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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24
webrtc/api/rtp_packet_info.cc
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@ -16,27 +16,22 @@
namespace webrtc {
RtpPacketInfo::RtpPacketInfo()
: ssrc_(0), rtp_timestamp_(0), receive_time_ms_(-1) {}
: ssrc_(0), rtp_timestamp_(0), receive_time_(Timestamp::MinusInfinity()) {}
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)
RtpPacketInfo::RtpPacketInfo(uint32_t ssrc,
std::vector<uint32_t> csrcs,
uint32_t rtp_timestamp,
Timestamp receive_time)
: ssrc_(ssrc),
csrcs_(std::move(csrcs)),
rtp_timestamp_(rtp_timestamp),
audio_level_(audio_level),
absolute_capture_time_(absolute_capture_time),
receive_time_ms_(receive_time_ms) {}
receive_time_(receive_time) {}
RtpPacketInfo::RtpPacketInfo(const RTPHeader& rtp_header,
int64_t receive_time_ms)
Timestamp receive_time)
: ssrc_(rtp_header.ssrc),
rtp_timestamp_(rtp_header.timestamp),
receive_time_ms_(receive_time_ms) {
receive_time_(receive_time) {
const auto& extension = rtp_header.extension;
const auto csrcs_count = std::min<size_t>(rtp_header.numCSRCs, kRtpCsrcSize);
@ -52,9 +47,10 @@ RtpPacketInfo::RtpPacketInfo(const RTPHeader& rtp_header,
bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs) {
return (lhs.ssrc() == rhs.ssrc()) && (lhs.csrcs() == rhs.csrcs()) &&
(lhs.rtp_timestamp() == rhs.rtp_timestamp()) &&
(lhs.receive_time() == rhs.receive_time()) &&
(lhs.audio_level() == rhs.audio_level()) &&
(lhs.absolute_capture_time() == rhs.absolute_capture_time()) &&
(lhs.receive_time_ms() == rhs.receive_time_ms());
(lhs.local_capture_clock_offset() == rhs.local_capture_clock_offset());
}
} // namespace webrtc