c7a8b08a7c
AudioSendStream will be replacing the send side of VoiceEngine channels and associated APIs. Hence, they will be used transform recorded audio into RTP/RTCP packets that can be transmitted to another party, according to given parameters. BUG=webrtc:4690 Review URL: https://codereview.webrtc.org/1397123003 Cr-Commit-Position: refs/heads/master@{#10307}
44 lines
1.3 KiB
C++
44 lines
1.3 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
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#define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
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#include "webrtc/audio_send_stream.h"
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namespace webrtc {
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namespace internal {
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class AudioSendStream : public webrtc::AudioSendStream {
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public:
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explicit AudioSendStream(const webrtc::AudioSendStream::Config& config);
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~AudioSendStream() override;
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// webrtc::SendStream implementation.
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void Start() override;
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void Stop() override;
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void SignalNetworkState(NetworkState state) override;
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bool DeliverRtcp(const uint8_t* packet, size_t length) override;
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// webrtc::AudioSendStream implementation.
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webrtc::AudioSendStream::Stats GetStats() const override;
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const webrtc::AudioSendStream::Config& config() const {
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return config_;
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}
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private:
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const webrtc::AudioSendStream::Config config_;
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};
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} // namespace internal
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} // namespace webrtc
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#endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
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