp/models/ggml-base.en.bin\
> +:language=en\
> +:queue=3000\
> +:destination=http\\://localhost\\:3000\
> +:format=json' -f null -
> +@end example
> +
> +@item
> +Transcribe the microphone input using the VAD option:
> +@example
> +ffmpeg -loglevel warning -f pulse -i default \
> +-af 'highpass=f=200,lowpass=f=3000,aformat=sample_rates=16000:channel_layouts=mono,whisper=model=../whisper.cpp/models/ggml-medium.bin\
> +:language=en\
> +:queue=10000\
> +:destination=-\
> +:format=json\
> +:vad_model=../whisper.cpp/models/ggml-silero-v5.1.2.bin' -f null -
> +@end example
> +
> +@end itemize
> +
>  @c man end AUDIO FILTERS
>
>  @chapter Audio Sources
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 9e9153f5b0..e133422ca4 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -188,6 +188,8 @@ OBJS-$(CONFIG_HILBERT_FILTER)                += asrc_hilbert.o
>  OBJS-$(CONFIG_SINC_FILTER)                   += asrc_sinc.o
>  OBJS-$(CONFIG_SINE_FILTER)                   += asrc_sine.o
>
> +OBJS-$(CONFIG_WHISPER_FILTER)                += af_whisper.o
> +
>  OBJS-$(CONFIG_ANULLSINK_FILTER)              += asink_anullsink.o
>
>  # video filters
> diff --git a/libavfilter/af_whisper.c b/libavfilter/af_whisper.c
> new file mode 100644
> index 0000000000..7bfdd3a9dc
> --- /dev/null
> +++ b/libavfilter/af_whisper.c
> @@ -0,0 +1,494 @@
> +/*
> + * Copyright (c) 2025 Vittorio Palmisano
> + *
> + * 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 <stdio.h>
> +#include <stdint.h>
> +#include <stdlib.h>
> +
> +#include "libavutil/avutil.h"
> +#include "libavutil/opt.h"
> +#include "libavutil/channel_layout.h"
> +#include "libavutil/samplefmt.h"
> +#include "libavfilter/avfilter.h"
> +#include "libavfilter/audio.h"
> +#include "libavutil/mem.h"
> +#include "libavutil/avstring.h"
> +#include "libavutil/internal.h"
> +#include "libavformat/avio.h"
> +#include "libavutil/thread.h"
> +
> +#include "formats.h"
> +
> +#include "whisper.h"
> +
> +typedef struct WhisperContext
> +{
> +    const AVClass *class;
> +    char *model_path;
> +    char *language;
> +    bool use_gpu;
> +    int gpu_device;
> +    char *vad_model_path;
> +    float vad_threshold;
> +    int vad_min_speech_duration;
> +    int vad_min_silence_duration;
> +
> +    int queue;
> +    char *destination;
> +    char *format;
> +
> +    struct whisper_context *ctx_wsp;
> +    struct whisper_state *whisper_state;
> +    struct whisper_vad_context *ctx_vad;
> +    struct whisper_vad_params vad_params;
> +
> +    float *audio_buffer;
> +    int audio_buffer_queue_size;
> +    int audio_buffer_fill_size;
> +    int audio_buffer_vad_size;
> +
> +    int eof;
> +    int64_t next_pts;
> +
> +    AVIOContext *avio_context;
> +    int index;
> +    int64_t timestamp;
> +} WhisperContext;
> +
> +static void cb_log_disable(enum ggml_log_level, const char *, void *) {}
> +
> +static int init(AVFilterContext *ctx)
> +{
> +    WhisperContext *wctx = ctx->priv;
> +
> +    ggml_backend_load_all();
> +    whisper_log_set(cb_log_disable, NULL);
> +
> +    // Init whisper context
> +    if (!wctx->model_path)
> +    {
> +        av_log(ctx, AV_LOG_ERROR, "No whisper model path specified. Use the 'model' option.\n");
> +        return AVERROR(EINVAL);
> +    }
> +
> +    struct whisper_context_params params = whisper_context_default_params();
> +    params.use_gpu = wctx->use_gpu;
> +    params.gpu_device = wctx->gpu_device;
> +
> +    wctx->ctx_wsp = whisper_init_from_file_with_params(wctx->model_path, params);
> +    if (wctx->ctx_wsp == NULL)
> +    {
> +        av_log(ctx, AV_LOG_ERROR, "Failed to initialize whisper context from model: %s\n", wctx->model_path);
> +        return AVERROR(EIO);
> +    }
> +
> +    wctx->whisper_state = whisper_init_state(wctx->ctx_wsp);
> +    if (wctx->whisper_state == NULL)
> +    {
> +        av_log(ctx, AV_LOG_ERROR, "Failed to get whisper state from context\n");
> +        whisper_free(wctx->ctx_wsp);
> +        wctx->ctx_wsp = NULL;
> +        return AVERROR(EIO);
> +    }
> +
> +    // Init VAD model context
> +    if (wctx->vad_model_path)
> +    {
> +        struct whisper_vad_context_params ctx_params = whisper_vad_default_context_params();
> +        ctx_params.n_threads = 4;
> +        // ctx_params.use_gpu = wctx->use_gpu; TODO (see: whisper_vad_init_context)
> +        ctx_params.gpu_device = wctx->gpu_device;
> +        wctx->ctx_vad = whisper_vad_init_from_file_with_params(
> +            wctx->vad_model_path,
> +            ctx_params);
> +
> +        wctx->vad_params = whisper_vad_default_params();
> +        wctx->vad_params.threshold = wctx->vad_threshold;
> +        wctx->vad_params.min_speech_duration_ms = wctx->vad_min_speech_duration;
> +        wctx->vad_params.min_silence_duration_ms = wctx->vad_min_silence_duration;
> +        wctx->vad_params.max_speech_duration_s = (float)(wctx->audio_buffer_queue_size / 1000.0f);
> +        wctx->vad_params.speech_pad_ms = 0;
> +        wctx->vad_params.samples_overlap = 0;
> +    }
> +
> +    // Init buffer
> +    wctx->audio_buffer_queue_size = WHISPER_SAMPLE_RATE * wctx->queue / 1000;
> +    wctx->audio_buffer = av_malloc(wctx->audio_buffer_queue_size * sizeof(float));
> +    if (!wctx->audio_buffer)
> +    {
> +        return AVERROR(ENOMEM);
> +    }
> +
> +    wctx->audio_buffer_fill_size = 0;
> +
> +    wctx->next_pts = AV_NOPTS_VALUE;
> +
> +    wctx->avio_context = NULL;
> +    if (wctx->destination && strcmp("", wctx->destination))
> +    {
> +        int ret = 0;
> +
> +        if (!strcmp("-", wctx->destination))
> +        {
> +            ret = avio_open(&wctx->avio_context, "pipe:1", AVIO_FLAG_WRITE);
> +        }
> +        else
> +        {
> +            ret = avio_open(&wctx->avio_context, wctx->destination, AVIO_FLAG_WRITE);
> +        }
> +
> +        if (ret < 0)
> +        {
> +            av_log(ctx, AV_LOG_ERROR, "Could not open %s: %s\n",
> +                   wctx->destination, av_err2str(ret));
> +            return ret;
> +        }
> +
> +        wctx->avio_context->direct = AVIO_FLAG_DIRECT;
> +    }
> +
> +    av_log(ctx, AV_LOG_INFO, "Whisper filter initialized: model: %s lang: %s queue: %d ms\n",
> +           wctx->model_path, wctx->language, wctx->queue);
> +
> +    return 0;
> +}
> +
> +static void uninit(AVFilterContext *ctx)
> +{
> +    WhisperContext *wctx = ctx->priv;
> +
> +    if (wctx->audio_buffer_fill_size > 0)
> +    {
> +        av_log(ctx, AV_LOG_WARNING, "Remaining audio buffer %d samples (%d seconds) after stopping\n",
> +               wctx->audio_buffer_fill_size,
> +               wctx->audio_buffer_fill_size / WHISPER_SAMPLE_RATE);
> +    }
> +
> +    if (wctx->ctx_vad)
> +    {
> +        whisper_vad_free(wctx->ctx_vad);
> +        wctx->ctx_vad = NULL;
> +    }
> +
> +    if (wctx->whisper_state)
> +    {
> +        whisper_free_state(wctx->whisper_state);
> +        wctx->whisper_state = NULL;
> +    }
> +
> +    if (wctx->ctx_wsp)
> +    {
> +        whisper_free(wctx->ctx_wsp);
> +        wctx->ctx_wsp = NULL;
> +    }
> +
> +    av_freep(&wctx->audio_buffer);
> +
> +    if (wctx->avio_context)
> +    {
> +        avio_closep(&wctx->avio_context);
> +    }
> +}
> +
> +static void run_transcription(AVFilterContext *ctx, AVDictionary **metadata, int end_pos)
> +{
> +    WhisperContext *wctx = ctx->priv;
> +    end_pos = FFMIN(end_pos, wctx->audio_buffer_fill_size);
> +
> +    if (!wctx->ctx_wsp || end_pos == 0)
> +    {
> +        return;
> +    }
> +
> +    if (!wctx->ctx_wsp)
> +    {
> +        return;
> +    }
> +
> +    float duration = (float)end_pos / WHISPER_SAMPLE_RATE;
> +
> +    av_log(ctx, AV_LOG_INFO, "run transcription %d/%d samples (%.2f seconds)...\n",
> +           end_pos, wctx->audio_buffer_fill_size,
> +           duration);
> +
> +    struct whisper_full_params params = whisper_full_default_params(WHISPER_SAMPLING_GREEDY);
> +    params.language = wctx->language;
> +    params.print_special = 0;
> +    params.print_progress = 0;
> +    params.print_realtime = 0;
> +    params.print_timestamps = 0;
> +
> +    if (whisper_full(wctx->ctx_wsp, params, wctx->audio_buffer, end_pos) != 0)
> +    {
> +        av_log(ctx, AV_LOG_ERROR, "Failed to process audio with whisper.cpp\n");
> +        return;
> +    }
> +
> +    const int n_segments = whisper_full_n_segments_from_state(wctx->whisper_state);
> +    char *segments_text = NULL;
> +
> +    for (int i = 0; i < n_segments; ++i)
> +    {
> +        const bool turn = whisper_full_get_segment_speaker_turn_next(wctx->ctx_wsp, i);
> +        const int64_t t0 = whisper_full_get_segment_t0(wctx->ctx_wsp, i) * 10;
> +        const int64_t t1 = whisper_full_get_segment_t1(wctx->ctx_wsp, i) * 10;
> +        const char *text = whisper_full_get_segment_text(wctx->ctx_wsp, i);
> +        char *text_cleaned = av_strireplace(text + 1, "[BLANK_AUDIO]", "");
> +
> +        if (av_strnlen(text_cleaned, 1) == 0)
> +        {
> +            av_free(text_cleaned);
> +            continue;
> +        }
> +        av_log(ctx, AV_LOG_INFO, "  [%ld-%ld%s]: \"%s\"\n", wctx->timestamp + t0, wctx->timestamp + t1, turn ? " (turn)" : "", text_cleaned);
> +
> +        if (segments_text)
> +        {
> +            char *new_text = av_asprintf("%s%s", segments_text, text_cleaned);
> +            av_free(segments_text);
> +            segments_text = new_text;
> +        }
> +        else
> +        {
> +            segments_text = av_strdup(text_cleaned);
> +        }
> +
> +        if (wctx->avio_context)
> +        {
> +            const int64_t start_t = wctx->timestamp + t0;
> +            const int64_t end_t = wctx->timestamp + t1;
> +            char *buf = NULL;
> +
> +            if (!av_strcasecmp(wctx->format, "srt"))
> +            {
> +                buf = av_asprintf("%d\n%02ld:%02ld:%02ld.%03ld --> %02ld:%02ld:%02ld.%03ld\n%s\n\n",
> +                                  wctx->index,
> +                                  start_t / 3600000, (start_t / 60000) % 60, (start_t / 1000) % 60, start_t % 1000,
> +                                  end_t / 3600000, (end_t / 60000) % 60, (end_t / 1000) % 60, end_t % 1000,
> +                                  text_cleaned);
> +            }
> +            else if (!av_strcasecmp(wctx->format, "json"))
> +            {
> +                buf = av_asprintf("{\"start\":%ld,\"end\":%ld,\"text\":\"%s\",\"turn\":%s}\n",
> +                                  start_t, end_t, text_cleaned, turn ? "true" : "false");
> +            }
> +            else
> +            {
> +                buf = av_strdup(text_cleaned);
> +            }
> +
> +            if (buf)
> +            {
> +                avio_write(wctx->avio_context, buf, strlen(buf));
> +                av_free(buf);
> +            }
> +        }
> +
> +        av_free(text_cleaned);
> +    }
> +
> +    wctx->index++;
> +    wctx->timestamp += (int64_t)(duration * 1000);
> +
> +    if (metadata && segments_text)
> +    {
> +        av_dict_set(metadata, "lavfi.whisper.text", segments_text, 0);
> +        char *duration_text = av_asprintf("%f", duration);
> +        av_dict_set(metadata, "lavfi.whisper.duration", duration_text, 0);
> +        av_free(duration_text);
> +    }
> +    if (segments_text)
> +    {
> +        av_free(segments_text);
> +    }
> +
> +    memcpy(wctx->audio_buffer, wctx->audio_buffer + end_pos, end_pos * sizeof(float));
> +    wctx->audio_buffer_fill_size -= end_pos;
> +    wctx->audio_buffer_vad_size = wctx->audio_buffer_fill_size;
> +}
> +
> +static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
> +{
> +    AVFilterContext *ctx = inlink->dst;
> +    WhisperContext *wctx = ctx->priv;
> +    AVFilterLink *outlink = ctx->outputs[0];
> +    AVDictionary **metadata = &frame->metadata;
> +
> +    const int samples = frame->nb_samples;
> +    const float *input_data = (const float *)frame->data[0];
> +
> +    if (wctx->audio_buffer_fill_size + samples > wctx->audio_buffer_queue_size)
> +    {
> +        run_transcription(ctx, metadata, wctx->audio_buffer_fill_size);
> +    }
> +
> +    memcpy(wctx->audio_buffer + wctx->audio_buffer_fill_size, input_data, samples * sizeof(float));
> +    wctx->audio_buffer_fill_size += samples;
> +
> +    if (wctx->ctx_vad && (wctx->audio_buffer_fill_size - wctx->audio_buffer_vad_size) >=
> +                             WHISPER_SAMPLE_RATE * (wctx->vad_min_speech_duration + wctx->vad_min_silence_duration) / 1000)
> +    {
> +        struct whisper_vad_segments *segments = whisper_vad_segments_from_samples(
> +            wctx->ctx_vad, wctx->vad_params, wctx->audio_buffer, wctx->audio_buffer_fill_size);
> +        wctx->audio_buffer_vad_size = wctx->audio_buffer_fill_size;
> +
> +        if (!segments)
> +        {
> +            av_log(ctx, AV_LOG_ERROR, "failed to detect VAD\n");
> +        }
> +        else
> +        {
> +            int n_segments = whisper_vad_segments_n_segments(segments);
> +
> +            if (n_segments > 0)
> +            {
> +                const int64_t start_ms = whisper_vad_segments_get_segment_t0(segments, n_segments - 1) * 10;
> +                const int64_t end_ms = whisper_vad_segments_get_segment_t1(segments, n_segments - 1) * 10;
> +                int end_pos = (int)(end_ms * WHISPER_SAMPLE_RATE / 1000);
> +
> +                if (end_pos < wctx->audio_buffer_fill_size)
> +                {
> +                    av_log(ctx, AV_LOG_INFO, "VAD detected %d segments, start: %ld ms, end: %ld ms (buffer: %d ms)\n",
> +                           n_segments, start_ms, end_ms, 1000 * wctx->audio_buffer_fill_size / WHISPER_SAMPLE_RATE);
> +                    run_transcription(ctx, metadata, end_pos);
> +                }
> +            }
> +
> +            whisper_vad_free_segments(segments);
> +        }
> +    }
> +    else if (wctx->audio_buffer_fill_size >= wctx->audio_buffer_queue_size)
> +    {
> +        run_transcription(ctx, metadata, wctx->audio_buffer_fill_size);
> +    }
> +
> +    wctx->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
> +    return ff_filter_frame(outlink, frame);
> +}
> +
> +static int push_last_frame(AVFilterLink *outlink)
> +{
> +    AVFilterContext *ctx = outlink->src;
> +    WhisperContext *wctx = ctx->priv;
> +    AVFrame *frame;
> +    int n_out = 1;
> +
> +    if (ctx->is_disabled || wctx->audio_buffer_fill_size == 0)
> +        return 0;
> +    frame = ff_get_audio_buffer(outlink, n_out);
> +    if (!frame)
> +        return AVERROR(ENOMEM);
> +
> +    av_samples_set_silence(frame->extended_data, 0,
> +                           n_out,
> +                           frame->ch_layout.nb_channels,
> +                           frame->format);
> +
> +    frame->pts = wctx->next_pts;
> +    if (wctx->next_pts != AV_NOPTS_VALUE)
> +        wctx->next_pts += av_rescale_q(n_out, (AVRational){1, outlink->sample_rate}, outlink->time_base);
> +
> +    run_transcription(ctx, &frame->metadata, wctx->audio_buffer_fill_size);
> +
> +    return ff_filter_frame(outlink, frame);
> +}
> +
> +static int activate(AVFilterContext *ctx)
> +{
> +    AVFilterLink *inlink = ctx->inputs[0];
> +    AVFilterLink *outlink = ctx->outputs[0];
> +    WhisperContext *wctx = ctx->priv;
> +    int64_t pts;
> +    int status;
> +
> +    FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
> +
> +    if (!wctx->eof && ff_inlink_queued_frames(inlink))
> +    {
> +        AVFrame *frame = NULL;
> +        int ret;
> +
> +        ret = ff_inlink_consume_frame(inlink, &frame);
> +        if (ret < 0)
> +            return ret;
> +        if (ret > 0)
> +            return filter_frame(inlink, frame);
> +    }
> +
> +    if (!wctx->eof && ff_inlink_acknowledge_status(inlink, &status, &pts))
> +        wctx->eof = status == AVERROR_EOF;
> +
> +    if (wctx->eof)
> +    {
> +        push_last_frame(outlink);
> +
> +        ff_outlink_set_status(outlink, AVERROR_EOF, wctx->next_pts);
> +        return 0;
> +    }
> +
> +    FF_FILTER_FORWARD_WANTED(outlink, inlink);
> +
> +    return FFERROR_NOT_READY;
> +}
> +
> +#define OFFSET(x) offsetof(WhisperContext, x)
> +#define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM
> +
> +static const AVOption whisper_options[] = {
> +    {"model", "Path to the whisper.cpp model file", OFFSET(model_path), AV_OPT_TYPE_STRING, .flags = FLAGS},
> +    {"language", "Language for transcription ('auto' for auto-detect)", OFFSET(language), AV_OPT_TYPE_STRING, {.str = "auto"}, .flags = FLAGS},
> +    {"queue", "Audio queue size in milliseconds", OFFSET(queue), AV_OPT_TYPE_INT, {.i64 = 3000}, 20, INT_MAX, .flags = FLAGS},

This should probably be AV_OPT_TYPE_DURATION.

> +    {"use_gpu", "Use GPU for processing", OFFSET(use_gpu), AV_OPT_TYPE_BOOL, {.i64 = 1}, 0, 1, .flags = FLAGS},
> +    {"gpu_device", "GPU device to use", OFFSET(gpu_device), AV_OPT_TYPE_INT, {.i64 = 0}, 0, INT_MAX, .flags = FLAGS},
> +    {"destination", "Output destination", OFFSET(destination), AV_OPT_TYPE_STRING, {.str = ""}, .flags = FLAGS},
> +    {"format", "Output format (text|srt|json)", OFFSET(format), AV_OPT_TYPE_STRING, {.str = "text"}, .flags = FLAGS},
> +    {"vad_model", "Path to the VAD model file", OFFSET(vad_model_path), AV_OPT_TYPE_STRING, .flags = FLAGS},
> +    {"vad_threshold", "VAD threshold", OFFSET(vad_threshold), AV_OPT_TYPE_FLOAT, {.dbl = 0.5}, 0.0, 1.0, .flags = FLAGS},
> +    {"vad_min_speech_duration", "Minimum speech duration in milliseconds for VAD", OFFSET(vad_min_speech_duration), AV_OPT_TYPE_INT, {.i64 = 50}, 20, INT_MAX, .flags = FLAGS},
> +    {"vad_min_silence_duration", "Minimum silence duration in milliseconds for VAD", OFFSET(vad_min_silence_duration), AV_OPT_TYPE_INT, {.i64 = 500}, 0, INT_MAX, .flags = FLAGS},

These should be AV_OPT_TYPE_DURATION too.

> +    {NULL}};
> +
> +static const AVClass whisper_class = {
> +    .class_name = "whisper",
> +    .item_name = av_default_item_name,
> +    .option = whisper_options,
> +    .version = LIBAVUTIL_VERSION_INT,
> +};
> +
> +static const AVFilterPad whisper_outputs[] = {
> +    {
> +        .name = "default",
> +        .type = AVMEDIA_TYPE_AUDIO,
> +    },
> +};
> +
> +const FFFilter ff_af_whisper = {
> +    .p.name = "whisper",
> +    .p.description = NULL_IF_CONFIG_SMALL("Transcribe audio using whisper.cpp."),
> +    .p.priv_class = &whisper_class,
> +    .p.flags = AVFILTER_FLAG_METADATA_ONLY,
> +    .init = init,
> +    .uninit = uninit,
> +    .activate = activate,
> +    .priv_size = sizeof(WhisperContext),
> +    FILTER_INPUTS(ff_audio_default_filterpad),
> +    FILTER_OUTPUTS(whisper_outputs),
> +    FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP),
> +};
> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
> index 409099bf1f..eaf0c8fe6f 100644
> --- a/libavfilter/allfilters.c
> +++ b/libavfilter/allfilters.c
> @@ -163,6 +163,8 @@ extern const FFFilter ff_af_virtualbass;
>  extern const FFFilter ff_af_volume;
>  extern const FFFilter ff_af_volumedetect;
>
> +extern const FFFilter ff_af_whisper;
> +
>  extern const FFFilter ff_asrc_aevalsrc;
>  extern const FFFilter ff_asrc_afdelaysrc;
>  extern const FFFilter ff_asrc_afireqsrc;
> -- 
> 2.43.0
>
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