diff --git a/api/rtp_packet_info.cc b/api/rtp_packet_info.cc index 312dc33f10..cba274ec38 100644 --- a/api/rtp_packet_info.cc +++ b/api/rtp_packet_info.cc @@ -27,20 +27,6 @@ RtpPacketInfo::RtpPacketInfo(uint32_t ssrc, rtp_timestamp_(rtp_timestamp), receive_time_(receive_time) {} -RtpPacketInfo::RtpPacketInfo( - uint32_t ssrc, - std::vector csrcs, - uint32_t rtp_timestamp, - absl::optional audio_level, - absl::optional absolute_capture_time, - Timestamp receive_time) - : ssrc_(ssrc), - csrcs_(std::move(csrcs)), - rtp_timestamp_(rtp_timestamp), - receive_time_(receive_time), - audio_level_(audio_level), - absolute_capture_time_(absolute_capture_time) {} - RtpPacketInfo::RtpPacketInfo(const RTPHeader& rtp_header, Timestamp receive_time) : ssrc_(rtp_header.ssrc), @@ -58,23 +44,6 @@ RtpPacketInfo::RtpPacketInfo(const RTPHeader& rtp_header, absolute_capture_time_ = extension.absolute_capture_time; } -RtpPacketInfo::RtpPacketInfo( - uint32_t ssrc, - std::vector csrcs, - uint32_t rtp_timestamp, - absl::optional audio_level, - absl::optional absolute_capture_time, - int64_t receive_time_ms) - : RtpPacketInfo(ssrc, - csrcs, - rtp_timestamp, - audio_level, - absolute_capture_time, - Timestamp::Millis(receive_time_ms)) {} -RtpPacketInfo::RtpPacketInfo(const RTPHeader& rtp_header, - int64_t receive_time_ms) - : RtpPacketInfo(rtp_header, Timestamp::Millis(receive_time_ms)) {} - bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs) { return (lhs.ssrc() == rhs.ssrc()) && (lhs.csrcs() == rhs.csrcs()) && (lhs.rtp_timestamp() == rhs.rtp_timestamp()) && diff --git a/api/rtp_packet_info.h b/api/rtp_packet_info.h index bc8784c6a8..8df12a36cf 100644 --- a/api/rtp_packet_info.h +++ b/api/rtp_packet_info.h @@ -37,27 +37,8 @@ class RTC_EXPORT RtpPacketInfo { uint32_t rtp_timestamp, Timestamp receive_time); - // TODO(bugs.webrtc.org/12722): Deprecated, remove once downstream projects - // are updated. - RtpPacketInfo(uint32_t ssrc, - std::vector csrcs, - uint32_t rtp_timestamp, - absl::optional audio_level, - absl::optional absolute_capture_time, - Timestamp receive_time); - RtpPacketInfo(const RTPHeader& rtp_header, Timestamp receive_time); - // TODO(bugs.webrtc.org/12722): Deprecated, remove once downstream projects - // are updated. - RtpPacketInfo(uint32_t ssrc, - std::vector csrcs, - uint32_t rtp_timestamp, - absl::optional audio_level, - absl::optional absolute_capture_time, - int64_t receive_time_ms); - RtpPacketInfo(const RTPHeader& rtp_header, int64_t receive_time_ms); - RtpPacketInfo(const RtpPacketInfo& other) = default; RtpPacketInfo(RtpPacketInfo&& other) = default; RtpPacketInfo& operator=(const RtpPacketInfo& other) = default; @@ -74,9 +55,6 @@ class RTC_EXPORT RtpPacketInfo { Timestamp receive_time() const { return receive_time_; } void set_receive_time(Timestamp value) { receive_time_ = value; } - // TODO(bugs.webrtc.org/12722): Deprecated, remove once downstream projects - // are updated. - int64_t receive_time_ms() const { return receive_time_.ms(); } absl::optional audio_level() const { return audio_level_; } RtpPacketInfo& set_audio_level(absl::optional value) {