Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(304)

Side by Side Diff: webrtc/logging/rtc_event_log/rtc_event_log.h

Issue 3013523002: Move StreamConfig into its own file (Closed)
Patch Set: Rebased Created 3 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/logging/BUILD.gn ('k') | webrtc/logging/rtc_event_log/rtc_stream_config.h » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 #ifndef WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_ 11 #ifndef WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_
12 #define WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_ 12 #define WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_
13 13
14 #include <memory> 14 #include <memory>
15 #include <string> 15 #include <string>
16 #include <vector> 16 #include <vector>
17 17
18 // TODO(eladalon): Get rid of this later in the CL-stack.
18 #include "webrtc/api/rtpparameters.h" 19 #include "webrtc/api/rtpparameters.h"
19 #include "webrtc/common_types.h" 20 #include "webrtc/common_types.h"
21 // TODO(eladalon): Get rid of this later in the CL-stack.
22 #include "webrtc/logging/rtc_event_log/rtc_stream_config.h"
20 #include "webrtc/rtc_base/platform_file.h" 23 #include "webrtc/rtc_base/platform_file.h"
21 24
22 namespace webrtc { 25 namespace webrtc {
23 26
24 // Forward declaration of storage class that is automatically generated from
25 // the protobuf file.
26 namespace rtclog { 27 namespace rtclog {
27 class EventStream; 28 class EventStream; // Storage class automatically generated from protobuf.
28
29 struct StreamConfig {
30 uint32_t local_ssrc = 0;
31 uint32_t remote_ssrc = 0;
32 uint32_t rtx_ssrc = 0;
33 std::string rsid;
34
35 bool remb = false;
36 std::vector<RtpExtension> rtp_extensions;
37
38 RtcpMode rtcp_mode = RtcpMode::kReducedSize;
39
40 struct Codec {
41 Codec(const std::string& payload_name,
42 int payload_type,
43 int rtx_payload_type)
44 : payload_name(payload_name),
45 payload_type(payload_type),
46 rtx_payload_type(rtx_payload_type) {}
47
48 std::string payload_name;
49 int payload_type;
50 int rtx_payload_type;
51 };
52 std::vector<Codec> codecs;
53 };
54
55 } // namespace rtclog 29 } // namespace rtclog
56 30
57 class Clock; 31 class Clock;
58 struct AudioEncoderRuntimeConfig; 32 struct AudioEncoderRuntimeConfig;
59 33
60 enum class MediaType; 34 enum class MediaType;
61 enum class BandwidthUsage; 35 enum class BandwidthUsage;
62 36
63 enum PacketDirection { kIncomingPacket = 0, kOutgoingPacket }; 37 enum PacketDirection { kIncomingPacket = 0, kOutgoingPacket };
64 enum ProbeFailureReason { 38 enum ProbeFailureReason {
(...skipping 158 matching lines...) Expand 10 before | Expand all | Expand 10 after
223 int min_probes, 197 int min_probes,
224 int min_bytes) override{}; 198 int min_bytes) override{};
225 void LogProbeResultSuccess(int id, int bitrate_bps) override{}; 199 void LogProbeResultSuccess(int id, int bitrate_bps) override{};
226 void LogProbeResultFailure(int id, 200 void LogProbeResultFailure(int id,
227 ProbeFailureReason failure_reason) override{}; 201 ProbeFailureReason failure_reason) override{};
228 }; 202 };
229 203
230 } // namespace webrtc 204 } // namespace webrtc
231 205
232 #endif // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_ 206 #endif // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_
OLDNEW
« no previous file with comments | « webrtc/logging/BUILD.gn ('k') | webrtc/logging/rtc_event_log/rtc_stream_config.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698