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avdevice/decklink_enc: Add support for compressed AC-3 output over SDI

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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>
Signed-off-by: Marton Balint <cus@passwd.hu>
This commit is contained in:
Devin Heitmueller 2023-04-07 17:36:14 -04:00 committed by Marton Balint
parent 30f1f89572
commit 12d1f7c4b7
1 changed files with 87 additions and 17 deletions
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104
libavdevice/decklink_enc.cpp
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@ -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,
bmdAudioSampleRate48kHz, NULL) != S_OK) {
av_log(avctx, AV_LOG_ERROR, "Could not schedule audio samples.\n");
return AVERROR(EIO);
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;
}
return 0;
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");
ret = AVERROR(EIO);
}
if (st->codecpar->codec_id == AV_CODEC_ID_AC3)
av_freep(&outbuf);
return ret;
}
extern "C" {