Chiebot-Mirror
/
FFmpeg
mirror of https://github.com/FFmpeg/FFmpeg.git
8
0
Fork 0
Code Issues Projects Releases Wiki Activity

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

Browse Source 
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>
pull/389/head
Devin Heitmueller 2 years ago committed by Marton Balint
parent 30f1f89572
commit 12d1f7c4b7
1 changed files with 85 additions and 15 deletions
Whitespace
Show all changes Ignore whitespace when comparing lines Ignore changes in amount of whitespace Ignore changes in whitespace at EOL
Split View
Diff Options
Show Stats Download Patch File Download Diff File
  1. 100
      libavdevice/decklink_enc.cpp

100
libavdevice/decklink_enc.cpp
Unescape View File

@ -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" {

Write Preview
Loading…
Cancel
Save