[FFmpeg-devel] [PATCH 1/1] avcodec/libopusdec: Enable FEC/PLC

[Tristan Matthews]

On Thu, Feb 18, 2021 at 11:39 AM Philip-Dylan Gleonec
<philip-dylan.gleonec at savoirfairelinux.com> wrote:
>
> Here is the reworked patch properly attached.
> Sorry about the duplicate mail, I just noticed I had a mishap with my
> mail client and the previous patch was scrubbed.
>
> Signed-off-by: Philip-Dylan Gleonec <philip-dylan.gleonec at savoirfairelinux.com>
> Co-developed-by: Steinar H. Gunderson <steinar+ffmpeg at gunderson.no>
> ---
>  libavcodec/libopusdec.c | 107 +++++++++++++++++++++++++++++++++++-----
>  1 file changed, 96 insertions(+), 11 deletions(-)
>
> diff --git a/libavcodec/libopusdec.c b/libavcodec/libopusdec.c
> index 082a431c6c..3de784dfbd 100644
> --- a/libavcodec/libopusdec.c
> +++ b/libavcodec/libopusdec.c
> @@ -43,10 +43,15 @@ struct libopus_context {
>  #ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST
>      int apply_phase_inv;
>  #endif
> +    int decode_fec;
> +    int64_t expected_next_pts;
>  };
>
>  #define OPUS_HEAD_SIZE 19
>
> +// Sample rate is constant as libopus always output at 48kHz
> +const AVRational opus_timebase = { 1, 48000 };
> +
>  static av_cold int libopus_decode_init(AVCodecContext *avc)
>  {
>      struct libopus_context *opus = avc->priv_data;
> @@ -134,6 +139,8 @@ static av_cold int libopus_decode_init(AVCodecContext *avc)
>      /* Decoder delay (in samples) at 48kHz */
>      avc->delay = avc->internal->skip_samples = opus->pre_skip;
>
> +    opus->expected_next_pts = AV_NOPTS_VALUE;
> +
>      return 0;
>  }
>
> @@ -155,25 +162,102 @@ static int libopus_decode(AVCodecContext *avc, void *data,
>  {
>      struct libopus_context *opus = avc->priv_data;
>      AVFrame *frame               = data;
> -    int ret, nb_samples;
> +    uint8_t *outptr;
> +    int ret, nb_samples = 0, nb_lost_samples = 0, nb_samples_left;
> +
> +    // If FEC is enabled, calculate number of lost samples
> +    if (opus->decode_fec &&
> +        opus->expected_next_pts != AV_NOPTS_VALUE &&
> +        pkt->pts != AV_NOPTS_VALUE &&
> +        pkt->pts != opus->expected_next_pts) {
> +        // Cap at recovering 120 ms of lost audio.
> +        nb_lost_samples = pkt->pts - opus->expected_next_pts;
> +        nb_lost_samples = FFMIN(nb_lost_samples, MAX_FRAME_SIZE);
> +        // pts is expressed in ms for some containers (e.g. mkv)
> +        // FEC only works for SILK frames (> 10ms)
> +        // Detect if nb_lost_samples is in ms, and convert in samples if it is
> +        if (nb_lost_samples > 0) {
> +            if (avc->pkt_timebase.den != 48000) {
> +                nb_lost_samples = av_rescale_q(nb_lost_samples, avc->pkt_timebase, opus_timebase);
> +            }
> +            // For FEC/PLC, frame_size has to be to have a multiple of 2.5 ms
> +            if (nb_lost_samples % (int)(2.5 / 1000 * opus_timebase.den)) {
> +                nb_lost_samples -= nb_lost_samples % (int)(2.5 / 1000 * opus_timebase.den);
> +            }
> +        }
> +    }
>
> -    frame->nb_samples = MAX_FRAME_SIZE;
> +    frame->nb_samples = MAX_FRAME_SIZE + nb_lost_samples;
>      if ((ret = ff_get_buffer(avc, frame, 0)) < 0)
>          return ret;
>
> +    outptr = frame->data[0];
> +    nb_samples_left = frame->nb_samples;
> +
> +    if (opus->decode_fec && nb_lost_samples > 0) {
> +        // Try to recover the lost samples with FEC data from this one.
> +        // If there's no FEC data, the decoder will do loss concealment instead.
> +        if (avc->sample_fmt == AV_SAMPLE_FMT_S16)
> +            ret = opus_multistream_decode(opus->dec, pkt->data, pkt->size,
> +                                                  (opus_int16 *)outptr,
> +                                                  nb_lost_samples, 1);
> +        else
> +            ret = opus_multistream_decode_float(opus->dec, pkt->data, pkt->size,
> +                                                       (float *)outptr,
> +                                                       nb_lost_samples, 1);
> +
> +        if (ret < 0) {
> +            if (opus->decode_fec) opus->expected_next_pts = pkt->pts + pkt->duration;
> +            av_log(avc, AV_LOG_ERROR, "Decoding error: %s\n",
> +                   opus_strerror(ret));
> +            return ff_opus_error_to_averror(ret);
> +        }
> +
> +        av_log(avc, AV_LOG_WARNING, "Recovered %d samples with FEC/PLC\n",
> +                   ret);
> +
> +        outptr += ret * avc->channels * av_get_bytes_per_sample(avc->sample_fmt);
> +        nb_samples_left -= ret;
> +        nb_samples += ret;
> +        if (pkt->pts != AV_NOPTS_VALUE) {
> +            frame->pts = pkt->pts - ret;
> +        }
> +    }
> +
> +    // Decode the actual, non-lost data.
>      if (avc->sample_fmt == AV_SAMPLE_FMT_S16)
> -        nb_samples = opus_multistream_decode(opus->dec, pkt->data, pkt->size,
> -                                             (opus_int16 *)frame->data[0],
> -                                             frame->nb_samples, 0);
> +        ret = opus_multistream_decode(opus->dec, pkt->data, pkt->size,
> +                                      (opus_int16 *)outptr,
> +                                      nb_samples_left, 0);
>      else
> -        nb_samples = opus_multistream_decode_float(opus->dec, pkt->data, pkt->size,
> -                                                   (float *)frame->data[0],
> -                                                   frame->nb_samples, 0);
> +        ret = opus_multistream_decode_float(opus->dec, pkt->data, pkt->size,
> +                                            (float *)outptr,
> +                                            nb_samples_left, 0);
>
> -    if (nb_samples < 0) {
> +    if (ret < 0) {
> +        if (opus->decode_fec) opus->expected_next_pts = pkt->pts + pkt->duration;
>          av_log(avc, AV_LOG_ERROR, "Decoding error: %s\n",
> -               opus_strerror(nb_samples));
> -        return ff_opus_error_to_averror(nb_samples);
> +               opus_strerror(ret));
> +        return ff_opus_error_to_averror(ret);
> +    }
> +    nb_samples += ret;
> +
> +    av_log(avc, AV_LOG_WARNING, "Decoded %d samples normally\n", ret);
> +

I would either drop this or only log it at AV_LOG_DEBUG level as it's
pretty verbose (especially given that it's the "normal" case).

Otherwise this seems great, nice work.

Best,
Tristan