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. Updated to reflect feedback from Aaron Levinson <alevinsn_...@levland.net> Signed-off-by: Devin Heitmueller <dheitmuel...@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 ff60050..c39fc0e 100644 --- a/libavdevice/decklink_enc.cpp +++ b/libavdevice/decklink_enc.cpp @@ -237,19 +237,32 @@ 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->channels != 2 && c->channels != 8 && c->channels != 16) { - av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels!" - " Only 2, 8 or 16 channels are supported.\n"); + + if (st->codecpar->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 (st->codecpar->codec_id == AV_CODEC_ID_PCM_S16LE) { + if (c->channels != 2 && c->channels != 8 && c->channels != 16) { + av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels!" + " Only 2, 8 or 16 channels are supported.\n"); + return -1; + } + if (c->sample_rate != bmdAudioSampleRate48kHz) { + av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate!" + " Only 48kHz is supported.\n"); + return -1; + } + ctx->channels = c->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->channels, + ctx->channels, bmdAudioOutputStreamTimestamped) != S_OK) { av_log(avctx, AV_LOG_ERROR, "Could not enable audio output!\n"); return -1; @@ -260,8 +273,7 @@ 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->channels; + avpriv_set_pts_info(st, 64, 1, bmdAudioSampleRate48kHz); ctx->audio = 1; @@ -553,25 +565,83 @@ static int decklink_write_video_packet(AVFormatContext *avctx, AVPacket *pkt) 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 payload_size = (pkt->size + 4) * sizeof(uint16_t); + int i; + + /* Encapsulate AC3 syncframe into SMPTE 337 packet */ + s337_payload = (uint8_t *) av_mallocz(payload_size); + 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 */ + + if (codec_id == AV_CODEC_ID_AC3) { + s337_payload[4] = 0x01; + } else { + av_free(s337_payload); + return AVERROR(EINVAL); + } + + 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; + *outsize = payload_size; + return 0; +} + 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; buffercount_type 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 */ + ret = create_s337_payload(pkt, st->codecpar->codec_id, + &outbuf, &sample_count); + if (ret != 0) + return ret; + } 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_free(outbuf); + + return ret; } extern "C" { -- 1.8.3.1 _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel