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-rw-r--r--plugins/ice/src/transport_parameters.vala1
-rw-r--r--plugins/rtp/CMakeLists.txt4
-rw-r--r--plugins/rtp/src/stream.vala119
3 files changed, 87 insertions, 37 deletions
diff --git a/plugins/ice/src/transport_parameters.vala b/plugins/ice/src/transport_parameters.vala
index 9aa3dda1..f684e411 100644
--- a/plugins/ice/src/transport_parameters.vala
+++ b/plugins/ice/src/transport_parameters.vala
@@ -273,6 +273,7 @@ public class Dino.Plugins.Ice.TransportParameters : JingleIceUdp.IceUdpTransport
if (decrypt_data == null) return;
} catch (Crypto.Error e) {
warning("%s while on_recv stream %u component %u", e.message, stream_id, component_id);
+ return;
}
}
may_consider_ready(stream_id, component_id);
diff --git a/plugins/rtp/CMakeLists.txt b/plugins/rtp/CMakeLists.txt
index 3264e24a..fa4f367c 100644
--- a/plugins/rtp/CMakeLists.txt
+++ b/plugins/rtp/CMakeLists.txt
@@ -16,6 +16,10 @@ if(GstRtp_VERSION VERSION_GREATER "1.16")
set(RTP_DEFINITIONS GST_1_16)
endif()
+if(Vala_VERSION VERSION_GREATER "0.50")
+ set(RTP_DEFINITIONS VALA_0_50)
+endif()
+
if(WebRTCAudioProcessing_VERSION GREATER "0.4")
message(STATUS "Ignoring WebRTCAudioProcessing, only versions < 0.4 supported so far")
unset(WebRTCAudioProcessing_FOUND)
diff --git a/plugins/rtp/src/stream.vala b/plugins/rtp/src/stream.vala
index 2cc40783..2c8b6356 100644
--- a/plugins/rtp/src/stream.vala
+++ b/plugins/rtp/src/stream.vala
@@ -343,36 +343,57 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
}
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) {
+ uint buffer_ssrc = 0, buffer_seq = 0;
Gst.RTP.Buffer rtp_buffer;
if (Gst.RTP.Buffer.map(buffer, Gst.MapFlags.READ, out rtp_buffer)) {
- if (our_ssrc != rtp_buffer.get_ssrc()) {
- warning("Sending buffer with SSRC %u when our ssrc is %u", rtp_buffer.get_ssrc(), our_ssrc);
- }
+ buffer_ssrc = rtp_buffer.get_ssrc();
+ buffer_seq = rtp_buffer.get_seq();
next_seqnum_offset = rtp_buffer.get_seq() + 1;
+ next_timestamp_offset_base = rtp_buffer.get_timestamp();
+ next_timestamp_offset_stamp = get_monotonic_time();
rtp_buffer.unmap();
}
- if (crypto_session.has_encrypt) {
- data = crypto_session.encrypt_rtp(data);
+ if (our_ssrc != buffer_ssrc) {
+ warning("Sending RTP %s buffer seq %u with SSRC %u when our ssrc is %u", media, buffer_seq, buffer_ssrc, our_ssrc);
+ } else {
+ debug("Sending RTP %s buffer seq %u with SSRC %u", media, buffer_seq, buffer_ssrc);
}
- on_send_rtp_data(new Bytes.take((owned) data));
+ }
+
+ prepare_local_crypto();
+
+ uint8[] data;
+ buffer.extract_dup(0, buffer.get_size(), out data);
+ if (sink == send_rtp) {
+ encrypt_and_send_rtp((owned) data);
} else if (sink == send_rtcp) {
encrypt_and_send_rtcp((owned) data);
}
return Gst.FlowReturn.OK;
}
+ private void encrypt_and_send_rtp(owned uint8[] data) {
+ Bytes bytes;
+ if (crypto_session.has_encrypt) {
+ bytes = new Bytes.take(crypto_session.encrypt_rtp(data));
+ } else {
+ bytes = new Bytes.take(data);
+ }
+ on_send_rtp_data(bytes);
+ }
+
private void encrypt_and_send_rtcp(owned uint8[] data) {
+ Bytes bytes;
if (crypto_session.has_encrypt) {
- data = crypto_session.encrypt_rtcp(data);
+ bytes = new Bytes.take(crypto_session.encrypt_rtcp(data));
+ } else {
+ bytes = new Bytes.take(data);
}
if (rtcp_mux) {
- on_send_rtp_data(new Bytes.take((owned) data));
+ on_send_rtp_data(bytes);
} else {
- on_send_rtcp_data(new Bytes.take((owned) data));
+ on_send_rtcp_data(bytes);
}
}
@@ -514,17 +535,38 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
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 GST_1_16
+ {
+ Gst.Buffer buffer = new Gst.Buffer.wrapped_bytes(bytes);
+ Gst.RTP.Buffer rtp_buffer;
+ uint buffer_ssrc = 0, buffer_seq = 0;
+ if (Gst.RTP.Buffer.map(buffer, Gst.MapFlags.READ, out rtp_buffer)) {
+ buffer_ssrc = rtp_buffer.get_ssrc();
+ buffer_seq = rtp_buffer.get_seq();
+ rtp_buffer.unmap();
}
+ debug("Received RTP %s buffer seq %u with SSRC %u", media, buffer_seq, buffer_ssrc);
}
+#endif
if (push_recv_data) {
- Gst.Buffer buffer = new Gst.Buffer.wrapped((owned) data);
+ prepare_remote_crypto();
+
+ Gst.Buffer buffer;
+ if (crypto_session.has_decrypt) {
+ try {
+ buffer = new Gst.Buffer.wrapped(crypto_session.decrypt_rtp(bytes.get_data()));
+ } catch (Error e) {
+ warning("%s (%d)", e.message, e.code);
+ return;
+ }
+ } else {
+#if GST_1_16
+ buffer = new Gst.Buffer.wrapped_bytes(bytes);
+#else
+ buffer = new Gst.Buffer.wrapped(bytes.get_data());
+#endif
+ }
+
Gst.RTP.Buffer rtp_buffer;
if (Gst.RTP.Buffer.map(buffer, Gst.MapFlags.READ, out rtp_buffer)) {
if (rtp_buffer.get_extension()) {
@@ -552,11 +594,7 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
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
+#if VALA_0_50
recv_rtp.push_buffer((owned) buffer);
#else
Gst.FlowReturn ret;
@@ -566,19 +604,26 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
}
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
+ prepare_remote_crypto();
+
+ Gst.Buffer buffer;
+ if (crypto_session.has_decrypt) {
+ try {
+ buffer = new Gst.Buffer.wrapped(crypto_session.decrypt_rtcp(bytes.get_data()));
+ } catch (Error e) {
+ warning("%s (%d)", e.message, e.code);
+ return;
+ }
+ } else {
+#if GST_1_16
+ buffer = new Gst.Buffer.wrapped_bytes(bytes);
+#else
+ buffer = new Gst.Buffer.wrapped(bytes.get_data());
+#endif
+ }
+
+#if VALA_0_50
recv_rtcp.push_buffer((owned) buffer);
#else
Gst.FlowReturn ret;
generated by cgit v1.2.3-70-g09d2 (git 2.47.0) at 2024-11-09 22:56:23 +0000