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Fix circular dependencies in webrtc_common.

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One reason for the circular deps is that common_types.h is a
historical dumping ground for various structs and defines that
are believed to be generally useful. I tried moving things out
that did not appear to be used downstream (StreamCounters,
RtpCounters etc) and moved the things that seemed used
(RtpHeader + supporting structs) to a new file api/rtp_headers.h.
This makes their place in the api more clear while moving out
the things that don't belong in the API in the first place.

I had to extract out typedefs.h from webrtc_common to resolve
another circular dependency. I believe checks includes typedefs,
but common depends on checks.

Bug: webrtc:7745
Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b
Reviewed-on: https://webrtc-review.googlesource.com/33001
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21295}
This commit is contained in:
Patrik Höglund
2017-12-15 14:40:10 +01:00
committed by Commit Bot
parent d5d0540b86
commit 3e113438b1
57 changed files with 559 additions and 316 deletions
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241
common_types.h
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@ -19,9 +19,6 @@
#include "api/array_view.h"
#include "api/optional.h"
#include "api/video/video_content_type.h"
#include "api/video/video_rotation.h"
#include "api/video/video_timing.h"
#include "rtc_base/checks.h"
#include "rtc_base/deprecation.h"
#include "typedefs.h" // NOLINT(build/include)
@ -685,244 +682,6 @@ struct PlayoutDelay {
int max_ms;
};
// Class to represent the value of RTP header extensions that are
// variable-length strings (e.g., RtpStreamId and RtpMid).
// Unlike std::string, it can be copied with memcpy and cleared with memset.
//
// Empty value represents unset header extension (use empty() to query).
class StringRtpHeaderExtension {
public:
// String RTP header extensions are limited to 16 bytes because it is the
// maximum length that can be encoded with one-byte header extensions.
static constexpr size_t kMaxSize = 16;
static bool IsLegalName(rtc::ArrayView<const char> name);
StringRtpHeaderExtension() { value_[0] = 0; }
explicit StringRtpHeaderExtension(rtc::ArrayView<const char> value) {
Set(value.data(), value.size());
}
StringRtpHeaderExtension(const StringRtpHeaderExtension&) = default;
StringRtpHeaderExtension& operator=(const StringRtpHeaderExtension&) =
default;
bool empty() const { return value_[0] == 0; }
const char* data() const { return value_; }
size_t size() const { return strnlen(value_, kMaxSize); }
void Set(rtc::ArrayView<const uint8_t> value) {
Set(reinterpret_cast<const char*>(value.data()), value.size());
}
void Set(const char* data, size_t size);
friend bool operator==(const StringRtpHeaderExtension& lhs,
const StringRtpHeaderExtension& rhs) {
return strncmp(lhs.value_, rhs.value_, kMaxSize) == 0;
}
friend bool operator!=(const StringRtpHeaderExtension& lhs,
const StringRtpHeaderExtension& rhs) {
return !(lhs == rhs);
}
private:
char value_[kMaxSize];
};
// StreamId represents RtpStreamId which is a string.
typedef StringRtpHeaderExtension StreamId;
// Mid represents RtpMid which is a string.
typedef StringRtpHeaderExtension Mid;
struct RTPHeaderExtension {
RTPHeaderExtension();
RTPHeaderExtension(const RTPHeaderExtension& other);
RTPHeaderExtension& operator=(const RTPHeaderExtension& other);
bool hasTransmissionTimeOffset;
int32_t transmissionTimeOffset;
bool hasAbsoluteSendTime;
uint32_t absoluteSendTime;
bool hasTransportSequenceNumber;
uint16_t transportSequenceNumber;
// Audio Level includes both level in dBov and voiced/unvoiced bit. See:
// https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/
bool hasAudioLevel;
bool voiceActivity;
uint8_t audioLevel;
// For Coordination of Video Orientation. See
// http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
// ts_126114v120700p.pdf
bool hasVideoRotation;
VideoRotation videoRotation;
// TODO(ilnik): Refactor this and one above to be rtc::Optional() and remove
// a corresponding bool flag.
bool hasVideoContentType;
VideoContentType videoContentType;
bool has_video_timing;
VideoSendTiming video_timing;
PlayoutDelay playout_delay = {-1, -1};
// For identification of a stream when ssrc is not signaled. See
// https://tools.ietf.org/html/draft-ietf-avtext-rid-09
// TODO(danilchap): Update url from draft to release version.
StreamId stream_id;
StreamId repaired_stream_id;
// For identifying the media section used to interpret this RTP packet. See
// https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-38
Mid mid;
};
struct RTPHeader {
RTPHeader();
RTPHeader(const RTPHeader& other);
RTPHeader& operator=(const RTPHeader& other);
bool markerBit;
uint8_t payloadType;
uint16_t sequenceNumber;
uint32_t timestamp;
uint32_t ssrc;
uint8_t numCSRCs;
uint32_t arrOfCSRCs[kRtpCsrcSize];
size_t paddingLength;
size_t headerLength;
int payload_type_frequency;
RTPHeaderExtension extension;
};
struct RtpPacketCounter {
RtpPacketCounter()
: header_bytes(0), payload_bytes(0), padding_bytes(0), packets(0) {}
void Add(const RtpPacketCounter& other) {
header_bytes += other.header_bytes;
payload_bytes += other.payload_bytes;
padding_bytes += other.padding_bytes;
packets += other.packets;
}
void Subtract(const RtpPacketCounter& other) {
RTC_DCHECK_GE(header_bytes, other.header_bytes);
header_bytes -= other.header_bytes;
RTC_DCHECK_GE(payload_bytes, other.payload_bytes);
payload_bytes -= other.payload_bytes;
RTC_DCHECK_GE(padding_bytes, other.padding_bytes);
padding_bytes -= other.padding_bytes;
RTC_DCHECK_GE(packets, other.packets);
packets -= other.packets;
}
void AddPacket(size_t packet_length, const RTPHeader& header) {
++packets;
header_bytes += header.headerLength;
padding_bytes += header.paddingLength;
payload_bytes +=
packet_length - (header.headerLength + header.paddingLength);
}
size_t TotalBytes() const {
return header_bytes + payload_bytes + padding_bytes;
}
size_t header_bytes; // Number of bytes used by RTP headers.
size_t payload_bytes; // Payload bytes, excluding RTP headers and padding.
size_t padding_bytes; // Number of padding bytes.
uint32_t packets; // Number of packets.
};
// Data usage statistics for a (rtp) stream.
struct StreamDataCounters {
StreamDataCounters();
void Add(const StreamDataCounters& other) {
transmitted.Add(other.transmitted);
retransmitted.Add(other.retransmitted);
fec.Add(other.fec);
if (other.first_packet_time_ms != -1 &&
(other.first_packet_time_ms < first_packet_time_ms ||
first_packet_time_ms == -1)) {
// Use oldest time.
first_packet_time_ms = other.first_packet_time_ms;
}
}
void Subtract(const StreamDataCounters& other) {
transmitted.Subtract(other.transmitted);
retransmitted.Subtract(other.retransmitted);
fec.Subtract(other.fec);
if (other.first_packet_time_ms != -1 &&
(other.first_packet_time_ms > first_packet_time_ms ||
first_packet_time_ms == -1)) {
// Use youngest time.
first_packet_time_ms = other.first_packet_time_ms;
}
}
int64_t TimeSinceFirstPacketInMs(int64_t now_ms) const {
return (first_packet_time_ms == -1) ? -1 : (now_ms - first_packet_time_ms);
}
// Returns the number of bytes corresponding to the actual media payload (i.e.
// RTP headers, padding, retransmissions and fec packets are excluded).
// Note this function does not have meaning for an RTX stream.
size_t MediaPayloadBytes() const {
return transmitted.payload_bytes - retransmitted.payload_bytes -
fec.payload_bytes;
}
int64_t first_packet_time_ms; // Time when first packet is sent/received.
RtpPacketCounter transmitted; // Number of transmitted packets/bytes.
RtpPacketCounter retransmitted; // Number of retransmitted packets/bytes.
RtpPacketCounter fec; // Number of redundancy packets/bytes.
};
// Callback, called whenever byte/packet counts have been updated.
class StreamDataCountersCallback {
public:
virtual ~StreamDataCountersCallback() {}
virtual void DataCountersUpdated(const StreamDataCounters& counters,
uint32_t ssrc) = 0;
};
// RTCP mode to use. Compound mode is described by RFC 4585 and reduced-size
// RTCP mode is described by RFC 5506.
enum class RtcpMode { kOff, kCompound, kReducedSize };
enum NetworkState {
kNetworkUp,
kNetworkDown,
};
struct RtpKeepAliveConfig final {
// If no packet has been sent for |timeout_interval_ms|, send a keep-alive
// packet. The keep-alive packet is an empty (no payload) RTP packet with a
// payload type of 20 as long as the other end has not negotiated the use of
// this value. If this value has already been negotiated, then some other
// unused static payload type from table 5 of RFC 3551 shall be used and set
// in |payload_type|.
int64_t timeout_interval_ms = -1;
uint8_t payload_type = 20;
bool operator==(const RtpKeepAliveConfig& o) const {
return timeout_interval_ms == o.timeout_interval_ms &&
payload_type == o.payload_type;
}
bool operator!=(const RtpKeepAliveConfig& o) const { return !(*this == o); }
};
// Currently only VP8/VP9 specific.
struct RtpPayloadState {
int16_t picture_id = -1;
};
} // namespace webrtc
#endif // COMMON_TYPES_H_