[FFmpeg-devel] [PATCH v3] decklink: Add support for compressed AC-3 output over SDI
Devin Heitmueller
devin.heitmueller at ltnglobal.com
Mon Mar 27 19:08:57 EEST 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.
Thanks to Marton Balint for providing feedback.
Signed-off-by: Devin Heitmueller <dheitmueller at ltnglobal.com>
---
libavdevice/decklink_enc.cpp | 94 ++++++++++++++++++++++++++++++------
1 file changed, 79 insertions(+), 15 deletions(-)
diff --git a/libavdevice/decklink_enc.cpp b/libavdevice/decklink_enc.cpp
index 8d423f6b6e..a63aeaa088 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 +280,45 @@ 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;
+
+ if (codec_id != AV_CODEC_ID_AC3)
+ return AVERROR(EINVAL);
+
+ /* 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; i += 2)
+ 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 +576,40 @@ 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 < 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" {
--
2.35.1.655.ga68dfadae5
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