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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17
webrtc/api/rtp_headers.h
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@ -103,15 +103,6 @@ struct RTPHeaderExtension {
(1 << kAbsSendTimeFraction));
}
TimeDelta GetAbsoluteSendTimeDelta(uint32_t previous_sendtime) const {
RTC_DCHECK(hasAbsoluteSendTime);
RTC_DCHECK(absoluteSendTime < (1ul << 24));
RTC_DCHECK(previous_sendtime < (1ul << 24));
int32_t delta =
static_cast<int32_t>((absoluteSendTime - previous_sendtime) << 8) >> 8;
return TimeDelta::Micros((delta * 1000000ll) / (1 << kAbsSendTimeFraction));
}
bool hasTransmissionTimeOffset;
int32_t transmissionTimeOffset;
bool hasAbsoluteSendTime;
@ -144,13 +135,12 @@ struct RTPHeaderExtension {
VideoPlayoutDelay playout_delay;
// 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.
// https://tools.ietf.org/html/rfc8852
std::string stream_id;
std::string 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
// https://tools.ietf.org/html/rfc8843
std::string mid;
absl::optional<ColorSpace> color_space;
@ -158,7 +148,7 @@ struct RTPHeaderExtension {
enum { kRtpCsrcSize = 15 }; // RFC 3550 page 13
struct RTPHeader {
struct RTC_EXPORT RTPHeader {
RTPHeader();
RTPHeader(const RTPHeader& other);
RTPHeader& operator=(const RTPHeader& other);
@ -172,7 +162,6 @@ struct RTPHeader {
uint32_t arrOfCSRCs[kRtpCsrcSize];
size_t paddingLength;
size_t headerLength;
int payload_type_frequency;
RTPHeaderExtension extension;
};