Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(8)

Side by Side Diff: webrtc/video_send_stream.h

Issue 2303273002: Expose Ivf logging through the native API (Closed)
Patch Set: Nit Created 4 years, 2 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/video_receive_stream.h ('k') | no next file » | 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) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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_VIDEO_SEND_STREAM_H_ 11 #ifndef WEBRTC_VIDEO_SEND_STREAM_H_
12 #define WEBRTC_VIDEO_SEND_STREAM_H_ 12 #define WEBRTC_VIDEO_SEND_STREAM_H_
13 13
14 #include <map> 14 #include <map>
15 #include <string> 15 #include <string>
16 #include <utility>
16 #include <vector> 17 #include <vector>
17 18
19 #include "webrtc/base/platform_file.h"
18 #include "webrtc/common_types.h" 20 #include "webrtc/common_types.h"
19 #include "webrtc/common_video/include/frame_callback.h" 21 #include "webrtc/common_video/include/frame_callback.h"
20 #include "webrtc/config.h" 22 #include "webrtc/config.h"
21 #include "webrtc/media/base/videosinkinterface.h" 23 #include "webrtc/media/base/videosinkinterface.h"
22 #include "webrtc/media/base/videosourceinterface.h" 24 #include "webrtc/media/base/videosourceinterface.h"
23 #include "webrtc/transport.h" 25 #include "webrtc/transport.h"
24 26
25 namespace webrtc { 27 namespace webrtc {
26 28
27 class LoadObserver; 29 class LoadObserver;
(...skipping 157 matching lines...) Expand 10 before | Expand all | Expand 10 after
185 virtual void SetSource( 187 virtual void SetSource(
186 rtc::VideoSourceInterface<webrtc::VideoFrame>* source) = 0; 188 rtc::VideoSourceInterface<webrtc::VideoFrame>* source) = 0;
187 189
188 // Set which streams to send. Must have at least as many SSRCs as configured 190 // Set which streams to send. Must have at least as many SSRCs as configured
189 // in the config. Encoder settings are passed on to the encoder instance along 191 // in the config. Encoder settings are passed on to the encoder instance along
190 // with the VideoStream settings. 192 // with the VideoStream settings.
191 virtual void ReconfigureVideoEncoder(VideoEncoderConfig config) = 0; 193 virtual void ReconfigureVideoEncoder(VideoEncoderConfig config) = 0;
192 194
193 virtual Stats GetStats() = 0; 195 virtual Stats GetStats() = 0;
194 196
197 // Takes ownership of each file, is responsible for closing them later.
198 // Calling this method will close and finalize any current logs.
199 // Some codecs produce multiple streams (VP8 only at present), each of these
200 // streams will log to a separate file. kMaxSimulcastStreams in common_types.h
201 // gives the max number of such streams. If there is no file for a stream, or
202 // the file is rtc::kInvalidPlatformFileValue, frames from that stream will
203 // not be logged.
204 // If a frame to be written would make the log too large the write fails and
205 // the log is closed and finalized. A |byte_limit| of 0 means no limit.
206 virtual void EnableEncodedFrameRecording(
207 const std::vector<rtc::PlatformFile>& files,
208 size_t byte_limit) = 0;
209 inline void DisableEncodedFrameRecording() {
210 EnableEncodedFrameRecording(std::vector<rtc::PlatformFile>(), 0);
211 }
212
195 protected: 213 protected:
196 virtual ~VideoSendStream() {} 214 virtual ~VideoSendStream() {}
197 }; 215 };
198 216
199 } // namespace webrtc 217 } // namespace webrtc
200 218
201 #endif // WEBRTC_VIDEO_SEND_STREAM_H_ 219 #endif // WEBRTC_VIDEO_SEND_STREAM_H_
OLDNEW
« no previous file with comments | « webrtc/video_receive_stream.h ('k') | no next file » | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698