[FFmpeg-devel] [PATCH] decklink: Add support for compressed AC-3 output over SDI
Devin Heitmueller
devin.heitmueller at ltnglobal.com
Thu Mar 9 16:08:08 EET 2023
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.
Signed-off-by: Devin Heitmueller <dheitmueller at ltnglobal.com>
---
libavdevice/decklink_enc.cpp | 97 ++++++++++++++++++++++++++++++------
1 file changed, 82 insertions(+), 15 deletions(-)
diff --git a/libavdevice/decklink_enc.cpp b/libavdevice/decklink_enc.cpp
index 8d423f6b6e..51a2bb4aad 100644
--- a/libavdevice/decklink_enc.cpp
+++ b/libavdevice/decklink_enc.cpp
@@ -243,19 +243,32 @@ 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 +279,49 @@ 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, bmdAudioSampleRate48kHz);
ctx->audio = 1;
return 0;
}
+static int create_s337_payload(AVPacket *pkt, enum AVCodecID codec_id, uint8_t **outbuf, int *outsize)
+{
+ uint8_t *s337_payload;
+ uint8_t *s337_payload_start;
+ int i;
+
+ if (codec_id != AV_CODEC_ID_AC3)
+ return AVERROR(EINVAL);
+
+ /* Encapsulate AC3 syncframe into SMPTE 337 packet */
+ *outsize = pkt->size + 8;
+ s337_payload = (uint8_t *) av_mallocz(*outsize);
+ if (s337_payload == NULL)
+ return AVERROR(ENOMEM);
+
+ /* Construct SMPTE S337 Burst preamble */
+ s337_payload[0] = 0x72; /* Sync Word 1 */
+ s337_payload[1] = 0xf8; /* Sync Word 1 */
+ s337_payload[2] = 0x1f; /* Sync Word 1 */
+ s337_payload[3] = 0x4e; /* Sync Word 1 */
+ s337_payload[4] = 0x01; /* Burst Info, including data type (1=ac3) */
+ s337_payload[5] = 0x00;
+ uint16_t bitcount = pkt->size * 8;
+ s337_payload[6] = bitcount & 0xff; /* Length code */
+ s337_payload[7] = bitcount >> 8; /* Length code */
+ s337_payload_start = &s337_payload[8];
+ for (i = 0; i < pkt->size; i += 2) {
+ s337_payload_start[0] = pkt->data[i+1];
+ s337_payload_start[1] = pkt->data[i];
+ s337_payload_start += 2;
+ }
+
+ *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 +579,40 @@ 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)
+ 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" {
--
2.35.1.655.ga68dfadae5
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