mirror of
https://github.com/mollyim/webrtc.git
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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}
85 lines
2.5 KiB
C++
85 lines
2.5 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_PACKET_SOURCE_INPUT_H_
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#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PACKET_SOURCE_INPUT_H_
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#include <map>
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#include <string>
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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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class RtpFileSource;
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class RtcEventLogSource;
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// An adapter class to dress up a PacketSource object as a NetEqInput.
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class NetEqPacketSourceInput : public NetEqInput {
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public:
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using RtpHeaderExtensionMap = std::map<int, webrtc::RTPExtensionType>;
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NetEqPacketSourceInput();
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absl::optional<int64_t> NextPacketTime() const override;
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std::unique_ptr<PacketData> PopPacket() override;
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absl::optional<RTPHeader> NextHeader() const override;
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bool ended() const override { return !next_output_event_ms_; }
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void SelectSsrc(uint32_t);
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protected:
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virtual PacketSource* source() = 0;
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void LoadNextPacket();
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absl::optional<int64_t> next_output_event_ms_;
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private:
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std::unique_ptr<Packet> packet_;
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};
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// Implementation of NetEqPacketSourceInput to be used with an RtpFileSource.
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class NetEqRtpDumpInput final : public NetEqPacketSourceInput {
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public:
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NetEqRtpDumpInput(const std::string& file_name,
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const RtpHeaderExtensionMap& hdr_ext_map);
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absl::optional<int64_t> NextOutputEventTime() const override;
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void AdvanceOutputEvent() override;
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protected:
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PacketSource* source() override;
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private:
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static constexpr int64_t kOutputPeriodMs = 10;
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std::unique_ptr<RtpFileSource> source_;
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};
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// Implementation of NetEqPacketSourceInput to be used with an
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// RtcEventLogSource.
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class NetEqEventLogInput final : public NetEqPacketSourceInput {
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public:
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NetEqEventLogInput(const std::string& file_name,
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const RtpHeaderExtensionMap& hdr_ext_map);
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absl::optional<int64_t> NextOutputEventTime() const override;
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void AdvanceOutputEvent() override;
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protected:
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PacketSource* source() override;
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private:
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std::unique_ptr<RtcEventLogSource> source_;
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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_PACKET_SOURCE_INPUT_H_
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