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10542f21c8
Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
65 lines
2 KiB
C++
65 lines
2 KiB
C++
/*
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* Copyright (c) 2015 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_RTC_EVENT_LOG_SOURCE_H_
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#define MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
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#include <memory>
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#include <string>
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#include <vector>
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#include "absl/types/optional.h"
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#include "logging/rtc_event_log/rtc_event_log_parser.h"
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#include "modules/audio_coding/neteq/tools/packet_source.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "rtc_base/constructor_magic.h"
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namespace webrtc {
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class RtpHeaderParser;
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namespace test {
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class Packet;
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class RtcEventLogSource : public PacketSource {
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public:
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// Creates an RtcEventLogSource reading from |file_name|. If the file cannot
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// be opened, or has the wrong format, NULL will be returned.
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static RtcEventLogSource* Create(const std::string& file_name,
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absl::optional<uint32_t> ssrc_filter);
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virtual ~RtcEventLogSource();
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std::unique_ptr<Packet> NextPacket() override;
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// Returns the timestamp of the next audio output event, in milliseconds. The
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// maximum value of int64_t is returned if there are no more audio output
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// events available.
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int64_t NextAudioOutputEventMs();
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private:
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RtcEventLogSource();
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bool OpenFile(const std::string& file_name,
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absl::optional<uint32_t> ssrc_filter);
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std::vector<std::unique_ptr<Packet>> rtp_packets_;
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size_t rtp_packet_index_ = 0;
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std::vector<int64_t> audio_outputs_;
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size_t audio_output_index_ = 0;
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RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogSource);
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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_RTC_EVENT_LOG_SOURCE_H_
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