tor-browser

The Tor Browser
git clone https://git.dasho.dev/tor-browser.git
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voip_engine_factory.cc (1733B)


      1 /*
      2 *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
      3 *
      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
      6 *  tree. An additional intellectual property rights grant can be found
      7 *  in the file PATENTS.  All contributing project authors may
      8 *  be found in the AUTHORS file in the root of the source tree.
      9 */
     10 
     11 #include "api/voip/voip_engine_factory.h"
     12 
     13 #include <memory>
     14 #include <utility>
     15 
     16 #include "api/audio/audio_processing.h"
     17 #include "api/environment/environment.h"
     18 #include "api/environment/environment_factory.h"
     19 #include "api/scoped_refptr.h"
     20 #include "api/voip/voip_engine.h"
     21 #include "audio/voip/voip_core.h"
     22 #include "rtc_base/checks.h"
     23 #include "rtc_base/logging.h"
     24 
     25 namespace webrtc {
     26 
     27 std::unique_ptr<VoipEngine> CreateVoipEngine(VoipEngineConfig config) {
     28  RTC_CHECK(config.encoder_factory);
     29  RTC_CHECK(config.decoder_factory);
     30  RTC_CHECK(config.audio_device_module);
     31 
     32  RTC_CHECK(config.task_queue_factory == nullptr || !config.env.has_value());
     33  Environment env =
     34      config.env.has_value()
     35          ? *config.env
     36          : CreateEnvironment(std::move(config.task_queue_factory));
     37 
     38  scoped_refptr<AudioProcessing> audio_processing;
     39  if (config.audio_processing_builder != nullptr) {
     40    audio_processing = std::move(config.audio_processing_builder)->Build(env);
     41  }
     42 
     43  if (audio_processing == nullptr) {
     44    RTC_DLOG(LS_INFO) << "No audio processing functionality provided.";
     45  }
     46 
     47  return std::make_unique<VoipCore>(
     48      env, std::move(config.encoder_factory), std::move(config.decoder_factory),
     49      std::move(config.audio_device_module), std::move(audio_processing));
     50 }
     51 
     52 }  // namespace webrtc