mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 14:20:45 +01:00
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++
/*
|
|
* 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 ORTC_TESTRTPPARAMETERS_H_
|
|
#define ORTC_TESTRTPPARAMETERS_H_
|
|
|
|
#include "api/ortc/rtptransportinterface.h"
|
|
#include "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 RtpTransportParameters MakeRtcpMuxParameters() {
|
|
RtpTransportParameters parameters;
|
|
parameters.rtcp.mux = true;
|
|
return 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 // ORTC_TESTRTPPARAMETERS_H_
|