[FFmpeg-devel] [PATCH v2 2/2] decklink: Add support for compressed AC-3 output over SDI
Devin Heitmueller
devin.heitmueller at ltnglobal.com
Mon Mar 27 19:08:51 EEST 2023
On Fri, Mar 24, 2023 at 5:07 PM Marton Balint <cus at passwd.hu> wrote:
>
>
>
> 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 at 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
Thanks for your feedback. A revised patch reflecting your changes
will be sent to the mailing list shortly.
Devin
--
Devin Heitmueller, Senior Software Engineer
LTN Global Communications
o: +1 (301) 363-1001
w: https://ltnglobal.com e: devin.heitmueller at ltnglobal.com
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