mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-12 21:30:45 +01:00
97597c0f51
Bug: webrtc:13362 Change-Id: Ifda893861a036a85c045cd366f9eab33c62ebde0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237221 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35310}
37 lines
1.3 KiB
C++
37 lines
1.3 KiB
C++
/*
|
|
* Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#include "api/voip/voip_engine_factory.h"
|
|
|
|
#include <utility>
|
|
|
|
#include "audio/voip/voip_core.h"
|
|
#include "rtc_base/logging.h"
|
|
|
|
namespace webrtc {
|
|
|
|
std::unique_ptr<VoipEngine> CreateVoipEngine(VoipEngineConfig config) {
|
|
RTC_CHECK(config.encoder_factory);
|
|
RTC_CHECK(config.decoder_factory);
|
|
RTC_CHECK(config.task_queue_factory);
|
|
RTC_CHECK(config.audio_device_module);
|
|
|
|
if (!config.audio_processing) {
|
|
RTC_DLOG(LS_INFO) << "No audio processing functionality provided.";
|
|
}
|
|
|
|
return std::make_unique<VoipCore>(std::move(config.encoder_factory),
|
|
std::move(config.decoder_factory),
|
|
std::move(config.task_queue_factory),
|
|
std::move(config.audio_device_module),
|
|
std::move(config.audio_processing));
|
|
}
|
|
|
|
} // namespace webrtc
|