[FFmpeg-devel] [PATCH] libavformat/rtpdec_opus
Jonathan Baudanza
jon at jonb.org
Mon May 12 05:00:39 EEST 2025
This commit will properly set the duration field of Opus AVPackets.
Currently, duration is set to 0 on Opus packets from the RTP demuxer.
The Ogg muxer depends on the duration field to properly compute the page granule
value. Without a proper duration, the granule will be wrong, and result in
negative pts values in ogg files.
See oggenc.c:657 (ogg_write_packet_internal)
This commit calculates using the opus_duration function, which was copied
from oggparseopus.c
I moved this functionality and the existing opus extradata functionality
(added by me in 6c24f2b) into a new rtpdec_opus.c file.
---
libavformat/Makefile | 1 +
libavformat/rtpdec.c | 56 +----------------
libavformat/rtpdec_formats.h | 1 +
libavformat/rtpdec_opus.c | 117 +++++++++++++++++++++++++++++++++++
4 files changed, 120 insertions(+), 55 deletions(-)
create mode 100644 libavformat/rtpdec_opus.c
diff --git a/libavformat/Makefile b/libavformat/Makefile
index 6c9992adab..ee68345858 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -62,6 +62,7 @@ OBJS-$(CONFIG_RTPDEC) += rdt.o \
rtpdec_mpeg12.o \
rtpdec_mpeg4.o \
rtpdec_mpegts.o \
+ rtpdec_opus.o \
rtpdec_qcelp.o \
rtpdec_qdm2.o \
rtpdec_qt.o \
diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c
index a7d5a79a83..5eff1552f0 100644
--- a/libavformat/rtpdec.c
+++ b/libavformat/rtpdec.c
@@ -61,12 +61,6 @@ static const RTPDynamicProtocolHandler speex_dynamic_handler = {
.codec_id = AV_CODEC_ID_SPEEX,
};
-static const RTPDynamicProtocolHandler opus_dynamic_handler = {
- .enc_name = "opus",
- .codec_type = AVMEDIA_TYPE_AUDIO,
- .codec_id = AV_CODEC_ID_OPUS,
-};
-
static const RTPDynamicProtocolHandler t140_dynamic_handler = { /* RFC 4103 */
.enc_name = "t140",
.codec_type = AVMEDIA_TYPE_SUBTITLE,
@@ -125,7 +119,7 @@ static const RTPDynamicProtocolHandler *const rtp_dynamic_protocol_handler_list[
&ff_vp9_dynamic_handler,
&gsm_dynamic_handler,
&l24_dynamic_handler,
- &opus_dynamic_handler,
+ &ff_opus_dynamic_handler,
&realmedia_mp3_dynamic_handler,
&speex_dynamic_handler,
&t140_dynamic_handler,
@@ -531,43 +525,6 @@ int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd,
return 0;
}
-static int opus_write_extradata(AVCodecParameters *codecpar)
-{
- uint8_t *bs;
- int ret;
-
- /* This function writes an extradata with a channel mapping family of 0.
- * This mapping family only supports mono and stereo layouts. And RFC7587
- * specifies that the number of channels in the SDP must be 2.
- */
- if (codecpar->ch_layout.nb_channels > 2) {
- return AVERROR_INVALIDDATA;
- }
-
- ret = ff_alloc_extradata(codecpar, 19);
- if (ret < 0)
- return ret;
-
- bs = (uint8_t *)codecpar->extradata;
-
- /* Opus magic */
- bytestream_put_buffer(&bs, "OpusHead", 8);
- /* Version */
- bytestream_put_byte (&bs, 0x1);
- /* Channel count */
- bytestream_put_byte (&bs, codecpar->ch_layout.nb_channels);
- /* Pre skip */
- bytestream_put_le16 (&bs, 0);
- /* Input sample rate */
- bytestream_put_le32 (&bs, 48000);
- /* Output gain */
- bytestream_put_le16 (&bs, 0x0);
- /* Mapping family */
- bytestream_put_byte (&bs, 0x0);
-
- return 0;
-}
-
/**
* open a new RTP parse context for stream 'st'. 'st' can be NULL for
* MPEG-2 TS streams.
@@ -576,7 +533,6 @@ RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
int payload_type, int queue_size)
{
RTPDemuxContext *s;
- int ret;
s = av_mallocz(sizeof(RTPDemuxContext));
if (!s)
@@ -600,16 +556,6 @@ RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
if (st->codecpar->sample_rate == 8000)
st->codecpar->sample_rate = 16000;
break;
- case AV_CODEC_ID_OPUS:
- ret = opus_write_extradata(st->codecpar);
- if (ret < 0) {
- av_log(s1, AV_LOG_ERROR,
- "Error creating opus extradata: %s\n",
- av_err2str(ret));
- av_free(s);
- return NULL;
- }
- break;
default:
break;
}
diff --git a/libavformat/rtpdec_formats.h b/libavformat/rtpdec_formats.h
index 72a8f16a90..1ff2a72d2a 100644
--- a/libavformat/rtpdec_formats.h
+++ b/libavformat/rtpdec_formats.h
@@ -77,6 +77,7 @@ extern const RTPDynamicProtocolHandler ff_mpeg4_generic_dynamic_handler;
extern const RTPDynamicProtocolHandler ff_mpegts_dynamic_handler;
extern const RTPDynamicProtocolHandler ff_ms_rtp_asf_pfa_handler;
extern const RTPDynamicProtocolHandler ff_ms_rtp_asf_pfv_handler;
+extern const RTPDynamicProtocolHandler ff_opus_dynamic_handler;
extern const RTPDynamicProtocolHandler ff_qcelp_dynamic_handler;
extern const RTPDynamicProtocolHandler ff_qdm2_dynamic_handler;
extern const RTPDynamicProtocolHandler ff_qt_rtp_aud_handler;
diff --git a/libavformat/rtpdec_opus.c b/libavformat/rtpdec_opus.c
new file mode 100644
index 0000000000..1588f6d715
--- /dev/null
+++ b/libavformat/rtpdec_opus.c
@@ -0,0 +1,117 @@
+/*
+ * RTP Depacketization of Opus, RFC 7587
+ * Copyright (c) 2025 Jonathan Baudanza <jon at jonb.org>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavcodec/bytestream.h"
+#include "libavutil/mem.h"
+#include "rtpdec_formats.h"
+#include "internal.h"
+
+static int opus_duration(const uint8_t *src, int size)
+{
+ unsigned nb_frames = 1;
+ unsigned toc = src[0];
+ unsigned toc_config = toc >> 3;
+ unsigned toc_count = toc & 3;
+ unsigned frame_size = toc_config < 12 ? FFMAX(480, 960 * (toc_config & 3)) :
+ toc_config < 16 ? 480 << (toc_config & 1) :
+ 120 << (toc_config & 3);
+ if (toc_count == 3) {
+ if (size<2)
+ return AVERROR_INVALIDDATA;
+ nb_frames = src[1] & 0x3F;
+ } else if (toc_count) {
+ nb_frames = 2;
+ }
+
+ return frame_size * nb_frames;
+}
+
+static int opus_write_extradata(AVCodecParameters *codecpar)
+{
+ uint8_t *bs;
+ int ret;
+
+ /* This function writes an extradata with a channel mapping family of 0.
+ * This mapping family only supports mono and stereo layouts. And RFC7587
+ * specifies that the number of channels in the SDP must be 2.
+ */
+ if (codecpar->ch_layout.nb_channels > 2) {
+ return AVERROR_INVALIDDATA;
+ }
+
+ ret = ff_alloc_extradata(codecpar, 19);
+ if (ret < 0)
+ return ret;
+
+ bs = (uint8_t *)codecpar->extradata;
+
+ /* Opus magic */
+ bytestream_put_buffer(&bs, "OpusHead", 8);
+ /* Version */
+ bytestream_put_byte (&bs, 0x1);
+ /* Channel count */
+ bytestream_put_byte (&bs, codecpar->ch_layout.nb_channels);
+ /* Pre skip */
+ bytestream_put_le16 (&bs, 0);
+ /* Input sample rate */
+ bytestream_put_le32 (&bs, 48000);
+ /* Output gain */
+ bytestream_put_le16 (&bs, 0x0);
+ /* Mapping family */
+ bytestream_put_byte (&bs, 0x0);
+
+ return 0;
+}
+
+static int opus_init(AVFormatContext *s, int st_index, PayloadContext *priv_data)
+{
+ return opus_write_extradata(s->streams[st_index]->codecpar);
+}
+
+static int opus_parse_packet(AVFormatContext *ctx, PayloadContext *data,
+ AVStream *st, AVPacket *pkt, uint32_t *timestamp,
+ const uint8_t *buf, int len, uint16_t seq,
+ int flags)
+{
+ int rv;
+ int duration;
+
+ if ((rv = av_new_packet(pkt, len)) < 0)
+ return rv;
+
+ memcpy(pkt->data, buf, len);
+ pkt->stream_index = st->index;
+
+ duration = opus_duration(buf, len);
+ if (duration != AVERROR_INVALIDDATA) {
+ pkt->duration = duration;
+ }
+
+ return 0;
+}
+
+const RTPDynamicProtocolHandler ff_opus_dynamic_handler = {
+ .enc_name = "opus",
+ .codec_type = AVMEDIA_TYPE_AUDIO,
+ .codec_id = AV_CODEC_ID_OPUS,
+ .parse_packet = opus_parse_packet,
+ .init = opus_init,
+};
--
2.41.0
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