AnotherFoxGuy/webrtc-audio-processing
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Bump to WebRTC M120 release

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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
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44
webrtc/api/rtp_packet_info.h
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@ -17,14 +17,16 @@
#include "absl/types/optional.h"
#include "api/rtp_headers.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
//
// Structure to hold information about a received |RtpPacket|. It is primarily
// Structure to hold information about a received `RtpPacket`. It is primarily
// used to carry per-packet information from when a packet is received until
// the information is passed to |SourceTracker|.
// the information is passed to `SourceTracker`.
//
class RTC_EXPORT RtpPacketInfo {
public:
@ -33,11 +35,9 @@ class RTC_EXPORT RtpPacketInfo {
RtpPacketInfo(uint32_t ssrc,
std::vector<uint32_t> csrcs,
uint32_t rtp_timestamp,
absl::optional<uint8_t> audio_level,
absl::optional<AbsoluteCaptureTime> absolute_capture_time,
int64_t receive_time_ms);
Timestamp receive_time);
RtpPacketInfo(const RTPHeader& rtp_header, int64_t receive_time_ms);
RtpPacketInfo(const RTPHeader& rtp_header, Timestamp receive_time);
RtpPacketInfo(const RtpPacketInfo& other) = default;
RtpPacketInfo(RtpPacketInfo&& other) = default;
@ -53,19 +53,32 @@ class RTC_EXPORT RtpPacketInfo {
uint32_t rtp_timestamp() const { return rtp_timestamp_; }
void set_rtp_timestamp(uint32_t value) { rtp_timestamp_ = value; }
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_; }
void set_audio_level(absl::optional<uint8_t> value) { audio_level_ = value; }
RtpPacketInfo& set_audio_level(absl::optional<uint8_t> value) {
audio_level_ = value;
return *this;
}
const absl::optional<AbsoluteCaptureTime>& absolute_capture_time() const {
return absolute_capture_time_;
}
void set_absolute_capture_time(
RtpPacketInfo& set_absolute_capture_time(
const absl::optional<AbsoluteCaptureTime>& value) {
absolute_capture_time_ = value;
return *this;
}
int64_t receive_time_ms() const { return receive_time_ms_; }
void set_receive_time_ms(int64_t value) { receive_time_ms_ = value; }
const absl::optional<TimeDelta>& local_capture_clock_offset() const {
return local_capture_clock_offset_;
}
RtpPacketInfo& set_local_capture_clock_offset(
absl::optional<TimeDelta> value) {
local_capture_clock_offset_ = value;
return *this;
}
private:
// Fields from the RTP header:
@ -74,6 +87,9 @@ class RTC_EXPORT RtpPacketInfo {
std::vector<uint32_t> csrcs_;
uint32_t rtp_timestamp_;
// Local `webrtc::Clock`-based timestamp of when the packet was received.
Timestamp receive_time_;
// Fields from the Audio Level header extension:
// https://tools.ietf.org/html/rfc6464#section-3
absl::optional<uint8_t> audio_level_;
@ -82,8 +98,12 @@ class RTC_EXPORT RtpPacketInfo {
// http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time
absl::optional<AbsoluteCaptureTime> absolute_capture_time_;
// Local |webrtc::Clock|-based timestamp of when the packet was received.
int64_t receive_time_ms_;
// 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_;
};
bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs);