|  | /* | 
|  | *  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), | 
|  | red_rtx_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; | 
|  |  | 
|  | // RTX payload type for RED payload. | 
|  | int red_rtx_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 IsSupportedForAudio(const std::string& name); | 
|  | static bool IsSupportedForVideo(const std::string& name); | 
|  |  | 
|  | static const char* kTOffset; | 
|  | static const char* kAbsSendTime; | 
|  | static const char* kVideoRotation; | 
|  | static const char* kAudioLevel; | 
|  | 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 class ContentType { | 
|  | kRealtimeVideo, | 
|  | kScreen, | 
|  | }; | 
|  |  | 
|  | 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; | 
|  | }; | 
|  |  | 
|  | // Controls the capacity of the packet buffer in NetEq. The capacity is the | 
|  | // maximum number of packets that the buffer can contain. If the limit is | 
|  | // exceeded, the buffer will be flushed. The capacity does not affect the actual | 
|  | // audio delay in the general case, since this is governed by the target buffer | 
|  | // level (calculated from the jitter profile). It is only in the rare case of | 
|  | // severe network freezes that a higher capacity will lead to a (transient) | 
|  | // increase in audio delay. | 
|  | struct NetEqCapacityConfig { | 
|  | NetEqCapacityConfig() : enabled(false), capacity(0) {} | 
|  | explicit NetEqCapacityConfig(int value) : enabled(true), capacity(value) {} | 
|  | bool enabled; | 
|  | int capacity; | 
|  | }; | 
|  |  | 
|  | struct NetEqFastAccelerate { | 
|  | NetEqFastAccelerate() : enabled(false) {} | 
|  | explicit NetEqFastAccelerate(bool value) : enabled(value) {} | 
|  | bool enabled; | 
|  | }; | 
|  |  | 
|  | }  // namespace webrtc | 
|  |  | 
|  | #endif  // WEBRTC_CONFIG_H_ |