8f27fcce79
Reasons for revert: 1. glaznev discovered potentially related problems using the Android AppRTCDemo. 2. We're trying to do an M41 webrtc roll in Chromium, and this CL is risky. > Support associated payload type when registering Rtx payload type. > > Major changes include, > - Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType. > - Receiver: Restore RTP packets by the new RTX-APT map. > - Sender: Send RTP packets by checking RTX-APT map. > - Add RTX payload type for RED in the default codec list. > > BUG=4024 > R=pbos@webrtc.org, stefan@webrtc.org > TBR=mflodman@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/26259004 > > Patch from Changbin Shao <changbin.shao@intel.com>. TBR=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/33829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8033 4adac7df-926f-26a2-2b94-8c16560cd09d
139 lines
4.0 KiB
C++
139 lines
4.0 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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struct SsrcStats {
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SsrcStats()
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: sent_width(0),
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sent_height(0),
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total_bitrate_bps(0),
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retransmit_bitrate_bps(0),
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avg_delay_ms(0),
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max_delay_ms(0) {}
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FrameCounts frame_counts;
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int sent_width;
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int sent_height;
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// TODO(holmer): Move bitrate_bps out to the webrtc::Call layer.
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int total_bitrate_bps;
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int retransmit_bitrate_bps;
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int avg_delay_ms;
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int max_delay_ms;
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StreamDataCounters rtp_stats;
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RtcpStatistics rtcp_stats;
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};
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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() : ulpfec_payload_type(-1), red_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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};
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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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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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: width(0),
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height(0),
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max_framerate(-1),
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min_bitrate_bps(-1),
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target_bitrate_bps(-1),
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max_bitrate_bps(-1),
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max_qp(-1) {}
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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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: content_type(kRealtimeVideo),
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encoder_specific_settings(NULL),
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min_transmit_bitrate_bps(0) {}
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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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