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In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
51 lines
1.9 KiB
C++
51 lines
1.9 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_REPLACEMENT_INPUT_H_
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#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_REPLACEMENT_INPUT_H_
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#include <memory>
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#include <set>
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#include "webrtc/modules/audio_coding/neteq/tools/neteq_input.h"
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namespace webrtc {
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namespace test {
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// This class converts the packets from a NetEqInput to fake encodings to be
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// decoded by a FakeDecodeFromFile decoder.
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class NetEqReplacementInput : public NetEqInput {
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public:
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NetEqReplacementInput(std::unique_ptr<NetEqInput> source,
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uint8_t replacement_payload_type,
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const std::set<uint8_t>& comfort_noise_types,
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const std::set<uint8_t>& forbidden_types);
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rtc::Optional<int64_t> NextPacketTime() const override;
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rtc::Optional<int64_t> NextOutputEventTime() const override;
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std::unique_ptr<PacketData> PopPacket() override;
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void AdvanceOutputEvent() override;
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bool ended() const override;
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rtc::Optional<RTPHeader> NextHeader() const override;
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private:
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void ReplacePacket();
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std::unique_ptr<NetEqInput> source_;
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const uint8_t replacement_payload_type_;
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const std::set<uint8_t> comfort_noise_types_;
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const std::set<uint8_t> forbidden_types_;
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std::unique_ptr<PacketData> packet_; // The next packet to deliver.
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uint32_t last_frame_size_timestamps_ = 960; // Initial guess: 20 ms @ 48 kHz.
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};
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} // namespace test
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_REPLACEMENT_INPUT_H_
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