e62202fedf
This implementation registers RTX-APT map inside RTP sender and receiver. While it only generates SDP with RTX associated with VP8 to make it compatible with previous Chrome versions. Should add following changes after reaches stable, * Use RTX-APT map for building and restoring RTP packets. * Add RTX support for RED or VP9 in Video engine. * Set RTX payload type for RED inside FecConfig in EndToEndTest. BUG=4024 R=mflodman@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/36889004 Cr-Commit-Position: refs/heads/master@{#9040}
117 lines
3.4 KiB
C++
117 lines
3.4 KiB
C++
/*
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* Copyright (c) 2013 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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// TODO(pbos): Move Config from common.h to here.
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#ifndef WEBRTC_CONFIG_H_
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#define WEBRTC_CONFIG_H_
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#include <string>
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#include <vector>
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#include "webrtc/common_types.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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// Settings for NACK, see RFC 4585 for details.
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struct NackConfig {
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NackConfig() : rtp_history_ms(0) {}
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// Send side: the time RTP packets are stored for retransmissions.
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// Receive side: the time the receiver is prepared to wait for
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// retransmissions.
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// Set to '0' to disable.
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int rtp_history_ms;
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};
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// Settings for forward error correction, see RFC 5109 for details. Set the
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// payload types to '-1' to disable.
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struct FecConfig {
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FecConfig()
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: ulpfec_payload_type(-1),
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red_payload_type(-1),
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red_rtx_payload_type(-1) {}
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std::string ToString() const;
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// Payload type used for ULPFEC packets.
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int ulpfec_payload_type;
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// Payload type used for RED packets.
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int red_payload_type;
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// RTX payload type for RED payload.
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int red_rtx_payload_type;
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};
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// RTP header extension to use for the video stream, see RFC 5285.
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struct RtpExtension {
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RtpExtension(const std::string& name, int id) : name(name), id(id) {}
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std::string ToString() const;
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static bool IsSupported(const std::string& name);
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static const char* kTOffset;
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static const char* kAbsSendTime;
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static const char* kVideoRotation;
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std::string name;
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int id;
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};
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struct VideoStream {
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VideoStream();
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~VideoStream();
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std::string ToString() const;
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size_t width;
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size_t height;
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int max_framerate;
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int min_bitrate_bps;
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int target_bitrate_bps;
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int max_bitrate_bps;
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int max_qp;
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// Bitrate thresholds for enabling additional temporal layers. Since these are
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// thresholds in between layers, we have one additional layer. One threshold
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// gives two temporal layers, one below the threshold and one above, two give
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// three, and so on.
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// The VideoEncoder may redistribute bitrates over the temporal layers so a
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// bitrate threshold of 100k and an estimate of 105k does not imply that we
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// get 100k in one temporal layer and 5k in the other, just that the bitrate
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// in the first temporal layer should not exceed 100k.
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// TODO(pbos): Apart from a special case for two-layer screencast these
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// thresholds are not propagated to the VideoEncoder. To be implemented.
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std::vector<int> temporal_layer_thresholds_bps;
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};
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struct VideoEncoderConfig {
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enum ContentType {
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kRealtimeVideo,
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kScreenshare,
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};
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VideoEncoderConfig();
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~VideoEncoderConfig();
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std::string ToString() const;
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std::vector<VideoStream> streams;
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ContentType content_type;
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void* encoder_specific_settings;
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// Padding will be used up to this bitrate regardless of the bitrate produced
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// by the encoder. Padding above what's actually produced by the encoder helps
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// maintaining a higher bitrate estimate. Padding will however not be sent
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// unless the estimated bandwidth indicates that the link can handle it.
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int min_transmit_bitrate_bps;
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};
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} // namespace webrtc
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#endif // WEBRTC_CONFIG_H_
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