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Issue 2106183002: Fix bug where min transmit bitrate wasn't working (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Addressed comments Created 4 years, 5 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 #ifndef WEBRTC_CALL_H_ 10 #ifndef WEBRTC_CALL_H_
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85 // AudioState which is possibly shared between multiple calls. 85 // AudioState which is possibly shared between multiple calls.
86 // TODO(solenberg): Change this to a shared_ptr once we can use C++11. 86 // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
87 rtc::scoped_refptr<AudioState> audio_state; 87 rtc::scoped_refptr<AudioState> audio_state;
88 88
89 // Audio Processing Module to be used in this call. 89 // Audio Processing Module to be used in this call.
90 // TODO(solenberg): Change this to a shared_ptr once we can use C++11. 90 // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
91 AudioProcessing* audio_processing = nullptr; 91 AudioProcessing* audio_processing = nullptr;
92 }; 92 };
93 93
94 struct Stats { 94 struct Stats {
95 int send_bandwidth_bps = 0; 95 int send_bandwidth_bps = 0; // Estimated available send bandwidth.
96 int recv_bandwidth_bps = 0; 96 int max_padding_bitrate_bps = 0; // Cumulative configured max padding.
97 int recv_bandwidth_bps = 0; // Estimated available receive bandwidth.
97 int64_t pacer_delay_ms = 0; 98 int64_t pacer_delay_ms = 0;
98 int64_t rtt_ms = -1; 99 int64_t rtt_ms = -1;
99 }; 100 };
100 101
101 static Call* Create(const Call::Config& config); 102 static Call* Create(const Call::Config& config);
102 103
103 virtual AudioSendStream* CreateAudioSendStream( 104 virtual AudioSendStream* CreateAudioSendStream(
104 const AudioSendStream::Config& config) = 0; 105 const AudioSendStream::Config& config) = 0;
105 virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0; 106 virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0;
106 107
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151 virtual bool StartEventLog(rtc::PlatformFile log_file, 152 virtual bool StartEventLog(rtc::PlatformFile log_file,
152 int64_t max_size_bytes) = 0; 153 int64_t max_size_bytes) = 0;
153 virtual void StopEventLog() = 0; 154 virtual void StopEventLog() = 0;
154 155
155 virtual ~Call() {} 156 virtual ~Call() {}
156 }; 157 };
157 158
158 } // namespace webrtc 159 } // namespace webrtc
159 160
160 #endif // WEBRTC_CALL_H_ 161 #endif // WEBRTC_CALL_H_
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