AnotherFoxGuy/webrtc-audio-processing
1
0
Fork 0
Code Pull Requests Activity

Bump to WebRTC M131 release

Browse Source 
Ongoing fixes and improvements, transient suppressor is gone. Also,
dropping isac because it doesn't seem to be useful, and is just build
system deadweight now.

Upstream references:

  Version: 131.0.6778.200
  WebRTC: 79aff54b0fa9238ce3518dd9eaf9610cd6f22e82
  Chromium: 2a19506ad24af755f2a215a4c61f775393e0db42
This commit is contained in:
Arun Raghavan
2024-12-24 19:32:07 -05:00
parent 8bdb53d91c
commit b5c48b97f6
263 changed files with 4628 additions and 20416 deletions
Download Patch File Download Diff File Expand all files Collapse all files

20
webrtc/api/rtp_packet_info.h
View File

@ -12,10 +12,10 @@
#define API_RTP_PACKET_INFO_H_
#include <cstdint>
#include <optional>
#include <utility>
#include <vector>
#include "absl/types/optional.h"
#include "api/rtp_headers.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
@ -56,26 +56,26 @@ class RTC_EXPORT RtpPacketInfo {
Timestamp receive_time() const { return receive_time_; }
void set_receive_time(Timestamp value) { receive_time_ = value; }
absl::optional<uint8_t> audio_level() const { return audio_level_; }
RtpPacketInfo& set_audio_level(absl::optional<uint8_t> value) {
std::optional<uint8_t> audio_level() const { return audio_level_; }
RtpPacketInfo& set_audio_level(std::optional<uint8_t> value) {
audio_level_ = value;
return *this;
}
const absl::optional<AbsoluteCaptureTime>& absolute_capture_time() const {
const std::optional<AbsoluteCaptureTime>& absolute_capture_time() const {
return absolute_capture_time_;
}
RtpPacketInfo& set_absolute_capture_time(
const absl::optional<AbsoluteCaptureTime>& value) {
const std::optional<AbsoluteCaptureTime>& value) {
absolute_capture_time_ = value;
return *this;
}
const absl::optional<TimeDelta>& local_capture_clock_offset() const {
const std::optional<TimeDelta>& local_capture_clock_offset() const {
return local_capture_clock_offset_;
}
RtpPacketInfo& set_local_capture_clock_offset(
absl::optional<TimeDelta> value) {
std::optional<TimeDelta> value) {
local_capture_clock_offset_ = value;
return *this;
}
@ -92,18 +92,18 @@ class RTC_EXPORT RtpPacketInfo {
// Fields from the Audio Level header extension:
// https://tools.ietf.org/html/rfc6464#section-3
absl::optional<uint8_t> audio_level_;
std::optional<uint8_t> audio_level_;
// Fields from the Absolute Capture Time header extension:
// http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time
absl::optional<AbsoluteCaptureTime> absolute_capture_time_;
std::optional<AbsoluteCaptureTime> absolute_capture_time_;
// Clock offset between the local clock and the capturer's clock.
// Do not confuse with `AbsoluteCaptureTime::estimated_capture_clock_offset`
// which instead represents the clock offset between a remote sender and the
// capturer. The following holds:
// Capture's NTP Clock = Local NTP Clock + Local-Capture Clock Offset
absl::optional<TimeDelta> local_capture_clock_offset_;
std::optional<TimeDelta> local_capture_clock_offset_;
};
bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs);