[FFmpeg-devel] [PATCH 03/11] decklink: Introduce support for capture of multiple audio streams
Devin Heitmueller
dheitmueller at ltnglobal.com
Tue Jan 9 03:16:50 EET 2018
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.
Updated to reflect comments from Carl Eugen Hoyos <ceffmpeg at gmail.com>,
Aaron Levinson <alevinsn at levland.net>, and Matthias Hunstock
<atze at fem.tu-ilmenau.de>.
Signed-off-by: Devin Heitmueller <dheitmueller at ltnglobal.com>
---
doc/indevs.texi | 8 ++-
libavdevice/decklink_common.cpp | 12 ++++
libavdevice/decklink_common.h | 8 ++-
libavdevice/decklink_common_c.h | 6 ++
libavdevice/decklink_dec.cpp | 136 +++++++++++++++++++++++++++++++---------
libavdevice/decklink_dec_c.c | 3 +
6 files changed, 142 insertions(+), 31 deletions(-)
diff --git a/doc/indevs.texi b/doc/indevs.texi
index 56066bf..4760d70 100644
--- a/doc/indevs.texi
+++ b/doc/indevs.texi
@@ -278,9 +278,15 @@ For SD sources, ffmpeg needs to be compiled with @code{--enable-libzvbi}. For
HD sources, on older (pre-4K) DeckLink card models you have to capture in 10
bit mode.
+ at item audio_mode
+Defines whether to capture a bundle of audio channels (the number of which is determined
+by the channels argument), or whether to capture a number of audio pairs (the number of
+which is determined by the maximum number of pairs supported by the card). Must be
+ at samp{bundled} or @samp{pairs}. Defaults to @samp{bundled}.
+
@item channels
Defines number of audio channels to capture. Must be @samp{2}, @samp{8} or @samp{16}.
-Defaults to @samp{2}.
+Defaults to @samp{2}. This parameter is ignored if audio_mode is set to pairs.
@item duplex_mode
Sets the decklink device duplex mode. Must be @samp{unset}, @samp{half} or @samp{full}.
diff --git a/libavdevice/decklink_common.cpp b/libavdevice/decklink_common.cpp
index c432189..e7daa63 100644
--- a/libavdevice/decklink_common.cpp
+++ b/libavdevice/decklink_common.cpp
@@ -446,6 +446,7 @@ int ff_decklink_init_device(AVFormatContext *avctx, const char* name)
struct decklink_cctx *cctx = (struct decklink_cctx *)avctx->priv_data;
struct decklink_ctx *ctx = (struct decklink_ctx *)cctx->ctx;
IDeckLink *dl = NULL;
+ int64_t maxAudioChannels;
IDeckLinkIterator *iter = CreateDeckLinkIteratorInstance();
if (!iter) {
av_log(avctx, AV_LOG_ERROR, "Could not create DeckLink iterator\n");
@@ -479,5 +480,16 @@ int ff_decklink_init_device(AVFormatContext *avctx, const char* name)
return AVERROR_EXTERNAL;
}
+ if (ctx->attr->GetInt(BMDDeckLinkMaximumAudioChannels, &maxAudioChannels) != S_OK) {
+ av_log(avctx, AV_LOG_WARNING, "Could not determine number of audio channels\n");
+ ctx->max_audio_channels = 0;
+ } else {
+ ctx->max_audio_channels = maxAudioChannels;
+ }
+ 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");
+ ctx->max_audio_channels = DECKLINK_MAX_AUDIO_CHANNELS;
+ }
+
return 0;
}
diff --git a/libavdevice/decklink_common.h b/libavdevice/decklink_common.h
index 143bbb9..b262780 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;
+ 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 368ac25..3a21bae 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_BUNDLED = 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 94dae26..6c1ff82 100644
--- a/libavdevice/decklink_dec.cpp
+++ b/libavdevice/decklink_dec.cpp
@@ -627,9 +627,56 @@ 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_BUNDLED) {
+ 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;
+ st->codecpar->channel_layout = AV_CH_LAYOUT_STEREO;
+ 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++) {
+ 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;
+ st->codecpar->channel_layout = AV_CH_LAYOUT_STEREO;
+ 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 +824,57 @@ 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_BUNDLED) {
+ 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 +1079,7 @@ 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 +1165,17 @@ 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_BUNDLED) {
+ 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", 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 1c6d826..fe8e9fd 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_BUNDLED}, 0, 1, DEC, "audio_mode"},
+ { "bundled", NULL, 0, AV_OPT_TYPE_CONST, { .i64 = AUDIO_MODE_BUNDLED}, 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"},
--
1.8.3.1
More information about the ffmpeg-devel
mailing list