blob: 4ad15d6836592b898eb271698c62b9bdf6fcc1e2 [file] [log] [blame]
* 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.
#include <string>
#include <vector>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/config.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class AudioSendStream {
struct Stats {};
struct Config {
std::string ToString() const;
// Receive-stream specific RTP settings.
struct Rtp {
std::string ToString() const;
// Sender SSRC.
uint32_t ssrc = 0;
// RTP header extensions used for the received stream.
std::vector<RtpExtension> extensions;
} rtp;
rtc::scoped_ptr<AudioEncoder> encoder;
int cng_payload_type = -1; // pt, or -1 to disable Comfort Noise Generator.
int red_payload_type = -1; // pt, or -1 to disable REDundant coding.
virtual Stats GetStats() const = 0;
virtual ~AudioSendStream() {}
} // namespace webrtc