blob: 962e0f2fb9ac844ff243c0425c1b9df4dca76a4d [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_types.h"
#include "webrtc/rtc_base/basictypes.h"
#include "webrtc/rtc_base/optional.h"
#include "webrtc/rtc_base/refcount.h"
#include "webrtc/rtc_base/scoped_ref_ptr.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 ULPFEC forward error correction.
// Set the payload types to '-1' to disable.
struct UlpfecConfig {
: ulpfec_payload_type(-1),
red_rtx_payload_type(-1) {}
std::string ToString() const;
bool operator==(const UlpfecConfig& other) 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() {}
RtpExtension(const std::string& uri, int id) : uri(uri), id(id) {}
RtpExtension(const std::string& uri, int id, bool encrypt) : uri(uri),
id(id), encrypt(encrypt) {}
std::string ToString() const;
bool operator==(const RtpExtension& rhs) const {
return uri == rhs.uri && id == && encrypt == rhs.encrypt;
static bool IsSupportedForAudio(const std::string& uri);
static bool IsSupportedForVideo(const std::string& uri);
// Return "true" if the given RTP header extension URI may be encrypted.
static bool IsEncryptionSupported(const std::string& uri);
// Returns the named header extension if found among all extensions,
// nullptr otherwise.
static const RtpExtension* FindHeaderExtensionByUri(
const std::vector<RtpExtension>& extensions,
const std::string& uri);
// Return a list of RTP header extensions with the non-encrypted extensions
// removed if both the encrypted and non-encrypted extension is present for
// the same URI.
static std::vector<RtpExtension> FilterDuplicateNonEncrypted(
const std::vector<RtpExtension>& extensions);
// 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 video content type. E.g. default or screenshare.
static const char kVideoContentTypeUri[];
static const int kVideoContentTypeDefaultId;
// Header extension for video timing.
static const char kVideoTimingUri[];
static const int kVideoTimingDefaultId;
// 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;
// Encryption of Header Extensions, see RFC 6904 for details:
static const char kEncryptHeaderExtensionsUri[];
// Inclusive min and max IDs for one-byte header extensions, per RFC5285.
static const int kMinId;
static const int kMaxId;
std::string uri;
int id = 0;
bool encrypt = false;
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(kthelgason): 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;
class VideoEncoderConfig {
// These are reference counted to permit copying VideoEncoderConfig and be
// kept alive until all encoder_specific_settings go out of scope.
// TODO(kthelgason): Consider removing the need for copying VideoEncoderConfig
// and use rtc::Optional for encoder_specific_settings instead.
class EncoderSpecificSettings : public rtc::RefCountInterface {
// TODO(pbos): Remove FillEncoderSpecificSettings as soon as VideoCodec is
// not in use and encoder implementations ask for codec-specific structs
// directly.
void FillEncoderSpecificSettings(VideoCodec* codec_struct) const;
virtual void FillVideoCodecVp8(VideoCodecVP8* vp8_settings) const;
virtual void FillVideoCodecVp9(VideoCodecVP9* vp9_settings) const;
virtual void FillVideoCodecH264(VideoCodecH264* h264_settings) const;
~EncoderSpecificSettings() override {}
friend class VideoEncoderConfig;
class H264EncoderSpecificSettings : public EncoderSpecificSettings {
explicit H264EncoderSpecificSettings(const VideoCodecH264& specifics);
void FillVideoCodecH264(VideoCodecH264* h264_settings) const override;
VideoCodecH264 specifics_;
class Vp8EncoderSpecificSettings : public EncoderSpecificSettings {
explicit Vp8EncoderSpecificSettings(const VideoCodecVP8& specifics);
void FillVideoCodecVp8(VideoCodecVP8* vp8_settings) const override;
VideoCodecVP8 specifics_;
class Vp9EncoderSpecificSettings : public EncoderSpecificSettings {
explicit Vp9EncoderSpecificSettings(const VideoCodecVP9& specifics);
void FillVideoCodecVp9(VideoCodecVP9* vp9_settings) const override;
VideoCodecVP9 specifics_;
enum class ContentType {
class VideoStreamFactoryInterface : public rtc::RefCountInterface {
// An implementation should return a std::vector<VideoStream> with the
// wanted VideoStream settings for the given video resolution.
// The size of the vector may not be larger than
// |encoder_config.number_of_streams|.
virtual std::vector<VideoStream> CreateEncoderStreams(
int width,
int height,
const VideoEncoderConfig& encoder_config) = 0;
~VideoStreamFactoryInterface() override {}
VideoEncoderConfig& operator=(VideoEncoderConfig&&) = default;
VideoEncoderConfig& operator=(const VideoEncoderConfig&) = delete;
// Mostly used by tests. Avoid creating copies if you can.
VideoEncoderConfig Copy() const { return VideoEncoderConfig(*this); }
std::string ToString() const;
rtc::scoped_refptr<VideoStreamFactoryInterface> video_stream_factory;
std::vector<SpatialLayer> spatial_layers;
ContentType content_type;
rtc::scoped_refptr<const EncoderSpecificSettings> 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;
int max_bitrate_bps;
// Max number of encoded VideoStreams to produce.
size_t number_of_streams;
// Access to the copy constructor is private to force use of the Copy()
// method for those exceptional cases where we do use it.
VideoEncoderConfig(const VideoEncoderConfig&);
} // namespace webrtc
#endif // WEBRTC_CONFIG_H_