On Thu, Feb 18, 2021 at 11:39 AM Philip-Dylan Gleonec
<philip-dylan.gleo...@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.gleo...@savoirfairelinux.com>
> Co-developed-by: Steinar H. Gunderson <steinar+ffm...@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
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