Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(298)

Side by Side Diff: webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.cc

Issue 2964773002: Revert "Update includes for webrtc/{base => rtc_base} rename (1/3)" (Closed)
Patch Set: Created 3 years, 5 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
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2016 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 <algorithm> 11 #include <algorithm>
12 12
13 #include "webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h " 13 #include "webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h "
14 #include "webrtc/rtc_base/checks.h" 14 #include "webrtc/base/checks.h"
15 15
16 namespace webrtc { 16 namespace webrtc {
17 17
18 ChannelController::Config::Config(size_t num_encoder_channels, 18 ChannelController::Config::Config(size_t num_encoder_channels,
19 size_t intial_channels_to_encode, 19 size_t intial_channels_to_encode,
20 int channel_1_to_2_bandwidth_bps, 20 int channel_1_to_2_bandwidth_bps,
21 int channel_2_to_1_bandwidth_bps) 21 int channel_2_to_1_bandwidth_bps)
22 : num_encoder_channels(num_encoder_channels), 22 : num_encoder_channels(num_encoder_channels),
23 intial_channels_to_encode(intial_channels_to_encode), 23 intial_channels_to_encode(intial_channels_to_encode),
24 channel_1_to_2_bandwidth_bps(channel_1_to_2_bandwidth_bps), 24 channel_1_to_2_bandwidth_bps(channel_1_to_2_bandwidth_bps),
(...skipping 27 matching lines...) Expand all
52 } else if (channels_to_encode_ == 1 && 52 } else if (channels_to_encode_ == 1 &&
53 *uplink_bandwidth_bps_ >= config_.channel_1_to_2_bandwidth_bps) { 53 *uplink_bandwidth_bps_ >= config_.channel_1_to_2_bandwidth_bps) {
54 channels_to_encode_ = 54 channels_to_encode_ =
55 std::min(static_cast<size_t>(2), config_.num_encoder_channels); 55 std::min(static_cast<size_t>(2), config_.num_encoder_channels);
56 } 56 }
57 } 57 }
58 config->num_channels = rtc::Optional<size_t>(channels_to_encode_); 58 config->num_channels = rtc::Optional<size_t>(channels_to_encode_);
59 } 59 }
60 60
61 } // namespace webrtc 61 } // namespace webrtc
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698