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

Issue 1362303002: Reland "Wire up send-side bandwidth estimation." (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 2 months ago
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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
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113 config_(config), 113 config_(config),
114 suspended_ssrcs_(suspended_ssrcs), 114 suspended_ssrcs_(suspended_ssrcs),
115 module_process_thread_(module_process_thread), 115 module_process_thread_(module_process_thread),
116 channel_group_(channel_group), 116 channel_group_(channel_group),
117 channel_id_(channel_id), 117 channel_id_(channel_id),
118 use_config_bitrate_(true), 118 use_config_bitrate_(true),
119 stats_proxy_(Clock::GetRealTimeClock(), config) { 119 stats_proxy_(Clock::GetRealTimeClock(), config) {
120 RTC_DCHECK(!config_.rtp.ssrcs.empty()); 120 RTC_DCHECK(!config_.rtp.ssrcs.empty());
121 RTC_CHECK(channel_group->CreateSendChannel( 121 RTC_CHECK(channel_group->CreateSendChannel(
122 channel_id_, &transport_adapter_, &stats_proxy_, 122 channel_id_, &transport_adapter_, &stats_proxy_,
123 config.pre_encode_callback, num_cpu_cores, config_.rtp.ssrcs)); 123 config.pre_encode_callback, num_cpu_cores, config_));
124 vie_channel_ = channel_group_->GetChannel(channel_id_); 124 vie_channel_ = channel_group_->GetChannel(channel_id_);
125 vie_encoder_ = channel_group_->GetEncoder(channel_id_); 125 vie_encoder_ = channel_group_->GetEncoder(channel_id_);
126 126
127 for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) { 127 for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
128 const std::string& extension = config_.rtp.extensions[i].name; 128 const std::string& extension = config_.rtp.extensions[i].name;
129 int id = config_.rtp.extensions[i].id; 129 int id = config_.rtp.extensions[i].id;
130 // One-byte-extension local identifiers are in the range 1-14 inclusive. 130 // One-byte-extension local identifiers are in the range 1-14 inclusive.
131 RTC_DCHECK_GE(id, 1); 131 RTC_DCHECK_GE(id, 1);
132 RTC_DCHECK_LE(id, 14); 132 RTC_DCHECK_LE(id, 14);
133 if (extension == RtpExtension::kTOffset) { 133 if (extension == RtpExtension::kTOffset) {
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502 vie_channel_->IsSendingFecEnabled()); 502 vie_channel_->IsSendingFecEnabled());
503 503
504 // Restart the media flow 504 // Restart the media flow
505 vie_encoder_->Restart(); 505 vie_encoder_->Restart();
506 506
507 return true; 507 return true;
508 } 508 }
509 509
510 } // namespace internal 510 } // namespace internal
511 } // namespace webrtc 511 } // namespace webrtc
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