Bump to WebRTC M120 release

Some API deprecation -- ExperimentalAgc and ExperimentalNs are gone.
We're continuing to carry iSAC even though it's gone upstream, but maybe
we'll want to drop that soon.
This commit is contained in:
Arun Raghavan
2023-12-12 10:42:58 -05:00
parent 9a202fb8c2
commit c6abf6cd3f
479 changed files with 20900 additions and 11996 deletions

View File

@ -16,11 +16,14 @@
#include <limits>
#include <string>
#include "api/units/time_delta.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
// Video timing timestamps in ms counted from capture_time_ms of a frame.
// This structure represents data sent in video-timing RTP header extension.
struct VideoSendTiming {
struct RTC_EXPORT VideoSendTiming {
enum TimingFrameFlags : uint8_t {
kNotTriggered = 0, // Timing info valid, but not to be transmitted.
// Used on send-side only.
@ -34,6 +37,7 @@ struct VideoSendTiming {
// https://webrtc.org/experiments/rtp-hdrext/video-timing/ extension stores
// 16-bit deltas of timestamps from packet capture time.
static uint16_t GetDeltaCappedMs(int64_t base_ms, int64_t time_ms);
static uint16_t GetDeltaCappedMs(TimeDelta delta);
uint16_t encode_start_delta_ms;
uint16_t encode_finish_delta_ms;
@ -41,21 +45,21 @@ struct VideoSendTiming {
uint16_t pacer_exit_delta_ms;
uint16_t network_timestamp_delta_ms;
uint16_t network2_timestamp_delta_ms;
uint8_t flags;
uint8_t flags = TimingFrameFlags::kInvalid;
};
// Used to report precise timings of a 'timing frames'. Contains all important
// timestamps for a lifetime of that specific frame. Reported as a string via
// GetStats(). Only frame which took the longest between two GetStats calls is
// reported.
struct TimingFrameInfo {
struct RTC_EXPORT TimingFrameInfo {
TimingFrameInfo();
// Returns end-to-end delay of a frame, if sender and receiver timestamps are
// synchronized, -1 otherwise.
int64_t EndToEndDelay() const;
// Returns true if current frame took longer to process than |other| frame.
// Returns true if current frame took longer to process than `other` frame.
// If other frame's clocks are not synchronized, current frame is always
// preferred.
bool IsLongerThan(const TimingFrameInfo& other) const;
@ -103,26 +107,43 @@ struct TimingFrameInfo {
// Minimum and maximum playout delay values from capture to render.
// These are best effort values.
//
// A value < 0 indicates no change from previous valid value.
//
// min = max = 0 indicates that the receiver should try and render
// frame as soon as possible.
//
// min = x, max = y indicates that the receiver is free to adapt
// in the range (x, y) based on network jitter.
struct VideoPlayoutDelay {
VideoPlayoutDelay() = default;
VideoPlayoutDelay(int min_ms, int max_ms) : min_ms(min_ms), max_ms(max_ms) {}
int min_ms = -1;
int max_ms = -1;
// This class ensures invariant 0 <= min <= max <= kMax.
class RTC_EXPORT VideoPlayoutDelay {
public:
// Maximum supported value for the delay limit.
static constexpr TimeDelta kMax = TimeDelta::Millis(10) * 0xFFF;
bool operator==(const VideoPlayoutDelay& rhs) const {
return min_ms == rhs.min_ms && max_ms == rhs.max_ms;
// Creates delay limits that indicates receiver should try to render frame
// as soon as possible.
static VideoPlayoutDelay Minimal() {
return VideoPlayoutDelay(TimeDelta::Zero(), TimeDelta::Zero());
}
};
// TODO(bugs.webrtc.org/7660): Old name, delete after downstream use is updated.
using PlayoutDelay = VideoPlayoutDelay;
// Creates valid, but unspecified limits.
VideoPlayoutDelay() = default;
VideoPlayoutDelay(const VideoPlayoutDelay&) = default;
VideoPlayoutDelay& operator=(const VideoPlayoutDelay&) = default;
VideoPlayoutDelay(TimeDelta min, TimeDelta max);
bool Set(TimeDelta min, TimeDelta max);
TimeDelta min() const { return min_; }
TimeDelta max() const { return max_; }
friend bool operator==(const VideoPlayoutDelay& lhs,
const VideoPlayoutDelay& rhs) {
return lhs.min_ == rhs.min_ && lhs.max_ == rhs.max_;
}
private:
TimeDelta min_ = TimeDelta::Zero();
TimeDelta max_ = kMax;
};
} // namespace webrtc