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d5ebc33562
NetEq packet source input doesn't have any other uses than rtp dump, so remove that layer. Bug: None Change-Id: I667bb4aead9f0f2fe8a1c0d6d911a4670ded67e7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300542 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39810}
32 lines
1.1 KiB
C++
32 lines
1.1 KiB
C++
/*
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* Copyright (c) 2016 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 MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_RTP_DUMP_INPUT_H_
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#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_RTP_DUMP_INPUT_H_
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#include <map>
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#include <memory>
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#include "absl/strings/string_view.h"
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#include "absl/types/optional.h"
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#include "modules/audio_coding/neteq/tools/neteq_input.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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namespace webrtc {
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namespace test {
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std::unique_ptr<NetEqInput> CreateNetEqRtpDumpInput(
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absl::string_view file_name,
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const std::map<int, RTPExtensionType>& hdr_ext_map,
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absl::optional<uint32_t> ssrc_filter);
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_RTP_DUMP_INPUT_H_
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