[FFmpeg-devel] [PATCH 3/8] decklink: Introduce support for capture of multiple audio streams
Aaron Levinson
alevinsn_dev at levland.net
Sat Dec 30 09:16:26 EET 2017
On 12/29/2017 10:12 AM, Devin Heitmueller wrote:
> Add support for the ability to capture all audio pairs available
> to the capture hardware. Each pair is exposed as a different audio
> stream, which matches up with the most common use cases for the
> broadcast space (i.e. where there is one stereo pair per audio
> language).
>
> To support the existing use case where multi-channel audio can be
> captured (i.e. 7.1), we introduced a new configuration option, which
> defaults to the existing behavior.
> ---
> libavdevice/decklink_common.cpp | 9 +++
> libavdevice/decklink_common.h | 8 ++-
> libavdevice/decklink_common_c.h | 6 ++
> libavdevice/decklink_dec.cpp | 134 +++++++++++++++++++++++++++++++---------
> libavdevice/decklink_dec_c.c | 3 +
> 5 files changed, 130 insertions(+), 30 deletions(-)
>
> diff --git a/libavdevice/decklink_common.cpp b/libavdevice/decklink_common.cpp
> index ba091dadea..91a626221d 100644
> --- a/libavdevice/decklink_common.cpp
> +++ b/libavdevice/decklink_common.cpp
> @@ -480,5 +480,14 @@ int ff_decklink_init_device(AVFormatContext *avctx, const char* name)
> return AVERROR_EXTERNAL;
> }
>
> + if (ctx->attr->GetInt(BMDDeckLinkMaximumAudioChannels, &ctx->max_audio_channels) != S_OK) {
> + av_log(avctx, AV_LOG_WARNING, "Could not determine number of audio channels\n");
> + ctx->max_audio_channels = 0;
> + }
> + if (ctx->max_audio_channels > DECKLINK_MAX_AUDIO_CHANNELS) {
> + av_log(avctx, AV_LOG_WARNING, "Decklink card reported support for more channels than ffmpeg supports\n");
"Decklink" -> "DeckLink", "ffmpeg" -> "FFmpeg". Also, I think it is
preferable to not state "FFmpeg" in this log message, as technically
this is in libavdevice. If, say, libav were to merge your changes into
their codebase, then this particular log message would make that
annoying. Could be something as simple as "Max audio channels for
DeckLink reduced from %d to %d.\n".
> + ctx->max_audio_channels = DECKLINK_MAX_AUDIO_CHANNELS;
> + }
> +
> return 0;
> }
> diff --git a/libavdevice/decklink_common.h b/libavdevice/decklink_common.h
> index 143bbb951a..06b241029e 100644
> --- a/libavdevice/decklink_common.h
> +++ b/libavdevice/decklink_common.h
> @@ -37,6 +37,10 @@
> #define DECKLINK_BOOL bool
> #endif
>
> +/* Maximum number of channels possible across variants of Blackmagic cards.
> + Actual number for any particular model of card may be lower */
> +#define DECKLINK_MAX_AUDIO_CHANNELS 32
> +
> class decklink_output_callback;
> class decklink_input_callback;
>
> @@ -71,6 +75,7 @@ struct decklink_ctx {
> int bmd_height;
> int bmd_field_dominance;
> int supports_vanc;
> + int64_t max_audio_channels;
Rationale for using an int64_t here when an int would likely be
sufficient? I understand that GetInt() takes an int64_t as input, but
you could use a temporary int64_t with GetInt() and store the value in a
int max_audio_channels.
>
> /* Capture buffer queue */
> AVPacketQueue queue;
> @@ -85,7 +90,8 @@ struct decklink_ctx {
> int64_t last_pts;
> unsigned long frameCount;
> unsigned int dropped;
> - AVStream *audio_st;
> + AVStream *audio_st[DECKLINK_MAX_AUDIO_CHANNELS];
> + int num_audio_streams;
> AVStream *video_st;
> AVStream *teletext_st;
> uint16_t cdp_sequence_num;
> diff --git a/libavdevice/decklink_common_c.h b/libavdevice/decklink_common_c.h
> index 368ac259e4..02011ed53b 100644
> --- a/libavdevice/decklink_common_c.h
> +++ b/libavdevice/decklink_common_c.h
> @@ -30,6 +30,11 @@ typedef enum DecklinkPtsSource {
> PTS_SRC_WALLCLOCK = 4,
> } DecklinkPtsSource;
>
> +typedef enum DecklinkAudioMode {
> + AUDIO_MODE_DISCRETE = 0,
> + AUDIO_MODE_PAIRS = 1,
> +} DecklinkAudioMode;
> > struct decklink_cctx {
> const AVClass *cclass;
>
> @@ -42,6 +47,7 @@ struct decklink_cctx {
> double preroll;
> int v210;
> int audio_channels;
> + int audio_mode;
> int audio_depth;
> int duplex_mode;
> DecklinkPtsSource audio_pts_source;
> diff --git a/libavdevice/decklink_dec.cpp b/libavdevice/decklink_dec.cpp
> index 94dae26003..8d4070eecd 100644
> --- a/libavdevice/decklink_dec.cpp
> +++ b/libavdevice/decklink_dec.cpp
> @@ -627,9 +627,54 @@ static int64_t get_pkt_pts(IDeckLinkVideoInputFrame *videoFrame,
> return pts;
> }
>
> +static int setup_audio(AVFormatContext *avctx)
> +{
> + struct decklink_cctx *cctx = (struct decklink_cctx *)avctx->priv_data;
> + struct decklink_ctx *ctx = (struct decklink_ctx *)cctx->ctx;
> + AVStream *st;
> + int ret = 0;
> +
> + if (cctx->audio_mode == AUDIO_MODE_DISCRETE) {
> + st = avformat_new_stream(avctx, NULL);
> + if (!st) {
> + av_log(avctx, AV_LOG_ERROR, "Cannot add stream\n");
> + ret = AVERROR(ENOMEM);
> + goto error;
> + }
> + st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
> + st->codecpar->codec_id = ctx->audio_depth == 32 ? AV_CODEC_ID_PCM_S32LE : AV_CODEC_ID_PCM_S16LE;
> + st->codecpar->sample_rate = bmdAudioSampleRate48kHz;
> + st->codecpar->channels = cctx->audio_channels;
> + avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
> + ctx->audio_st[0] = st;
> + ctx->num_audio_streams++;
> + } else {
> + for (int i = 0; i < ctx->max_audio_channels / 2; i++) {
Technically, you declared max_audio_channels as an int64_t, but i is
declared as an int.
> + st = avformat_new_stream(avctx, NULL);
> + if (!st) {
> + av_log(avctx, AV_LOG_ERROR, "Cannot add stream %d\n", i);
> + ret = AVERROR(ENOMEM);
> + goto error;
> + }
> + st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
> + st->codecpar->codec_id = ctx->audio_depth == 32 ? AV_CODEC_ID_PCM_S32LE : AV_CODEC_ID_PCM_S16LE;
> + st->codecpar->sample_rate = bmdAudioSampleRate48kHz;
> + st->codecpar->channels = 2;
> + avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
> + ctx->audio_st[i] = st;
> + ctx->num_audio_streams++;
> + }
> + cctx->audio_channels = ctx->max_audio_channels;
> + }
> +
> +error:
> + return ret;
> +}
> +
> HRESULT decklink_input_callback::VideoInputFrameArrived(
> IDeckLinkVideoInputFrame *videoFrame, IDeckLinkAudioInputPacket *audioFrame)
> {
> + decklink_cctx *cctx = (struct decklink_cctx *)avctx->priv_data;
> void *frameBytes;
> void *audioFrameBytes;
> BMDTimeValue frameTime;
> @@ -777,24 +822,57 @@ HRESULT decklink_input_callback::VideoInputFrameArrived(
>
> // Handle Audio Frame
> if (audioFrame) {
> - AVPacket pkt;
> - BMDTimeValue audio_pts;
> - av_init_packet(&pkt);
> -
> - //hack among hacks
> - pkt.size = audioFrame->GetSampleFrameCount() * ctx->audio_st->codecpar->channels * (ctx->audio_depth / 8);
> audioFrame->GetBytes(&audioFrameBytes);
> - audioFrame->GetPacketTime(&audio_pts, ctx->audio_st->time_base.den);
> - pkt.pts = get_pkt_pts(videoFrame, audioFrame, wallclock, ctx->audio_pts_source, ctx->audio_st->time_base, &initial_audio_pts);
> - pkt.dts = pkt.pts;
>
> - //fprintf(stderr,"Audio Frame size %d ts %d\n", pkt.size, pkt.pts);
> - pkt.flags |= AV_PKT_FLAG_KEY;
> - pkt.stream_index = ctx->audio_st->index;
> - pkt.data = (uint8_t *)audioFrameBytes;
> + if (cctx->audio_mode == AUDIO_MODE_DISCRETE) {
> + AVPacket pkt;
> + BMDTimeValue audio_pts;
> + av_init_packet(&pkt);
>
> - if (avpacket_queue_put(&ctx->queue, &pkt) < 0) {
> - ++ctx->dropped;
> + //hack among hacks
> + pkt.size = audioFrame->GetSampleFrameCount() * ctx->audio_st[0]->codecpar->channels * (ctx->audio_depth / 8);
> + audioFrame->GetBytes(&audioFrameBytes);
> + audioFrame->GetPacketTime(&audio_pts, ctx->audio_st[0]->time_base.den);
> + pkt.pts = get_pkt_pts(videoFrame, audioFrame, wallclock, ctx->audio_pts_source, ctx->audio_st[0]->time_base, &initial_audio_pts);
> + pkt.dts = pkt.pts;
> +
> + pkt.flags |= AV_PKT_FLAG_KEY;
> + pkt.stream_index = ctx->audio_st[0]->index;
> + pkt.data = (uint8_t *)audioFrameBytes;
> +
> + if (avpacket_queue_put(&ctx->queue, &pkt) < 0) {
> + ++ctx->dropped;
> + }
> + } else {
> + /* Need to deinterleave audio */
> + int audio_offset = 0;
> + int audio_stride = cctx->audio_channels * ctx->audio_depth / 8;
> + for (int i = 0; i < ctx->num_audio_streams; i++) {
> + int sample_size = ctx->audio_st[i]->codecpar->channels *
> + ctx->audio_st[i]->codecpar->bits_per_coded_sample / 8;
> + AVPacket pkt;
> + int ret = av_new_packet(&pkt, audioFrame->GetSampleFrameCount() * sample_size);
> + if (ret != 0)
> + continue;
> +
> + pkt.pts = get_pkt_pts(videoFrame, audioFrame, wallclock, ctx->audio_pts_source,
> + ctx->audio_st[i]->time_base, &initial_audio_pts);
> + pkt.dts = pkt.pts;
> + pkt.flags |= AV_PKT_FLAG_KEY;
> + pkt.stream_index = ctx->audio_st[i]->index;
> +
> + uint8_t *audio_in = ((uint8_t *) audioFrameBytes) + audio_offset;
> + for (int x = 0; x < pkt.size; x += sample_size) {
> + memcpy(&pkt.data[x], audio_in, sample_size);
> + audio_in += audio_stride;
> + }
> +
> + if (avpacket_queue_put(&ctx->queue, &pkt) < 0)
> + ++ctx->dropped;
> +
> + av_packet_unref(&pkt);
> + audio_offset += sample_size;
> + }
> }
> }
>
> @@ -999,18 +1077,7 @@ av_cold int ff_decklink_read_header(AVFormatContext *avctx)
> #endif
>
> /* Setup streams. */
> - st = avformat_new_stream(avctx, NULL);
> - if (!st) {
> - av_log(avctx, AV_LOG_ERROR, "Cannot add stream\n");
> - ret = AVERROR(ENOMEM);
> - goto error;
> - }
> - st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
> - st->codecpar->codec_id = cctx->audio_depth == 32 ? AV_CODEC_ID_PCM_S32LE : AV_CODEC_ID_PCM_S16LE;
> - st->codecpar->sample_rate = bmdAudioSampleRate48kHz;
> - st->codecpar->channels = cctx->audio_channels;
> - avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
> - ctx->audio_st=st;
> + setup_audio(avctx);
>
> st = avformat_new_stream(avctx, NULL);
> if (!st) {
> @@ -1096,8 +1163,17 @@ av_cold int ff_decklink_read_header(AVFormatContext *avctx)
> ctx->teletext_st = st;
> }
>
> - av_log(avctx, AV_LOG_VERBOSE, "Using %d input audio channels\n", ctx->audio_st->codecpar->channels);
> - result = ctx->dli->EnableAudioInput(bmdAudioSampleRate48kHz, cctx->audio_depth == 32 ? bmdAudioSampleType32bitInteger : bmdAudioSampleType16bitInteger, ctx->audio_st->codecpar->channels);
> + if (cctx->audio_mode == AUDIO_MODE_DISCRETE) {
> + av_log(avctx, AV_LOG_VERBOSE, "Using %d input audio channels\n", ctx->audio_st[0]->codecpar->channels);
> + result = ctx->dli->EnableAudioInput(bmdAudioSampleRate48kHz,
> + ctx->audio_depth == 32 ? bmdAudioSampleType32bitInteger : bmdAudioSampleType16bitInteger,
> + ctx->audio_st[0]->codecpar->channels);
> + } else {
> + av_log(avctx, AV_LOG_VERBOSE, "Using %d input audio channels\n", (int)ctx->max_audio_channels);
> + result = ctx->dli->EnableAudioInput(bmdAudioSampleRate48kHz,
> + ctx->audio_depth == 32 ? bmdAudioSampleType32bitInteger : bmdAudioSampleType16bitInteger,
> + ctx->max_audio_channels);
> + }
>
> if (result != S_OK) {
> av_log(avctx, AV_LOG_ERROR, "Cannot enable audio input\n");
> diff --git a/libavdevice/decklink_dec_c.c b/libavdevice/decklink_dec_c.c
> index 1c6d826945..d3d8c848cf 100644
> --- a/libavdevice/decklink_dec_c.c
> +++ b/libavdevice/decklink_dec_c.c
> @@ -44,6 +44,9 @@ static const AVOption options[] = {
> { "standard", NULL, 0, AV_OPT_TYPE_CONST, { .i64 = 0x7fff9fffeLL}, 0, 0, DEC, "teletext_lines"},
> { "all", NULL, 0, AV_OPT_TYPE_CONST, { .i64 = 0x7ffffffffLL}, 0, 0, DEC, "teletext_lines"},
> { "channels", "number of audio channels", OFFSET(audio_channels), AV_OPT_TYPE_INT , { .i64 = 2 }, 2, 16, DEC },
> + { "audio_mode", "audio mode", OFFSET(audio_mode), AV_OPT_TYPE_INT, { .i64 = AUDIO_MODE_DISCRETE}, 0, 1, DEC, "audio_mode"},
> + { "discrete", NULL, 0, AV_OPT_TYPE_CONST, { .i64 = AUDIO_MODE_DISCRETE}, 0, 0, DEC, "audio_mode"},
> + { "pairs", NULL, 0, AV_OPT_TYPE_CONST, { .i64 = AUDIO_MODE_PAIRS}, 0, 0, DEC, "audio_mode"},
> { "duplex_mode", "duplex mode", OFFSET(duplex_mode), AV_OPT_TYPE_INT, { .i64 = 0}, 0, 2, DEC, "duplex_mode"},
> { "unset", NULL, 0, AV_OPT_TYPE_CONST, { .i64 = 0}, 0, 0, DEC, "duplex_mode"},
> { "half", NULL, 0, AV_OPT_TYPE_CONST, { .i64 = 1}, 0, 0, DEC, "duplex_mode"},
>
Aaron Levinson
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