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2b742f7eaa
This reverts commit d93b7b91e0.
Reason for revert: Breaks downstream tests
Original change's description:
> Unifying the handling of the events in NetEqInput.
>
> Bug: webrtc:14763
> Change-Id: I9615a9ce41c9b577c4ebd4cdcc9885bfbc5dac48
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293040
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39706}
Bug: webrtc:14763
Change-Id: If076c8fc59a38f011dfa20829f2dd91dd2f914b2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299420
Owners-Override: Björn Terelius <terelius@webrtc.org>
Auto-Submit: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#39711}
74 lines
2.3 KiB
C++
74 lines
2.3 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 <memory>
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#include <string>
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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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class RtpFileSource;
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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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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(absl::string_view file_name,
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const RtpHeaderExtensionMap& hdr_ext_map,
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absl::optional<uint32_t> ssrc_filter);
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absl::optional<int64_t> NextOutputEventTime() const override;
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absl::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override {
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return absl::nullopt;
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}
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void AdvanceOutputEvent() override;
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void AdvanceSetMinimumDelay() 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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} // 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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