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}
61 lines
1.6 KiB
C++
61 lines
1.6 KiB
C++
/*
|
|
* Copyright (c) 2015 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.
|
|
*/
|
|
|
|
#ifndef WEBRTC_AUDIO_RECEIVE_STREAM_H_
|
|
#define WEBRTC_AUDIO_RECEIVE_STREAM_H_
|
|
|
|
#include <map>
|
|
#include <string>
|
|
#include <vector>
|
|
|
|
#include "webrtc/config.h"
|
|
#include "webrtc/typedefs.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class AudioDecoder;
|
|
|
|
class AudioReceiveStream {
|
|
public:
|
|
struct Stats {};
|
|
|
|
struct Config {
|
|
std::string ToString() const;
|
|
|
|
// Receive-stream specific RTP settings.
|
|
struct Rtp {
|
|
std::string ToString() const;
|
|
|
|
// Synchronization source (stream identifier) to be received.
|
|
uint32_t remote_ssrc = 0;
|
|
|
|
// Sender SSRC used for sending RTCP (such as receiver reports).
|
|
uint32_t local_ssrc = 0;
|
|
|
|
// RTP header extensions used for the received stream.
|
|
std::vector<RtpExtension> extensions;
|
|
} rtp;
|
|
|
|
// Decoders for every payload that we can receive. Call owns the
|
|
// AudioDecoder instances once the Config is submitted to
|
|
// Call::CreateReceiveStream().
|
|
// TODO(solenberg): Use unique_ptr<> once our std lib fully supports C++11.
|
|
std::map<uint8_t, AudioDecoder*> decoder_map;
|
|
};
|
|
|
|
virtual Stats GetStats() const = 0;
|
|
|
|
protected:
|
|
virtual ~AudioReceiveStream() {}
|
|
};
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_AUDIO_RECEIVE_STREAM_H_
|