b602123a5a
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'modules/audio_coding'
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: Ic980ee605148fdb160666d4aa03cc87175e48fe8
Reviewed-on: https://webrtc-review.googlesource.com/84130
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23659}
52 lines
1.9 KiB
C++
52 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 MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_REPLACEMENT_INPUT_H_
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#define 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 "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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absl::optional<int64_t> NextPacketTime() const override;
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absl::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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absl::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 // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_REPLACEMENT_INPUT_H_
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