[FFmpeg-devel] [PATCH v6] avdevice/decklink_enc: Add support for compressed AC-3 output over SDI
Marton Balint
cus at passwd.hu
Sat Apr 8 19:53:21 EEST 2023
On Fri, 7 Apr 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, will apply.
Regards,
Marton
>
> Thanks to Marton Balint for providing feedback.
>
> Signed-off-by: Devin Heitmueller <dheitmueller at ltnglobal.com>
> ---
> libavdevice/decklink_enc.cpp | 100 ++++++++++++++++++++++++++++++++++++-------
> 1 file changed, 85 insertions(+), 15 deletions(-)
>
> diff --git a/libavdevice/decklink_enc.cpp b/libavdevice/decklink_enc.cpp
> index 62676ea..92bfdb2 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 +280,52 @@ 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;
> }
>
> +/* Wrap the AC-3 packet into an S337 payload that is in S16LE format which can be easily
> + injected into the PCM stream. Note: despite the function name, only AC-3 is implemented */
> +static int create_s337_payload(AVPacket *pkt, uint8_t **outbuf, int *outsize)
> +{
> + /* Note: if the packet size is not divisible by four, we need to make the actual
> + payload larger to ensure it ends on an two channel S16LE boundary */
> + int payload_size = FFALIGN(pkt->size, 4) + 8;
> + uint16_t bitcount = pkt->size * 8;
> + uint8_t *s337_payload;
> + PutByteContext pb;
> +
> + /* Sanity check: According to SMPTE ST 340:2015 Sec 4.1, the AC-3 sync frame will
> + exactly match the 1536 samples of baseband (PCM) audio that it represents. */
> + if (pkt->size > 1536)
> + return AVERROR(EINVAL);
> +
> + /* Encapsulate AC3 syncframe into SMPTE 337 packet */
> + s337_payload = (uint8_t *) av_malloc(payload_size);
> + 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 (int i = 0; i < (pkt->size - 1); i += 2)
> + bytestream2_put_le16u(&pb, (pkt->data[i] << 8) | pkt->data[i+1]);
> +
> + /* Ensure final payload is aligned on 4-byte boundary */
> + if (pkt->size & 1)
> + bytestream2_put_le16u(&pb, pkt->data[pkt->size - 1] << 8);
> + if ((pkt->size & 3 == 1) || (pkt->size & 3 == 2))
> + bytestream2_put_le16u(&pb, 0);
> +
> + *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;
> @@ -617,21 +669,39 @@ 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, &outbuf, &outbuf_size);
> + if (ret < 0)
> + 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;
> }
>
> extern "C" {
> --
> 1.8.3.1
>
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