fdd1057949
1. standard plumbing CVO through vie layer. 2. added a rtp_cvo.h which has both conversion functions from rtp header byte to/from VideoRotation. WebRTCVideoEngine will later pass the rotation info in SendFrame() through VieVideoFrameI420. BUG=4145 R=mflodman@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/46429007 Cr-Commit-Position: refs/heads/master@{#8703} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8703 4adac7df-926f-26a2-2b94-8c16560cd09d
111 lines
3.3 KiB
C++
111 lines
3.3 KiB
C++
/*
|
|
* Copyright (c) 2013 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.
|
|
*/
|
|
|
|
// TODO(pbos): Move Config from common.h to here.
|
|
|
|
#ifndef WEBRTC_CONFIG_H_
|
|
#define WEBRTC_CONFIG_H_
|
|
|
|
#include <string>
|
|
#include <vector>
|
|
|
|
#include "webrtc/common_types.h"
|
|
#include "webrtc/typedefs.h"
|
|
|
|
namespace webrtc {
|
|
|
|
// Settings for NACK, see RFC 4585 for details.
|
|
struct NackConfig {
|
|
NackConfig() : rtp_history_ms(0) {}
|
|
// Send side: the time RTP packets are stored for retransmissions.
|
|
// Receive side: the time the receiver is prepared to wait for
|
|
// retransmissions.
|
|
// Set to '0' to disable.
|
|
int rtp_history_ms;
|
|
};
|
|
|
|
// Settings for forward error correction, see RFC 5109 for details. Set the
|
|
// payload types to '-1' to disable.
|
|
struct FecConfig {
|
|
FecConfig() : ulpfec_payload_type(-1), red_payload_type(-1) {}
|
|
std::string ToString() const;
|
|
// Payload type used for ULPFEC packets.
|
|
int ulpfec_payload_type;
|
|
|
|
// Payload type used for RED packets.
|
|
int red_payload_type;
|
|
};
|
|
|
|
// RTP header extension to use for the video stream, see RFC 5285.
|
|
struct RtpExtension {
|
|
RtpExtension(const std::string& name, int id) : name(name), id(id) {}
|
|
std::string ToString() const;
|
|
static bool IsSupported(const std::string& name);
|
|
|
|
static const char* kTOffset;
|
|
static const char* kAbsSendTime;
|
|
static const char* kVideoRotation;
|
|
std::string name;
|
|
int id;
|
|
};
|
|
|
|
struct VideoStream {
|
|
VideoStream();
|
|
~VideoStream();
|
|
std::string ToString() const;
|
|
|
|
size_t width;
|
|
size_t height;
|
|
int max_framerate;
|
|
|
|
int min_bitrate_bps;
|
|
int target_bitrate_bps;
|
|
int max_bitrate_bps;
|
|
|
|
int max_qp;
|
|
|
|
// Bitrate thresholds for enabling additional temporal layers. Since these are
|
|
// thresholds in between layers, we have one additional layer. One threshold
|
|
// gives two temporal layers, one below the threshold and one above, two give
|
|
// three, and so on.
|
|
// The VideoEncoder may redistribute bitrates over the temporal layers so a
|
|
// bitrate threshold of 100k and an estimate of 105k does not imply that we
|
|
// get 100k in one temporal layer and 5k in the other, just that the bitrate
|
|
// in the first temporal layer should not exceed 100k.
|
|
// TODO(pbos): Apart from a special case for two-layer screencast these
|
|
// thresholds are not propagated to the VideoEncoder. To be implemented.
|
|
std::vector<int> temporal_layer_thresholds_bps;
|
|
};
|
|
|
|
struct VideoEncoderConfig {
|
|
enum ContentType {
|
|
kRealtimeVideo,
|
|
kScreenshare,
|
|
};
|
|
|
|
VideoEncoderConfig();
|
|
~VideoEncoderConfig();
|
|
std::string ToString() const;
|
|
|
|
std::vector<VideoStream> streams;
|
|
ContentType content_type;
|
|
void* encoder_specific_settings;
|
|
|
|
// Padding will be used up to this bitrate regardless of the bitrate produced
|
|
// by the encoder. Padding above what's actually produced by the encoder helps
|
|
// maintaining a higher bitrate estimate. Padding will however not be sent
|
|
// unless the estimated bandwidth indicates that the link can handle it.
|
|
int min_transmit_bitrate_bps;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_CONFIG_H_
|