Re: [FFmpeg-devel] [PATCH v2 2/2] decklink: Add support for compressed AC-3 output over SDI

[Marton Balint <cus@passwd.hu>]

On Fri, 17 Mar 2023, Devin Heitmueller wrote:

> Extend the decklink output to include support for compressed AC-3,
> encapsulated using the SMPTE ST 377:2015 standard.
>
> This functionality can be exercised by using the "copy" codec when
> the input audio stream is AC-3.  For example:
>
> ./ffmpeg -i ~/foo.ts -codec:a copy -f decklink 'UltraStudio Mini Monitor'
>
> Note that the default behavior continues to be to do PCM output,
> which means without specifying the copy codec a stream containing
> AC-3 will be decoded and downmixed to stereo audio before output.
>
> Thanks to Marton Balint for providing feedback.
>
> Signed-off-by: Devin Heitmueller <dheitmueller@ltnglobal.com>
> ---
> libavdevice/decklink_enc.cpp | 90 ++++++++++++++++++++++++++++++------
> 1 file changed, 75 insertions(+), 15 deletions(-)
>
> diff --git a/libavdevice/decklink_enc.cpp b/libavdevice/decklink_enc.cpp
> index 8d423f6b6e..8d80f00247 100644
> --- a/libavdevice/decklink_enc.cpp
> +++ b/libavdevice/decklink_enc.cpp
> @@ -32,6 +32,7 @@ extern "C" {
>
> extern "C" {
> #include "libavformat/avformat.h"
> +#include "libavcodec/bytestream.h"
> #include "libavutil/internal.h"
> #include "libavutil/imgutils.h"
> #include "avdevice.h"
> @@ -243,19 +244,32 @@ static int decklink_setup_audio(AVFormatContext *avctx, AVStream *st)
>         av_log(avctx, AV_LOG_ERROR, "Only one audio stream is supported!\n");
>         return -1;
>     }
> -    if (c->sample_rate != 48000) {
> -        av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate!"
> -               " Only 48kHz is supported.\n");
> -        return -1;
> -    }
> -    if (c->ch_layout.nb_channels != 2 && c->ch_layout.nb_channels != 8 && c->ch_layout.nb_channels != 16) {
> -        av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels!"
> -               " Only 2, 8 or 16 channels are supported.\n");
> +
> +    if (c->codec_id == AV_CODEC_ID_AC3) {
> +        /* Regardless of the number of channels in the codec, we're only
> +           using 2 SDI audio channels at 48000Hz */
> +        ctx->channels = 2;
> +    } else if (c->codec_id == AV_CODEC_ID_PCM_S16LE) {
> +        if (c->sample_rate != 48000) {
> +            av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate!"
> +                   " Only 48kHz is supported.\n");
> +            return -1;
> +        }
> +        if (c->ch_layout.nb_channels != 2 && c->ch_layout.nb_channels != 8 && c->ch_layout.nb_channels != 16) {
> +            av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels!"
> +                   " Only 2, 8 or 16 channels are supported.\n");
> +            return -1;
> +        }
> +        ctx->channels = c->ch_layout.nb_channels;
> +    } else {
> +        av_log(avctx, AV_LOG_ERROR, "Unsupported codec specified!"
> +               " Only PCM_S16LE and AC-3 are supported.\n");
>         return -1;
>     }
> +
>     if (ctx->dlo->EnableAudioOutput(bmdAudioSampleRate48kHz,
>                                     bmdAudioSampleType16bitInteger,
> -                                    c->ch_layout.nb_channels,
> +                                    ctx->channels,
>                                     bmdAudioOutputStreamTimestamped) != S_OK) {
>         av_log(avctx, AV_LOG_ERROR, "Could not enable audio output!\n");
>         return -1;
> @@ -266,14 +280,41 @@ static int decklink_setup_audio(AVFormatContext *avctx, AVStream *st)
>     }
>
>     /* The device expects the sample rate to be fixed. */
> -    avpriv_set_pts_info(st, 64, 1, c->sample_rate);
> -    ctx->channels = c->ch_layout.nb_channels;
> +    avpriv_set_pts_info(st, 64, 1, 48000);
>
>     ctx->audio = 1;
>
>     return 0;
> }
>
> +static int create_s337_payload(AVPacket *pkt, enum AVCodecID codec_id, uint8_t **outbuf, int *outsize)
> +{
> +    int payload_size = pkt->size + 8;
> +    uint16_t bitcount = pkt->size * 8;
> +    uint8_t *s337_payload;
> +    PutByteContext pb;
> +    int i;
> +
> +    if (codec_id != AV_CODEC_ID_AC3)
> +        return AVERROR(EINVAL);

Maybe some sanity check here for pkt->size upper limit to avoid overflows?

> +
> +    /* Encapsulate AC3 syncframe into SMPTE 337 packet */
> +    s337_payload = (uint8_t *) av_mallocz(payload_size);

Why not simply av_malloc?

> +    if (s337_payload == NULL)
> +        return AVERROR(ENOMEM);
> +    bytestream2_init_writer(&pb, s337_payload, payload_size);
> +    bytestream2_put_le16u(&pb, 0xf872); /* Sync word 1 */
> +    bytestream2_put_le16u(&pb, 0x4e1f); /* Sync word 1 */
> +    bytestream2_put_le16u(&pb, 0x0001); /* Burst Info, including data type (1=ac3) */
> +    bytestream2_put_le16u(&pb, bitcount); /* Length code */
> +    for (i = 0; i < pkt->size; i += 2)

for (int i =

> +        bytestream2_put_le16u(&pb, (pkt->data[i] << 8) | pkt->data[i+1]);
> +
> +    *outsize = payload_size;
> +    *outbuf = s337_payload;
> +    return 0;
> +}
> +
> av_cold int ff_decklink_write_trailer(AVFormatContext *avctx)
> {
>     struct decklink_cctx *cctx = (struct decklink_cctx *)avctx->priv_data;
> @@ -531,21 +572,40 @@ static int decklink_write_audio_packet(AVFormatContext *avctx, AVPacket *pkt)
> {
>     struct decklink_cctx *cctx = (struct decklink_cctx *)avctx->priv_data;
>     struct decklink_ctx *ctx = (struct decklink_ctx *)cctx->ctx;
> -    int sample_count = pkt->size / (ctx->channels << 1);
> +    AVStream *st = avctx->streams[pkt->stream_index];
> +    int sample_count;
>     uint32_t buffered;
> +    uint8_t *outbuf = NULL;
> +    int ret = 0;
>
>     ctx->dlo->GetBufferedAudioSampleFrameCount(&buffered);
>     if (pkt->pts > 1 && !buffered)
>         av_log(avctx, AV_LOG_WARNING, "There's no buffered audio."
>                " Audio will misbehave!\n");
>
> -    if (ctx->dlo->ScheduleAudioSamples(pkt->data, sample_count, pkt->pts,
> +    if (st->codecpar->codec_id == AV_CODEC_ID_AC3) {
> +        /* Encapsulate AC3 syncframe into SMPTE 337 packet */
> +        int outbuf_size;
> +        ret = create_s337_payload(pkt, st->codecpar->codec_id,
> +                                  &outbuf, &outbuf_size);
> +        if (ret)

if (ret < 0) is preferred

> +            return ret;
> +        sample_count = outbuf_size / 4;
> +    } else {
> +        sample_count = pkt->size / (ctx->channels << 1);
> +        outbuf = pkt->data;
> +    }
> +
> +    if (ctx->dlo->ScheduleAudioSamples(outbuf, sample_count, pkt->pts,
>                                        bmdAudioSampleRate48kHz, NULL) != S_OK) {
>         av_log(avctx, AV_LOG_ERROR, "Could not schedule audio samples.\n");
> -        return AVERROR(EIO);
> +        ret = AVERROR(EIO);
>     }
>
> -    return 0;
> +    if (st->codecpar->codec_id == AV_CODEC_ID_AC3)
> +        av_freep(&outbuf);
> +
> +    return ret;
> }
>

Thanks,
Marton
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