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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
72 lines
2.9 KiB
C++
72 lines
2.9 KiB
C++
/*
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* Copyright 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef ORTC_TESTRTPPARAMETERS_H_
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#define ORTC_TESTRTPPARAMETERS_H_
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#include "api/ortc/rtptransportinterface.h"
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#include "api/rtpparameters.h"
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namespace webrtc {
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// Helper methods to create RtpParameters to use for sending/receiving.
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//
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// "MakeMinimal" methods contain the minimal necessary information for an
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// RtpSender or RtpReceiver to function. The "MakeFull" methods are the
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// opposite, and include all features that would normally be offered by a
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// PeerConnection, and in some cases additional ones.
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//
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// These methods are intended to be used for end-to-end testing (such as in
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// ortcfactory_integrationtest.cc), or unit testing that doesn't care about the
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// specific contents of the parameters. Tests should NOT assume that these
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// methods will not change; tests that are testing that a specific value in the
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// parameters is applied properly should construct the parameters in the test
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// itself.
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inline RtpTransportParameters MakeRtcpMuxParameters() {
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RtpTransportParameters parameters;
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parameters.rtcp.mux = true;
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return parameters;
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}
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RtpParameters MakeMinimalOpusParameters();
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RtpParameters MakeMinimalIsacParameters();
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RtpParameters MakeMinimalOpusParametersWithSsrc(uint32_t ssrc);
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RtpParameters MakeMinimalIsacParametersWithSsrc(uint32_t ssrc);
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RtpParameters MakeMinimalVp8Parameters();
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RtpParameters MakeMinimalVp9Parameters();
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RtpParameters MakeMinimalVp8ParametersWithSsrc(uint32_t ssrc);
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RtpParameters MakeMinimalVp9ParametersWithSsrc(uint32_t ssrc);
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// Will create an encoding with no SSRC (meaning "match first SSRC seen" for a
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// receiver, or "pick one automatically" for a sender).
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RtpParameters MakeMinimalOpusParametersWithNoSsrc();
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RtpParameters MakeMinimalIsacParametersWithNoSsrc();
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RtpParameters MakeMinimalVp8ParametersWithNoSsrc();
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RtpParameters MakeMinimalVp9ParametersWithNoSsrc();
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// Make audio parameters with all the available properties configured and
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// features used, and with multiple codecs offered. Obtained by taking a
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// snapshot of a default PeerConnection offer (and adding other things, like
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// bitrate limit).
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RtpParameters MakeFullOpusParameters();
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RtpParameters MakeFullIsacParameters();
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// Make video parameters with all the available properties configured and
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// features used, and with multiple codecs offered. Obtained by taking a
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// snapshot of a default PeerConnection offer (and adding other things, like
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// bitrate limit).
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RtpParameters MakeFullVp8Parameters();
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RtpParameters MakeFullVp9Parameters();
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} // namespace webrtc
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#endif // ORTC_TESTRTPPARAMETERS_H_
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