| /* |
| * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_ |
| #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_ |
| |
| #include <memory> |
| #include <string> |
| |
| #include "webrtc/base/constructormagic.h" |
| #include "webrtc/call/rtc_event_log_parser.h" |
| #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| |
| namespace webrtc { |
| |
| class RtpHeaderParser; |
| |
| namespace test { |
| |
| class Packet; |
| |
| class RtcEventLogSource : public PacketSource { |
| public: |
| // Creates an RtcEventLogSource reading from |file_name|. If the file cannot |
| // be opened, or has the wrong format, NULL will be returned. |
| static RtcEventLogSource* Create(const std::string& file_name); |
| |
| virtual ~RtcEventLogSource(); |
| |
| // Registers an RTP header extension and binds it to |id|. |
| virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id); |
| |
| std::unique_ptr<Packet> NextPacket() override; |
| |
| // Returns the timestamp of the next audio output event, in milliseconds. The |
| // maximum value of int64_t is returned if there are no more audio output |
| // events available. |
| int64_t NextAudioOutputEventMs(); |
| |
| private: |
| RtcEventLogSource(); |
| |
| bool OpenFile(const std::string& file_name); |
| |
| size_t rtp_packet_index_ = 0; |
| size_t audio_output_index_ = 0; |
| |
| ParsedRtcEventLog parsed_stream_; |
| std::unique_ptr<RtpHeaderParser> parser_; |
| |
| RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogSource); |
| }; |
| |
| } // namespace test |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_ |