Signed-off-by: Michael Riedl <michael.ri...@nativewaves.com> --- configure | 2 + libavformat/Makefile | 1 + libavformat/allformats.c | 1 + libavformat/webrtc_mux.c | 273 +++++++++++++++++++++++++++++++++++++++ 4 files changed, 277 insertions(+) create mode 100644 libavformat/webrtc_mux.c
diff --git a/configure b/configure index 02c6f7f2c5d..05cfbbb2376 100755 --- a/configure +++ b/configure @@ -3557,6 +3557,8 @@ wav_demuxer_select="riffdec" wav_muxer_select="riffenc" webm_chunk_muxer_select="webm_muxer" webm_dash_manifest_demuxer_select="matroska_demuxer" +whip_muxer_deps="libdatachannel rtp_muxer" +whip_muxer_select="http_protocol rtpenc_chain" whep_demuxer_deps="libdatachannel sdp_demuxer" whep_demuxer_select="http_protocol" wtv_demuxer_select="mpegts_demuxer riffdec" diff --git a/libavformat/Makefile b/libavformat/Makefile index f790fa8cae4..000fd308be2 100644 --- a/libavformat/Makefile +++ b/libavformat/Makefile @@ -621,6 +621,7 @@ OBJS-$(CONFIG_WEBM_CHUNK_MUXER) += webm_chunk.o OBJS-$(CONFIG_WEBP_MUXER) += webpenc.o OBJS-$(CONFIG_WEBVTT_DEMUXER) += webvttdec.o subtitles.o OBJS-$(CONFIG_WEBVTT_MUXER) += webvttenc.o +OBJS-$(CONFIG_WHIP_MUXER) += webrtc.o webrtc_mux.o OBJS-$(CONFIG_WHEP_DEMUXER) += webrtc.o webrtc_demux.o OBJS-$(CONFIG_WSAUD_DEMUXER) += westwood_aud.o OBJS-$(CONFIG_WSAUD_MUXER) += westwood_audenc.o diff --git a/libavformat/allformats.c b/libavformat/allformats.c index 7acb05634c8..2ad2a6dcba2 100644 --- a/libavformat/allformats.c +++ b/libavformat/allformats.c @@ -504,6 +504,7 @@ extern const FFOutputFormat ff_webm_chunk_muxer; extern const FFOutputFormat ff_webp_muxer; extern const AVInputFormat ff_webvtt_demuxer; extern const FFOutputFormat ff_webvtt_muxer; +extern const FFOutputFormat ff_whip_muxer; extern const AVInputFormat ff_whep_demuxer; extern const AVInputFormat ff_wsaud_demuxer; extern const FFOutputFormat ff_wsaud_muxer; diff --git a/libavformat/webrtc_mux.c b/libavformat/webrtc_mux.c new file mode 100644 index 00000000000..1fe30ecb278 --- /dev/null +++ b/libavformat/webrtc_mux.c @@ -0,0 +1,273 @@ +/* + * WebRTC-HTTP ingestion protocol (WHIP) muxer using libdatachannel + * + * Copyright (C) 2023 NativeWaves GmbH <cont...@nativewaves.com> + * This work is supported by FFG project 47168763. + * + * 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 "avformat.h" +#include "internal.h" +#include "libavutil/avstring.h" +#include "libavutil/time.h" +#include "mux.h" +#include "rtpenc.h" +#include "rtpenc_chain.h" +#include "rtsp.h" +#include "webrtc.h" +#include "version.h" + +typedef struct WHIPContext { + AVClass *av_class; + DataChannelContext data_channel; +} WHIPContext; + + +static void whip_deinit(AVFormatContext* avctx); +static int whip_init(AVFormatContext* avctx) +{ + WHIPContext*const ctx = (WHIPContext*const)avctx->priv_data; + AVStream* stream; + const AVCodecParameters* codecpar; + int i, ret; + char media_stream_id[37] = { 0 }; + rtcTrackInit track_init; + const AVChannelLayout supported_layout = AV_CHANNEL_LAYOUT_STEREO; + const RTPMuxContext* rtp_mux_ctx; + DataChannelTrack* track; + char sdp_stream[SDP_MAX_SIZE] = { 0 }; + char* fmtp; + + ctx->data_channel.avctx = avctx; + webrtc_init_logger(); + ret = webrtc_init_connection(&ctx->data_channel); + if (ret < 0) { + av_log(avctx, AV_LOG_ERROR, "Failed to initialize connection\n"); + goto fail; + } + + if (!(ctx->data_channel.tracks = av_mallocz(sizeof(DataChannelTrack) * avctx->nb_streams))) { + av_log(avctx, AV_LOG_ERROR, "Failed to allocate tracks\n"); + ret = AVERROR(ENOMEM); + goto fail; + } + + /* configure tracks */ + ret = webrtc_generate_media_stream_id(media_stream_id); + if (ret < 0) { + av_log(avctx, AV_LOG_ERROR, "Failed to generate media stream id\n"); + goto fail; + } + + for (i = 0; i < avctx->nb_streams; ++i) { + stream = avctx->streams[i]; + codecpar = stream->codecpar; + track = &ctx->data_channel.tracks[i]; + + switch (codecpar->codec_type) + { + case AVMEDIA_TYPE_VIDEO: + /* based on rtpenc */ + avpriv_set_pts_info(stream, 32, 1, 90000); + break; + case AVMEDIA_TYPE_AUDIO: + if (codecpar->sample_rate != 48000) { + av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate. Only 48kHz is supported\n"); + ret = AVERROR(EINVAL); + goto fail; + } + if (av_channel_layout_compare(&codecpar->ch_layout, &supported_layout) != 0) { + av_log(avctx, AV_LOG_ERROR, "Unsupported channel layout. Only stereo is supported\n"); + ret = AVERROR(EINVAL); + goto fail; + } + /* based on rtpenc */ + avpriv_set_pts_info(stream, 32, 1, codecpar->sample_rate); + break; + default: + continue; + } + + ret = webrtc_init_urlcontext(&ctx->data_channel, i); + if (ret < 0) { + av_log(avctx, AV_LOG_ERROR, "webrtc_init_urlcontext failed\n"); + goto fail; + } + + ret = ff_rtp_chain_mux_open(&track->rtp_ctx, avctx, stream, track->rtp_url_context, RTP_MAX_PACKET_SIZE, i); + if (ret < 0) { + av_log(avctx, AV_LOG_ERROR, "ff_rtp_chain_mux_open failed\n"); + goto fail; + } + rtp_mux_ctx = (const RTPMuxContext*)ctx->data_channel.tracks[i].rtp_ctx->priv_data; + + memset(&track_init, 0, sizeof(rtcTrackInit)); + track_init.direction = RTC_DIRECTION_SENDONLY; + track_init.payloadType = rtp_mux_ctx->payload_type; + track_init.ssrc = rtp_mux_ctx->ssrc; + track_init.mid = av_asprintf("%d", i); + track_init.name = LIBAVFORMAT_IDENT; + track_init.msid = media_stream_id; + track_init.trackId = av_asprintf("%s-video-%d", media_stream_id, i); + + ret = webrtc_convert_codec(codecpar->codec_id, &track_init.codec); + if (ret < 0) { + av_log(avctx, AV_LOG_ERROR, "Failed to convert codec\n"); + goto fail; + } + + /* parse fmtp from global header */ + ret = ff_sdp_write_media(sdp_stream, sizeof(sdp_stream), stream, i, NULL, NULL, 0, 0, NULL); + if (ret < 0) { + av_log(avctx, AV_LOG_ERROR, "Failed to write sdp\n"); + goto fail; + } + fmtp = strstr(sdp_stream, "a=fmtp:"); + if (fmtp) { + track_init.profile = av_strndup(fmtp + 10, strchr(fmtp, '\r') - fmtp - 10); + track_init.profile = av_asprintf("%s;level-asymmetry-allowed=1", track_init.profile); + memset(sdp_stream, 0, sizeof(sdp_stream)); + } + + track->track_id = rtcAddTrackEx(ctx->data_channel.peer_connection, &track_init); + if (track->track_id < 0) { + av_log(avctx, AV_LOG_ERROR, "Failed to add track\n"); + ret = AVERROR(EINVAL); + goto fail; + } + } + + return 0; + +fail: + return ret; +} + +static int whip_write_header(AVFormatContext* avctx) +{ + WHIPContext*const ctx = (WHIPContext*const)avctx->priv_data; + int ret; + int64_t timeout; + + ret = webrtc_create_resource(&ctx->data_channel); + if (ret < 0) { + av_log(avctx, AV_LOG_ERROR, "Failed to create resource\n"); + goto fail; + } + + /* wait for connection to be established */ + timeout = av_gettime_relative() + ctx->data_channel.connection_timeout; + while (ctx->data_channel.state != RTC_CONNECTED) { + if (ctx->data_channel.state == RTC_FAILED || ctx->data_channel.state == RTC_CLOSED || av_gettime_relative() > timeout) { + av_log(avctx, AV_LOG_ERROR, "Failed to open connection\n"); + ret = AVERROR_EXTERNAL; + goto fail; + } + + av_log(avctx, AV_LOG_VERBOSE, "Waiting for PeerConnection to open\n"); + av_usleep(1000); + } + + return 0; + +fail: + return ret; +} + +static int whip_write_packet(AVFormatContext* avctx, AVPacket* pkt) +{ + WHIPContext*const ctx = (WHIPContext*const)avctx->priv_data; + AVFormatContext* rtpctx = ctx->data_channel.tracks[pkt->stream_index].rtp_ctx; + pkt->stream_index = 0; + + if (ctx->data_channel.state != RTC_CONNECTED) { + av_log(avctx, AV_LOG_ERROR, "Connection is not open\n"); + return AVERROR(EINVAL); + } + + return av_write_frame(rtpctx, pkt); +} + +static int whip_write_trailer(AVFormatContext* avctx) +{ + WHIPContext*const ctx = (WHIPContext*const)avctx->priv_data; + return webrtc_close_resource(&ctx->data_channel); +} + +static void whip_deinit(AVFormatContext* avctx) +{ + WHIPContext*const ctx = (WHIPContext*const)avctx->priv_data; + webrtc_deinit(&ctx->data_channel); +} + +static int whip_check_bitstream(AVFormatContext *s, AVStream *st, const AVPacket *pkt) +{ + /* insert SPS/PPS into every keyframe otherwise browsers won't play the stream */ + if (st->codecpar->extradata_size && st->codecpar->codec_type == AVMEDIA_TYPE_VIDEO) + return ff_stream_add_bitstream_filter(st, "dump_extra", "freq=keyframe"); + return 1; +} + +static int whip_query_codec(enum AVCodecID codec_id, int std_compliance) +{ + switch (codec_id) + { + case AV_CODEC_ID_OPUS: + case AV_CODEC_ID_AAC: + case AV_CODEC_ID_PCM_MULAW: + case AV_CODEC_ID_PCM_ALAW: + case AV_CODEC_ID_H264: + case AV_CODEC_ID_HEVC: + case AV_CODEC_ID_AV1: + case AV_CODEC_ID_VP9: + return 1; + default: + return 0; + } +} + +#define OFFSET(x) offsetof(WHIPContext, x) +#define ENC AV_OPT_FLAG_ENCODING_PARAM +static const AVOption options[] = { + WEBRTC_OPTIONS(ENC, OFFSET(data_channel)), + { NULL }, +}; + +static const AVClass whip_muxer_class = { + .class_name = "WHIP muxer", + .item_name = av_default_item_name, + .option = options, + .version = LIBAVUTIL_VERSION_INT, +}; + +const FFOutputFormat ff_whip_muxer = { + .p.name = "whip", + .p.long_name = NULL_IF_CONFIG_SMALL("WebRTC-HTTP ingestion protocol (WHIP) muxer"), + .p.audio_codec = AV_CODEC_ID_OPUS, // supported by major browsers + .p.video_codec = AV_CODEC_ID_H264, + .p.flags = AVFMT_NOFILE | AVFMT_GLOBALHEADER | AVFMT_EXPERIMENTAL, + .p.priv_class = &whip_muxer_class, + .priv_data_size = sizeof(WHIPContext), + .write_packet = whip_write_packet, + .write_header = whip_write_header, + .write_trailer = whip_write_trailer, + .init = whip_init, + .deinit = whip_deinit, + .query_codec = whip_query_codec, + .check_bitstream = whip_check_bitstream, +}; -- 2.39.2 _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-requ...@ffmpeg.org with subject "unsubscribe".