Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(239)

Side by Side Diff: webrtc/video/video_send_stream.cc

Issue 1411573007: Removed vie_defines.h (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 1 month 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
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 #include "webrtc/video/video_send_stream.h" 11 #include "webrtc/video/video_send_stream.h"
12 12
13 #include <algorithm> 13 #include <algorithm>
14 #include <sstream> 14 #include <sstream>
15 #include <string> 15 #include <string>
16 #include <vector> 16 #include <vector>
17 17
18 #include "webrtc/base/checks.h" 18 #include "webrtc/base/checks.h"
19 #include "webrtc/base/logging.h" 19 #include "webrtc/base/logging.h"
20 #include "webrtc/base/trace_event.h" 20 #include "webrtc/base/trace_event.h"
21 #include "webrtc/call/congestion_controller.h" 21 #include "webrtc/call/congestion_controller.h"
22 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" 22 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
23 #include "webrtc/modules/pacing/include/packet_router.h" 23 #include "webrtc/modules/pacing/include/packet_router.h"
24 #include "webrtc/video/video_capture_input.h" 24 #include "webrtc/video/video_capture_input.h"
25 #include "webrtc/video_engine/call_stats.h" 25 #include "webrtc/video_engine/call_stats.h"
26 #include "webrtc/video_engine/encoder_state_feedback.h" 26 #include "webrtc/video_engine/encoder_state_feedback.h"
27 #include "webrtc/video_engine/payload_router.h" 27 #include "webrtc/video_engine/payload_router.h"
28 #include "webrtc/video_engine/vie_channel.h" 28 #include "webrtc/video_engine/vie_channel.h"
29 #include "webrtc/video_engine/vie_defines.h"
30 #include "webrtc/video_engine/vie_encoder.h" 29 #include "webrtc/video_engine/vie_encoder.h"
31 #include "webrtc/video_send_stream.h" 30 #include "webrtc/video_send_stream.h"
32 31
33 namespace webrtc { 32 namespace webrtc {
34 33
35 class BitrateAllocator; 34 class BitrateAllocator;
36 class PacedSender; 35 class PacedSender;
37 class RtcpIntraFrameObserver; 36 class RtcpIntraFrameObserver;
38 class TransportFeedbackObserver; 37 class TransportFeedbackObserver;
39 38
(...skipping 471 matching lines...) Expand 10 before | Expand all | Expand 10 after
511 int64_t rtt_ms; 510 int64_t rtt_ms;
512 if (vie_channel_->GetSendRtcpStatistics(&frac_lost, &cumulative_lost, 511 if (vie_channel_->GetSendRtcpStatistics(&frac_lost, &cumulative_lost,
513 &extended_max_sequence_number, 512 &extended_max_sequence_number,
514 &jitter, &rtt_ms) == 0) { 513 &jitter, &rtt_ms) == 0) {
515 return rtt_ms; 514 return rtt_ms;
516 } 515 }
517 return -1; 516 return -1;
518 } 517 }
519 518
520 bool VideoSendStream::SetSendCodec(VideoCodec video_codec) { 519 bool VideoSendStream::SetSendCodec(VideoCodec video_codec) {
520 const int kEncoderMinBitrate = 30;
pbos-webrtc 2015/10/29 15:39:40 static
mflodman 2015/11/06 11:38:36 Done.
521 if (video_codec.maxBitrate == 0) { 521 if (video_codec.maxBitrate == 0) {
522 // Unset max bitrate -> cap to one bit per pixel. 522 // Unset max bitrate -> cap to one bit per pixel.
523 video_codec.maxBitrate = 523 video_codec.maxBitrate =
524 (video_codec.width * video_codec.height * video_codec.maxFramerate) / 524 (video_codec.width * video_codec.height * video_codec.maxFramerate) /
525 1000; 525 1000;
526 } 526 }
527 527
528 if (video_codec.minBitrate < kViEMinCodecBitrate) 528 if (video_codec.minBitrate < kEncoderMinBitrate)
529 video_codec.minBitrate = kViEMinCodecBitrate; 529 video_codec.minBitrate = kEncoderMinBitrate;
530 if (video_codec.maxBitrate < kViEMinCodecBitrate) 530 if (video_codec.maxBitrate < kEncoderMinBitrate)
531 video_codec.maxBitrate = kViEMinCodecBitrate; 531 video_codec.maxBitrate = kEncoderMinBitrate;
532 532
533 // Stop the media flow while reconfiguring. 533 // Stop the media flow while reconfiguring.
534 vie_encoder_->Pause(); 534 vie_encoder_->Pause();
535 535
536 if (vie_encoder_->SetEncoder(video_codec) != 0) { 536 if (vie_encoder_->SetEncoder(video_codec) != 0) {
537 LOG(LS_ERROR) << "Failed to set encoder."; 537 LOG(LS_ERROR) << "Failed to set encoder.";
538 return false; 538 return false;
539 } 539 }
540 540
541 if (vie_channel_->SetSendCodec(video_codec, false) != 0) { 541 if (vie_channel_->SetSendCodec(video_codec, false) != 0) {
(...skipping 14 matching lines...) Expand all
556 vie_channel_->IsSendingFecEnabled()); 556 vie_channel_->IsSendingFecEnabled());
557 557
558 // Restart the media flow 558 // Restart the media flow
559 vie_encoder_->Restart(); 559 vie_encoder_->Restart();
560 560
561 return true; 561 return true;
562 } 562 }
563 563
564 } // namespace internal 564 } // namespace internal
565 } // namespace webrtc 565 } // namespace webrtc
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698