| Message ID | 20230317150232.17804-2-dheitmueller@ltnglobal.com |
|---|---|
| State | New |
| Headers | show |
| Series | [FFmpeg-devel,v2,1/2] avcodec: Fix warnings with signed/unsigned compare in bitstream.h | expand |
On Fri, 17 Mar 2023, Devin Heitmueller wrote: > 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> > --- > libavdevice/decklink_enc.cpp | 90 ++++++++++++++++++++++++++++++------ > 1 file changed, 75 insertions(+), 15 deletions(-) > > diff --git a/libavdevice/decklink_enc.cpp b/libavdevice/decklink_enc.cpp > index 8d423f6b6e..8d80f00247 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 +280,41 @@ 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; > + int i; > + > + if (codec_id != AV_CODEC_ID_AC3) > + return AVERROR(EINVAL); Maybe some sanity check here for pkt->size upper limit to avoid overflows? > + > + /* Encapsulate AC3 syncframe into SMPTE 337 packet */ > + s337_payload = (uint8_t *) av_mallocz(payload_size); Why not simply av_malloc? > + 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 (i = 0; i < pkt->size; i += 2) for (int i = > + 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 +572,40 @@ 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) if (ret < 0) is preferred > + 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; > } > Thanks, Marton
On Fri, Mar 24, 2023 at 5:07 PM Marton Balint <cus@passwd.hu> wrote: > > > > On Fri, 17 Mar 2023, Devin Heitmueller wrote: > > > 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> > > --- > > libavdevice/decklink_enc.cpp | 90 ++++++++++++++++++++++++++++++------ > > 1 file changed, 75 insertions(+), 15 deletions(-) > > > > diff --git a/libavdevice/decklink_enc.cpp b/libavdevice/decklink_enc.cpp > > index 8d423f6b6e..8d80f00247 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 +280,41 @@ 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; > > + int i; > > + > > + if (codec_id != AV_CODEC_ID_AC3) > > + return AVERROR(EINVAL); > > Maybe some sanity check here for pkt->size upper limit to avoid overflows? > > > + > > + /* Encapsulate AC3 syncframe into SMPTE 337 packet */ > > + s337_payload = (uint8_t *) av_mallocz(payload_size); > > Why not simply av_malloc? > > > + 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 (i = 0; i < pkt->size; i += 2) > > for (int i = > > > + 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 +572,40 @@ 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) > > if (ret < 0) is preferred > > > + 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; > > } > > > > Thanks, > Marton Thanks for your feedback. A revised patch reflecting your changes will be sent to the mailing list shortly. Devin
diff --git a/libavdevice/decklink_enc.cpp b/libavdevice/decklink_enc.cpp index 8d423f6b6e..8d80f00247 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 +280,41 @@ 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; + int i; + + if (codec_id != AV_CODEC_ID_AC3) + return AVERROR(EINVAL); + + /* Encapsulate AC3 syncframe into SMPTE 337 packet */ + s337_payload = (uint8_t *) av_mallocz(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 (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 +572,40 @@ 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" {
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> --- libavdevice/decklink_enc.cpp | 90 ++++++++++++++++++++++++++++++------ 1 file changed, 75 insertions(+), 15 deletions(-)