04f4931ef0
BUG=4690 R=jmarusic@webrtc.org, kwiberg@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50029004 Cr-Commit-Position: refs/heads/master@{#9392}
55 lines
1.4 KiB
C++
55 lines
1.4 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_SEND_STREAM_H_
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#define WEBRTC_AUDIO_SEND_STREAM_H_
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#include <string>
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#include <vector>
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/config.h"
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#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class AudioSendStream {
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public:
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struct Stats {};
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struct Config {
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std::string ToString() const;
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// Receive-stream specific RTP settings.
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struct Rtp {
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std::string ToString() const;
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// Sender SSRC.
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uint32_t ssrc = 0;
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// RTP header extensions used for the received stream.
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std::vector<RtpExtension> extensions;
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} rtp;
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rtc::scoped_ptr<AudioEncoder> encoder;
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int cng_payload_type = -1; // pt, or -1 to disable Comfort Noise Generator.
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int red_payload_type = -1; // pt, or -1 to disable REDundant coding.
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};
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virtual Stats GetStats() const = 0;
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protected:
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virtual ~AudioSendStream() {}
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};
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} // namespace webrtc
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#endif // WEBRTC_AUDIO_SEND_STREAM_H_
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