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-rw-r--r--cmake/FindGstRtp.cmake14
-rw-r--r--plugins/rtp/CMakeLists.txt4
-rw-r--r--plugins/rtp/src/codec_util.vala115
-rw-r--r--plugins/rtp/src/device.vala9
-rw-r--r--plugins/rtp/src/module.vala133
-rw-r--r--plugins/rtp/src/plugin.vala14
-rw-r--r--plugins/rtp/src/stream.vala220
-rw-r--r--plugins/rtp/src/video_widget.vala10
-rw-r--r--plugins/rtp/vapi/gstreamer-rtp-1.0.vapi625
-rw-r--r--xmpp-vala/src/module/xep/0167_jingle_rtp/content_parameters.vala40
-rw-r--r--xmpp-vala/src/module/xep/0167_jingle_rtp/jingle_rtp_module.vala5
-rw-r--r--xmpp-vala/src/module/xep/0167_jingle_rtp/payload_type.vala49
-rw-r--r--xmpp-vala/src/module/xep/0167_jingle_rtp/stream.vala7
13 files changed, 1126 insertions, 119 deletions
diff --git a/cmake/FindGstRtp.cmake b/cmake/FindGstRtp.cmake
new file mode 100644
index 00000000..0756a985
--- /dev/null
+++ b/cmake/FindGstRtp.cmake
@@ -0,0 +1,14 @@
+include(PkgConfigWithFallback)
+find_pkg_config_with_fallback(GstRtp
+ PKG_CONFIG_NAME gstreamer-rtp-1.0
+ LIB_NAMES gstrtp
+ LIB_DIR_HINTS gstreamer-1.0
+ INCLUDE_NAMES gst/rtp/rtp.h
+ INCLUDE_DIR_SUFFIXES gstreamer-1.0 gstreamer-1.0/include gstreamer-rtp-1.0 gstreamer-rtp-1.0/include
+ DEPENDS Gst
+)
+
+include(FindPackageHandleStandardArgs)
+find_package_handle_standard_args(GstRtp
+ REQUIRED_VARS GstRtp_LIBRARY
+ VERSION_VAR GstRtp_VERSION)
diff --git a/plugins/rtp/CMakeLists.txt b/plugins/rtp/CMakeLists.txt
index 76d6e66d..92ec1b97 100644
--- a/plugins/rtp/CMakeLists.txt
+++ b/plugins/rtp/CMakeLists.txt
@@ -1,3 +1,4 @@
+find_package(GstRtp REQUIRED)
find_packages(RTP_PACKAGES REQUIRED
Gee
GLib
@@ -27,6 +28,7 @@ CUSTOM_VAPIS
${CMAKE_BINARY_DIR}/exports/xmpp-vala.vapi
${CMAKE_BINARY_DIR}/exports/dino.vapi
${CMAKE_BINARY_DIR}/exports/qlite.vapi
+ ${CMAKE_CURRENT_SOURCE_DIR}/vapi/gstreamer-rtp-1.0.vapi
PACKAGES
${RTP_PACKAGES}
DEFINITIONS
@@ -35,7 +37,7 @@ DEFINITIONS
add_definitions(${VALA_CFLAGS} -DG_LOG_DOMAIN="rtp" -I${CMAKE_CURRENT_SOURCE_DIR}/src)
add_library(rtp SHARED ${RTP_VALA_C})
-target_link_libraries(rtp libdino crypto-vala ${RTP_PACKAGES})
+target_link_libraries(rtp libdino crypto-vala ${RTP_PACKAGES} gstreamer-rtp-1.0)
set_target_properties(rtp PROPERTIES PREFIX "")
set_target_properties(rtp PROPERTIES LIBRARY_OUTPUT_DIRECTORY ${CMAKE_BINARY_DIR}/plugins/)
diff --git a/plugins/rtp/src/codec_util.vala b/plugins/rtp/src/codec_util.vala
index 6bd465c1..7537c11d 100644
--- a/plugins/rtp/src/codec_util.vala
+++ b/plugins/rtp/src/codec_util.vala
@@ -6,7 +6,7 @@ public class Dino.Plugins.Rtp.CodecUtil {
private Set<string> supported_elements = new HashSet<string>();
private Set<string> unsupported_elements = new HashSet<string>();
- public static Gst.Caps get_caps(string media, JingleRtp.PayloadType payload_type) {
+ public static Gst.Caps get_caps(string media, JingleRtp.PayloadType payload_type, bool incoming) {
Gst.Caps caps = new Gst.Caps.simple("application/x-rtp",
"media", typeof(string), media,
"payload", typeof(int), payload_type.id);
@@ -19,6 +19,15 @@ public class Dino.Plugins.Rtp.CodecUtil {
if (payload_type.name != null) {
s.set("encoding-name", typeof(string), payload_type.name.up());
}
+ if (incoming) {
+ foreach (JingleRtp.RtcpFeedback rtcp_fb in payload_type.rtcp_fbs) {
+ if (rtcp_fb.subtype == null) {
+ s.set(@"rtcp-fb-$(rtcp_fb.type_)", typeof(bool), true);
+ } else {
+ s.set(@"rtcp-fb-$(rtcp_fb.type_)-$(rtcp_fb.subtype)", typeof(bool), true);
+ }
+ }
+ }
return caps;
}
@@ -122,32 +131,82 @@ public class Dino.Plugins.Rtp.CodecUtil {
return new string[0];
}
- public static string? get_encode_prefix(string media, string codec, string encode) {
+ 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_suffix(string media, string codec, string encode) {
+ public static string? get_encode_args(string media, string codec, string encode, JingleRtp.PayloadType? payload_type) {
// H264
- const string h264_suffix = " ! video/x-h264,profile=constrained-baseline ! h264parse";
- if (encode == "msdkh264enc") return @" bitrate=256 rate-control=vbr target-usage=7$h264_suffix";
- if (encode == "vaapih264enc") return @" bitrate=256 quality-level=7 tune=low-power$h264_suffix";
- if (encode == "x264enc") return @" byte-stream=1 bitrate=256 profile=baseline speed-preset=ultrafast tune=zerolatency$h264_suffix";
- if (media == "video" && codec == "h264") return h264_suffix;
+ if (encode == "msdkh264enc") return @" rate-control=vbr";
+ if (encode == "vaapih264enc") return @" tune=low-power";
+ if (encode == "x264enc") return @" byte-stream=1 profile=baseline speed-preset=ultrafast tune=zerolatency";
// VP8
- if (encode == "msdkvp8enc") return " bitrate=256 rate-control=vbr target-usage=7";
- if (encode == "vaapivp8enc") return " bitrate=256 rate-control=vbr quality-level=7";
- if (encode == "vp8enc") return " target-bitrate=256000 deadline=1 error-resilient=1";
+ if (encode == "msdkvp8enc") return " rate-control=vbr";
+ if (encode == "vaapivp8enc") return " rate-control=vbr";
+ if (encode == "vp8enc") return " deadline=1 error-resilient=1";
// OPUS
- if (encode == "opusenc") return " audio-type=voice";
+ if (encode == "opusenc") {
+ if (payload_type != null && payload_type.parameters.has("useinbandfec", "1")) return " audio-type=voice inband-fec=true";
+ return " audio-type=voice";
+ }
return null;
}
- public static string? get_decode_prefix(string media, string codec, string decode) {
+ public static string? get_encode_suffix(string media, string codec, string encode, JingleRtp.PayloadType? payload_type) {
+ // H264
+ if (media == "video" && codec == "h264") return " ! video/x-h264,profile=constrained-baseline ! h264parse";
+ return null;
+ }
+
+ public uint update_bitrate(string media, JingleRtp.PayloadType payload_type, Gst.Element encode_element, uint bitrate) {
+ Gst.Bin? encode_bin = encode_element as Gst.Bin;
+ if (encode_bin == null) return 0;
+ string? codec = get_codec_from_payload(media, payload_type);
+ string? encode_name = get_encode_element_name(media, codec);
+ if (encode_name == null) return 0;
+ Gst.Element encode = encode_bin.get_by_name(@"$(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;
}
@@ -195,21 +254,24 @@ public class Dino.Plugins.Rtp.CodecUtil {
unsupported_elements.add(element_name);
}
- public string? get_decode_bin_description(string media, string? codec, string? element_name = null, string? name = null) {
+ 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) ?? "";
- string resample = media == "audio" ? @" ! audioresample name=$base_name-resample" : "";
- return @"$depay name=$base_name-rtp-depay ! $decode_prefix$decode name=$base_name-decode ! $(media)convert name=$base_name-convert$resample";
+ string decode_prefix = get_decode_prefix(media, codec, decode, payload_type) ?? "";
+ string decode_args = get_decode_args(media, codec, decode, payload_type) ?? "";
+ string decode_suffix = get_decode_suffix(media, codec, decode, payload_type) ?? "";
+ string depay_args = get_depay_args(media, codec, decode, payload_type) ?? "";
+ string resample = media == "audio" ? @" ! audioresample name=$(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-$(Random.next_int())";
- string? desc = get_decode_bin_description(media, codec, null, base_name);
+ string? desc = get_decode_bin_description(media, codec, payload_type, null, base_name);
if (desc == null) return null;
debug("Pipeline to decode %s %s: %s", media, codec, desc);
Gst.Element bin = Gst.parse_bin_from_description(desc, true);
@@ -217,22 +279,23 @@ public class Dino.Plugins.Rtp.CodecUtil {
return bin;
}
- public string? get_encode_bin_description(string media, string? codec, string? element_name = null, uint pt = 96, string? name = null) {
+ public string? get_encode_bin_description(string media, string? codec, JingleRtp.PayloadType? payload_type, string? element_name = null, string? name = null) {
if (codec == null) return null;
- string base_name = name ?? @"encode-$codec-$(Random.next_int())";
+ string base_name = name ?? @"encode_$(codec)_$(Random.next_int())";
string pay = get_pay_element_name(media, codec);
string encode = element_name ?? get_encode_element_name(media, codec);
if (pay == null || encode == null) return null;
- string encode_prefix = get_encode_prefix(media, codec, encode) ?? "";
- string encode_suffix = get_encode_suffix(media, codec, encode) ?? "";
- string resample = media == "audio" ? @" ! audioresample name=$base_name-resample" : "";
- return @"$(media)convert name=$base_name-convert$resample ! $encode_prefix$encode$encode_suffix ! $pay pt=$pt name=$base_name-rtp-pay";
+ string encode_prefix = get_encode_prefix(media, codec, encode, payload_type) ?? "";
+ string encode_args = get_encode_args(media, codec, encode, payload_type) ?? "";
+ string encode_suffix = get_encode_suffix(media, codec, encode, payload_type) ?? "";
+ string resample = media == "audio" ? @" ! audioresample name=$(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-$(Random.next_int())";
- string? desc = get_encode_bin_description(media, codec, null, payload_type.id, base_name);
+ string base_name = name ?? @"encode_$(codec)_$(Random.next_int())";
+ string? desc = get_encode_bin_description(media, codec, payload_type, null, base_name);
if (desc == null) return null;
debug("Pipeline to encode %s %s: %s", media, codec, desc);
Gst.Element bin = Gst.parse_bin_from_description(desc, true);
diff --git a/plugins/rtp/src/device.vala b/plugins/rtp/src/device.vala
index 3c9a38d2..785f853a 100644
--- a/plugins/rtp/src/device.vala
+++ b/plugins/rtp/src/device.vala
@@ -126,19 +126,20 @@ public class Dino.Plugins.Rtp.Device : MediaDevice, Object {
element = device.create_element(id);
pipe.add(element);
if (is_source) {
- filter = Gst.ElementFactory.make("capsfilter", @"$id-caps-filter");
+ element.@set("do-timestamp", true);
+ filter = Gst.ElementFactory.make("capsfilter", @"caps_filter_$id");
filter.@set("caps", get_best_caps());
pipe.add(filter);
element.link(filter);
if (media == "audio" && plugin.echoprobe != null) {
- dsp = Gst.ElementFactory.make("webrtcdsp", @"$id-dsp");
+ dsp = Gst.ElementFactory.make("webrtcdsp", @"dsp_$id");
if (dsp != null) {
dsp.@set("probe", plugin.echoprobe.name);
pipe.add(dsp);
filter.link(dsp);
}
}
- tee = Gst.ElementFactory.make("tee", @"$id-tee");
+ tee = Gst.ElementFactory.make("tee", @"tee_$id");
tee.@set("allow-not-linked", true);
pipe.add(tee);
(dsp ?? filter).link(tee);
@@ -148,7 +149,7 @@ public class Dino.Plugins.Rtp.Device : MediaDevice, Object {
element.@set("sync", false);
}
if (is_sink && media == "audio") {
- filter = Gst.ElementFactory.make("capsfilter", @"$id-caps-filter");
+ filter = Gst.ElementFactory.make("capsfilter", @"caps_filter_$id");
filter.@set("caps", get_best_caps());
pipe.add(filter);
if (plugin.echoprobe != null) {
diff --git a/plugins/rtp/src/module.vala b/plugins/rtp/src/module.vala
index 231a9dde..52cc1880 100644
--- a/plugins/rtp/src/module.vala
+++ b/plugins/rtp/src/module.vala
@@ -63,7 +63,7 @@ public class Dino.Plugins.Rtp.Module : JingleRtp.Module {
return supported;
}
- private async bool supports(string media, JingleRtp.PayloadType payload_type) {
+ private async bool is_payload_supported(string media, JingleRtp.PayloadType payload_type) {
string codec = CodecUtil.get_codec_from_payload(media, payload_type);
if (codec == null) return false;
if (unsupported_codecs.contains(codec)) return false;
@@ -77,7 +77,7 @@ public class Dino.Plugins.Rtp.Module : JingleRtp.Module {
return false;
}
- string encode_bin = codec_util.get_encode_bin_description(media, codec, encode_element);
+ string encode_bin = codec_util.get_encode_bin_description(media, codec, null, encode_element);
while (!(yield pipeline_works(media, encode_bin))) {
debug("%s not suited for encoding %s", encode_element, codec);
codec_util.mark_element_unsupported(encode_element);
@@ -87,11 +87,11 @@ public class Dino.Plugins.Rtp.Module : JingleRtp.Module {
unsupported_codecs.add(codec);
return false;
}
- encode_bin = codec_util.get_encode_bin_description(media, codec, encode_element);
+ encode_bin = codec_util.get_encode_bin_description(media, codec, null, encode_element);
}
debug("using %s to encode %s", encode_element, codec);
- string decode_bin = codec_util.get_decode_bin_description(media, codec, decode_element);
+ string decode_bin = codec_util.get_decode_bin_description(media, codec, null, decode_element);
while (!(yield pipeline_works(media, @"$encode_bin ! $decode_bin"))) {
debug("%s not suited for decoding %s", decode_element, codec);
codec_util.mark_element_unsupported(decode_element);
@@ -101,7 +101,7 @@ public class Dino.Plugins.Rtp.Module : JingleRtp.Module {
unsupported_codecs.add(codec);
return false;
}
- decode_bin = codec_util.get_decode_bin_description(media, codec, decode_element);
+ decode_bin = codec_util.get_decode_bin_description(media, codec, null, decode_element);
}
debug("using %s to decode %s", decode_element, codec);
@@ -109,8 +109,21 @@ public class Dino.Plugins.Rtp.Module : JingleRtp.Module {
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 supports(media, payload_type)) {
+ if (yield is_payload_supported(media, payload_type)) {
list.add(payload_type);
}
}
@@ -118,58 +131,34 @@ public class Dino.Plugins.Rtp.Module : JingleRtp.Module {
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") {
- yield add_if_supported(list, media, new JingleRtp.PayloadType() {
- channels = 2,
- clockrate = 48000,
- name = "opus",
- id = 99
- });
- yield add_if_supported(list, media, new JingleRtp.PayloadType() {
- channels = 1,
- clockrate = 32000,
- name = "speex",
- id = 100
- });
- yield add_if_supported(list, media, new JingleRtp.PayloadType() {
- channels = 1,
- clockrate = 16000,
- name = "speex",
- id = 101
- });
- yield add_if_supported(list, media, new JingleRtp.PayloadType() {
- channels = 1,
- clockrate = 8000,
- name = "speex",
- id = 102
- });
- yield add_if_supported(list, media, new JingleRtp.PayloadType() {
- channels = 1,
- clockrate = 8000,
- name = "PCMU",
- id = 0
- });
- yield add_if_supported(list, media, new JingleRtp.PayloadType() {
- channels = 1,
- clockrate = 8000,
- name = "PCMA",
- id = 8
- });
+ 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") {
- yield add_if_supported(list, media, new JingleRtp.PayloadType() {
- clockrate = 90000,
- name = "H264",
- id = 96
- });
- yield add_if_supported(list, media, new JingleRtp.PayloadType() {
- clockrate = 90000,
- name = "VP9",
- id = 97
- });
- yield add_if_supported(list, media, new JingleRtp.PayloadType() {
- clockrate = 90000,
- name = "VP8",
- id = 98
- });
+ 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);
}
@@ -179,11 +168,15 @@ public class Dino.Plugins.Rtp.Module : JingleRtp.Module {
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 supports(media, type)) return type;
+ 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 supports(media, type)) return type;
+ if (yield is_payload_supported(media, type)) return adjust_payload_type(media, type.clone());
}
} else {
warning("Unsupported media type: %s", media);
@@ -191,6 +184,28 @@ public class Dino.Plugins.Rtp.Module : JingleRtp.Module {
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);
}
diff --git a/plugins/rtp/src/plugin.vala b/plugins/rtp/src/plugin.vala
index 40ad1e0f..f0ad7db2 100644
--- a/plugins/rtp/src/plugin.vala
+++ b/plugins/rtp/src/plugin.vala
@@ -65,6 +65,9 @@ public class Dino.Plugins.Rtp.Plugin : RootInterface, VideoCallPlugin, Object {
}
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);
@@ -160,6 +163,17 @@ public class Dino.Plugins.Rtp.Plugin : RootInterface, VideoCallPlugin, Object {
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;
diff --git a/plugins/rtp/src/stream.vala b/plugins/rtp/src/stream.vala
index 3a63f3fa..23634aa3 100644
--- a/plugins/rtp/src/stream.vala
+++ b/plugins/rtp/src/stream.vala
@@ -19,9 +19,12 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
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 {
@@ -85,15 +88,15 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
}
// Create app elements
- send_rtp = Gst.ElementFactory.make("appsink", @"rtp-sink-$rtpid") as Gst.App.Sink;
+ send_rtp = Gst.ElementFactory.make("appsink", @"rtp_sink_$rtpid") as Gst.App.Sink;
send_rtp.async = false;
- send_rtp.caps = CodecUtil.get_caps(media, payload_type);
+ send_rtp.caps = CodecUtil.get_caps(media, payload_type, false);
send_rtp.emit_signals = true;
send_rtp.sync = false;
send_rtp.new_sample.connect(on_new_sample);
pipe.add(send_rtp);
- send_rtcp = Gst.ElementFactory.make("appsink", @"rtcp-sink-$rtpid") as Gst.App.Sink;
+ send_rtcp = Gst.ElementFactory.make("appsink", @"rtcp_sink_$rtpid") as Gst.App.Sink;
send_rtcp.async = false;
send_rtcp.caps = new Gst.Caps.empty_simple("application/x-rtcp");
send_rtcp.emit_signals = true;
@@ -101,14 +104,14 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
send_rtcp.new_sample.connect(on_new_sample);
pipe.add(send_rtcp);
- recv_rtp = Gst.ElementFactory.make("appsrc", @"rtp-src-$rtpid") as Gst.App.Src;
- recv_rtp.caps = CodecUtil.get_caps(media, payload_type);
+ 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 = Gst.ElementFactory.make("appsrc", @"rtcp_src_$rtpid") as Gst.App.Src;
recv_rtcp.caps = new Gst.Caps.empty_simple("application/x-rtcp");
recv_rtcp.do_timestamp = true;
recv_rtcp.format = Gst.Format.TIME;
@@ -122,7 +125,8 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
recv_rtcp.get_static_pad("src").link(recv_rtcp_sink_pad);
// Connect input
- encode = codec_util.get_encode_bin(media, payload_type, @"encode-$rtpid");
+ encode = 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_$rtpid");
encode.get_static_pad("src").link(send_rtp_sink_pad);
@@ -131,7 +135,8 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
}
// Connect output
- decode = codec_util.get_decode_bin(media, payload_type, @"decode-$rtpid");
+ 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);
@@ -144,6 +149,110 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
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() {
@@ -167,22 +276,26 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
if (crypto_session.has_encrypt) {
data = crypto_session.encrypt_rtp(data);
}
- on_send_rtp_data(new Bytes.take(data));
+ on_send_rtp_data(new Bytes.take((owned) data));
} else if (sink == send_rtcp) {
- if (crypto_session.has_encrypt) {
- data = crypto_session.encrypt_rtcp(data);
- }
- if (rtcp_mux) {
- on_send_rtp_data(new Bytes.take(data));
- } else {
- on_send_rtcp_data(new Bytes.take(data));
- }
+ 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;
}
@@ -211,6 +324,7 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
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) {
@@ -243,6 +357,7 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
decode.set_state(Gst.State.NULL);
pipe.remove(decode);
decode = null;
+ decode_depay = null;
output = null;
// Disconnect output device
@@ -276,6 +391,8 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
send_rtcp_src_pad = null;
send_rtp_src_pad = null;
recv_rtp_src_pad = null;
+
+ session = null;
}
private void prepare_remote_crypto() {
@@ -285,6 +402,9 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
}
}
+ 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);
@@ -301,6 +421,33 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
}
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
@@ -449,6 +596,8 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
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);
@@ -456,11 +605,15 @@ public class Dino.Plugins.Rtp.VideoStream : Stream {
}
public override void create() {
+ video_orientation_changed_handler = video_orientation_changed.connect(on_video_orientation_changed);
plugin.pause();
- output_tee = Gst.ElementFactory.make("tee", null);
+ 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);
- add_output(output_tee);
+ rotate.link(output_tee);
+ add_output(rotate);
base.create();
foreach (Gst.Element output in outputs) {
output_tee.link(output);
@@ -468,19 +621,44 @@ public class Dino.Plugins.Rtp.VideoStream : Stream {
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) {
+ if (element == output_tee || element == rotate) {
base.add_output(element);
return;
}
@@ -491,7 +669,7 @@ public class Dino.Plugins.Rtp.VideoStream : Stream {
}
public override void remove_output(Gst.Element element) {
- if (element == output_tee) {
+ if (element == output_tee || element == rotate) {
base.remove_output(element);
return;
}
diff --git a/plugins/rtp/src/video_widget.vala b/plugins/rtp/src/video_widget.vala
index fa5ba138..351069a7 100644
--- a/plugins/rtp/src/video_widget.vala
+++ b/plugins/rtp/src/video_widget.vala
@@ -19,7 +19,7 @@ public class Dino.Plugins.Rtp.VideoWidget : Gtk.Bin, Dino.Plugins.VideoCallWidge
this.plugin = plugin;
id = last_id++;
- element = Gst.ElementFactory.make("gtksink", @"video-widget-$id");
+ element = Gst.ElementFactory.make("gtksink", @"video_widget_$id");
if (element != null) {
Gtk.Widget widget;
element.@get("widget", out widget);
@@ -51,8 +51,8 @@ public class Dino.Plugins.Rtp.VideoWidget : Gtk.Bin, Dino.Plugins.VideoCallWidge
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";
+ 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);
@@ -68,8 +68,8 @@ public class Dino.Plugins.Rtp.VideoWidget : Gtk.Bin, Dino.Plugins.VideoCallWidge
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-$id-prepare";
+ convert = Gst.parse_bin_from_description(@"videoflip method=horizontal-flip name=video_widget_$(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);
diff --git a/plugins/rtp/vapi/gstreamer-rtp-1.0.vapi b/plugins/rtp/vapi/gstreamer-rtp-1.0.vapi
new file mode 100644
index 00000000..30490896
--- /dev/null
+++ b/plugins/rtp/vapi/gstreamer-rtp-1.0.vapi
@@ -0,0 +1,625 @@
+// Fixme: This is fetched from development code of Vala upstream which fixed a few bugs.
+/* gstreamer-rtp-1.0.vapi generated by vapigen, do not modify. */
+
+[CCode (cprefix = "Gst", gir_namespace = "GstRtp", gir_version = "1.0", lower_case_cprefix = "gst_")]
+namespace Gst {
+ namespace RTCP {
+ [CCode (cheader_filename = "gst/rtp/rtp.h", has_type_id = false)]
+ [GIR (name = "RTCPBuffer")]
+ public struct Buffer {
+ public weak Gst.Buffer buffer;
+ public bool add_packet (Gst.RTCP.Type type, Gst.RTCP.Packet packet);
+ public bool get_first_packet (Gst.RTCP.Packet packet);
+ public uint get_packet_count ();
+ public static bool map (Gst.Buffer buffer, Gst.MapFlags flags, out Gst.RTCP.Buffer rtcp);
+ public static Gst.Buffer @new (uint mtu);
+ public static Gst.Buffer new_copy_data ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "guint")] uint8[] data);
+ public static Gst.Buffer new_take_data ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "guint")] owned uint8[] data);
+ public bool unmap ();
+ public static bool validate (Gst.Buffer buffer);
+ public static bool validate_data ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "guint")] uint8[] data);
+ [Version (since = "1.6")]
+ public static bool validate_data_reduced ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "guint")] uint8[] data);
+ [Version (since = "1.6")]
+ public static bool validate_reduced (Gst.Buffer buffer);
+ }
+ [CCode (cheader_filename = "gst/rtp/rtp.h", has_type_id = false)]
+ [GIR (name = "RTCPPacket")]
+ public struct Packet {
+ public weak Gst.RTCP.Buffer? rtcp;
+ public uint offset;
+ [Version (since = "1.10")]
+ public bool add_profile_specific_ext ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "guint")] uint8[] data);
+ public bool add_rb (uint32 ssrc, uint8 fractionlost, int32 packetslost, uint32 exthighestseq, uint32 jitter, uint32 lsr, uint32 dlsr);
+ [Version (since = "1.10")]
+ public uint8 app_get_data ();
+ [Version (since = "1.10")]
+ public uint16 app_get_data_length ();
+ [Version (since = "1.10")]
+ public unowned string app_get_name ();
+ [Version (since = "1.10")]
+ public uint32 app_get_ssrc ();
+ [Version (since = "1.10")]
+ public uint8 app_get_subtype ();
+ [Version (since = "1.10")]
+ public bool app_set_data_length (uint16 wordlen);
+ [Version (since = "1.10")]
+ public void app_set_name (string name);
+ [Version (since = "1.10")]
+ public void app_set_ssrc (uint32 ssrc);
+ [Version (since = "1.10")]
+ public void app_set_subtype (uint8 subtype);
+ public bool bye_add_ssrc (uint32 ssrc);
+ public bool bye_add_ssrcs ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "guint")] uint32[] ssrc);
+ public uint32 bye_get_nth_ssrc (uint nth);
+ public string bye_get_reason ();
+ public uint8 bye_get_reason_len ();
+ public uint bye_get_ssrc_count ();
+ public bool bye_set_reason (string reason);
+ [Version (since = "1.10")]
+ public bool copy_profile_specific_ext ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "guint")] out uint8[] data);
+ public uint8 fb_get_fci ();
+ public uint16 fb_get_fci_length ();
+ public uint32 fb_get_media_ssrc ();
+ public uint32 fb_get_sender_ssrc ();
+ public Gst.RTCP.FBType fb_get_type ();
+ public bool fb_set_fci_length (uint16 wordlen);
+ public void fb_set_media_ssrc (uint32 ssrc);
+ public void fb_set_sender_ssrc (uint32 ssrc);
+ public void fb_set_type (Gst.RTCP.FBType type);
+ public uint8 get_count ();
+ public uint16 get_length ();
+ public bool get_padding ();
+ [Version (since = "1.10")]
+ public bool get_profile_specific_ext ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "guint")] out unowned uint8[] data);
+ [Version (since = "1.10")]
+ public uint16 get_profile_specific_ext_length ();
+ public void get_rb (uint nth, out uint32 ssrc, out uint8 fractionlost, out int32 packetslost, out uint32 exthighestseq, out uint32 jitter, out uint32 lsr, out uint32 dlsr);
+ public uint get_rb_count ();
+ public Gst.RTCP.Type get_type ();
+ public bool move_to_next ();
+ public bool remove ();
+ public uint32 rr_get_ssrc ();
+ public void rr_set_ssrc (uint32 ssrc);
+ public bool sdes_add_entry (Gst.RTCP.SDESType type, [CCode (array_length_cname = "len", array_length_pos = 1.5, array_length_type = "guint8")] uint8[] data);
+ public bool sdes_add_item (uint32 ssrc);
+ public bool sdes_copy_entry (out Gst.RTCP.SDESType type, [CCode (array_length_cname = "len", array_length_pos = 1.5, array_length_type = "guint8")] out uint8[] data);
+ public bool sdes_first_entry ();
+ public bool sdes_first_item ();
+ public bool sdes_get_entry (out Gst.RTCP.SDESType type, [CCode (array_length_cname = "len", array_length_pos = 1.5, array_length_type = "guint8")] out unowned uint8[] data);
+ public uint sdes_get_item_count ();
+ public uint32 sdes_get_ssrc ();
+ public bool sdes_next_entry ();
+ public bool sdes_next_item ();
+ public void set_rb (uint nth, uint32 ssrc, uint8 fractionlost, int32 packetslost, uint32 exthighestseq, uint32 jitter, uint32 lsr, uint32 dlsr);
+ public void sr_get_sender_info (out uint32 ssrc, out uint64 ntptime, out uint32 rtptime, out uint32 packet_count, out uint32 octet_count);
+ public void sr_set_sender_info (uint32 ssrc, uint64 ntptime, uint32 rtptime, uint32 packet_count, uint32 octet_count);
+ [Version (since = "1.16")]
+ public bool xr_first_rb ();
+ [Version (since = "1.16")]
+ public uint16 xr_get_block_length ();
+ [Version (since = "1.16")]
+ public Gst.RTCP.XRType xr_get_block_type ();
+ [Version (since = "1.16")]
+ public bool xr_get_dlrr_block (uint nth, out uint32 ssrc, out uint32 last_rr, out uint32 delay);
+ [Version (since = "1.16")]
+ public bool xr_get_prt_by_seq (uint16 seq, out uint32 receipt_time);
+ [Version (since = "1.16")]
+ public bool xr_get_prt_info (out uint32 ssrc, out uint8 thinning, out uint16 begin_seq, out uint16 end_seq);
+ [Version (since = "1.16")]
+ public bool xr_get_rle_info (out uint32 ssrc, out uint8 thinning, out uint16 begin_seq, out uint16 end_seq, out uint32 chunk_count);
+ [Version (since = "1.16")]
+ public bool xr_get_rle_nth_chunk (uint nth, out uint16 chunk);
+ [Version (since = "1.16")]
+ public bool xr_get_rrt (out uint64 timestamp);
+ [Version (since = "1.16")]
+ public uint32 xr_get_ssrc ();
+ [Version (since = "1.16")]
+ public bool xr_get_summary_info (out uint32 ssrc, out uint16 begin_seq, out uint16 end_seq);
+ [Version (since = "1.16")]
+ public bool xr_get_summary_jitter (out uint32 min_jitter, out uint32 max_jitter, out uint32 mean_jitter, out uint32 dev_jitter);
+ [Version (since = "1.16")]
+ public bool xr_get_summary_pkt (out uint32 lost_packets, out uint32 dup_packets);
+ [Version (since = "1.16")]
+ public bool xr_get_summary_ttl (out bool is_ipv4, out uint8 min_ttl, out uint8 max_ttl, out uint8 mean_ttl, out uint8 dev_ttl);
+ [Version (since = "1.16")]
+ public bool xr_get_voip_burst_metrics (out uint8 burst_density, out uint8 gap_density, out uint16 burst_duration, out uint16 gap_duration);
+ [Version (since = "1.16")]
+ public bool xr_get_voip_configuration_params (out uint8 gmin, out uint8 rx_config);
+ [Version (since = "1.16")]
+ public bool xr_get_voip_delay_metrics (out uint16 roundtrip_delay, out uint16 end_system_delay);
+ [Version (since = "1.16")]
+ public bool xr_get_voip_jitter_buffer_params (out uint16 jb_nominal, out uint16 jb_maximum, out uint16 jb_abs_max);
+ [Version (since = "1.16")]
+ public bool xr_get_voip_metrics_ssrc (out uint32 ssrc);
+ [Version (since = "1.16")]
+ public bool xr_get_voip_packet_metrics (out uint8 loss_rate, out uint8 discard_rate);
+ [Version (since = "1.16")]
+ public bool xr_get_voip_quality_metrics (out uint8 r_factor, out uint8 ext_r_factor, out uint8 mos_lq, out uint8 mos_cq);
+ [Version (since = "1.16")]
+ public bool xr_get_voip_signal_metrics (out uint8 signal_level, out uint8 noise_level, out uint8 rerl, out uint8 gmin);
+ [Version (since = "1.16")]
+ public bool xr_next_rb ();
+ }
+ [CCode (cheader_filename = "gst/rtp/rtp.h", cprefix = "GST_RTCP_", type_id = "gst_rtcpfb_type_get_type ()")]
+ [GIR (name = "RTCPFBType")]
+ public enum FBType {
+ FB_TYPE_INVALID,
+ RTPFB_TYPE_NACK,
+ RTPFB_TYPE_TMMBR,
+ RTPFB_TYPE_TMMBN,
+ RTPFB_TYPE_RTCP_SR_REQ,
+ RTPFB_TYPE_TWCC,
+ PSFB_TYPE_PLI,
+ PSFB_TYPE_SLI,
+ PSFB_TYPE_RPSI,
+ PSFB_TYPE_AFB,
+ PSFB_TYPE_FIR,
+ PSFB_TYPE_TSTR,
+ PSFB_TYPE_TSTN,
+ PSFB_TYPE_VBCN
+ }
+ [CCode (cheader_filename = "gst/rtp/rtp.h", cprefix = "GST_RTCP_SDES_", type_id = "gst_rtcpsdes_type_get_type ()")]
+ [GIR (name = "RTCPSDESType")]
+ public enum SDESType {
+ INVALID,
+ END,
+ CNAME,
+ NAME,
+ EMAIL,
+ PHONE,
+ LOC,
+ TOOL,
+ NOTE,
+ PRIV;
+ [CCode (cname = "gst_rtcp_sdes_name_to_type")]
+ public static Gst.RTCP.SDESType from_string (string name);
+ [CCode (cname = "gst_rtcp_sdes_type_to_name")]
+ public unowned string to_string ();
+ }
+ [CCode (cheader_filename = "gst/rtp/rtp.h", cprefix = "GST_RTCP_TYPE_", type_id = "gst_rtcp_type_get_type ()")]
+ [GIR (name = "RTCPType")]
+ public enum Type {
+ INVALID,
+ SR,
+ RR,
+ SDES,
+ BYE,
+ APP,
+ RTPFB,
+ PSFB,
+ XR
+ }
+ [CCode (cheader_filename = "gst/rtp/rtp.h", cprefix = "GST_RTCP_XR_TYPE_", type_id = "gst_rtcpxr_type_get_type ()")]
+ [GIR (name = "RTCPXRType")]
+ [Version (since = "1.16")]
+ public enum XRType {
+ INVALID,
+ LRLE,
+ DRLE,
+ PRT,
+ RRT,
+ DLRR,
+ SSUMM,
+ VOIP_METRICS
+ }
+ [CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTCP_MAX_BYE_SSRC_COUNT")]
+ public const int MAX_BYE_SSRC_COUNT;
+ [CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTCP_MAX_RB_COUNT")]
+ public const int MAX_RB_COUNT;
+ [CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTCP_MAX_SDES")]
+ public const int MAX_SDES;
+ [CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTCP_MAX_SDES_ITEM_COUNT")]
+ public const int MAX_SDES_ITEM_COUNT;
+ [CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTCP_REDUCED_SIZE_VALID_MASK")]
+ public const int REDUCED_SIZE_VALID_MASK;
+ [CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTCP_VALID_MASK")]
+ public const int VALID_MASK;
+ [CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTCP_VALID_VALUE")]
+ public const int VALID_VALUE;
+ [CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTCP_VERSION")]
+ public const int VERSION;
+ [CCode (cheader_filename = "gst/rtp/rtp.h")]
+ public static uint64 ntp_to_unix (uint64 ntptime);
+ [CCode (cheader_filename = "gst/rtp/rtp.h")]
+ public static uint64 unix_to_ntp (uint64 unixtime);
+ }
+ namespace RTP {
+ [CCode (cheader_filename = "gst/rtp/rtp.h", type_id = "gst_rtp_base_audio_payload_get_type ()")]
+ [GIR (name = "RTPBaseAudioPayload")]
+ public class BaseAudioPayload : Gst.RTP.BasePayload {
+ public Gst.ClockTime base_ts;
+ public int frame_duration;
+ public int frame_size;
+ public int sample_size;
+ [CCode (has_construct_function = false)]
+ protected BaseAudioPayload ();
+ public Gst.FlowReturn flush (uint payload_len, Gst.ClockTime timestamp);
+ public Gst.Base.Adapter get_adapter ();
+ public Gst.FlowReturn push ([CCode (array_length_cname = "payload_len", array_length_pos = 1.5, array_length_type = "guint")] uint8[] data, Gst.ClockTime timestamp);
+ public void set_frame_based ();
+ public void set_frame_options (int frame_duration, int frame_size);
+ public void set_sample_based ();
+ public void set_sample_options (int sample_size);
+ public void set_samplebits_options (int sample_size);
+ [NoAccessorMethod]
+ public bool buffer_list { get; set; }
+ }
+ [CCode (cheader_filename = "gst/rtp/rtp.h", type_id = "gst_rtp_base_depayload_get_type ()")]
+ [GIR (name = "RTPBaseDepayload")]
+ public abstract class BaseDepayload : Gst.Element {
+ public uint clock_rate;
+ public bool need_newsegment;
+ public weak Gst.Segment segment;
+ public weak Gst.Pad sinkpad;
+ public weak Gst.Pad srcpad;
+ [CCode (has_construct_function = false)]
+ protected BaseDepayload ();
+ [NoWrapper]
+ public virtual bool handle_event (Gst.Event event);
+ [Version (since = "1.16")]
+ public bool is_source_info_enabled ();
+ [NoWrapper]
+ public virtual bool packet_lost (Gst.Event event);
+ [NoWrapper]
+ public virtual Gst.Buffer process (Gst.Buffer @in);
+ [NoWrapper]
+ public virtual Gst.Buffer process_rtp_packet (Gst.RTP.Buffer rtp_buffer);
+ public Gst.FlowReturn push (Gst.Buffer out_buf);
+ public Gst.FlowReturn push_list (Gst.BufferList out_list);
+ [NoWrapper]
+ public virtual bool set_caps (Gst.Caps caps);
+ [Version (since = "1.16")]
+ public void set_source_info_enabled (bool enable);
+ [NoAccessorMethod]
+ [Version (since = "1.20")]
+ public bool auto_header_extension { get; set; }
+ [NoAccessorMethod]
+ [Version (since = "1.18")]
+ public int max_reorder { get; set; }
+ [NoAccessorMethod]
+ [Version (since = "1.16")]
+ public bool source_info { get; set; }
+ [NoAccessorMethod]
+ public Gst.Structure stats { owned get; }
+ [Version (since = "1.20")]
+ public signal void add_extension (owned Gst.RTP.HeaderExtension ext);
+ [Version (since = "1.20")]
+ public signal void clear_extensions ();
+ [Version (since = "1.20")]
+ public signal Gst.RTP.HeaderExtension request_extension (uint ext_id, string? ext_uri);
+ }
+ [CCode (cheader_filename = "gst/rtp/rtp.h", type_id = "gst_rtp_base_payload_get_type ()")]
+ [GIR (name = "RTPBasePayload")]
+ public abstract class BasePayload : Gst.Element {
+ [CCode (has_construct_function = false)]
+ protected BasePayload ();
+ [Version (since = "1.16")]
+ public Gst.Buffer allocate_output_buffer (uint payload_len, uint8 pad_len, uint8 csrc_count);
+ [NoWrapper]
+ public virtual Gst.Caps get_caps (Gst.Pad pad, Gst.Caps filter);
+ [Version (since = "1.16")]
+ public uint get_source_count (Gst.Buffer buffer);
+ [NoWrapper]
+ public virtual Gst.FlowReturn handle_buffer (Gst.Buffer buffer);
+ public bool is_filled (uint size, Gst.ClockTime duration);
+ [Version (since = "1.16")]
+ public bool is_source_info_enabled ();
+ public Gst.FlowReturn push (Gst.Buffer buffer);
+ public Gst.FlowReturn push_list (Gst.BufferList list);
+ [NoWrapper]
+ public virtual bool query (Gst.Pad pad, Gst.Query query);
+ [NoWrapper]
+ public virtual bool set_caps (Gst.Caps caps);
+ public void set_options (string media, bool @dynamic, string encoding_name, uint32 clock_rate);
+ [Version (since = "1.20")]
+ public bool set_outcaps_structure (Gst.Structure? s);
+ [Version (since = "1.16")]
+ public void set_source_info_enabled (bool enable);
+ [NoWrapper]
+ public virtual bool sink_event (Gst.Event event);
+ [NoWrapper]
+ public virtual bool src_event (Gst.Event event);
+ [NoAccessorMethod]
+ [Version (since = "1.20")]
+ public bool auto_header_extension { get; set; }
+ [NoAccessorMethod]
+ public int64 max_ptime { get; set; }
+ [NoAccessorMethod]
+ public int64 min_ptime { get; set; }
+ [NoAccessorMethod]
+ public uint mtu { get; set; }
+ [NoAccessorMethod]
+ [Version (since = "1.16")]
+ public bool onvif_no_rate_control { get; set; }
+ [NoAccessorMethod]
+ public bool perfect_rtptime { get; set; }
+ [NoAccessorMethod]
+ public uint pt { get; set; }
+ [NoAccessorMethod]
+ public int64 ptime_multiple { get; set; }
+ [NoAccessorMethod]
+ [Version (since = "1.18")]
+ public bool scale_rtptime { get; set; }
+ [NoAccessorMethod]
+ public uint seqnum { get; }
+ [NoAccessorMethod]
+ public int seqnum_offset { get; set; }
+ [NoAccessorMethod]
+ [Version (since = "1.16")]
+ public bool source_info { get; set; }
+ [NoAccessorMethod]
+ public uint ssrc { get; set; }
+ [NoAccessorMethod]
+ public Gst.Structure stats { owned get; }
+ [NoAccessorMethod]
+ public uint timestamp { get; }
+ [NoAccessorMethod]
+ public uint timestamp_offset { get; set; }
+ [Version (since = "1.20")]
+ public signal void add_extension (owned Gst.RTP.HeaderExtension ext);
+ [Version (since = "1.20")]
+ public signal void clear_extensions ();
+ [Version (since = "1.20")]
+ public signal Gst.RTP.HeaderExtension request_extension (uint ext_id, string ext_uri);
+ }
+ [CCode (cheader_filename = "gst/rtp/rtp.h", type_id = "gst_rtp_header_extension_get_type ()")]
+ [GIR (name = "RTPHeaderExtension")]
+ [Version (since = "1.20")]
+ public abstract class HeaderExtension : Gst.Element {
+ public uint ext_id;
+ [CCode (has_construct_function = false)]
+ protected HeaderExtension ();
+ public static Gst.RTP.HeaderExtension? create_from_uri (string uri);
+ public uint get_id ();
+ public virtual size_t get_max_size (Gst.Buffer input_meta);
+ public string get_sdp_caps_field_name ();
+ public virtual Gst.RTP.HeaderExtensionFlags get_supported_flags ();
+ public unowned string get_uri ();
+ public virtual bool read (Gst.RTP.HeaderExtensionFlags read_flags, [CCode (array_length_cname = "size", array_length_pos = 2.5, array_length_type = "gsize", type = "const guint8*")] uint8[] data, Gst.Buffer buffer);
+ public virtual bool set_attributes_from_caps (Gst.Caps caps);
+ public bool set_attributes_from_caps_simple_sdp (Gst.Caps caps);
+ public virtual bool set_caps_from_attributes (Gst.Caps caps);
+ public bool set_caps_from_attributes_simple_sdp (Gst.Caps caps);
+ public void set_id (uint ext_id);
+ public virtual bool set_non_rtp_sink_caps (Gst.Caps caps);
+ [CCode (cname = "gst_rtp_header_extension_class_set_uri")]
+ public class void set_uri (string uri);
+ public void set_wants_update_non_rtp_src_caps (bool state);
+ public virtual bool update_non_rtp_src_caps (Gst.Caps caps);
+ public virtual size_t write (Gst.Buffer input_meta, Gst.RTP.HeaderExtensionFlags write_flags, Gst.Buffer output, [CCode (array_length_cname = "size", array_length_pos = 4.1, array_length_type = "gsize", type = "guint8*")] uint8[] data);
+ }
+ [CCode (cheader_filename = "gst/rtp/rtp.h", has_type_id = false)]
+ [GIR (name = "RTPBuffer")]
+ public struct Buffer {
+ public weak Gst.Buffer buffer;
+ public uint state;
+ [CCode (array_length = false)]
+ public weak void* data[4];
+ [CCode (array_length = false)]
+ public weak size_t size[4];
+ public bool add_extension_onebyte_header (uint8 id, [CCode (array_length_cname = "size", array_length_pos = 2.1, array_length_type = "guint")] uint8[] data);
+ public bool add_extension_twobytes_header (uint8 appbits, uint8 id, [CCode (array_length_cname = "size", array_length_pos = 3.1, array_length_type = "guint")] uint8[] data);
+ [CCode (cname = "gst_buffer_add_rtp_source_meta")]
+ [Version (since = "1.16")]
+ public static unowned Gst.RTP.SourceMeta? add_rtp_source_meta (Gst.Buffer buffer, uint32? ssrc, uint32? csrc, uint csrc_count);
+ public static void allocate_data (Gst.Buffer buffer, uint payload_len, uint8 pad_len, uint8 csrc_count);
+ public static uint calc_header_len (uint8 csrc_count);
+ public static uint calc_packet_len (uint payload_len, uint8 pad_len, uint8 csrc_count);
+ public static uint calc_payload_len (uint packet_len, uint8 pad_len, uint8 csrc_count);
+ public static int compare_seqnum (uint16 seqnum1, uint16 seqnum2);
+ public static uint32 default_clock_rate (uint8 payload_type);
+ public static uint64 ext_timestamp (ref uint64 exttimestamp, uint32 timestamp);
+ public uint32 get_csrc (uint8 idx);
+ public uint8 get_csrc_count ();
+ public bool get_extension ();
+ [Version (since = "1.2")]
+ public GLib.Bytes get_extension_bytes (out uint16 bits);
+ public bool get_extension_data (out uint16 bits, [CCode (array_length = false)] out unowned uint8[] data, out uint wordlen);
+ public bool get_extension_onebyte_header (uint8 id, uint nth, [CCode (array_length_cname = "size", array_length_pos = 3.1, array_length_type = "guint")] out unowned uint8[] data);
+ [Version (since = "1.18")]
+ public static bool get_extension_onebyte_header_from_bytes (GLib.Bytes bytes, uint16 bit_pattern, uint8 id, uint nth, [CCode (array_length_cname = "size", array_length_pos = 5.1, array_length_type = "guint")] out unowned uint8[] data);
+ public bool get_extension_twobytes_header (out uint8 appbits, uint8 id, uint nth, [CCode (array_length_cname = "size", array_length_pos = 4.1, array_length_type = "guint")] out unowned uint8[] data);
+ public uint get_header_len ();
+ public bool get_marker ();
+ public uint get_packet_len ();
+ public bool get_padding ();
+ [CCode (array_length = false)]
+ public unowned uint8[] get_payload ();
+ public Gst.Buffer get_payload_buffer ();
+ [Version (since = "1.2")]
+ public GLib.Bytes get_payload_bytes ();
+ public uint get_payload_len ();
+ public Gst.Buffer get_payload_subbuffer (uint offset, uint len);
+ public uint8 get_payload_type ();
+ [CCode (cname = "gst_buffer_get_rtp_source_meta")]
+ [Version (since = "1.16")]
+ public static unowned Gst.RTP.SourceMeta? get_rtp_source_meta (Gst.Buffer buffer);
+ public uint16 get_seq ();
+ public uint32 get_ssrc ();
+ public uint32 get_timestamp ();
+ public uint8 get_version ();
+ public static bool map (Gst.Buffer buffer, Gst.MapFlags flags, out Gst.RTP.Buffer rtp);
+ public static Gst.Buffer new_allocate (uint payload_len, uint8 pad_len, uint8 csrc_count);
+ public static Gst.Buffer new_allocate_len (uint packet_len, uint8 pad_len, uint8 csrc_count);
+ public static Gst.Buffer new_copy_data ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "gsize")] uint8[] data);
+ public static Gst.Buffer new_take_data ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "gsize")] owned uint8[] data);
+ public void pad_to (uint len);
+ public void set_csrc (uint8 idx, uint32 csrc);
+ public void set_extension (bool extension);
+ public bool set_extension_data (uint16 bits, uint16 length);
+ public void set_marker (bool marker);
+ public void set_packet_len (uint len);
+ public void set_padding (bool padding);
+ public void set_payload_type (uint8 payload_type);
+ public void set_seq (uint16 seq);
+ public void set_ssrc (uint32 ssrc);
+ public void set_timestamp (uint32 timestamp);
+ public void set_version (uint8 version);
+ public void unmap ();
+ }
+ [CCode (cheader_filename = "gst/rtp/rtp.h", has_type_id = false)]
+ [GIR (name = "RTPPayloadInfo")]
+ public struct PayloadInfo {
+ public uint8 payload_type;
+ public weak string media;
+ public weak string encoding_name;
+ public uint clock_rate;
+ public weak string encoding_parameters;
+ public uint bitrate;
+ }
+ [CCode (cheader_filename = "gst/rtp/rtp.h", has_type_id = false)]
+ [GIR (name = "RTPSourceMeta")]
+ [Version (since = "1.16")]
+ public struct SourceMeta {
+ public Gst.Meta meta;
+ public uint32 ssrc;
+ public bool ssrc_valid;
+ [CCode (array_length = false)]
+ public weak uint32 csrc[15];
+ public uint csrc_count;
+ public bool append_csrc ([CCode (array_length_cname = "csrc_count", array_length_pos = 1.1, array_length_type = "guint", type = "const guint32*")] uint32[] csrc);
+ public uint get_source_count ();
+ public bool set_ssrc (uint32? ssrc);
+ }
+ [CCode (cheader_filename = "gst/rtp/rtp.h", cprefix = "GST_RTP_BUFFER_FLAG_", type_id = "gst_rtp_buffer_flags_get_type ()")]
+ [Flags]
+ [GIR (name = "RTPBufferFlags")]
+ [Version (since = "1.10")]
+ public enum BufferFlags {
+ RETRANSMISSION,
+ REDUNDANT,
+ LAST
+ }
+ [CCode (cheader_filename = "gst/rtp/rtp.h", cprefix = "GST_RTP_BUFFER_MAP_FLAG_", type_id = "gst_rtp_buffer_map_flags_get_type ()")]
+ [Flags]
+ [GIR (name = "RTPBufferMapFlags")]
+ [Version (since = "1.6.1")]
+ public enum BufferMapFlags {
+ SKIP_PADDING,
+ LAST
+ }
+ [CCode (cheader_filename = "gst/rtp/rtp.h", cprefix = "GST_RTP_HEADER_EXTENSION_", type_id = "gst_rtp_header_extension_flags_get_type ()")]
+ [Flags]
+ [GIR (name = "RTPHeaderExtensionFlags")]
+ [Version (since = "1.20")]
+ public enum HeaderExtensionFlags {
+ ONE_BYTE,
+ TWO_BYTE
+ }
+ [CCode (cheader_filename = "gst/rtp/rtp.h", cprefix = "GST_RTP_PAYLOAD_", type_id = "gst_rtp_payload_get_type ()")]
+ [GIR (name = "RTPPayload")]
+ public enum Payload {
+ PCMU,
+ @1016,
+ G721,
+ GSM,
+ G723,
+ DVI4_8000,
+ DVI4_16000,
+ LPC,
+ PCMA,
+ G722,
+ L16_STEREO,
+ L16_MONO,
+ QCELP,
+ CN,
+ MPA,
+ G728,
+ DVI4_11025,
+ DVI4_22050,
+ G729,
+ CELLB,
+ JPEG,
+ NV,
+ H261,
+ MPV,
+ MP2T,
+ H263;
+ public const string @1016_STRING;
+ public const string CELLB_STRING;
+ public const string CN_STRING;
+ public const string DVI4_11025_STRING;
+ public const string DVI4_16000_STRING;
+ public const string DVI4_22050_STRING;
+ public const string DVI4_8000_STRING;
+ public const string DYNAMIC_STRING;
+ public const string G721_STRING;
+ public const string G722_STRING;
+ public const int G723_53;
+ public const string G723_53_STRING;
+ public const int G723_63;
+ public const string G723_63_STRING;
+ public const string G723_STRING;
+ public const string G728_STRING;
+ public const string G729_STRING;
+ public const string GSM_STRING;
+ public const string H261_STRING;
+ public const string H263_STRING;
+ public const string JPEG_STRING;
+ public const string L16_MONO_STRING;
+ public const string L16_STEREO_STRING;
+ public const string LPC_STRING;
+ public const string MP2T_STRING;
+ public const string MPA_STRING;
+ public const string MPV_STRING;
+ public const string NV_STRING;
+ public const string PCMA_STRING;
+ public const string PCMU_STRING;
+ public const string QCELP_STRING;
+ public const int TS41;
+ public const string TS41_STRING;
+ public const int TS48;
+ public const string TS48_STRING;
+ }
+ [CCode (cheader_filename = "gst/rtp/rtp.h", cprefix = "GST_RTP_PROFILE_", type_id = "gst_rtp_profile_get_type ()")]
+ [GIR (name = "RTPProfile")]
+ [Version (since = "1.6")]
+ public enum Profile {
+ UNKNOWN,
+ AVP,
+ SAVP,
+ AVPF,
+ SAVPF
+ }
+ [CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTP_HDREXT_BASE")]
+ public const string HDREXT_BASE;
+ [CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTP_HDREXT_ELEMENT_CLASS")]
+ [Version (since = "1.20")]
+ public const string HDREXT_ELEMENT_CLASS;
+ [CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTP_HDREXT_NTP_56")]
+ public const string HDREXT_NTP_56;
+ [CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTP_HDREXT_NTP_56_SIZE")]
+ public const int HDREXT_NTP_56_SIZE;
+ [CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTP_HDREXT_NTP_64")]
+ public const string HDREXT_NTP_64;
+ [CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTP_HDREXT_NTP_64_SIZE")]
+ public const int HDREXT_NTP_64_SIZE;
+ [CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTP_HEADER_EXTENSION_URI_METADATA_KEY")]
+ [Version (since = "1.20")]
+ public const string HEADER_EXTENSION_URI_METADATA_KEY;
+ [CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTP_SOURCE_META_MAX_CSRC_COUNT")]
+ public const int SOURCE_META_MAX_CSRC_COUNT;
+ [CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTP_VERSION")]
+ public const int VERSION;
+ [CCode (cheader_filename = "gst/rtp/rtp.h")]
+ [Version (since = "1.20")]
+ public static GLib.List<Gst.RTP.HeaderExtension> get_header_extension_list ();
+ [CCode (cheader_filename = "gst/rtp/rtp.h")]
+ public static bool hdrext_get_ntp_56 ([CCode (array_length_cname = "size", array_length_pos = 1.5, array_length_type = "guint")] uint8[] data, out uint64 ntptime);
+ [CCode (cheader_filename = "gst/rtp/rtp.h")]
+ public static bool hdrext_get_ntp_64 ([CCode (array_length_cname = "size", array_length_pos = 1.5, array_length_type = "guint")] uint8[] data, out uint64 ntptime);
+ [CCode (cheader_filename = "gst/rtp/rtp.h")]
+ public static bool hdrext_set_ntp_56 (void* data, uint size, uint64 ntptime);
+ [CCode (cheader_filename = "gst/rtp/rtp.h")]
+ public static bool hdrext_set_ntp_64 (void* data, uint size, uint64 ntptime);
+ [CCode (cheader_filename = "gst/rtp/rtp.h")]
+ public static unowned Gst.RTP.PayloadInfo? payload_info_for_name (string media, string encoding_name);
+ [CCode (cheader_filename = "gst/rtp/rtp.h")]
+ public static unowned Gst.RTP.PayloadInfo? payload_info_for_pt (uint8 payload_type);
+ [CCode (cheader_filename = "gst/rtp/rtp.h")]
+ public static GLib.Type source_meta_api_get_type ();
+ [CCode (cheader_filename = "gst/rtp/rtp.h")]
+ public static unowned Gst.MetaInfo? source_meta_get_info ();
+ }
+}
diff --git a/xmpp-vala/src/module/xep/0167_jingle_rtp/content_parameters.vala b/xmpp-vala/src/module/xep/0167_jingle_rtp/content_parameters.vala
index d6f1acd2..d4440169 100644
--- a/xmpp-vala/src/module/xep/0167_jingle_rtp/content_parameters.vala
+++ b/xmpp-vala/src/module/xep/0167_jingle_rtp/content_parameters.vala
@@ -17,6 +17,7 @@ public class Xmpp.Xep.JingleRtp.Parameters : Jingle.ContentParameters, Object {
public bool encryption_required { get; private set; default = false; }
public PayloadType? agreed_payload_type { get; private set; }
public Gee.List<PayloadType> payload_types = new ArrayList<PayloadType>(PayloadType.equals_func);
+ public Gee.List<HeaderExtension> header_extensions = new ArrayList<HeaderExtension>();
public Gee.List<Crypto> remote_cryptos = new ArrayList<Crypto>();
public Crypto? local_crypto = null;
public Crypto? remote_crypto = null;
@@ -54,9 +55,12 @@ public class Xmpp.Xep.JingleRtp.Parameters : Jingle.ContentParameters, Object {
this.remote_cryptos.add(Crypto.parse(crypto));
}
}
- foreach (StanzaNode payloadType in node.get_subnodes("payload-type")) {
+ foreach (StanzaNode payloadType in node.get_subnodes(PayloadType.NAME)) {
this.payload_types.add(PayloadType.parse(payloadType));
}
+ foreach (StanzaNode subnode in node.get_subnodes(HeaderExtension.NAME, HeaderExtension.NS_URI)) {
+ this.header_extensions.add(HeaderExtension.parse(subnode));
+ }
}
public async void handle_proposed_content(XmppStream stream, Jingle.Session session, Jingle.Content content) {
@@ -66,6 +70,11 @@ public class Xmpp.Xep.JingleRtp.Parameters : Jingle.ContentParameters, Object {
content.reject();
return;
}
+ // Drop unsupported header extensions
+ var iter = header_extensions.iterator();
+ while(iter.next()) {
+ if (!parent.is_header_extension_supported(media, iter.@get())) iter.remove();
+ }
remote_crypto = parent.pick_remote_crypto(remote_cryptos);
if (local_crypto == null && remote_crypto != null) {
local_crypto = parent.pick_local_crypto(remote_crypto);
@@ -151,7 +160,7 @@ public class Xmpp.Xep.JingleRtp.Parameters : Jingle.ContentParameters, Object {
Gee.List<StanzaNode> crypto_nodes = description_node.get_deep_subnodes("encryption", "crypto");
if (crypto_nodes.size == 0) {
- warning("Counterpart didn't include any cryptos");
+ debug("Counterpart didn't include any cryptos");
if (encryption_required) {
return;
}
@@ -182,6 +191,9 @@ public class Xmpp.Xep.JingleRtp.Parameters : Jingle.ContentParameters, Object {
ret.put_node(payload_type.to_xml());
}
}
+ foreach (HeaderExtension ext in header_extensions) {
+ ret.put_node(ext.to_xml());
+ }
if (local_crypto != null) {
ret.put_node(new StanzaNode.build("encryption", NS_URI)
.put_node(local_crypto.to_xml()));
@@ -191,4 +203,28 @@ public class Xmpp.Xep.JingleRtp.Parameters : Jingle.ContentParameters, Object {
}
return ret;
}
+}
+
+public class Xmpp.Xep.JingleRtp.HeaderExtension {
+ public const string NS_URI = "urn:xmpp:jingle:apps:rtp:rtp-hdrext:0";
+ public const string NAME = "rtp-hdrext";
+
+ public uint8 id { get; private set; }
+ public string uri { get; private set; }
+
+ public HeaderExtension(uint8 id, string uri) {
+ this.id = id;
+ this.uri = uri;
+ }
+
+ public static HeaderExtension parse(StanzaNode node) {
+ return new HeaderExtension((uint8) node.get_attribute_int("id"), node.get_attribute("uri"));
+ }
+
+ public StanzaNode to_xml() {
+ return new StanzaNode.build(NAME, NS_URI)
+ .add_self_xmlns()
+ .put_attribute("id", id.to_string())
+ .put_attribute("uri", uri);
+ }
} \ No newline at end of file
diff --git a/xmpp-vala/src/module/xep/0167_jingle_rtp/jingle_rtp_module.vala b/xmpp-vala/src/module/xep/0167_jingle_rtp/jingle_rtp_module.vala
index 3adad114..6eb6289b 100644
--- a/xmpp-vala/src/module/xep/0167_jingle_rtp/jingle_rtp_module.vala
+++ b/xmpp-vala/src/module/xep/0167_jingle_rtp/jingle_rtp_module.vala
@@ -24,6 +24,8 @@ public abstract class Module : XmppStreamModule {
public abstract Crypto? pick_remote_crypto(Gee.List<Crypto> cryptos);
public abstract Crypto? pick_local_crypto(Crypto? remote);
public abstract Stream create_stream(Jingle.Content content);
+ public abstract bool is_header_extension_supported(string media, HeaderExtension ext);
+ public abstract Gee.List<HeaderExtension> get_suggested_header_extensions(string media);
public abstract void close_stream(Stream stream);
public async Jingle.Session start_call(XmppStream stream, Jid receiver_full_jid, bool video, string? sid = null) throws Jingle.Error {
@@ -40,6 +42,7 @@ public abstract class Module : XmppStreamModule {
// Create audio content
Parameters audio_content_parameters = new Parameters(this, "audio", yield get_supported_payloads("audio"));
audio_content_parameters.local_crypto = generate_local_crypto();
+ audio_content_parameters.header_extensions.add_all(get_suggested_header_extensions("audio"));
Jingle.Transport? audio_transport = yield jingle_module.select_transport(stream, content_type.required_transport_type, content_type.required_components, receiver_full_jid, Set.empty());
if (audio_transport == null) {
throw new Jingle.Error.NO_SHARED_PROTOCOLS("No suitable audio transports");
@@ -57,6 +60,7 @@ public abstract class Module : XmppStreamModule {
// Create video content
Parameters video_content_parameters = new Parameters(this, "video", yield get_supported_payloads("video"));
video_content_parameters.local_crypto = generate_local_crypto();
+ video_content_parameters.header_extensions.add_all(get_suggested_header_extensions("video"));
Jingle.Transport? video_transport = yield stream.get_module(Jingle.Module.IDENTITY).select_transport(stream, content_type.required_transport_type, content_type.required_components, receiver_full_jid, Set.empty());
if (video_transport == null) {
throw new Jingle.Error.NO_SHARED_PROTOCOLS("No suitable video transports");
@@ -98,6 +102,7 @@ public abstract class Module : XmppStreamModule {
// Content for video does not yet exist -> create it
Parameters video_content_parameters = new Parameters(this, "video", yield get_supported_payloads("video"));
video_content_parameters.local_crypto = generate_local_crypto();
+ video_content_parameters.header_extensions.add_all(get_suggested_header_extensions("video"));
Jingle.Transport? video_transport = yield stream.get_module(Jingle.Module.IDENTITY).select_transport(stream, content_type.required_transport_type, content_type.required_components, receiver_full_jid, Set.empty());
if (video_transport == null) {
throw new Jingle.Error.NO_SHARED_PROTOCOLS("No suitable video transports");
diff --git a/xmpp-vala/src/module/xep/0167_jingle_rtp/payload_type.vala b/xmpp-vala/src/module/xep/0167_jingle_rtp/payload_type.vala
index 452f1d65..faba38c9 100644
--- a/xmpp-vala/src/module/xep/0167_jingle_rtp/payload_type.vala
+++ b/xmpp-vala/src/module/xep/0167_jingle_rtp/payload_type.vala
@@ -3,6 +3,8 @@ using Xmpp;
using Xmpp.Xep;
public class Xmpp.Xep.JingleRtp.PayloadType {
+ public const string NAME = "payload-type";
+
public uint8 id { get; set; }
public string? name { get; set; }
public uint8 channels { get; set; default = 1; }
@@ -10,6 +12,7 @@ public class Xmpp.Xep.JingleRtp.PayloadType {
public uint32 maxptime { get; set; }
public uint32 ptime { get; set; }
public Map<string, string> parameters = new HashMap<string, string>();
+ public Gee.List<RtcpFeedback> rtcp_fbs = new ArrayList<RtcpFeedback>();
public static PayloadType parse(StanzaNode node) {
PayloadType payloadType = new PayloadType();
@@ -22,11 +25,14 @@ public class Xmpp.Xep.JingleRtp.PayloadType {
foreach (StanzaNode parameter in node.get_subnodes("parameter")) {
payloadType.parameters[parameter.get_attribute("name")] = parameter.get_attribute("value");
}
+ foreach (StanzaNode subnode in node.get_subnodes(RtcpFeedback.NAME, RtcpFeedback.NS_URI)) {
+ payloadType.rtcp_fbs.add(RtcpFeedback.parse(subnode));
+ }
return payloadType;
}
public StanzaNode to_xml() {
- StanzaNode node = new StanzaNode.build("payload-type", NS_URI)
+ StanzaNode node = new StanzaNode.build(NAME, NS_URI)
.put_attribute("id", id.to_string());
if (channels != 1) node.put_attribute("channels", channels.to_string());
if (clockrate != 0) node.put_attribute("clockrate", clockrate.to_string());
@@ -38,9 +44,25 @@ public class Xmpp.Xep.JingleRtp.PayloadType {
.put_attribute("name", parameter)
.put_attribute("value", parameters[parameter]));
}
+ foreach (RtcpFeedback rtcp_fb in rtcp_fbs) {
+ node.put_node(rtcp_fb.to_xml());
+ }
return node;
}
+ public PayloadType clone() {
+ PayloadType clone = new PayloadType();
+ clone.id = id;
+ clone.name = name;
+ clone.channels = channels;
+ clone.clockrate = clockrate;
+ clone.maxptime = maxptime;
+ clone.ptime = ptime;
+ clone.parameters.set_all(parameters);
+ clone.rtcp_fbs.add_all(rtcp_fbs);
+ return clone;
+ }
+
public static bool equals_func(PayloadType a, PayloadType b) {
return a.id == b.id &&
a.name == b.name &&
@@ -49,4 +71,29 @@ public class Xmpp.Xep.JingleRtp.PayloadType {
a.maxptime == b.maxptime &&
a.ptime == b.ptime;
}
+}
+
+public class Xmpp.Xep.JingleRtp.RtcpFeedback {
+ public const string NS_URI = "urn:xmpp:jingle:apps:rtp:rtcp-fb:0";
+ public const string NAME = "rtcp-fb";
+
+ public string type_ { get; private set; }
+ public string? subtype { get; private set; }
+
+ public RtcpFeedback(string type, string? subtype = null) {
+ this.type_ = type;
+ this.subtype = subtype;
+ }
+
+ public static RtcpFeedback parse(StanzaNode node) {
+ return new RtcpFeedback(node.get_attribute("type"), node.get_attribute("subtype"));
+ }
+
+ public StanzaNode to_xml() {
+ StanzaNode node = new StanzaNode.build(NAME, NS_URI)
+ .add_self_xmlns()
+ .put_attribute("type", type_);
+ if (subtype != null) node.put_attribute("subtype", subtype);
+ return node;
+ }
} \ No newline at end of file
diff --git a/xmpp-vala/src/module/xep/0167_jingle_rtp/stream.vala b/xmpp-vala/src/module/xep/0167_jingle_rtp/stream.vala
index adae11f5..65be8a0a 100644
--- a/xmpp-vala/src/module/xep/0167_jingle_rtp/stream.vala
+++ b/xmpp-vala/src/module/xep/0167_jingle_rtp/stream.vala
@@ -33,6 +33,13 @@ public abstract class Xmpp.Xep.JingleRtp.Stream : Object {
}
return null;
}}
+ public Gee.List<JingleRtp.HeaderExtension>? header_extensions { get {
+ var content_params = content.content_params;
+ if (content_params is Parameters) {
+ return ((Parameters)content_params).header_extensions;
+ }
+ return null;
+ }}
public bool sending { get {
return content.session.senders_include_us(content.senders);
}}
generated by cgit v1.2.3-70-g09d2 (git 2.47.1) at 2024-12-25 06:35:38 +0000