blob: b2dfe2d272642fa5b8f73bd4a0e75b4a3c4925d1 [file] [log] [blame]
* 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.
#include <string>
#include <vector>
#include "webrtc/common.h"
#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) {}
std::string ToString() const;
// 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 {
: ulpfec_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, see RFC 5285.
struct RtpExtension {
RtpExtension() : id(0) {}
RtpExtension(const std::string& uri, int id) : uri(uri), id(id) {}
std::string ToString() const;
bool operator==(const RtpExtension& rhs) const {
return uri == rhs.uri && id ==;
static bool IsSupportedForAudio(const std::string& uri);
static bool IsSupportedForVideo(const std::string& uri);
// Header extension for audio levels, as defined in:
static const char* kAudioLevelUri;
static const int kAudioLevelDefaultId;
// Header extension for RTP timestamp offset, see RFC 5450 for details:
static const char* kTimestampOffsetUri;
static const int kTimestampOffsetDefaultId;
// Header extension for absolute send time, see url for details:
static const char* kAbsSendTimeUri;
static const int kAbsSendTimeDefaultId;
// Header extension for coordination of video orientation, see url for
// details:
static const char* kVideoRotationUri;
static const int kVideoRotationDefaultId;
// Header extension for transport sequence number, see url for details:
static const char* kTransportSequenceNumberUri;
static const int kTransportSequenceNumberDefaultId;
static const char* kPlayoutDelayUri;
static const int kPlayoutDelayDefaultId;
std::string uri;
int id;
struct 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 {
std::string ToString() const;
std::vector<VideoStream> streams;
std::vector<SpatialLayer> spatial_layers;
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;
bool expect_encode_from_texture;
// 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) {}
static const ConfigOptionID identifier = ConfigOptionID::kNetEqCapacityConfig;
bool enabled;
int capacity;
struct NetEqFastAccelerate {
NetEqFastAccelerate() : enabled(false) {}
explicit NetEqFastAccelerate(bool value) : enabled(value) {}
static const ConfigOptionID identifier = ConfigOptionID::kNetEqFastAccelerate;
bool enabled;
struct VoicePacing {
VoicePacing() : enabled(false) {}
explicit VoicePacing(bool value) : enabled(value) {}
static const ConfigOptionID identifier = ConfigOptionID::kVoicePacing;
bool enabled;
} // namespace webrtc
#endif // WEBRTC_CONFIG_H_