blob: 70d6480b10c1cc3e9d140b93a90b614d1008413e [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 <map>
#include <string>
#include <vector>
#include "webrtc/config.h"
#include "webrtc/stream.h"
#include "webrtc/transport.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class AudioDecoder;
class AudioReceiveStream : public ReceiveStream {
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;
Transport* receive_transport = nullptr;
Transport* rtcp_send_transport = nullptr;
// Underlying VoiceEngine handle, used to map AudioReceiveStream to lower-
// level components.
// TODO(solenberg): Remove when VoiceEngine channels are created outside
// of Call.
int voe_channel_id = -1;
// Identifier for an A/V synchronization group. Empty string to disable.
// TODO(pbos): Synchronize streams in a sync group, not just one video
// stream to one audio stream. Tracked by issue webrtc:4762.
std::string sync_group;
// 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;
// TODO(pbos): Remove config option once combined A/V BWE is always on.
bool combined_audio_video_bwe = false;
virtual Stats GetStats() const = 0;
} // namespace webrtc