diff options
Diffstat (limited to 'chromium/third_party/webrtc/modules/audio_coding/neteq/rtcp.h')
| -rw-r--r-- | chromium/third_party/webrtc/modules/audio_coding/neteq/rtcp.h | 140 |
1 files changed, 48 insertions, 92 deletions
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/rtcp.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/rtcp.h index 5e066eb38f2..2a765efa588 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/rtcp.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/rtcp.h @@ -8,95 +8,51 @@ * be found in the AUTHORS file in the root of the source tree. */ -/* - * RTCP statistics reporting. - */ - -#ifndef RTCP_H -#define RTCP_H - -#include "typedefs.h" - -typedef struct -{ - uint16_t cycles; /* The number of wrap-arounds for the sequence number */ - uint16_t max_seq; /* The maximum sequence number received - (starts from 0 again after wrap around) */ - uint16_t base_seq; /* The sequence number of the first packet that arrived */ - uint32_t received; /* The number of packets that has been received */ - uint32_t rec_prior; /* Number of packets received when last report was generated */ - uint32_t exp_prior; /* Number of packets that should have been received if no - packets were lost. Stored value from last report. */ - uint32_t jitter; /* Jitter statistics at this instance (calculated according to RFC) */ - int32_t transit; /* Clock difference for previous packet (RTPtimestamp - LOCALtime_rec) */ -} WebRtcNetEQ_RTCP_t; - -/**************************************************************************** - * WebRtcNetEQ_RTCPInit(...) - * - * This function calculates the parameters that are needed for the RTCP - * report. - * - * Input: - * - RTCP_inst : RTCP instance, that contains information about the - * packets that have been received etc. - * - seqNo : Packet number of the first received frame. - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcNetEQ_RTCPInit(WebRtcNetEQ_RTCP_t *RTCP_inst, uint16_t uw16_seqNo); - -/**************************************************************************** - * WebRtcNetEQ_RTCPUpdate(...) - * - * This function calculates the parameters that are needed for the RTCP - * report. - * - * Input: - * - RTCP_inst : RTCP instance, that contains information about the - * packets that have been received etc. - * - seqNo : Packet number of the first received frame. - * - timeStamp : Time stamp from the RTP header. - * - recTime : Time (in RTP timestamps) when this packet was received. - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcNetEQ_RTCPUpdate(WebRtcNetEQ_RTCP_t *RTCP_inst, uint16_t uw16_seqNo, - uint32_t uw32_timeStamp, uint32_t uw32_recTime); - -/**************************************************************************** - * WebRtcNetEQ_RTCPGetStats(...) - * - * This function calculates the parameters that are needed for the RTCP - * report. - * - * Input: - * - RTCP_inst : RTCP instance, that contains information about the - * packets that have been received etc. - * - doNotReset : If non-zero, the fraction lost statistics will not - * be reset. - * - * Output: - * - RTCP_inst : Updated RTCP information (some statistics are - * reset when generating this report) - * - fraction_lost : Number of lost RTP packets divided by the number of - * expected packets, since the last RTCP Report. - * - cum_lost : Cumulative number of lost packets during this - * session. - * - ext_max : Extended highest sequence number received. - * - jitter : Inter-arrival jitter. - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcNetEQ_RTCPGetStats(WebRtcNetEQ_RTCP_t *RTCP_inst, - uint16_t *puw16_fraction_lost, - uint32_t *puw32_cum_lost, uint32_t *puw32_ext_max, - uint32_t *puw32_jitter, int16_t doNotReset); - -#endif +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_RTCP_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_RTCP_H_ + +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/neteq/interface/neteq.h" +#include "webrtc/typedefs.h" + +namespace webrtc { + +// Forward declaration. +struct RTPHeader; + +class Rtcp { + public: + Rtcp() { + Init(0); + } + + ~Rtcp() {} + + // Resets the RTCP statistics, and sets the first received sequence number. + void Init(uint16_t start_sequence_number); + + // Updates the RTCP statistics with a new received packet. + void Update(const RTPHeader& rtp_header, uint32_t receive_timestamp); + + // Returns the current RTCP statistics. If |no_reset| is true, the statistics + // are not reset, otherwise they are. + void GetStatistics(bool no_reset, RtcpStatistics* stats); + + private: + uint16_t cycles_; // The number of wrap-arounds for the sequence number. + uint16_t max_seq_no_; // The maximum sequence number received. Starts over + // from 0 after wrap-around. + uint16_t base_seq_no_; // The sequence number of the first received packet. + uint32_t received_packets_; // The number of packets that have been received. + uint32_t received_packets_prior_; // Number of packets received when last + // report was generated. + uint32_t expected_prior_; // Expected number of packets, at the time of the + // last report. + uint32_t jitter_; // Current jitter value. + int32_t transit_; // Clock difference for previous packet. + + DISALLOW_COPY_AND_ASSIGN(Rtcp); +}; + +} // namespace webrtc +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_RTCP_H_ |