diff options
Diffstat (limited to 'plugins/rtp/src')
| -rw-r--r-- | plugins/rtp/src/codec_util.vala | 307 | ||||
| -rw-r--r-- | plugins/rtp/src/device.vala | 272 | ||||
| -rw-r--r-- | plugins/rtp/src/module.vala | 237 | ||||
| -rw-r--r-- | plugins/rtp/src/participant.vala | 39 | ||||
| -rw-r--r-- | plugins/rtp/src/plugin.vala | 449 | ||||
| -rw-r--r-- | plugins/rtp/src/register_plugin.vala | 3 | ||||
| -rw-r--r-- | plugins/rtp/src/stream.vala | 681 | ||||
| -rw-r--r-- | plugins/rtp/src/video_widget.vala | 110 | ||||
| -rw-r--r-- | plugins/rtp/src/voice_processor.vala | 176 | ||||
| -rw-r--r-- | plugins/rtp/src/voice_processor_native.cpp | 148 |
10 files changed, 2422 insertions, 0 deletions
diff --git a/plugins/rtp/src/codec_util.vala b/plugins/rtp/src/codec_util.vala new file mode 100644 index 00000000..6a2438f1 --- /dev/null +++ b/plugins/rtp/src/codec_util.vala @@ -0,0 +1,307 @@ +using Gee; +using Xmpp; +using Xmpp.Xep; + +public class Dino.Plugins.Rtp.CodecUtil { + private Set<string> supported_elements = new HashSet<string>(); + private Set<string> unsupported_elements = new HashSet<string>(); + + public static Gst.Caps get_caps(string media, JingleRtp.PayloadType payload_type, bool incoming) { + Gst.Caps caps = new Gst.Caps.simple("application/x-rtp", + "media", typeof(string), media, + "payload", typeof(int), payload_type.id); + //"channels", typeof(int), payloadType.channels, + //"max-ptime", typeof(int), payloadType.maxptime); + unowned Gst.Structure s = caps.get_structure(0); + if (payload_type.clockrate != 0) { + s.set("clock-rate", typeof(int), payload_type.clockrate); + } + if (payload_type.name != null) { + s.set("encoding-name", typeof(string), payload_type.name.up()); + } + if (incoming) { + foreach (JingleRtp.RtcpFeedback rtcp_fb in payload_type.rtcp_fbs) { + if (rtcp_fb.subtype == null) { + s.set(@"rtcp-fb-$(rtcp_fb.type_)", typeof(bool), true); + } else { + s.set(@"rtcp-fb-$(rtcp_fb.type_)-$(rtcp_fb.subtype)", typeof(bool), true); + } + } + } + return caps; + } + + public static string? get_codec_from_payload(string media, JingleRtp.PayloadType payload_type) { + if (payload_type.name != null) return payload_type.name.down(); + if (media == "audio") { + switch (payload_type.id) { + case 0: + return "pcmu"; + case 8: + return "pcma"; + } + } + return null; + } + + public static string? get_media_type_from_payload(string media, JingleRtp.PayloadType payload_type) { + return get_media_type(media, get_codec_from_payload(media, payload_type)); + } + + public static string? get_media_type(string media, string? codec) { + if (codec == null) return null; + if (media == "audio") { + switch (codec) { + case "pcma": + return "audio/x-alaw"; + case "pcmu": + return "audio/x-mulaw"; + } + } + return @"$media/x-$codec"; + } + + public static string? get_rtp_pay_element_name_from_payload(string media, JingleRtp.PayloadType payload_type) { + return get_pay_candidate(media, get_codec_from_payload(media, payload_type)); + } + + public static string? get_pay_candidate(string media, string? codec) { + if (codec == null) return null; + return @"rtp$(codec)pay"; + } + + public static string? get_rtp_depay_element_name_from_payload(string media, JingleRtp.PayloadType payload_type) { + return get_depay_candidate(media, get_codec_from_payload(media, payload_type)); + } + + public static string? get_depay_candidate(string media, string? codec) { + if (codec == null) return null; + return @"rtp$(codec)depay"; + } + + public static string[] get_encode_candidates(string media, string? codec) { + if (codec == null) return new string[0]; + if (media == "audio") { + switch (codec) { + case "opus": + return new string[] {"opusenc"}; + case "speex": + return new string[] {"speexenc"}; + case "pcma": + return new string[] {"alawenc"}; + case "pcmu": + return new string[] {"mulawenc"}; + } + } else if (media == "video") { + switch (codec) { + case "h264": + return new string[] {/*"msdkh264enc", */"vaapih264enc", "x264enc"}; + case "vp9": + return new string[] {/*"msdkvp9enc", */"vaapivp9enc" /*, "vp9enc" */}; + case "vp8": + return new string[] {/*"msdkvp8enc", */"vaapivp8enc", "vp8enc"}; + } + } + return new string[0]; + } + + public static string[] get_decode_candidates(string media, string? codec) { + if (codec == null) return new string[0]; + if (media == "audio") { + switch (codec) { + case "opus": + return new string[] {"opusdec"}; + case "speex": + return new string[] {"speexdec"}; + case "pcma": + return new string[] {"alawdec"}; + case "pcmu": + return new string[] {"mulawdec"}; + } + } else if (media == "video") { + switch (codec) { + case "h264": + return new string[] {/*"msdkh264dec", */"vaapih264dec"}; + case "vp9": + return new string[] {/*"msdkvp9dec", */"vaapivp9dec", "vp9dec"}; + case "vp8": + return new string[] {/*"msdkvp8dec", */"vaapivp8dec", "vp8dec"}; + } + } + return new string[0]; + } + + public static string? get_encode_prefix(string media, string codec, string encode, JingleRtp.PayloadType? payload_type) { + if (encode == "msdkh264enc") return "video/x-raw,format=NV12 ! "; + if (encode == "vaapih264enc") return "video/x-raw,format=NV12 ! "; + return null; + } + + public static string? get_encode_args(string media, string codec, string encode, JingleRtp.PayloadType? payload_type) { + // H264 + if (encode == "msdkh264enc") return @" rate-control=vbr"; + if (encode == "vaapih264enc") return @" tune=low-power"; + if (encode == "x264enc") return @" byte-stream=1 profile=baseline speed-preset=ultrafast tune=zerolatency"; + + // VP8 + if (encode == "msdkvp8enc") return " rate-control=vbr"; + if (encode == "vaapivp8enc") return " rate-control=vbr"; + if (encode == "vp8enc") return " deadline=1 error-resilient=1"; + + // OPUS + if (encode == "opusenc") { + if (payload_type != null && payload_type.parameters.has("useinbandfec", "1")) return " audio-type=voice inband-fec=true"; + return " audio-type=voice"; + } + + return null; + } + + public static string? get_encode_suffix(string media, string codec, string encode, JingleRtp.PayloadType? payload_type) { + // H264 + if (media == "video" && codec == "h264") return " ! video/x-h264,profile=constrained-baseline ! h264parse"; + return null; + } + + public uint update_bitrate(string media, JingleRtp.PayloadType payload_type, Gst.Element encode_element, uint bitrate) { + Gst.Bin? encode_bin = encode_element as Gst.Bin; + if (encode_bin == null) return 0; + string? codec = get_codec_from_payload(media, payload_type); + string? encode_name = get_encode_element_name(media, codec); + if (encode_name == null) return 0; + Gst.Element encode = encode_bin.get_by_name(@"$(encode_bin.name)_encode"); + + bitrate = uint.min(2048000, bitrate); + + switch (encode_name) { + case "msdkh264enc": + case "vaapih264enc": + case "x264enc": + case "msdkvp8enc": + case "vaapivp8enc": + bitrate = uint.min(2048000, bitrate); + encode.set("bitrate", bitrate); + return bitrate; + case "vp8enc": + bitrate = uint.min(2147483, bitrate); + encode.set("target-bitrate", bitrate * 1000); + return bitrate; + } + + return 0; + } + + public static string? get_decode_prefix(string media, string codec, string decode, JingleRtp.PayloadType? payload_type) { + return null; + } + + public static string? get_decode_args(string media, string codec, string decode, JingleRtp.PayloadType? payload_type) { + if (decode == "opusdec" && payload_type != null && payload_type.parameters.has("useinbandfec", "1")) return " use-inband-fec=true"; + if (decode == "vaapivp9dec" || decode == "vaapivp8dec" || decode == "vaapih264dec") return " max-errors=100"; + return null; + } + + public static string? get_decode_suffix(string media, string codec, string encode, JingleRtp.PayloadType? payload_type) { + return null; + } + + public static string? get_depay_args(string media, string codec, string encode, JingleRtp.PayloadType? payload_type) { + if (codec == "vp8") return " wait-for-keyframe=true"; + return null; + } + + public bool is_element_supported(string element_name) { + if (unsupported_elements.contains(element_name)) return false; + if (supported_elements.contains(element_name)) return true; + var test_element = Gst.ElementFactory.make(element_name, @"test-$element_name"); + if (test_element != null) { + supported_elements.add(element_name); + return true; + } else { + debug("%s is not supported on this platform", element_name); + unsupported_elements.add(element_name); + return false; + } + } + + public string? get_encode_element_name(string media, string? codec) { + if (!is_element_supported(get_pay_element_name(media, codec))) return null; + foreach (string candidate in get_encode_candidates(media, codec)) { + if (is_element_supported(candidate)) return candidate; + } + return null; + } + + public string? get_pay_element_name(string media, string? codec) { + string candidate = get_pay_candidate(media, codec); + if (is_element_supported(candidate)) return candidate; + return null; + } + + public string? get_decode_element_name(string media, string? codec) { + foreach (string candidate in get_decode_candidates(media, codec)) { + if (is_element_supported(candidate)) return candidate; + } + return null; + } + + public string? get_depay_element_name(string media, string? codec) { + string candidate = get_depay_candidate(media, codec); + if (is_element_supported(candidate)) return candidate; + return null; + } + + public void mark_element_unsupported(string element_name) { + unsupported_elements.add(element_name); + } + + public string? get_decode_bin_description(string media, string? codec, JingleRtp.PayloadType? payload_type, string? element_name = null, string? name = null) { + if (codec == null) return null; + string base_name = name ?? @"encode-$codec-$(Random.next_int())"; + string depay = get_depay_element_name(media, codec); + string decode = element_name ?? get_decode_element_name(media, codec); + if (depay == null || decode == null) return null; + string decode_prefix = get_decode_prefix(media, codec, decode, payload_type) ?? ""; + string decode_args = get_decode_args(media, codec, decode, payload_type) ?? ""; + string decode_suffix = get_decode_suffix(media, codec, decode, payload_type) ?? ""; + string depay_args = get_depay_args(media, codec, decode, payload_type) ?? ""; + string resample = media == "audio" ? @" ! audioresample name=$(base_name)_resample" : ""; + return @"$depay$depay_args name=$(base_name)_rtp_depay ! $decode_prefix$decode$decode_args name=$(base_name)_$(codec)_decode$decode_suffix ! $(media)convert name=$(base_name)_convert$resample"; + } + + public Gst.Element? get_decode_bin(string media, JingleRtp.PayloadType payload_type, string? name = null) { + string? codec = get_codec_from_payload(media, payload_type); + string base_name = name ?? @"encode-$codec-$(Random.next_int())"; + string? desc = get_decode_bin_description(media, codec, payload_type, null, base_name); + if (desc == null) return null; + debug("Pipeline to decode %s %s: %s", media, codec, desc); + Gst.Element bin = Gst.parse_bin_from_description(desc, true); + bin.name = name; + return bin; + } + + public string? get_encode_bin_description(string media, string? codec, JingleRtp.PayloadType? payload_type, string? element_name = null, string? name = null) { + if (codec == null) return null; + string base_name = name ?? @"encode_$(codec)_$(Random.next_int())"; + string pay = get_pay_element_name(media, codec); + string encode = element_name ?? get_encode_element_name(media, codec); + if (pay == null || encode == null) return null; + string encode_prefix = get_encode_prefix(media, codec, encode, payload_type) ?? ""; + string encode_args = get_encode_args(media, codec, encode, payload_type) ?? ""; + string encode_suffix = get_encode_suffix(media, codec, encode, payload_type) ?? ""; + string resample = media == "audio" ? @" ! audioresample name=$(base_name)_resample" : ""; + return @"$(media)convert name=$(base_name)_convert$resample ! $encode_prefix$encode$encode_args name=$(base_name)_encode$encode_suffix ! $pay pt=$(payload_type != null ? payload_type.id : 96) name=$(base_name)_rtp_pay"; + } + + public Gst.Element? get_encode_bin(string media, JingleRtp.PayloadType payload_type, string? name = null) { + string? codec = get_codec_from_payload(media, payload_type); + string base_name = name ?? @"encode_$(codec)_$(Random.next_int())"; + string? desc = get_encode_bin_description(media, codec, payload_type, null, base_name); + if (desc == null) return null; + debug("Pipeline to encode %s %s: %s", media, codec, desc); + Gst.Element bin = Gst.parse_bin_from_description(desc, true); + bin.name = name; + return bin; + } + +}
\ No newline at end of file diff --git a/plugins/rtp/src/device.vala b/plugins/rtp/src/device.vala new file mode 100644 index 00000000..e25271b1 --- /dev/null +++ b/plugins/rtp/src/device.vala @@ -0,0 +1,272 @@ +public class Dino.Plugins.Rtp.Device : MediaDevice, Object { + public Plugin plugin { get; private set; } + public Gst.Device device { get; private set; } + + private string device_name; + public string id { get { + return device_name; + }} + private string device_display_name; + public string display_name { get { + return device_display_name; + }} + public string detail_name { get { + return device.properties.get_string("alsa.card_name") ?? device.properties.get_string("alsa.id") ?? id; + }} + public Gst.Pipeline pipe { get { + return plugin.pipe; + }} + public string? media { get { + if (device.device_class.has_prefix("Audio/")) { + return "audio"; + } else if (device.device_class.has_prefix("Video/")) { + return "video"; + } else { + return null; + } + }} + public bool is_source { get { + return device.device_class.has_suffix("/Source"); + }} + public bool is_sink { get { + return device.device_class.has_suffix("/Sink"); + }} + + private Gst.Element element; + private Gst.Element tee; + private Gst.Element dsp; + private Gst.Element mixer; + private Gst.Element filter; + private Gst.Element rate; + private int links = 0; + + public Device(Plugin plugin, Gst.Device device) { + this.plugin = plugin; + update(device); + } + + public bool matches(Gst.Device device) { + if (this.device.name == device.name) return true; + return false; + } + + public void update(Gst.Device device) { + this.device = device; + this.device_name = device.name; + this.device_display_name = device.display_name; + } + + public Gst.Element? link_sink() { + if (element == null) create(); + links++; + if (mixer != null) return mixer; + if (is_sink && media == "audio") return filter; + return element; + } + + public Gst.Element? link_source() { + if (element == null) create(); + links++; + if (tee != null) return tee; + return element; + } + + public void unlink() { + if (links <= 0) { + critical("Link count below zero."); + return; + } + links--; + if (links == 0) { + destroy(); + } + } + + private Gst.Caps get_best_caps() { + if (media == "audio") { + return Gst.Caps.from_string("audio/x-raw,rate=48000,channels=1"); + } else if (media == "video" && device.caps.get_size() > 0) { + int best_index = 0; + Value? best_fraction = null; + int best_fps = 0; + int best_width = 0; + int best_height = 0; + for (int i = 0; i < device.caps.get_size(); i++) { + unowned Gst.Structure? that = device.caps.get_structure(i); + if (!that.has_name("video/x-raw")) continue; + int num = 0, den = 0, width = 0, height = 0; + if (!that.has_field("framerate")) continue; + Value framerate = that.get_value("framerate"); + if (framerate.type() == typeof(Gst.Fraction)) { + num = Gst.Value.get_fraction_numerator(framerate); + den = Gst.Value.get_fraction_denominator(framerate); + } else if (framerate.type() == typeof(Gst.ValueList)) { + for(uint j = 0; j < Gst.ValueList.get_size(framerate); j++) { + Value fraction = Gst.ValueList.get_value(framerate, j); + int in_num = Gst.Value.get_fraction_numerator(fraction); + int in_den = Gst.Value.get_fraction_denominator(fraction); + int fps = den > 0 ? (num/den) : 0; + int in_fps = in_den > 0 ? (in_num/in_den) : 0; + if (in_fps > fps) { + best_fraction = fraction; + num = in_num; + den = in_den; + } + } + } else { + debug("Unknown type for framerate: %s", framerate.type_name()); + } + if (den == 0) continue; + if (!that.has_field("width") || !that.get_int("width", out width)) continue; + if (!that.has_field("height") || !that.get_int("height", out height)) continue; + int fps = num/den; + if (best_fps < fps || best_fps == fps && best_width < width || best_fps == fps && best_width == width && best_height < height) { + best_fps = fps; + best_width = width; + best_height = height; + best_index = i; + } + } + Gst.Caps res = caps_copy_nth(device.caps, best_index); + unowned Gst.Structure? that = res.get_structure(0); + Value framerate = that.get_value("framerate"); + if (framerate.type() == typeof(Gst.ValueList)) { + that.set_value("framerate", best_fraction); + } + debug("Selected caps %s", res.to_string()); + return res; + } else if (device.caps.get_size() > 0) { + return caps_copy_nth(device.caps, 0); + } else { + return new Gst.Caps.any(); + } + } + + // Backport from gst_caps_copy_nth added in GStreamer 1.16 + private static Gst.Caps caps_copy_nth(Gst.Caps source, uint index) { + Gst.Caps target = new Gst.Caps.empty(); + target.flags = source.flags; + target.append_structure_full(source.get_structure(index).copy(), source.get_features(index).copy()); + return target; + } + + private void create() { + debug("Creating device %s", id); + plugin.pause(); + element = device.create_element(id); + pipe.add(element); + if (is_source) { + element.@set("do-timestamp", true); + filter = Gst.ElementFactory.make("capsfilter", @"caps_filter_$id"); + filter.@set("caps", get_best_caps()); + pipe.add(filter); + element.link(filter); +#if WITH_VOICE_PROCESSOR + if (media == "audio" && plugin.echoprobe != null) { + dsp = new VoiceProcessor(plugin.echoprobe as EchoProbe, element as Gst.Audio.StreamVolume); + dsp.name = @"dsp_$id"; + pipe.add(dsp); + filter.link(dsp); + } +#endif + tee = Gst.ElementFactory.make("tee", @"tee_$id"); + tee.@set("allow-not-linked", true); + pipe.add(tee); + (dsp ?? filter).link(tee); + } + if (is_sink) { + element.@set("async", false); + element.@set("sync", false); + } + if (is_sink && media == "audio") { + filter = Gst.ElementFactory.make("capsfilter", @"caps_filter_$id"); + filter.@set("caps", get_best_caps()); + pipe.add(filter); + if (plugin.echoprobe != null) { + rate = Gst.ElementFactory.make("audiorate", @"rate_$id"); + rate.@set("tolerance", 100000000); + pipe.add(rate); + filter.link(rate); + rate.link(plugin.echoprobe); + plugin.echoprobe.link(element); + } else { + filter.link(element); + } + } + plugin.unpause(); + } + + private void destroy() { + if (mixer != null) { + if (is_sink && media == "audio" && plugin.echoprobe != null) { + plugin.echoprobe.unlink(mixer); + } + int linked_sink_pads = 0; + mixer.foreach_sink_pad((_, pad) => { + if (pad.is_linked()) linked_sink_pads++; + return true; + }); + if (linked_sink_pads > 0) { + warning("%s-mixer still has %i sink pads while being destroyed", id, linked_sink_pads); + } + mixer.set_locked_state(true); + mixer.set_state(Gst.State.NULL); + mixer.unlink(element); + pipe.remove(mixer); + mixer = null; + } else if (is_sink && media == "audio") { + if (filter != null) { + filter.set_locked_state(true); + filter.set_state(Gst.State.NULL); + filter.unlink(rate ?? ((Gst.Element)plugin.echoprobe) ?? element); + pipe.remove(filter); + filter = null; + } + if (rate != null) { + rate.set_locked_state(true); + rate.set_state(Gst.State.NULL); + rate.unlink(plugin.echoprobe); + pipe.remove(rate); + rate = null; + } + if (plugin.echoprobe != null) { + plugin.echoprobe.unlink(element); + } + } + element.set_locked_state(true); + element.set_state(Gst.State.NULL); + if (filter != null) element.unlink(filter); + else if (is_source) element.unlink(tee); + pipe.remove(element); + element = null; + if (filter != null) { + filter.set_locked_state(true); + filter.set_state(Gst.State.NULL); + filter.unlink(dsp ?? tee); + pipe.remove(filter); + filter = null; + } + if (dsp != null) { + dsp.set_locked_state(true); + dsp.set_state(Gst.State.NULL); + dsp.unlink(tee); + pipe.remove(dsp); + dsp = null; + } + if (tee != null) { + int linked_src_pads = 0; + tee.foreach_src_pad((_, pad) => { + if (pad.is_linked()) linked_src_pads++; + return true; + }); + if (linked_src_pads != 0) { + warning("%s-tee still has %d src pads while being destroyed", id, linked_src_pads); + } + tee.set_locked_state(true); + tee.set_state(Gst.State.NULL); + pipe.remove(tee); + tee = null; + } + debug("Destroyed device %s", id); + } +}
\ No newline at end of file diff --git a/plugins/rtp/src/module.vala b/plugins/rtp/src/module.vala new file mode 100644 index 00000000..19a7501d --- /dev/null +++ b/plugins/rtp/src/module.vala @@ -0,0 +1,237 @@ +using Gee; +using Xmpp; +using Xmpp.Xep; + +public class Dino.Plugins.Rtp.Module : JingleRtp.Module { + private Set<string> supported_codecs = new HashSet<string>(); + private Set<string> unsupported_codecs = new HashSet<string>(); + public Plugin plugin { get; private set; } + public CodecUtil codec_util { get { + return plugin.codec_util; + }} + + public Module(Plugin plugin) { + base(); + this.plugin = plugin; + } + + private async bool pipeline_works(string media, string element_desc) { + var supported = false; + string pipeline_desc = @"$(media)testsrc is-live=true ! $element_desc ! appsink name=output"; + try { + var pipeline = Gst.parse_launch(pipeline_desc); + var output = (pipeline as Gst.Bin).get_by_name("output") as Gst.App.Sink; + SourceFunc callback = pipeline_works.callback; + var finished = false; + output.emit_signals = true; + output.new_sample.connect(() => { + if (!finished) { + finished = true; + supported = true; + Idle.add(() => { + callback(); + return Source.REMOVE; + }); + } + return Gst.FlowReturn.EOS; + }); + pipeline.bus.add_watch(Priority.DEFAULT, (_, message) => { + if (message.type == Gst.MessageType.ERROR && !finished) { + Error e; + string d; + message.parse_error(out e, out d); + debug("pipeline [%s] failed: %s", pipeline_desc, e.message); + debug(d); + finished = true; + callback(); + } + return true; + }); + Timeout.add(2000, () => { + if (!finished) { + finished = true; + callback(); + } + return Source.REMOVE; + }); + pipeline.set_state(Gst.State.PLAYING); + yield; + pipeline.set_state(Gst.State.NULL); + } catch (Error e) { + debug("pipeline [%s] failed: %s", pipeline_desc, e.message); + } + return supported; + } + + private async bool is_payload_supported(string media, JingleRtp.PayloadType payload_type) { + string? codec = CodecUtil.get_codec_from_payload(media, payload_type); + if (codec == null) return false; + if (unsupported_codecs.contains(codec)) return false; + if (supported_codecs.contains(codec)) return true; + + string? encode_element = codec_util.get_encode_element_name(media, codec); + string? decode_element = codec_util.get_decode_element_name(media, codec); + if (encode_element == null || decode_element == null) { + debug("No suitable encoder or decoder found for %s", codec); + unsupported_codecs.add(codec); + return false; + } + + string encode_bin = codec_util.get_encode_bin_description(media, codec, null, encode_element); + while (!(yield pipeline_works(media, encode_bin))) { + debug("%s not suited for encoding %s", encode_element, codec); + codec_util.mark_element_unsupported(encode_element); + encode_element = codec_util.get_encode_element_name(media, codec); + if (encode_element == null) { + debug("No suitable encoder found for %s", codec); + unsupported_codecs.add(codec); + return false; + } + encode_bin = codec_util.get_encode_bin_description(media, codec, null, encode_element); + } + debug("using %s to encode %s", encode_element, codec); + + string decode_bin = codec_util.get_decode_bin_description(media, codec, null, decode_element); + while (!(yield pipeline_works(media, @"$encode_bin ! $decode_bin"))) { + debug("%s not suited for decoding %s", decode_element, codec); + codec_util.mark_element_unsupported(decode_element); + decode_element = codec_util.get_decode_element_name(media, codec); + if (decode_element == null) { + debug("No suitable decoder found for %s", codec); + unsupported_codecs.add(codec); + return false; + } + decode_bin = codec_util.get_decode_bin_description(media, codec, null, decode_element); + } + debug("using %s to decode %s", decode_element, codec); + + supported_codecs.add(codec); + return true; + } + + public override bool is_header_extension_supported(string media, JingleRtp.HeaderExtension ext) { + if (media == "video" && ext.uri == "urn:3gpp:video-orientation") return true; + return false; + } + + public override Gee.List<JingleRtp.HeaderExtension> get_suggested_header_extensions(string media) { + Gee.List<JingleRtp.HeaderExtension> exts = new ArrayList<JingleRtp.HeaderExtension>(); + if (media == "video") { + exts.add(new JingleRtp.HeaderExtension(1, "urn:3gpp:video-orientation")); + } + return exts; + } + + public async void add_if_supported(Gee.List<JingleRtp.PayloadType> list, string media, JingleRtp.PayloadType payload_type) { + if (yield is_payload_supported(media, payload_type)) { + list.add(payload_type); + } + } + + public override async Gee.List<JingleRtp.PayloadType> get_supported_payloads(string media) { + Gee.List<JingleRtp.PayloadType> list = new ArrayList<JingleRtp.PayloadType>(JingleRtp.PayloadType.equals_func); + if (media == "audio") { + var opus = new JingleRtp.PayloadType() { channels = 2, clockrate = 48000, name = "opus", id = 99 }; + opus.parameters["useinbandfec"] = "1"; + var speex32 = new JingleRtp.PayloadType() { channels = 1, clockrate = 32000, name = "speex", id = 100 }; + var speex16 = new JingleRtp.PayloadType() { channels = 1, clockrate = 16000, name = "speex", id = 101 }; + var speex8 = new JingleRtp.PayloadType() { channels = 1, clockrate = 8000, name = "speex", id = 102 }; + var pcmu = new JingleRtp.PayloadType() { channels = 1, clockrate = 8000, name = "PCMU", id = 0 }; + var pcma = new JingleRtp.PayloadType() { channels = 1, clockrate = 8000, name = "PCMA", id = 8 }; + yield add_if_supported(list, media, opus); + yield add_if_supported(list, media, speex32); + yield add_if_supported(list, media, speex16); + yield add_if_supported(list, media, speex8); + yield add_if_supported(list, media, pcmu); + yield add_if_supported(list, media, pcma); + } else if (media == "video") { + var h264 = new JingleRtp.PayloadType() { clockrate = 90000, name = "H264", id = 96 }; + var vp9 = new JingleRtp.PayloadType() { clockrate = 90000, name = "VP9", id = 97 }; + var vp8 = new JingleRtp.PayloadType() { clockrate = 90000, name = "VP8", id = 98 }; + var rtcp_fbs = new ArrayList<JingleRtp.RtcpFeedback>(); + rtcp_fbs.add(new JingleRtp.RtcpFeedback("goog-remb")); + rtcp_fbs.add(new JingleRtp.RtcpFeedback("ccm", "fir")); + rtcp_fbs.add(new JingleRtp.RtcpFeedback("nack")); + rtcp_fbs.add(new JingleRtp.RtcpFeedback("nack", "pli")); + h264.rtcp_fbs.add_all(rtcp_fbs); + vp9.rtcp_fbs.add_all(rtcp_fbs); + vp8.rtcp_fbs.add_all(rtcp_fbs); + yield add_if_supported(list, media, h264); + yield add_if_supported(list, media, vp9); + yield add_if_supported(list, media, vp8); + } else { + warning("Unsupported media type: %s", media); + } + return list; + } + + public override async JingleRtp.PayloadType? pick_payload_type(string media, Gee.List<JingleRtp.PayloadType> payloads) { + if (media == "audio") { + foreach (JingleRtp.PayloadType type in payloads) { + if (yield is_payload_supported(media, type)) return adjust_payload_type(media, type.clone()); + } + } else if (media == "video") { + // We prefer H.264 (best support for hardware acceleration and good overall codec quality) + JingleRtp.PayloadType? h264 = payloads.first_match((it) => it.name.up() == "H264"); + if (h264 != null && yield is_payload_supported(media, h264)) return adjust_payload_type(media, h264.clone()); + // Take first of the list that we do support otherwise + foreach (JingleRtp.PayloadType type in payloads) { + if (yield is_payload_supported(media, type)) return adjust_payload_type(media, type.clone()); + } + } else { + warning("Unsupported media type: %s", media); + } + return null; + } + + public JingleRtp.PayloadType adjust_payload_type(string media, JingleRtp.PayloadType type) { + var iter = type.rtcp_fbs.iterator(); + while (iter.next()) { + var fb = iter.@get(); + switch (fb.type_) { + case "goog-remb": + if (fb.subtype != null) iter.remove(); + break; + case "ccm": + if (fb.subtype != "fir") iter.remove(); + break; + case "nack": + if (fb.subtype != null && fb.subtype != "pli") iter.remove(); + break; + default: + iter.remove(); + break; + } + } + return type; + } + + public override JingleRtp.Stream create_stream(Jingle.Content content) { + return plugin.open_stream(content); + } + + public override void close_stream(JingleRtp.Stream stream) { + var rtp_stream = stream as Rtp.Stream; + plugin.close_stream(rtp_stream); + } + + public override JingleRtp.Crypto? generate_local_crypto() { + uint8[] key_and_salt = new uint8[30]; + Crypto.randomize(key_and_salt); + return JingleRtp.Crypto.create(JingleRtp.Crypto.AES_CM_128_HMAC_SHA1_80, key_and_salt); + } + + public override JingleRtp.Crypto? pick_remote_crypto(Gee.List<JingleRtp.Crypto> cryptos) { + foreach (JingleRtp.Crypto crypto in cryptos) { + if (crypto.is_valid) return crypto; + } + return null; + } + + public override JingleRtp.Crypto? pick_local_crypto(JingleRtp.Crypto? remote) { + if (remote == null || !remote.is_valid) return null; + uint8[] key_and_salt = new uint8[30]; + Crypto.randomize(key_and_salt); + return remote.rekey(key_and_salt); + } +}
\ No newline at end of file diff --git a/plugins/rtp/src/participant.vala b/plugins/rtp/src/participant.vala new file mode 100644 index 00000000..1ca13191 --- /dev/null +++ b/plugins/rtp/src/participant.vala @@ -0,0 +1,39 @@ +using Gee; +using Xmpp; + +public class Dino.Plugins.Rtp.Participant { + public Jid full_jid { get; private set; } + + protected Gst.Pipeline pipe; + private Map<Stream, uint32> ssrcs = new HashMap<Stream, uint32>(); + + public Participant(Gst.Pipeline pipe, Jid full_jid) { + this.pipe = pipe; + this.full_jid = full_jid; + } + + public uint32 get_ssrc(Stream stream) { + if (ssrcs.has_key(stream)) { + return ssrcs[stream]; + } + return 0; + } + + public void set_ssrc(Stream stream, uint32 ssrc) { + if (ssrcs.has_key(stream)) { + warning("Learning ssrc %ul for %s in %s when it is already known as %ul", ssrc, full_jid.to_string(), stream.to_string(), ssrcs[stream]); + } else { + stream.on_destroy.connect(unset_ssrc); + } + ssrcs[stream] = ssrc; + } + + public void unset_ssrc(Stream stream) { + ssrcs.unset(stream); + stream.on_destroy.disconnect(unset_ssrc); + } + + public string to_string() { + return @"participant $full_jid"; + } +}
\ No newline at end of file diff --git a/plugins/rtp/src/plugin.vala b/plugins/rtp/src/plugin.vala new file mode 100644 index 00000000..19a266b1 --- /dev/null +++ b/plugins/rtp/src/plugin.vala @@ -0,0 +1,449 @@ +using Gee; +using Xmpp; +using Xmpp.Xep; + +public class Dino.Plugins.Rtp.Plugin : RootInterface, VideoCallPlugin, Object { + public Dino.Application app { get; private set; } + public CodecUtil codec_util { get; private set; } + public Gst.DeviceMonitor device_monitor { get; private set; } + public Gst.Pipeline pipe { get; private set; } + public Gst.Bin rtpbin { get; private set; } + public Gst.Element echoprobe { get; private set; } + + private Gee.List<Stream> streams = new ArrayList<Stream>(); + private Gee.List<Device> devices = new ArrayList<Device>(); + // private Gee.List<Participant> participants = new ArrayList<Participant>(); + + public void registered(Dino.Application app) { + this.app = app; + this.codec_util = new CodecUtil(); + app.startup.connect(startup); + app.add_option_group(Gst.init_get_option_group()); + app.stream_interactor.module_manager.initialize_account_modules.connect((account, list) => { + list.add(new Module(this)); + }); + app.plugin_registry.video_call_plugin = this; + } + + private int pause_count = 0; + public void pause() { +// if (pause_count == 0) { +// debug("Pausing pipe for modifications"); +// pipe.set_state(Gst.State.PAUSED); +// } + pause_count++; + } + public void unpause() { + pause_count--; + if (pause_count == 0) { + debug("Continue pipe after modifications"); + pipe.set_state(Gst.State.PLAYING); + } + if (pause_count < 0) warning("Pause count below zero!"); + } + + public void startup() { + device_monitor = new Gst.DeviceMonitor(); + device_monitor.show_all = true; + device_monitor.get_bus().add_watch(Priority.DEFAULT, on_device_monitor_message); + device_monitor.start(); + foreach (Gst.Device device in device_monitor.get_devices()) { + if (device.properties.has_name("pipewire-proplist") && device.device_class.has_prefix("Audio/")) continue; + if (device.properties.get_string("device.class") == "monitor") continue; + if (devices.any_match((it) => it.matches(device))) continue; + devices.add(new Device(this, device)); + } + + pipe = new Gst.Pipeline(null); + + // RTP + rtpbin = Gst.ElementFactory.make("rtpbin", null) as Gst.Bin; + if (rtpbin == null) { + warning("RTP not supported"); + pipe = null; + return; + } + rtpbin.pad_added.connect(on_rtp_pad_added); + rtpbin.@set("latency", 100); + rtpbin.@set("do-lost", true); + rtpbin.@set("do-sync-event", true); + rtpbin.@set("drop-on-latency", true); + rtpbin.connect("signal::request-pt-map", request_pt_map, this); + pipe.add(rtpbin); + +#if WITH_VOICE_PROCESSOR + // Audio echo probe + echoprobe = new EchoProbe(); + if (echoprobe != null) pipe.add(echoprobe); +#endif + + // Pipeline + pipe.auto_flush_bus = true; + pipe.bus.add_watch(GLib.Priority.DEFAULT, (_, message) => { + on_pipe_bus_message(message); + return true; + }); + pipe.set_state(Gst.State.PLAYING); + } + + private static Gst.Caps? request_pt_map(Gst.Element rtpbin, uint session, uint pt, Plugin plugin) { + debug("request-pt-map"); + return null; + } + + private void on_rtp_pad_added(Gst.Pad pad) { + debug("pad added: %s", pad.name); + if (pad.name.has_prefix("recv_rtp_src_")) { + string[] split = pad.name.split("_"); + uint8 rtpid = (uint8)int.parse(split[3]); + foreach (Stream stream in streams) { + if (stream.rtpid == rtpid) { + stream.on_ssrc_pad_added(split[4], pad); + } + } + } + if (pad.name.has_prefix("send_rtp_src_")) { + string[] split = pad.name.split("_"); + uint8 rtpid = (uint8)int.parse(split[3]); + debug("pad %s for stream %hhu", pad.name, rtpid); + foreach (Stream stream in streams) { + if (stream.rtpid == rtpid) { + stream.on_send_rtp_src_added(pad); + } + } + } + } + + private void on_pipe_bus_message(Gst.Message message) { + switch (message.type) { + case Gst.MessageType.ERROR: + Error error; + string str; + message.parse_error(out error, out str); + warning("Error in pipeline: %s", error.message); + debug(str); + break; + case Gst.MessageType.WARNING: + Error error; + string str; + message.parse_warning(out error, out str); + warning("Warning in pipeline: %s", error.message); + debug(str); + break; + case Gst.MessageType.CLOCK_LOST: + debug("Clock lost. Restarting"); + pipe.set_state(Gst.State.READY); + pipe.set_state(Gst.State.PLAYING); + break; + case Gst.MessageType.STATE_CHANGED: + // Ignore + break; + case Gst.MessageType.STREAM_STATUS: + Gst.StreamStatusType status; + Gst.Element owner; + message.parse_stream_status(out status, out owner); + if (owner != null) { + debug("%s stream changed status to %s", owner.name, status.to_string()); + } + break; + case Gst.MessageType.ELEMENT: + unowned Gst.Structure struc = message.get_structure(); + if (struc != null && message.src is Gst.Element) { + debug("Message from %s in pipeline: %s", ((Gst.Element)message.src).name, struc.to_string()); + } + break; + case Gst.MessageType.NEW_CLOCK: + debug("New clock."); + break; + case Gst.MessageType.TAG: + // Ignore + break; + case Gst.MessageType.QOS: + // Ignore + break; + case Gst.MessageType.LATENCY: + if (message.src != null && message.src.name != null && message.src is Gst.Element) { + Gst.Query latency_query = new Gst.Query.latency(); + if (((Gst.Element)message.src).query(latency_query)) { + bool live; + Gst.ClockTime min_latency, max_latency; + latency_query.parse_latency(out live, out min_latency, out max_latency); + debug("Latency message from %s: live=%s, min_latency=%s, max_latency=%s", message.src.name, live.to_string(), min_latency.to_string(), max_latency.to_string()); + } + } + break; + default: + debug("Pipe bus message: %s", message.type.to_string()); + break; + } + } + + private bool on_device_monitor_message(Gst.Bus bus, Gst.Message message) { + Gst.Device old_device = null; + Gst.Device device = null; + Device old = null; + switch (message.type) { + case Gst.MessageType.DEVICE_ADDED: + message.parse_device_added(out device); + if (device.properties.has_name("pipewire-proplist") && device.device_class.has_prefix("Audio/")) return Source.CONTINUE; + if (device.properties.get_string("device.class") == "monitor") return Source.CONTINUE; + if (devices.any_match((it) => it.matches(device))) return Source.CONTINUE; + devices.add(new Device(this, device)); + break; +#if GST_1_16 + case Gst.MessageType.DEVICE_CHANGED: + message.parse_device_changed(out device, out old_device); + if (device.properties.has_name("pipewire-proplist") && device.device_class.has_prefix("Audio/")) return Source.CONTINUE; + if (device.properties.get_string("device.class") == "monitor") return Source.CONTINUE; + old = devices.first_match((it) => it.matches(old_device)); + if (old != null) old.update(device); + break; +#endif + case Gst.MessageType.DEVICE_REMOVED: + message.parse_device_removed(out device); + if (device.properties.has_name("pipewire-proplist") && device.device_class.has_prefix("Audio/")) return Source.CONTINUE; + if (device.properties.get_string("device.class") == "monitor") return Source.CONTINUE; + old = devices.first_match((it) => it.matches(device)); + if (old != null) devices.remove(old); + break; + } + if (device != null) { + switch (device.device_class) { + case "Audio/Source": + devices_changed("audio", false); + break; + case "Audio/Sink": + devices_changed("audio", true); + break; + case "Video/Source": + devices_changed("video", false); + break; + case "Video/Sink": + devices_changed("video", true); + break; + } + } + return Source.CONTINUE; + } + + public uint8 next_free_id() { + uint8 rtpid = 0; + while (streams.size < 100 && streams.any_match((stream) => stream.rtpid == rtpid)) { + rtpid++; + } + return rtpid; + } + + // public Participant get_participant(Jid full_jid, bool self) { +// foreach (Participant participant in participants) { +// if (participant.full_jid.equals(full_jid)) { +// return participant; +// } +// } +// Participant participant; +// if (self) { +// participant = new SelfParticipant(pipe, full_jid); +// } else { +// participant = new Participant(pipe, full_jid); +// } +// participants.add(participant); +// return participant; +// } + + public Stream open_stream(Xmpp.Xep.Jingle.Content content) { + var content_params = content.content_params as Xmpp.Xep.JingleRtp.Parameters; + if (content_params == null) return null; + Stream stream; + if (content_params.media == "video") { + stream = new VideoStream(this, content); + } else { + stream = new Stream(this, content); + } + streams.add(stream); + return stream; + } + + public void close_stream(Stream stream) { + streams.remove(stream); + stream.destroy(); + } + + public void shutdown() { + device_monitor.stop(); + pipe.set_state(Gst.State.NULL); + rtpbin = null; + pipe = null; + Gst.deinit(); + } + + public bool supports(string media) { + if (rtpbin == null) return false; + + if (media == "audio") { + if (get_devices("audio", false).is_empty) return false; + if (get_devices("audio", true).is_empty) return false; + } + + if (media == "video") { + if (Gst.ElementFactory.make("gtksink", null) == null) return false; + if (get_devices("video", false).is_empty) return false; + } + + return true; + } + + public VideoCallWidget? create_widget(WidgetType type) { + if (type == WidgetType.GTK) { + return new VideoWidget(this); + } + return null; + } + + public Gee.List<MediaDevice> get_devices(string media, bool incoming) { + if (media == "video" && !incoming) { + return get_video_sources(); + } + + ArrayList<MediaDevice> result = new ArrayList<MediaDevice>(); + foreach (Device device in devices) { + if (device.media == media && (incoming && device.is_sink || !incoming && device.is_source)) { + result.add(device); + } + } + if (media == "audio") { + // Reorder sources + result.sort((media_left, media_right) => { + Device left = media_left as Device; + Device right = media_right as Device; + if (left == null) return 1; + if (right == null) return -1; + + bool left_is_pipewire = left.device.properties.has_name("pipewire-proplist"); + bool right_is_pipewire = right.device.properties.has_name("pipewire-proplist"); + + bool left_is_default = false; + left.device.properties.get_boolean("is-default", out left_is_default); + bool right_is_default = false; + right.device.properties.get_boolean("is-default", out right_is_default); + + // Prefer pipewire + if (left_is_pipewire && !right_is_pipewire) return -1; + if (right_is_pipewire && !left_is_pipewire) return 1; + + // Prefer pulse audio default device + if (left_is_default && !right_is_default) return -1; + if (right_is_default && !left_is_default) return 1; + + + return 0; + }); + } + return result; + } + + public Gee.List<MediaDevice> get_video_sources() { + ArrayList<MediaDevice> pipewire_devices = new ArrayList<MediaDevice>(); + ArrayList<MediaDevice> other_devices = new ArrayList<MediaDevice>(); + + foreach (Device device in devices) { + if (device.media != "video") continue; + if (device.is_sink) continue; + + bool is_color = false; + for (int i = 0; i < device.device.caps.get_size(); i++) { + unowned Gst.Structure structure = device.device.caps.get_structure(i); + if (structure.has_field("format") && !structure.get_string("format").has_prefix("GRAY")) { + is_color = true; + } + } + + // Don't allow grey-scale devices + if (!is_color) continue; + + if (device.device.properties.has_name("pipewire-proplist")) { + pipewire_devices.add(device); + } else { + other_devices.add(device); + } + } + + // If we have any pipewire devices, present only those. Don't want duplicated devices from pipewire and video for linux. + ArrayList<MediaDevice> devices = pipewire_devices.size > 0 ? pipewire_devices : other_devices; + + // Reorder sources + devices.sort((media_left, media_right) => { + Device left = media_left as Device; + Device right = media_right as Device; + if (left == null) return 1; + if (right == null) return -1; + + int left_fps = 0; + for (int i = 0; i < left.device.caps.get_size(); i++) { + unowned Gst.Structure structure = left.device.caps.get_structure(i); + int num = 0, den = 0; + if (structure.has_field("framerate") && structure.get_fraction("framerate", out num, out den)) left_fps = int.max(left_fps, num / den); + } + + int right_fps = 0; + for (int i = 0; i < left.device.caps.get_size(); i++) { + unowned Gst.Structure structure = left.device.caps.get_structure(i); + int num = 0, den = 0; + if (structure.has_field("framerate") && structure.get_fraction("framerate", out num, out den)) right_fps = int.max(right_fps, num / den); + } + + // More FPS is better + if (left_fps > right_fps) return -1; + if (right_fps > left_fps) return 1; + + return 0; + }); + + return devices; + } + + public Device? get_preferred_device(string media, bool incoming) { + foreach (MediaDevice media_device in get_devices(media, incoming)) { + Device? device = media_device as Device; + if (device != null) return device; + } + warning("No preferred device for %s %s. Media will not be processed.", incoming ? "incoming" : "outgoing", media); + return null; + } + + public MediaDevice? get_device(Xmpp.Xep.JingleRtp.Stream stream, bool incoming) { + Stream plugin_stream = stream as Stream; + if (plugin_stream == null) return null; + if (incoming) { + return plugin_stream.output_device ?? get_preferred_device(stream.media, incoming); + } else { + return plugin_stream.input_device ?? get_preferred_device(stream.media, incoming); + } + } + + private void dump_dot() { + string name = @"pipe-$(pipe.clock.get_time())-$(pipe.current_state)"; + Gst.Debug.bin_to_dot_file(pipe, Gst.DebugGraphDetails.ALL, name); + debug("Stored pipe details as %s", name); + } + + public void set_pause(Xmpp.Xep.JingleRtp.Stream stream, bool pause) { + Stream plugin_stream = stream as Stream; + if (plugin_stream == null) return; + if (pause) { + plugin_stream.pause(); + } else { + plugin_stream.unpause(); + } + } + + public void set_device(Xmpp.Xep.JingleRtp.Stream stream, MediaDevice? device) { + Device real_device = device as Device; + Stream plugin_stream = stream as Stream; + if (real_device == null || plugin_stream == null) return; + if (real_device.is_source) { + plugin_stream.input_device = real_device; + } else if (real_device.is_sink) { + plugin_stream.output_device = real_device; + } + } +} diff --git a/plugins/rtp/src/register_plugin.vala b/plugins/rtp/src/register_plugin.vala new file mode 100644 index 00000000..a80137ea --- /dev/null +++ b/plugins/rtp/src/register_plugin.vala @@ -0,0 +1,3 @@ +public Type register_plugin(Module module) { + return typeof (Dino.Plugins.Rtp.Plugin); +} diff --git a/plugins/rtp/src/stream.vala b/plugins/rtp/src/stream.vala new file mode 100644 index 00000000..bd8a279f --- /dev/null +++ b/plugins/rtp/src/stream.vala @@ -0,0 +1,681 @@ +using Gee; +using Xmpp; + +public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream { + public uint8 rtpid { get; private set; } + + public Plugin plugin { get; private set; } + public Gst.Pipeline pipe { get { + return plugin.pipe; + }} + public Gst.Element rtpbin { get { + return plugin.rtpbin; + }} + public CodecUtil codec_util { get { + return plugin.codec_util; + }} + private Gst.App.Sink send_rtp; + private Gst.App.Sink send_rtcp; + private Gst.App.Src recv_rtp; + private Gst.App.Src recv_rtcp; + private Gst.Element encode; + private Gst.RTP.BasePayload encode_pay; + private Gst.Element decode; + private Gst.RTP.BaseDepayload decode_depay; + private Gst.Element input; + private Gst.Element output; + private Gst.Element session; + + private Device _input_device; + public Device input_device { get { return _input_device; } set { + if (!paused) { + if (this._input_device != null) { + this._input_device.unlink(); + this._input_device = null; + } + set_input(value != null ? value.link_source() : null); + } + this._input_device = value; + }} + private Device _output_device; + public Device output_device { get { return _output_device; } set { + if (output != null) remove_output(output); + if (value != null) add_output(value.link_sink()); + this._output_device = value; + }} + + public bool created { get; private set; default = false; } + public bool paused { get; private set; default = false; } + private bool push_recv_data = false; + private string participant_ssrc = null; + + private Gst.Pad recv_rtcp_sink_pad; + private Gst.Pad recv_rtp_sink_pad; + private Gst.Pad recv_rtp_src_pad; + private Gst.Pad send_rtcp_src_pad; + private Gst.Pad send_rtp_sink_pad; + private Gst.Pad send_rtp_src_pad; + + private Crypto.Srtp.Session? crypto_session = new Crypto.Srtp.Session(); + + public Stream(Plugin plugin, Xmpp.Xep.Jingle.Content content) { + base(content); + this.plugin = plugin; + this.rtpid = plugin.next_free_id(); + + content.notify["senders"].connect_after(on_senders_changed); + } + + public void on_senders_changed() { + if (sending && input == null) { + input_device = plugin.get_preferred_device(media, false); + } + if (receiving && output == null) { + output_device = plugin.get_preferred_device(media, true); + } + } + + public override void create() { + plugin.pause(); + + // Create i/o if needed + + if (input == null && input_device == null && sending) { + input_device = plugin.get_preferred_device(media, false); + } + if (output == null && output_device == null && receiving && media == "audio") { + output_device = plugin.get_preferred_device(media, true); + } + + // Create app elements + send_rtp = Gst.ElementFactory.make("appsink", @"rtp_sink_$rtpid") as Gst.App.Sink; + send_rtp.async = false; + send_rtp.caps = CodecUtil.get_caps(media, payload_type, false); + send_rtp.emit_signals = true; + send_rtp.sync = false; + send_rtp.new_sample.connect(on_new_sample); + pipe.add(send_rtp); + + send_rtcp = Gst.ElementFactory.make("appsink", @"rtcp_sink_$rtpid") as Gst.App.Sink; + send_rtcp.async = false; + send_rtcp.caps = new Gst.Caps.empty_simple("application/x-rtcp"); + send_rtcp.emit_signals = true; + send_rtcp.sync = false; + send_rtcp.new_sample.connect(on_new_sample); + pipe.add(send_rtcp); + + recv_rtp = Gst.ElementFactory.make("appsrc", @"rtp_src_$rtpid") as Gst.App.Src; + recv_rtp.caps = CodecUtil.get_caps(media, payload_type, true); + recv_rtp.do_timestamp = true; + recv_rtp.format = Gst.Format.TIME; + recv_rtp.is_live = true; + pipe.add(recv_rtp); + + recv_rtcp = Gst.ElementFactory.make("appsrc", @"rtcp_src_$rtpid") as Gst.App.Src; + recv_rtcp.caps = new Gst.Caps.empty_simple("application/x-rtcp"); + recv_rtcp.do_timestamp = true; + recv_rtcp.format = Gst.Format.TIME; + recv_rtcp.is_live = true; + pipe.add(recv_rtcp); + + // Connect RTCP + send_rtcp_src_pad = rtpbin.get_request_pad(@"send_rtcp_src_$rtpid"); + send_rtcp_src_pad.link(send_rtcp.get_static_pad("sink")); + recv_rtcp_sink_pad = rtpbin.get_request_pad(@"recv_rtcp_sink_$rtpid"); + recv_rtcp.get_static_pad("src").link(recv_rtcp_sink_pad); + + // Connect input + encode = codec_util.get_encode_bin(media, payload_type, @"encode_$rtpid"); + encode_pay = (Gst.RTP.BasePayload)((Gst.Bin)encode).get_by_name(@"encode_$(rtpid)_rtp_pay"); + pipe.add(encode); + send_rtp_sink_pad = rtpbin.get_request_pad(@"send_rtp_sink_$rtpid"); + encode.get_static_pad("src").link(send_rtp_sink_pad); + if (input != null) { + input.link(encode); + } + + // Connect output + decode = codec_util.get_decode_bin(media, payload_type, @"decode_$rtpid"); + decode_depay = (Gst.RTP.BaseDepayload)((Gst.Bin)encode).get_by_name(@"decode_$(rtpid)_rtp_depay"); + pipe.add(decode); + if (output != null) { + decode.link(output); + } + + // Connect RTP + recv_rtp_sink_pad = rtpbin.get_request_pad(@"recv_rtp_sink_$rtpid"); + recv_rtp.get_static_pad("src").link(recv_rtp_sink_pad); + + created = true; + push_recv_data = true; + plugin.unpause(); + + GLib.Signal.emit_by_name(rtpbin, "get-session", rtpid, out session); + if (session != null && payload_type.rtcp_fbs.any_match((it) => it.type_ == "goog-remb")) { + Object internal_session; + session.@get("internal-session", out internal_session); + if (internal_session != null) { + internal_session.connect("signal::on-feedback-rtcp", on_feedback_rtcp, this); + } + Timeout.add(1000, () => remb_adjust()); + } + if (media == "video") { + codec_util.update_bitrate(media, payload_type, encode, 256); + } + } + + private uint remb = 256; + private int last_packets_lost = -1; + private uint64 last_packets_received; + private uint64 last_octets_received; + private bool remb_adjust() { + unowned Gst.Structure? stats; + if (session == null) { + debug("Session for %u finished, turning off remb adjustment", rtpid); + return Source.REMOVE; + } + session.get("stats", out stats); + if (stats == null) { + warning("No stats for session %u", rtpid); + return Source.REMOVE; + } + unowned ValueArray? source_stats; + stats.get("source-stats", typeof(ValueArray), out source_stats); + if (source_stats == null) { + warning("No source-stats for session %u", rtpid); + return Source.REMOVE; + } + foreach (Value value in source_stats.values) { + unowned Gst.Structure source_stat = (Gst.Structure) value.get_boxed(); + uint ssrc; + if (!source_stat.get_uint("ssrc", out ssrc)) continue; + if (ssrc.to_string() == participant_ssrc) { + int packets_lost; + uint64 packets_received, octets_received; + source_stat.get_int("packets-lost", out packets_lost); + source_stat.get_uint64("packets-received", out packets_received); + source_stat.get_uint64("octets-received", out octets_received); + int new_lost = packets_lost - last_packets_lost; + uint64 new_received = packets_received - last_packets_received; + uint64 new_octets = octets_received - last_octets_received; + if (new_received == 0) continue; + last_packets_lost = packets_lost; + last_packets_received = packets_received; + last_octets_received = octets_received; + double loss_rate = (double)new_lost / (double)(new_lost + new_received); + if (new_lost <= 0 || loss_rate < 0.02) { + remb = (uint)(1.08 * (double)remb); + } else if (loss_rate > 0.1) { + remb = (uint)((1.0 - 0.5 * loss_rate) * (double)remb); + } + remb = uint.max(remb, (uint)((new_octets * 8) / 1000)); + remb = uint.max(16, remb); // Never go below 16 + uint8[] data = new uint8[] { + 143, 206, 0, 5, + 0, 0, 0, 0, + 0, 0, 0, 0, + 'R', 'E', 'M', 'B', + 1, 0, 0, 0, + 0, 0, 0, 0 + }; + data[4] = (uint8)((encode_pay.ssrc >> 24) & 0xff); + data[5] = (uint8)((encode_pay.ssrc >> 16) & 0xff); + data[6] = (uint8)((encode_pay.ssrc >> 8) & 0xff); + data[7] = (uint8)(encode_pay.ssrc & 0xff); + uint8 br_exp = 0; + uint32 br_mant = remb * 1000; + uint8 bits = (uint8)Math.log2(br_mant); + if (bits > 16) { + br_exp = (uint8)bits - 16; + br_mant = br_mant >> br_exp; + } + data[17] = (uint8)((br_exp << 2) | ((br_mant >> 16) & 0x3)); + data[18] = (uint8)((br_mant >> 8) & 0xff); + data[19] = (uint8)(br_mant & 0xff); + data[20] = (uint8)((ssrc >> 24) & 0xff); + data[21] = (uint8)((ssrc >> 16) & 0xff); + data[22] = (uint8)((ssrc >> 8) & 0xff); + data[23] = (uint8)(ssrc & 0xff); + encrypt_and_send_rtcp(data); + } + } + return Source.CONTINUE; + } + + private static void on_feedback_rtcp(Gst.Element session, uint type, uint fbtype, uint sender_ssrc, uint media_ssrc, Gst.Buffer? fci, Stream self) { + if (type == 206 && fbtype == 15 && fci != null && sender_ssrc.to_string() == self.participant_ssrc) { + // https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03 + uint8[] data; + fci.extract_dup(0, fci.get_size(), out data); + if (data[0] != 'R' || data[1] != 'E' || data[2] != 'M' || data[3] != 'B') return; + uint8 br_exp = data[5] >> 2; + uint32 br_mant = (((uint32)data[5] & 0x3) << 16) + ((uint32)data[6] << 8) + (uint32)data[7]; + uint bitrate = (br_mant << br_exp) / 1000; + self.codec_util.update_bitrate(self.media, self.payload_type, self.encode, bitrate * 8); + } + } + + private void prepare_local_crypto() { + if (local_crypto != null && local_crypto.is_valid && !crypto_session.has_encrypt) { + crypto_session.set_encryption_key(local_crypto.crypto_suite, local_crypto.key, local_crypto.salt); + debug("Setting up encryption with key params %s", local_crypto.key_params); + } + } + + private Gst.FlowReturn on_new_sample(Gst.App.Sink sink) { + if (sink == null) { + debug("Sink is null"); + return Gst.FlowReturn.EOS; + } + Gst.Sample sample = sink.pull_sample(); + Gst.Buffer buffer = sample.get_buffer(); + uint8[] data; + buffer.extract_dup(0, buffer.get_size(), out data); + prepare_local_crypto(); + if (sink == send_rtp) { + if (crypto_session.has_encrypt) { + data = crypto_session.encrypt_rtp(data); + } + on_send_rtp_data(new Bytes.take((owned) data)); + } else if (sink == send_rtcp) { + encrypt_and_send_rtcp((owned) data); + } else { + warning("unknown sample"); + } + return Gst.FlowReturn.OK; + } + + private void encrypt_and_send_rtcp(owned uint8[] data) { + if (crypto_session.has_encrypt) { + data = crypto_session.encrypt_rtcp(data); + } + if (rtcp_mux) { + on_send_rtp_data(new Bytes.take((owned) data)); + } else { + on_send_rtcp_data(new Bytes.take((owned) data)); + } + } + + private static Gst.PadProbeReturn drop_probe() { + return Gst.PadProbeReturn.DROP; + } + + public override void destroy() { + // Stop network communication + push_recv_data = false; + recv_rtp.end_of_stream(); + recv_rtcp.end_of_stream(); + send_rtp.new_sample.disconnect(on_new_sample); + send_rtcp.new_sample.disconnect(on_new_sample); + + // Disconnect input device + if (input != null) { + input.unlink(encode); + input = null; + } + if (this._input_device != null) { + if (!paused) this._input_device.unlink(); + this._input_device = null; + } + + // Disconnect encode + encode.set_locked_state(true); + encode.set_state(Gst.State.NULL); + encode.get_static_pad("src").unlink(send_rtp_sink_pad); + pipe.remove(encode); + encode = null; + encode_pay = null; + + // Disconnect RTP sending + if (send_rtp_src_pad != null) { + send_rtp_src_pad.add_probe(Gst.PadProbeType.BLOCK, drop_probe); + send_rtp_src_pad.unlink(send_rtp.get_static_pad("sink")); + } + send_rtp.set_locked_state(true); + send_rtp.set_state(Gst.State.NULL); + pipe.remove(send_rtp); + send_rtp = null; + + // Disconnect decode + if (recv_rtp_src_pad != null) { + recv_rtp_src_pad.add_probe(Gst.PadProbeType.BLOCK, drop_probe); + recv_rtp_src_pad.unlink(decode.get_static_pad("sink")); + } + + // Disconnect RTP receiving + recv_rtp.set_locked_state(true); + recv_rtp.set_state(Gst.State.NULL); + recv_rtp.get_static_pad("src").unlink(recv_rtp_sink_pad); + pipe.remove(recv_rtp); + recv_rtp = null; + + // Disconnect output + if (output != null) { + decode.unlink(output); + } + decode.set_locked_state(true); + decode.set_state(Gst.State.NULL); + pipe.remove(decode); + decode = null; + decode_depay = null; + output = null; + + // Disconnect output device + if (this._output_device != null) { + this._output_device.unlink(); + this._output_device = null; + } + + // Disconnect RTCP receiving + recv_rtcp.get_static_pad("src").unlink(recv_rtcp_sink_pad); + recv_rtcp.set_locked_state(true); + recv_rtcp.set_state(Gst.State.NULL); + pipe.remove(recv_rtcp); + recv_rtcp = null; + + // Disconnect RTCP sending + send_rtcp_src_pad.unlink(send_rtcp.get_static_pad("sink")); + send_rtcp.set_locked_state(true); + send_rtcp.set_state(Gst.State.NULL); + pipe.remove(send_rtcp); + send_rtcp = null; + + // Release rtp pads + rtpbin.release_request_pad(send_rtp_sink_pad); + send_rtp_sink_pad = null; + rtpbin.release_request_pad(recv_rtp_sink_pad); + recv_rtp_sink_pad = null; + rtpbin.release_request_pad(recv_rtcp_sink_pad); + recv_rtcp_sink_pad = null; + rtpbin.release_request_pad(send_rtcp_src_pad); + send_rtcp_src_pad = null; + send_rtp_src_pad = null; + recv_rtp_src_pad = null; + + session = null; + } + + private void prepare_remote_crypto() { + if (remote_crypto != null && remote_crypto.is_valid && !crypto_session.has_decrypt) { + crypto_session.set_decryption_key(remote_crypto.crypto_suite, remote_crypto.key, remote_crypto.salt); + debug("Setting up decryption with key params %s", remote_crypto.key_params); + } + } + + private uint16 previous_video_orientation_degree = uint16.MAX; + public signal void video_orientation_changed(uint16 degree); + + public override void on_recv_rtp_data(Bytes bytes) { + if (rtcp_mux && bytes.length >= 2 && bytes.get(1) >= 192 && bytes.get(1) < 224) { + on_recv_rtcp_data(bytes); + return; + } + prepare_remote_crypto(); + uint8[] data = bytes.get_data(); + if (crypto_session.has_decrypt) { + try { + data = crypto_session.decrypt_rtp(data); + } catch (Error e) { + warning("%s (%d)", e.message, e.code); + } + } + if (push_recv_data) { + Gst.Buffer buffer = new Gst.Buffer.wrapped((owned) data); + Gst.RTP.Buffer rtp_buffer; + if (Gst.RTP.Buffer.map(buffer, Gst.MapFlags.READ, out rtp_buffer)) { + if (rtp_buffer.get_extension()) { + Xmpp.Xep.JingleRtp.HeaderExtension? ext = header_extensions.first_match((it) => it.uri == "urn:3gpp:video-orientation"); + if (ext != null) { + unowned uint8[] extension_data; + if (rtp_buffer.get_extension_onebyte_header(ext.id, 0, out extension_data) && extension_data.length == 1) { + bool camera = (extension_data[0] & 0x8) > 0; + bool flip = (extension_data[0] & 0x4) > 0; + uint8 rotation = extension_data[0] & 0x3; + uint16 rotation_degree = uint16.MAX; + switch(rotation) { + case 0: rotation_degree = 0; break; + case 1: rotation_degree = 90; break; + case 2: rotation_degree = 180; break; + case 3: rotation_degree = 270; break; + } + if (rotation_degree != previous_video_orientation_degree) { + video_orientation_changed(rotation_degree); + previous_video_orientation_degree = rotation_degree; + } + } + } + } + rtp_buffer.unmap(); + } + + // FIXME: VAPI file in Vala < 0.49.1 has a bug that results in broken ownership of buffer in push_buffer() + // We workaround by using the plain signal. The signal unfortunately will cause an unnecessary copy of + // the underlying buffer, so and some point we should move over to the new version (once we require + // Vala >= 0.50) +#if FIXED_APPSRC_PUSH_BUFFER_IN_VAPI + recv_rtp.push_buffer((owned) buffer); +#else + Gst.FlowReturn ret; + GLib.Signal.emit_by_name(recv_rtp, "push-buffer", buffer, out ret); +#endif + } + } + + public override void on_recv_rtcp_data(Bytes bytes) { + prepare_remote_crypto(); + uint8[] data = bytes.get_data(); + if (crypto_session.has_decrypt) { + try { + data = crypto_session.decrypt_rtcp(data); + } catch (Error e) { + warning("%s (%d)", e.message, e.code); + } + } + if (push_recv_data) { + Gst.Buffer buffer = new Gst.Buffer.wrapped((owned) data); + // See above +#if FIXED_APPSRC_PUSH_BUFFER_IN_VAPI + recv_rtcp.push_buffer((owned) buffer); +#else + Gst.FlowReturn ret; + GLib.Signal.emit_by_name(recv_rtcp, "push-buffer", buffer, out ret); +#endif + } + } + + public override void on_rtp_ready() { + // If full frame has been sent before the connection was ready, the counterpart would only display our video after the next full frame. + // Send a full frame to let the counterpart display our video asap + rtpbin.send_event(new Gst.Event.custom( + Gst.EventType.CUSTOM_UPSTREAM, + new Gst.Structure("GstForceKeyUnit", "all-headers", typeof(bool), true, null)) + ); + } + + public override void on_rtcp_ready() { + int rtp_session_id = (int) rtpid; + uint64 max_delay = int.MAX; + Object rtp_session; + bool rtp_sent; + GLib.Signal.emit_by_name(rtpbin, "get-internal-session", rtp_session_id, out rtp_session); + GLib.Signal.emit_by_name(rtp_session, "send-rtcp-full", max_delay, out rtp_sent); + debug("RTCP is ready, resending rtcp: %s", rtp_sent.to_string()); + } + + public void on_ssrc_pad_added(string ssrc, Gst.Pad pad) { + debug("New ssrc %s with pad %s", ssrc, pad.name); + if (participant_ssrc != null && participant_ssrc != ssrc) { + warning("Got second ssrc on stream (old: %s, new: %s), ignoring", participant_ssrc, ssrc); + return; + } + participant_ssrc = ssrc; + recv_rtp_src_pad = pad; + if (decode != null) { + plugin.pause(); + debug("Link %s to %s decode for %s", recv_rtp_src_pad.name, media, name); + recv_rtp_src_pad.link(decode.get_static_pad("sink")); + plugin.unpause(); + } + } + + public void on_send_rtp_src_added(Gst.Pad pad) { + send_rtp_src_pad = pad; + if (send_rtp != null) { + plugin.pause(); + debug("Link %s to %s send_rtp for %s", send_rtp_src_pad.name, media, name); + send_rtp_src_pad.link(send_rtp.get_static_pad("sink")); + plugin.unpause(); + } + } + + public void set_input(Gst.Element? input) { + set_input_and_pause(input, paused); + } + + private void set_input_and_pause(Gst.Element? input, bool paused) { + if (created && this.input != null) { + this.input.unlink(encode); + this.input = null; + } + + this.input = input; + this.paused = paused; + + if (created && sending && !paused && input != null) { + plugin.pause(); + input.link(encode); + plugin.unpause(); + } + } + + public void pause() { + if (paused) return; + set_input_and_pause(null, true); + if (input_device != null) input_device.unlink(); + } + + public void unpause() { + if (!paused) return; + set_input_and_pause(input_device != null ? input_device.link_source() : null, false); + } + + ulong block_probe_handler_id = 0; + public virtual void add_output(Gst.Element element) { + if (output != null) { + critical("add_output() invoked more than once"); + return; + } + this.output = element; + if (created) { + plugin.pause(); + decode.link(element); + if (block_probe_handler_id != 0) { + decode.get_static_pad("src").remove_probe(block_probe_handler_id); + } + plugin.unpause(); + } + } + + public virtual void remove_output(Gst.Element element) { + if (output != element) { + critical("remove_output() invoked without prior add_output()"); + return; + } + if (created) { + block_probe_handler_id = decode.get_static_pad("src").add_probe(Gst.PadProbeType.BLOCK, drop_probe); + decode.unlink(element); + } + if (this._output_device != null) { + this._output_device.unlink(); + this._output_device = null; + } + this.output = null; + } +} + +public class Dino.Plugins.Rtp.VideoStream : Stream { + private Gee.List<Gst.Element> outputs = new ArrayList<Gst.Element>(); + private Gst.Element output_tee; + private Gst.Element rotate; + private ulong video_orientation_changed_handler; + + public VideoStream(Plugin plugin, Xmpp.Xep.Jingle.Content content) { + base(plugin, content); + if (media != "video") critical("VideoStream created for non-video media"); + } + + public override void create() { + video_orientation_changed_handler = video_orientation_changed.connect(on_video_orientation_changed); + plugin.pause(); + rotate = Gst.ElementFactory.make("videoflip", @"video_rotate_$rtpid"); + pipe.add(rotate); + output_tee = Gst.ElementFactory.make("tee", @"video_tee_$rtpid"); + output_tee.@set("allow-not-linked", true); + pipe.add(output_tee); + rotate.link(output_tee); + add_output(rotate); + base.create(); + foreach (Gst.Element output in outputs) { + output_tee.link(output); + } + plugin.unpause(); + } + + private void on_video_orientation_changed(uint16 degree) { + if (rotate != null) { + switch (degree) { + case 0: + rotate.@set("method", 0); + break; + case 90: + rotate.@set("method", 1); + break; + case 180: + rotate.@set("method", 2); + break; + case 270: + rotate.@set("method", 3); + break; + } + } + } + + public override void destroy() { + foreach (Gst.Element output in outputs) { + output_tee.unlink(output); + } + base.destroy(); + rotate.set_locked_state(true); + rotate.set_state(Gst.State.NULL); + rotate.unlink(output_tee); + pipe.remove(rotate); + rotate = null; + output_tee.set_locked_state(true); + output_tee.set_state(Gst.State.NULL); + pipe.remove(output_tee); + output_tee = null; + disconnect(video_orientation_changed_handler); + } + + public override void add_output(Gst.Element element) { + if (element == output_tee || element == rotate) { + base.add_output(element); + return; + } + outputs.add(element); + if (output_tee != null) { + output_tee.link(element); + } + } + + public override void remove_output(Gst.Element element) { + if (element == output_tee || element == rotate) { + base.remove_output(element); + return; + } + outputs.remove(element); + if (output_tee != null) { + output_tee.unlink(element); + } + } +}
\ No newline at end of file diff --git a/plugins/rtp/src/video_widget.vala b/plugins/rtp/src/video_widget.vala new file mode 100644 index 00000000..351069a7 --- /dev/null +++ b/plugins/rtp/src/video_widget.vala @@ -0,0 +1,110 @@ +public class Dino.Plugins.Rtp.VideoWidget : Gtk.Bin, Dino.Plugins.VideoCallWidget { + private static uint last_id = 0; + + public uint id { get; private set; } + public Gst.Element element { get; private set; } + public Gtk.Widget widget { get; private set; } + + public Plugin plugin { get; private set; } + public Gst.Pipeline pipe { get { + return plugin.pipe; + }} + + private bool attached; + private Device? connected_device; + private Stream? connected_stream; + private Gst.Element convert; + + public VideoWidget(Plugin plugin) { + this.plugin = plugin; + + id = last_id++; + element = Gst.ElementFactory.make("gtksink", @"video_widget_$id"); + if (element != null) { + Gtk.Widget widget; + element.@get("widget", out widget); + element.@set("async", false); + element.@set("sync", false); + this.widget = widget; + add(widget); + widget.visible = true; + + // Listen for resolution changes + element.get_static_pad("sink").notify["caps"].connect(() => { + if (element.get_static_pad("sink").caps == null) return; + + int width, height; + element.get_static_pad("sink").caps.get_structure(0).get_int("width", out width); + element.get_static_pad("sink").caps.get_structure(0).get_int("height", out height); + resolution_changed(width, height); + }); + } else { + warning("Could not create GTK video sink. Won't display videos."); + } + } + + public void display_stream(Xmpp.Xep.JingleRtp.Stream stream) { + if (element == null) return; + detach(); + if (stream.media != "video") return; + connected_stream = stream as Stream; + if (connected_stream == null) return; + plugin.pause(); + pipe.add(element); + convert = Gst.parse_bin_from_description(@"videoconvert name=video_widget_$(id)_convert", true); + convert.name = @"video_widget_$(id)_prepare"; + pipe.add(convert); + convert.link(element); + connected_stream.add_output(convert); + element.set_locked_state(false); + plugin.unpause(); + attached = true; + } + + public void display_device(MediaDevice media_device) { + if (element == null) return; + detach(); + connected_device = media_device as Device; + if (connected_device == null) return; + plugin.pause(); + pipe.add(element); + convert = Gst.parse_bin_from_description(@"videoflip method=horizontal-flip name=video_widget_$(id)_flip ! videoconvert name=video_widget_$(id)_convert", true); + convert.name = @"video_widget_$(id)_prepare"; + pipe.add(convert); + convert.link(element); + connected_device.link_source().link(convert); + element.set_locked_state(false); + plugin.unpause(); + attached = true; + } + + public void detach() { + if (element == null) return; + if (attached) { + if (connected_stream != null) { + connected_stream.remove_output(convert); + connected_stream = null; + } + if (connected_device != null) { + connected_device.link_source().unlink(element); + connected_device.unlink(); // We get a new ref to recover the element, so unlink twice + connected_device.unlink(); + connected_device = null; + } + convert.set_locked_state(true); + convert.set_state(Gst.State.NULL); + pipe.remove(convert); + convert = null; + element.set_locked_state(true); + element.set_state(Gst.State.NULL); + pipe.remove(element); + attached = false; + } + } + + public override void dispose() { + detach(); + widget = null; + element = null; + } +}
\ No newline at end of file diff --git a/plugins/rtp/src/voice_processor.vala b/plugins/rtp/src/voice_processor.vala new file mode 100644 index 00000000..66e95d72 --- /dev/null +++ b/plugins/rtp/src/voice_processor.vala @@ -0,0 +1,176 @@ +using Gst; + +namespace Dino.Plugins.Rtp { +public static extern Buffer adjust_to_running_time(Base.Transform transform, Buffer buf); +} + +public class Dino.Plugins.Rtp.EchoProbe : Audio.Filter { + private static StaticPadTemplate sink_template = {"sink", PadDirection.SINK, PadPresence.ALWAYS, {null, "audio/x-raw,rate=48000,channels=1,layout=interleaved,format=S16LE"}}; + private static StaticPadTemplate src_template = {"src", PadDirection.SRC, PadPresence.ALWAYS, {null, "audio/x-raw,rate=48000,channels=1,layout=interleaved,format=S16LE"}}; + public Audio.Info audio_info { get; private set; } + public signal void on_new_buffer(Buffer buffer); + private uint period_samples; + private uint period_size; + private Base.Adapter adapter = new Base.Adapter(); + + static construct { + add_static_pad_template(sink_template); + add_static_pad_template(src_template); + set_static_metadata("Acoustic Echo Canceller probe", "Generic/Audio", "Gathers playback buffers for echo cancellation", "Dino Team <contact@dino.im>"); + } + + construct { + set_passthrough(true); + } + + public override bool setup(Audio.Info info) { + audio_info = info; + period_samples = info.rate / 100; // 10ms buffers + period_size = period_samples * info.bpf; + return true; + } + + + public override FlowReturn transform_ip(Buffer buf) { + lock (adapter) { + adapter.push(adjust_to_running_time(this, buf)); + while (adapter.available() > period_size) { + on_new_buffer(adapter.take_buffer(period_size)); + } + } + return FlowReturn.OK; + } + + public override bool stop() { + adapter.clear(); + return true; + } +} + +public class Dino.Plugins.Rtp.VoiceProcessor : Audio.Filter { + private static StaticPadTemplate sink_template = {"sink", PadDirection.SINK, PadPresence.ALWAYS, {null, "audio/x-raw,rate=48000,channels=1,layout=interleaved,format=S16LE"}}; + private static StaticPadTemplate src_template = {"src", PadDirection.SRC, PadPresence.ALWAYS, {null, "audio/x-raw,rate=48000,channels=1,layout=interleaved,format=S16LE"}}; + public Audio.Info audio_info { get; private set; } + private ulong process_outgoing_buffer_handler_id; + private uint adjust_delay_timeout_id; + private uint period_samples; + private uint period_size; + private Base.Adapter adapter = new Base.Adapter(); + private EchoProbe? echo_probe; + private Audio.StreamVolume? stream_volume; + private ClockTime last_reverse; + private void* native; + + static construct { + add_static_pad_template(sink_template); + add_static_pad_template(src_template); + set_static_metadata("Voice Processor (AGC, AEC, filters, etc.)", "Generic/Audio", "Pre-processes voice with WebRTC Audio Processing Library", "Dino Team <contact@dino.im>"); + } + + construct { + set_passthrough(false); + } + + public VoiceProcessor(EchoProbe? echo_probe = null, Audio.StreamVolume? stream_volume = null) { + this.echo_probe = echo_probe; + this.stream_volume = stream_volume; + } + + private static extern void* init_native(int stream_delay); + private static extern void setup_native(void* native); + private static extern void destroy_native(void* native); + private static extern void analyze_reverse_stream(void* native, Audio.Info info, Buffer buffer); + private static extern void process_stream(void* native, Audio.Info info, Buffer buffer); + private static extern void adjust_stream_delay(void* native); + private static extern void notify_gain_level(void* native, int gain_level); + private static extern int get_suggested_gain_level(void* native); + private static extern bool get_stream_has_voice(void* native); + + public override bool setup(Audio.Info info) { + debug("VoiceProcessor.setup(%s)", info.to_caps().to_string()); + audio_info = info; + period_samples = info.rate / 100; // 10ms buffers + period_size = period_samples * info.bpf; + adapter.clear(); + setup_native(native); + return true; + } + + public override bool start() { + native = init_native(150); + if (process_outgoing_buffer_handler_id == 0 && echo_probe != null) { + process_outgoing_buffer_handler_id = echo_probe.on_new_buffer.connect(process_outgoing_buffer); + } + if (stream_volume == null && sinkpad.get_peer() != null && sinkpad.get_peer().get_parent_element() is Audio.StreamVolume) { + stream_volume = sinkpad.get_peer().get_parent_element() as Audio.StreamVolume; + } + return true; + } + + private bool adjust_delay() { + if (native != null) { + adjust_stream_delay(native); + return Source.CONTINUE; + } else { + adjust_delay_timeout_id = 0; + return Source.REMOVE; + } + } + + private void process_outgoing_buffer(Buffer buffer) { + if (buffer.pts != uint64.MAX) { + last_reverse = buffer.pts; + } + analyze_reverse_stream(native, echo_probe.audio_info, buffer); + if (adjust_delay_timeout_id == 0 && echo_probe != null) { + adjust_delay_timeout_id = Timeout.add(1000, adjust_delay); + } + } + + public override FlowReturn submit_input_buffer(bool is_discont, Buffer input) { + lock (adapter) { + if (is_discont) { + adapter.clear(); + } + adapter.push(adjust_to_running_time(this, input)); + } + return FlowReturn.OK; + } + + public override FlowReturn generate_output(out Buffer output_buffer) { + lock (adapter) { + if (adapter.available() >= period_size) { + output_buffer = (Gst.Buffer) adapter.take_buffer(period_size).make_writable(); + int old_gain_level = 0; + if (stream_volume != null) { + old_gain_level = (int) (stream_volume.get_volume(Audio.StreamVolumeFormat.LINEAR) * 255.0); + notify_gain_level(native, old_gain_level); + } + process_stream(native, audio_info, output_buffer); + if (stream_volume != null) { + int new_gain_level = get_suggested_gain_level(native); + if (old_gain_level != new_gain_level) { + debug("Gain: %i -> %i", old_gain_level, new_gain_level); + stream_volume.set_volume(Audio.StreamVolumeFormat.LINEAR, ((double)new_gain_level) / 255.0); + } + } + } + } + return FlowReturn.OK; + } + + public override bool stop() { + if (process_outgoing_buffer_handler_id != 0) { + echo_probe.disconnect(process_outgoing_buffer_handler_id); + process_outgoing_buffer_handler_id = 0; + } + if (adjust_delay_timeout_id != 0) { + Source.remove(adjust_delay_timeout_id); + adjust_delay_timeout_id = 0; + } + adapter.clear(); + destroy_native(native); + native = null; + return true; + } +}
\ No newline at end of file diff --git a/plugins/rtp/src/voice_processor_native.cpp b/plugins/rtp/src/voice_processor_native.cpp new file mode 100644 index 00000000..8a052cf8 --- /dev/null +++ b/plugins/rtp/src/voice_processor_native.cpp @@ -0,0 +1,148 @@ +#include <algorithm> +#include <gst/gst.h> +#include <gst/audio/audio.h> +#include <webrtc/modules/audio_processing/include/audio_processing.h> +#include <webrtc/modules/interface/module_common_types.h> +#include <webrtc/system_wrappers/include/trace.h> + +#define SAMPLE_RATE 48000 +#define SAMPLE_CHANNELS 1 + +struct _DinoPluginsRtpVoiceProcessorNative { + webrtc::AudioProcessing *apm; + gint stream_delay; + gint last_median; + gint last_poor_delays; +}; + +extern "C" void *dino_plugins_rtp_adjust_to_running_time(GstBaseTransform *transform, GstBuffer *buffer) { + GstBuffer *copy = gst_buffer_copy(buffer); + GST_BUFFER_PTS(copy) = gst_segment_to_running_time(&transform->segment, GST_FORMAT_TIME, GST_BUFFER_PTS(buffer)); + return copy; +} + +extern "C" void *dino_plugins_rtp_voice_processor_init_native(gint stream_delay) { + _DinoPluginsRtpVoiceProcessorNative *native = new _DinoPluginsRtpVoiceProcessorNative(); + webrtc::Config config; + config.Set<webrtc::ExtendedFilter>(new webrtc::ExtendedFilter(true)); + config.Set<webrtc::ExperimentalAgc>(new webrtc::ExperimentalAgc(true, 85)); + native->apm = webrtc::AudioProcessing::Create(config); + native->stream_delay = stream_delay; + native->last_median = 0; + native->last_poor_delays = 0; + return native; +} + +extern "C" void dino_plugins_rtp_voice_processor_setup_native(void *native_ptr) { + _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr; + webrtc::AudioProcessing *apm = native->apm; + webrtc::ProcessingConfig pconfig; + pconfig.streams[webrtc::ProcessingConfig::kInputStream] = + webrtc::StreamConfig(SAMPLE_RATE, SAMPLE_CHANNELS, false); + pconfig.streams[webrtc::ProcessingConfig::kOutputStream] = + webrtc::StreamConfig(SAMPLE_RATE, SAMPLE_CHANNELS, false); + pconfig.streams[webrtc::ProcessingConfig::kReverseInputStream] = + webrtc::StreamConfig(SAMPLE_RATE, SAMPLE_CHANNELS, false); + pconfig.streams[webrtc::ProcessingConfig::kReverseOutputStream] = + webrtc::StreamConfig(SAMPLE_RATE, SAMPLE_CHANNELS, false); + apm->Initialize(pconfig); + apm->high_pass_filter()->Enable(true); + apm->echo_cancellation()->enable_drift_compensation(false); + apm->echo_cancellation()->set_suppression_level(webrtc::EchoCancellation::kModerateSuppression); + apm->echo_cancellation()->enable_delay_logging(true); + apm->echo_cancellation()->Enable(true); + apm->noise_suppression()->set_level(webrtc::NoiseSuppression::kModerate); + apm->noise_suppression()->Enable(true); + apm->gain_control()->set_analog_level_limits(0, 255); + apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveAnalog); + apm->gain_control()->set_target_level_dbfs(3); + apm->gain_control()->set_compression_gain_db(9); + apm->gain_control()->enable_limiter(true); + apm->gain_control()->Enable(true); + apm->voice_detection()->set_likelihood(webrtc::VoiceDetection::Likelihood::kLowLikelihood); + apm->voice_detection()->Enable(true); +} + +extern "C" void +dino_plugins_rtp_voice_processor_analyze_reverse_stream(void *native_ptr, GstAudioInfo *info, GstBuffer *buffer) { + _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr; + webrtc::StreamConfig config(SAMPLE_RATE, SAMPLE_CHANNELS, false); + webrtc::AudioProcessing *apm = native->apm; + + GstMapInfo map; + gst_buffer_map(buffer, &map, GST_MAP_READ); + + webrtc::AudioFrame frame; + frame.num_channels_ = info->channels; + frame.sample_rate_hz_ = info->rate; + frame.samples_per_channel_ = gst_buffer_get_size(buffer) / info->bpf; + memcpy(frame.data_, map.data, frame.samples_per_channel_ * info->bpf); + + int err = apm->AnalyzeReverseStream(&frame); + if (err < 0) g_warning("voice_processor_native.cpp: ProcessReverseStream %i", err); + + gst_buffer_unmap(buffer, &map); +} + +extern "C" void dino_plugins_rtp_voice_processor_notify_gain_level(void *native_ptr, gint gain_level) { + _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr; + webrtc::AudioProcessing *apm = native->apm; + apm->gain_control()->set_stream_analog_level(gain_level); +} + +extern "C" gint dino_plugins_rtp_voice_processor_get_suggested_gain_level(void *native_ptr) { + _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr; + webrtc::AudioProcessing *apm = native->apm; + return apm->gain_control()->stream_analog_level(); +} + +extern "C" bool dino_plugins_rtp_voice_processor_get_stream_has_voice(void *native_ptr) { + _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr; + webrtc::AudioProcessing *apm = native->apm; + return apm->voice_detection()->stream_has_voice(); +} + +extern "C" void dino_plugins_rtp_voice_processor_adjust_stream_delay(void *native_ptr) { + _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr; + webrtc::AudioProcessing *apm = native->apm; + int median, std, poor_delays; + float fraction_poor_delays; + apm->echo_cancellation()->GetDelayMetrics(&median, &std, &fraction_poor_delays); + poor_delays = (int)(fraction_poor_delays * 100.0); + if (fraction_poor_delays < 0 || (native->last_median == median && native->last_poor_delays == poor_delays)) return; + g_debug("voice_processor_native.cpp: Stream delay metrics: median=%i std=%i poor_delays=%i%%", median, std, poor_delays); + native->last_median = median; + native->last_poor_delays = poor_delays; + if (poor_delays > 90) { + native->stream_delay = std::min(std::max(0, native->stream_delay + std::min(48, std::max(median, -48))), 384); + g_debug("voice_processor_native.cpp: set stream_delay=%i", native->stream_delay); + } +} + +extern "C" void +dino_plugins_rtp_voice_processor_process_stream(void *native_ptr, GstAudioInfo *info, GstBuffer *buffer) { + _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr; + webrtc::StreamConfig config(SAMPLE_RATE, SAMPLE_CHANNELS, false); + webrtc::AudioProcessing *apm = native->apm; + + GstMapInfo map; + gst_buffer_map(buffer, &map, GST_MAP_READWRITE); + + webrtc::AudioFrame frame; + frame.num_channels_ = info->channels; + frame.sample_rate_hz_ = info->rate; + frame.samples_per_channel_ = info->rate / 100; + memcpy(frame.data_, map.data, frame.samples_per_channel_ * info->bpf); + + apm->set_stream_delay_ms(native->stream_delay); + int err = apm->ProcessStream(&frame); + if (err >= 0) memcpy(map.data, frame.data_, frame.samples_per_channel_ * info->bpf); + if (err < 0) g_warning("voice_processor_native.cpp: ProcessStream %i", err); + + gst_buffer_unmap(buffer, &map); +} + +extern "C" void dino_plugins_rtp_voice_processor_destroy_native(void *native_ptr) { + _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr; + delete native; +}
\ No newline at end of file |