Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
This CL adds the following interfaces:
* RtpTransportController
* RtpTransport
* RtpSender
* RtpReceiver
They're implemented on top of the "BaseChannel" object, which is normally used
in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result
of this, there are several limitations:
* You can only have one of each type of sender and receiver (audio/video) on top
of the same transport controller.
* The sender/receiver with the same media type must use the same RTP transport.
* You can't change the transport after creating the sender or receiver.
* Some of the parameters aren't supported.
Later, these "adapter" objects will be gradually replaced by real objects that don't
have these limitations, as "BaseChannel", "MediaChannel" and related code is
restructured. In this CL, we essentially have:
ORTC adapter objects -> BaseChannel -> Media engine
PeerConnection -> BaseChannel -> Media engine
And later we hope to have simply:
PeerConnection -> "Real" ORTC objects -> Media engine
See the linked bug for more context.
BUG=webrtc:7013
TBR=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/2675173003
Cr-Commit-Position: refs/heads/master@{#16842}
diff --git a/webrtc/ortc/testrtpparameters.h b/webrtc/ortc/testrtpparameters.h
new file mode 100644
index 0000000..87108ca
--- /dev/null
+++ b/webrtc/ortc/testrtpparameters.h
@@ -0,0 +1,72 @@
+/*
+ * Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_ORTC_TESTRTPPARAMETERS_H_
+#define WEBRTC_ORTC_TESTRTPPARAMETERS_H_
+
+#include "webrtc/api/ortc/rtptransportinterface.h"
+#include "webrtc/api/rtpparameters.h"
+
+namespace webrtc {
+
+// Helper methods to create RtpParameters to use for sending/receiving.
+//
+// "MakeMinimal" methods contain the minimal necessary information for an
+// RtpSender or RtpReceiver to function. The "MakeFull" methods are the
+// opposite, and include all features that would normally be offered by a
+// PeerConnection, and in some cases additional ones.
+//
+// These methods are intended to be used for end-to-end testing (such as in
+// ortcfactory_integrationtest.cc), or unit testing that doesn't care about the
+// specific contents of the parameters. Tests should NOT assume that these
+// methods will not change; tests that are testing that a specific value in the
+// parameters is applied properly should construct the parameters in the test
+// itself.
+
+inline RtcpParameters MakeRtcpMuxParameters() {
+ RtcpParameters rtcp_parameters;
+ rtcp_parameters.mux = true;
+ return rtcp_parameters;
+}
+
+RtpParameters MakeMinimalOpusParameters();
+RtpParameters MakeMinimalIsacParameters();
+RtpParameters MakeMinimalOpusParametersWithSsrc(uint32_t ssrc);
+RtpParameters MakeMinimalIsacParametersWithSsrc(uint32_t ssrc);
+
+RtpParameters MakeMinimalVp8Parameters();
+RtpParameters MakeMinimalVp9Parameters();
+RtpParameters MakeMinimalVp8ParametersWithSsrc(uint32_t ssrc);
+RtpParameters MakeMinimalVp9ParametersWithSsrc(uint32_t ssrc);
+
+// Will create an encoding with no SSRC (meaning "match first SSRC seen" for a
+// receiver, or "pick one automatically" for a sender).
+RtpParameters MakeMinimalOpusParametersWithNoSsrc();
+RtpParameters MakeMinimalIsacParametersWithNoSsrc();
+RtpParameters MakeMinimalVp8ParametersWithNoSsrc();
+RtpParameters MakeMinimalVp9ParametersWithNoSsrc();
+
+// Make audio parameters with all the available properties configured and
+// features used, and with multiple codecs offered. Obtained by taking a
+// snapshot of a default PeerConnection offer (and adding other things, like
+// bitrate limit).
+RtpParameters MakeFullOpusParameters();
+RtpParameters MakeFullIsacParameters();
+
+// Make video parameters with all the available properties configured and
+// features used, and with multiple codecs offered. Obtained by taking a
+// snapshot of a default PeerConnection offer (and adding other things, like
+// bitrate limit).
+RtpParameters MakeFullVp8Parameters();
+RtpParameters MakeFullVp9Parameters();
+
+} // namespace webrtc
+
+#endif // WEBRTC_ORTC_TESTRTPPARAMETERS_H_
