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5accd370e70b94517a39e622c75b794cc7a28829
webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver.cc
phoglund@webrtc.org 5accd370e7 RTP Receiver is now only deals with a receiver strategy. Cleaned up dependencies. 
BUG=
TESTED=vie/voe_auto_test, rtp_rtcp_unittests

Review URL: https://webrtc-codereview.appspot.com/1058004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3397 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-22 12:31:01 +00:00

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver.h"
#include <cassert>
#include <math.h>
#include <stdlib.h>
#include <string.h>
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/trace.h"
namespace webrtc {
using ModuleRTPUtility::AudioPayload;
using ModuleRTPUtility::GetCurrentRTP;
using ModuleRTPUtility::Payload;
using ModuleRTPUtility::RTPPayloadParser;
using ModuleRTPUtility::StringCompare;
using ModuleRTPUtility::VideoPayload;
RTPReceiver::RTPReceiver(const WebRtc_Word32 id,
Clock* clock,
ModuleRtpRtcpImpl* owner,
RtpAudioFeedback* incoming_audio_messages_callback,
RtpData* incoming_payload_callback,
RtpFeedback* incoming_messages_callback,
RTPReceiverStrategy* rtp_media_receiver,
RTPPayloadRegistry* rtp_payload_registry)
: Bitrate(clock),
rtp_payload_registry_(rtp_payload_registry),
rtp_media_receiver_(rtp_media_receiver),
id_(id),
rtp_rtcp_(*owner),
cb_rtp_feedback_(incoming_messages_callback),
critical_section_rtp_receiver_(
CriticalSectionWrapper::CreateCriticalSection()),
last_receive_time_(0),
last_received_payload_length_(0),
packet_timeout_ms_(0),
rtp_header_extension_map_(),
ssrc_(0),
num_csrcs_(0),
current_remote_csrc_(),
num_energy_(0),
current_remote_energy_(),
use_ssrc_filter_(false),
ssrc_filter_(0),
jitter_q4_(0),
jitter_max_q4_(0),
cumulative_loss_(0),
jitter_q4_transmission_time_offset_(0),
local_time_last_received_timestamp_(0),
last_received_frame_time_ms_(0),
last_received_timestamp_(0),
last_received_sequence_number_(0),
last_received_transmission_time_offset_(0),
received_seq_first_(0),
received_seq_max_(0),
received_seq_wraps_(0),
received_packet_oh_(12), // RTP header.
received_byte_count_(0),
received_old_packet_count_(0),
received_inorder_packet_count_(0),
last_report_inorder_packets_(0),
last_report_old_packets_(0),
last_report_seq_max_(0),
last_report_fraction_lost_(0),
last_report_cumulative_lost_(0),
last_report_extended_high_seq_num_(0),
last_report_jitter_(0),
last_report_jitter_transmission_time_offset_(0),
nack_method_(kNackOff),
rtx_(false),
ssrc_rtx_(0) {
assert(incoming_audio_messages_callback &&
incoming_messages_callback &&
incoming_payload_callback);
memset(current_remote_csrc_, 0, sizeof(current_remote_csrc_));
memset(current_remote_energy_, 0, sizeof(current_remote_energy_));
WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
}
RTPReceiver::~RTPReceiver() {
for (int i = 0; i < num_csrcs_; ++i) {
cb_rtp_feedback_->OnIncomingCSRCChanged(id_, current_remote_csrc_[i],
false);
}
delete critical_section_rtp_receiver_;
WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id_, "%s deleted", __FUNCTION__);
}
RtpVideoCodecTypes RTPReceiver::VideoCodecType() const {
ModuleRTPUtility::PayloadUnion media_specific;
rtp_media_receiver_->GetLastMediaSpecificPayload(&media_specific);
return media_specific.Video.videoCodecType;
}
WebRtc_UWord32 RTPReceiver::MaxConfiguredBitrate() const {
ModuleRTPUtility::PayloadUnion media_specific;
rtp_media_receiver_->GetLastMediaSpecificPayload(&media_specific);
return media_specific.Video.maxRate;
}
bool RTPReceiver::REDPayloadType(const WebRtc_Word8 payload_type) const {
return rtp_payload_registry_->red_payload_type() == payload_type;
}
WebRtc_Word8 RTPReceiver::REDPayloadType() const {
return rtp_payload_registry_->red_payload_type();
}
WebRtc_Word32 RTPReceiver::SetPacketTimeout(const WebRtc_UWord32 timeout_ms) {
CriticalSectionScoped lock(critical_section_rtp_receiver_);
packet_timeout_ms_ = timeout_ms;
return 0;
}
bool RTPReceiver::HaveNotReceivedPackets() const {
return last_receive_time_ == 0;
}
void RTPReceiver::PacketTimeout() {
bool packet_time_out = false;
{
CriticalSectionScoped lock(critical_section_rtp_receiver_);
if (packet_timeout_ms_ == 0) {
// Not configured.
return;
}
if (HaveNotReceivedPackets()) {
// Not active.
return;
}
WebRtc_Word64 now = clock_->TimeInMilliseconds();
if (now - last_receive_time_ > packet_timeout_ms_) {
packet_time_out = true;
last_receive_time_ = 0; // Only one callback.
rtp_payload_registry_->ResetLastReceivedPayloadTypes();
}
}
if (packet_time_out) {
cb_rtp_feedback_->OnPacketTimeout(id_);
}
}
void RTPReceiver::ProcessDeadOrAlive(const bool rtcp_alive,
const WebRtc_Word64 now) {
RTPAliveType alive = kRtpDead;
if (last_receive_time_ + 1000 > now) {
// Always alive if we have received a RTP packet the last second.
alive = kRtpAlive;
} else {
if (rtcp_alive) {
alive = rtp_media_receiver_->ProcessDeadOrAlive(
last_received_payload_length_);
} else {
// No RTP packet for 1 sec and no RTCP: dead.
}
}
cb_rtp_feedback_->OnPeriodicDeadOrAlive(id_, alive);
}
WebRtc_UWord16 RTPReceiver::PacketOHReceived() const {
CriticalSectionScoped lock(critical_section_rtp_receiver_);
return received_packet_oh_;
}
WebRtc_UWord32 RTPReceiver::PacketCountReceived() const {
CriticalSectionScoped lock(critical_section_rtp_receiver_);
return received_inorder_packet_count_;
}
WebRtc_UWord32 RTPReceiver::ByteCountReceived() const {
CriticalSectionScoped lock(critical_section_rtp_receiver_);
return received_byte_count_;
}
WebRtc_Word32 RTPReceiver::RegisterReceivePayload(
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const WebRtc_Word8 payload_type,
const WebRtc_UWord32 frequency,
const WebRtc_UWord8 channels,
const WebRtc_UWord32 rate) {
CriticalSectionScoped lock(critical_section_rtp_receiver_);
return rtp_payload_registry_->RegisterReceivePayload(
payload_name, payload_type, frequency, channels, rate);
}
WebRtc_Word32 RTPReceiver::DeRegisterReceivePayload(
const WebRtc_Word8 payload_type) {
CriticalSectionScoped lock(critical_section_rtp_receiver_);
return rtp_payload_registry_->DeRegisterReceivePayload(payload_type);
}
WebRtc_Word32 RTPReceiver::ReceivePayloadType(
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const WebRtc_UWord32 frequency,
const WebRtc_UWord8 channels,
const WebRtc_UWord32 rate,
WebRtc_Word8* payload_type) const {
CriticalSectionScoped lock(critical_section_rtp_receiver_);
return rtp_payload_registry_->ReceivePayloadType(
payload_name, frequency, channels, rate, payload_type);
}
WebRtc_Word32 RTPReceiver::ReceivePayload(
const WebRtc_Word8 payload_type,
char payload_name[RTP_PAYLOAD_NAME_SIZE],
WebRtc_UWord32* frequency,
WebRtc_UWord8* channels,
WebRtc_UWord32* rate) const {
CriticalSectionScoped lock(critical_section_rtp_receiver_);
return rtp_payload_registry_->ReceivePayload(
payload_type, payload_name, frequency, channels, rate);
}
WebRtc_Word32 RTPReceiver::RegisterRtpHeaderExtension(
const RTPExtensionType type,
const WebRtc_UWord8 id) {
CriticalSectionScoped cs(critical_section_rtp_receiver_);
return rtp_header_extension_map_.Register(type, id);
}
WebRtc_Word32 RTPReceiver::DeregisterRtpHeaderExtension(
const RTPExtensionType type) {
CriticalSectionScoped cs(critical_section_rtp_receiver_);
return rtp_header_extension_map_.Deregister(type);
}
void RTPReceiver::GetHeaderExtensionMapCopy(RtpHeaderExtensionMap* map) const {
CriticalSectionScoped cs(critical_section_rtp_receiver_);
rtp_header_extension_map_.GetCopy(map);
}
NACKMethod RTPReceiver::NACK() const {
CriticalSectionScoped lock(critical_section_rtp_receiver_);
return nack_method_;
}
// Turn negative acknowledgment requests on/off.
WebRtc_Word32 RTPReceiver::SetNACKStatus(const NACKMethod method) {
CriticalSectionScoped lock(critical_section_rtp_receiver_);
nack_method_ = method;
return 0;
}
void RTPReceiver::SetRTXStatus(const bool enable,
const WebRtc_UWord32 ssrc) {
CriticalSectionScoped lock(critical_section_rtp_receiver_);
rtx_ = enable;
ssrc_rtx_ = ssrc;
}
void RTPReceiver::RTXStatus(bool* enable, WebRtc_UWord32* ssrc) const {
CriticalSectionScoped lock(critical_section_rtp_receiver_);
*enable = rtx_;
*ssrc = ssrc_rtx_;
}
WebRtc_UWord32 RTPReceiver::SSRC() const {
CriticalSectionScoped lock(critical_section_rtp_receiver_);
return ssrc_;
}
// Get remote CSRC.
WebRtc_Word32 RTPReceiver::CSRCs(
WebRtc_UWord32 array_of_csrcs[kRtpCsrcSize]) const {
CriticalSectionScoped lock(critical_section_rtp_receiver_);
assert(num_csrcs_ <= kRtpCsrcSize);
if (num_csrcs_ > 0) {
memcpy(array_of_csrcs, current_remote_csrc_,
sizeof(WebRtc_UWord32)*num_csrcs_);
}
return num_csrcs_;
}
WebRtc_Word32 RTPReceiver::Energy(
WebRtc_UWord8 array_of_energy[kRtpCsrcSize]) const {
CriticalSectionScoped lock(critical_section_rtp_receiver_);
assert(num_energy_ <= kRtpCsrcSize);
if (num_energy_ > 0) {
memcpy(array_of_energy, current_remote_energy_,
sizeof(WebRtc_UWord8)*num_csrcs_);
}
return num_energy_;
}
WebRtc_Word32 RTPReceiver::IncomingRTPPacket(
WebRtcRTPHeader* rtp_header,
const WebRtc_UWord8* packet,
const WebRtc_UWord16 packet_length) {
// The rtp_header argument contains the parsed RTP header.
int length = packet_length - rtp_header->header.paddingLength;
// Sanity check.
if ((length - rtp_header->header.headerLength) < 0) {
WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
"%s invalid argument",
__FUNCTION__);
return -1;
}
if (rtx_) {
if (ssrc_rtx_ == rtp_header->header.ssrc) {
// Sanity check.
if (rtp_header->header.headerLength + 2 > packet_length) {
return -1;
}
rtp_header->header.ssrc = ssrc_;
rtp_header->header.sequenceNumber =
(packet[rtp_header->header.headerLength] << 8) +
packet[1 + rtp_header->header.headerLength];
// Count the RTX header as part of the RTP header.
rtp_header->header.headerLength += 2;
}
}
if (use_ssrc_filter_) {
if (rtp_header->header.ssrc != ssrc_filter_) {
WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_,
"%s drop packet due to SSRC filter",
__FUNCTION__);
return -1;
}
}
if (last_receive_time_ == 0) {
// Trigger only once.
if (length - rtp_header->header.headerLength == 0) {
// Keep-alive packet.
cb_rtp_feedback_->OnReceivedPacket(id_, kPacketKeepAlive);
} else {
cb_rtp_feedback_->OnReceivedPacket(id_, kPacketRtp);
}
}
WebRtc_Word8 first_payload_byte = 0;
if (length > 0) {
first_payload_byte = packet[rtp_header->header.headerLength];
}
// Trigger our callbacks.
CheckSSRCChanged(rtp_header);
bool is_red = false;
ModuleRTPUtility::PayloadUnion specific_payload = {};
if (CheckPayloadChanged(rtp_header,
first_payload_byte,
is_red,
&specific_payload) == -1) {
if (length - rtp_header->header.headerLength == 0) {
// OK, keep-alive packet.
WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
"%s received keepalive",
__FUNCTION__);
return 0;
}
WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_,
"%s received invalid payloadtype",
__FUNCTION__);
return -1;
}
CheckCSRC(rtp_header);
WebRtc_UWord16 payload_data_length =
ModuleRTPUtility::GetPayloadDataLength(rtp_header, packet_length);
bool is_first_packet_in_frame =
SequenceNumber() + 1 == rtp_header->header.sequenceNumber &&
TimeStamp() != rtp_header->header.timestamp;
bool is_first_packet = is_first_packet_in_frame || HaveNotReceivedPackets();
WebRtc_Word32 ret_val = rtp_media_receiver_->ParseRtpPacket(
rtp_header, specific_payload, is_red, packet, packet_length,
clock_->TimeInMilliseconds(), is_first_packet);
if (ret_val < 0) {
return ret_val;
}
CriticalSectionScoped lock(critical_section_rtp_receiver_);
// This compares to received_seq_max_. We store the last received after we
// have done the callback.
bool old_packet = RetransmitOfOldPacket(rtp_header->header.sequenceNumber,
rtp_header->header.timestamp);
// This updates received_seq_max_ and other members.
UpdateStatistics(rtp_header, payload_data_length, old_packet);
// Need to be updated after RetransmitOfOldPacket and
// RetransmitOfOldPacketUpdateStatistics.
last_receive_time_ = clock_->TimeInMilliseconds();
last_received_payload_length_ = payload_data_length;
if (!old_packet) {
if (last_received_timestamp_ != rtp_header->header.timestamp) {
last_received_timestamp_ = rtp_header->header.timestamp;
last_received_frame_time_ms_ = clock_->TimeInMilliseconds();
}
last_received_sequence_number_ = rtp_header->header.sequenceNumber;
last_received_transmission_time_offset_ =
rtp_header->extension.transmissionTimeOffset;
}
return ret_val;
}
// Implementation note: we expect to have the critical_section_rtp_receiver_
// critsect when we call this.
void RTPReceiver::UpdateStatistics(const WebRtcRTPHeader* rtp_header,
const WebRtc_UWord16 bytes,
const bool old_packet) {
WebRtc_UWord32 frequency_hz = rtp_media_receiver_->GetFrequencyHz();
Bitrate::Update(bytes);
received_byte_count_ += bytes;
if (received_seq_max_ == 0 && received_seq_wraps_ == 0) {
// This is the first received report.
received_seq_first_ = rtp_header->header.sequenceNumber;
received_seq_max_ = rtp_header->header.sequenceNumber;
received_inorder_packet_count_ = 1;
local_time_last_received_timestamp_ =
GetCurrentRTP(clock_, frequency_hz); // Time in samples.
return;
}
// Count only the new packets received.
if (InOrderPacket(rtp_header->header.sequenceNumber)) {
const WebRtc_UWord32 RTPtime =
GetCurrentRTP(clock_, frequency_hz); // Time in samples.
received_inorder_packet_count_++;
// Wrong if we use RetransmitOfOldPacket.
WebRtc_Word32 seq_diff =
rtp_header->header.sequenceNumber - received_seq_max_;
if (seq_diff < 0) {
// Wrap around detected.
received_seq_wraps_++;
}
// new max
received_seq_max_ = rtp_header->header.sequenceNumber;
if (rtp_header->header.timestamp != last_received_timestamp_ &&
received_inorder_packet_count_ > 1) {
WebRtc_Word32 time_diff_samples =
(RTPtime - local_time_last_received_timestamp_) -
(rtp_header->header.timestamp - last_received_timestamp_);
time_diff_samples = abs(time_diff_samples);
// lib_jingle sometimes deliver crazy jumps in TS for the same stream.
// If this happens, don't update jitter value. Use 5 secs video frequency
// as the treshold.
if (time_diff_samples < 450000) {
// Note we calculate in Q4 to avoid using float.
WebRtc_Word32 jitter_diff_q4 = (time_diff_samples << 4) - jitter_q4_;
jitter_q4_ += ((jitter_diff_q4 + 8) >> 4);
}
// Extended jitter report, RFC 5450.
// Actual network jitter, excluding the source-introduced jitter.
WebRtc_Word32 time_diff_samples_ext =
(RTPtime - local_time_last_received_timestamp_) -
((rtp_header->header.timestamp +
rtp_header->extension.transmissionTimeOffset) -
(last_received_timestamp_ +
last_received_transmission_time_offset_));
time_diff_samples_ext = abs(time_diff_samples_ext);
if (time_diff_samples_ext < 450000) {
WebRtc_Word32 jitter_diffQ4TransmissionTimeOffset =
(time_diff_samples_ext << 4) - jitter_q4_transmission_time_offset_;
jitter_q4_transmission_time_offset_ +=
((jitter_diffQ4TransmissionTimeOffset + 8) >> 4);
}
}
local_time_last_received_timestamp_ = RTPtime;
} else {
if (old_packet) {
received_old_packet_count_++;
} else {
received_inorder_packet_count_++;
}
}
WebRtc_UWord16 packet_oh =
rtp_header->header.headerLength + rtp_header->header.paddingLength;
// Our measured overhead. Filter from RFC 5104 4.2.1.2:
// avg_OH (new) = 15/16*avg_OH (old) + 1/16*pckt_OH,
received_packet_oh_ = (15 * received_packet_oh_ + packet_oh) >> 4;
}
// Implementation note: we expect to have the critical_section_rtp_receiver_
// critsect when we call this.
bool RTPReceiver::RetransmitOfOldPacket(
const WebRtc_UWord16 sequence_number,
const WebRtc_UWord32 rtp_time_stamp) const {
if (InOrderPacket(sequence_number)) {
return false;
}
WebRtc_UWord32 frequency_khz = rtp_media_receiver_->GetFrequencyHz() / 1000;
WebRtc_Word64 time_diff_ms = clock_->TimeInMilliseconds() -
last_receive_time_;
// Diff in time stamp since last received in order.
WebRtc_Word32 rtp_time_stamp_diff_ms =
static_cast<WebRtc_Word32>(rtp_time_stamp - last_received_timestamp_) /
frequency_khz;
WebRtc_UWord16 min_rtt = 0;
WebRtc_Word32 max_delay_ms = 0;
rtp_rtcp_.RTT(ssrc_, NULL, NULL, &min_rtt, NULL);
if (min_rtt == 0) {
// Jitter variance in samples.
float jitter = jitter_q4_ >> 4;
// Jitter standard deviation in samples.
float jitter_std = sqrt(jitter);
// 2 times the standard deviation => 95% confidence.
// And transform to milliseconds by dividing by the frequency in kHz.
max_delay_ms = static_cast<WebRtc_Word32>((2 * jitter_std) / frequency_khz);
// Min max_delay_ms is 1.
if (max_delay_ms == 0) {
max_delay_ms = 1;
}
} else {
max_delay_ms = (min_rtt / 3) + 1;
}
if (time_diff_ms > rtp_time_stamp_diff_ms + max_delay_ms) {
return true;
}
return false;
}
bool RTPReceiver::InOrderPacket(const WebRtc_UWord16 sequence_number) const {
if (received_seq_max_ >= sequence_number) {
// Detect wrap-around.
if (!(received_seq_max_ > 0xff00 && sequence_number < 0x0ff)) {
if (received_seq_max_ - NACK_PACKETS_MAX_SIZE > sequence_number) {
// We have a restart of the remote side.
} else {
// we received a retransmit of a packet we already have.
return false;
}
}
} else {
// Detect wrap-around.
if (sequence_number > 0xff00 && received_seq_max_ < 0x0ff) {
if (received_seq_max_ - NACK_PACKETS_MAX_SIZE > sequence_number) {
// We have a restart of the remote side
} else {
// We received a retransmit of a packet we already have
return false;
}
}
}
return true;
}
WebRtc_UWord16 RTPReceiver::SequenceNumber() const {
CriticalSectionScoped lock(critical_section_rtp_receiver_);
return last_received_sequence_number_;
}
WebRtc_UWord32 RTPReceiver::TimeStamp() const {
CriticalSectionScoped lock(critical_section_rtp_receiver_);
return last_received_timestamp_;
}
int32_t RTPReceiver::LastReceivedTimeMs() const {
CriticalSectionScoped lock(critical_section_rtp_receiver_);
return last_received_frame_time_ms_;
}
WebRtc_UWord32 RTPReceiver::PayloadTypeToPayload(
const WebRtc_UWord8 payload_type,
Payload*& payload) const {
CriticalSectionScoped lock(critical_section_rtp_receiver_);
return rtp_payload_registry_->PayloadTypeToPayload(payload_type, payload);
}
// Compute time stamp of the last incoming packet that is the first packet of
// its frame.
WebRtc_Word32 RTPReceiver::EstimatedRemoteTimeStamp(
WebRtc_UWord32& timestamp) const {
CriticalSectionScoped lock(critical_section_rtp_receiver_);
WebRtc_UWord32 frequency_hz = rtp_media_receiver_->GetFrequencyHz();
if (local_time_last_received_timestamp_ == 0) {
WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_,
"%s invalid state", __FUNCTION__);
return -1;
}
// Time in samples.
WebRtc_UWord32 diff = GetCurrentRTP(clock_, frequency_hz) -
local_time_last_received_timestamp_;
timestamp = last_received_timestamp_ + diff;
return 0;
}
// Get the currently configured SSRC filter.
WebRtc_Word32 RTPReceiver::SSRCFilter(WebRtc_UWord32& allowed_ssrc) const {
CriticalSectionScoped lock(critical_section_rtp_receiver_);
if (use_ssrc_filter_) {
allowed_ssrc = ssrc_filter_;
return 0;
}
WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_,
"%s invalid state", __FUNCTION__);
return -1;
}
// Set a SSRC to be used as a filter for incoming RTP streams.
WebRtc_Word32 RTPReceiver::SetSSRCFilter(
const bool enable, const WebRtc_UWord32 allowed_ssrc) {
CriticalSectionScoped lock(critical_section_rtp_receiver_);
use_ssrc_filter_ = enable;
if (enable) {
ssrc_filter_ = allowed_ssrc;
} else {
ssrc_filter_ = 0;
}
return 0;
}
// Implementation note: must not hold critsect when called.
void RTPReceiver::CheckSSRCChanged(const WebRtcRTPHeader* rtp_header) {
bool new_ssrc = false;
bool re_initialize_decoder = false;
char payload_name[RTP_PAYLOAD_NAME_SIZE];
WebRtc_UWord32 frequency = kDefaultVideoFrequency;
WebRtc_UWord8 channels = 1;
WebRtc_UWord32 rate = 0;
{
CriticalSectionScoped lock(critical_section_rtp_receiver_);
WebRtc_Word8 last_received_payload_type =
rtp_payload_registry_->last_received_payload_type();
if (ssrc_ != rtp_header->header.ssrc ||
(last_received_payload_type == -1 && ssrc_ == 0)) {
// We need the payload_type_ to make the call if the remote SSRC is 0.
new_ssrc = true;
ResetStatistics();
last_received_timestamp_ = 0;
last_received_sequence_number_ = 0;
last_received_transmission_time_offset_ = 0;
last_received_frame_time_ms_ = 0;
// Do we have a SSRC? Then the stream is restarted.
if (ssrc_) {
// Do we have the same codec? Then re-initialize coder.
if (rtp_header->header.payloadType == last_received_payload_type) {
re_initialize_decoder = true;
Payload* payload;
if (rtp_payload_registry_->PayloadTypeToPayload(
rtp_header->header.payloadType, payload) != 0) {
return;
}
assert(payload);
payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
if (payload->audio) {
frequency = payload->typeSpecific.Audio.frequency;
channels = payload->typeSpecific.Audio.channels;
rate = payload->typeSpecific.Audio.rate;
} else {
frequency = kDefaultVideoFrequency;
}
}
}
ssrc_ = rtp_header->header.ssrc;
}
}
if (new_ssrc) {
// We need to get this to our RTCP sender and receiver.
// We need to do this outside critical section.
rtp_rtcp_.SetRemoteSSRC(rtp_header->header.ssrc);
cb_rtp_feedback_->OnIncomingSSRCChanged(id_, rtp_header->header.ssrc);
}
if (re_initialize_decoder) {
if (-1 == cb_rtp_feedback_->OnInitializeDecoder(
id_, rtp_header->header.payloadType, payload_name, frequency,
channels, rate)) {
// New stream, same codec.
WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
"Failed to create decoder for payload type:%d",
rtp_header->header.payloadType);
}
}
}
// Implementation note: must not hold critsect when called.
// TODO(phoglund): Move as much as possible of this code path into the media
// specific receivers. Basically this method goes through a lot of trouble to
// compute something which is only used by the media specific parts later. If
// this code path moves we can get rid of some of the rtp_receiver ->
// media_specific interface (such as CheckPayloadChange, possibly get/set
// last known payload).
WebRtc_Word32 RTPReceiver::CheckPayloadChanged(
const WebRtcRTPHeader* rtp_header,
const WebRtc_Word8 first_payload_byte,
bool& is_red,
ModuleRTPUtility::PayloadUnion* specific_payload) {
bool re_initialize_decoder = false;
char payload_name[RTP_PAYLOAD_NAME_SIZE];
WebRtc_Word8 payload_type = rtp_header->header.payloadType;
{
CriticalSectionScoped lock(critical_section_rtp_receiver_);
WebRtc_Word8 last_received_payload_type =
rtp_payload_registry_->last_received_payload_type();
if (payload_type != last_received_payload_type) {
if (REDPayloadType(payload_type)) {
// Get the real codec payload type.
payload_type = first_payload_byte & 0x7f;
is_red = true;
if (REDPayloadType(payload_type)) {
// Invalid payload type, traced by caller. If we proceeded here,
// this would be set as |_last_received_payload_type|, and we would no
// longer catch corrupt packets at this level.
return -1;
}
// When we receive RED we need to check the real payload type.
if (payload_type == last_received_payload_type) {
rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
return 0;
}
}
bool should_reset_statistics = false;
bool should_discard_changes = false;
rtp_media_receiver_->CheckPayloadChanged(
payload_type, specific_payload, &should_reset_statistics,
&should_discard_changes);
if (should_reset_statistics) {
ResetStatistics();
}
if (should_discard_changes) {
is_red = false;
return 0;
}
Payload* payload;
if (rtp_payload_registry_->PayloadTypeToPayload(payload_type,
payload) != 0) {
// Not a registered payload type.
return -1;
}
assert(payload);
payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
rtp_payload_registry_->set_last_received_payload_type(payload_type);
re_initialize_decoder = true;
rtp_media_receiver_->SetLastMediaSpecificPayload(payload->typeSpecific);
rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
if (!payload->audio) {
if (VideoCodecType() == kRtpFecVideo) {
// Only reset the decoder on media packets.
re_initialize_decoder = false;
} else {
bool media_type_unchanged =
rtp_payload_registry_->ReportMediaPayloadType(payload_type);
if (media_type_unchanged) {
// Only reset the decoder if the media codec type has changed.
re_initialize_decoder = false;
}
}
}
if (re_initialize_decoder) {
ResetStatistics();
}
} else {
rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
is_red = false;
}
} // End critsect.
if (re_initialize_decoder) {
if (-1 == rtp_media_receiver_->InvokeOnInitializeDecoder(
cb_rtp_feedback_, id_, payload_type, payload_name,
*specific_payload)) {
return -1; // Wrong payload type.
}
}
return 0;
}
// Implementation note: must not hold critsect when called.
void RTPReceiver::CheckCSRC(const WebRtcRTPHeader* rtp_header) {
WebRtc_Word32 num_csrcs_diff = 0;
WebRtc_UWord32 old_remote_csrc[kRtpCsrcSize];
WebRtc_UWord8 old_num_csrcs = 0;
{
CriticalSectionScoped lock(critical_section_rtp_receiver_);
if (!rtp_media_receiver_->ShouldReportCsrcChanges(
rtp_header->header.payloadType)) {
return;
}
num_energy_ = rtp_header->type.Audio.numEnergy;
if (rtp_header->type.Audio.numEnergy > 0 &&
rtp_header->type.Audio.numEnergy <= kRtpCsrcSize) {
memcpy(current_remote_energy_,
rtp_header->type.Audio.arrOfEnergy,
rtp_header->type.Audio.numEnergy);
}
old_num_csrcs = num_csrcs_;
if (old_num_csrcs > 0) {
// Make a copy of old.
memcpy(old_remote_csrc, current_remote_csrc_,
num_csrcs_ * sizeof(WebRtc_UWord32));
}
const WebRtc_UWord8 num_csrcs = rtp_header->header.numCSRCs;
if ((num_csrcs > 0) && (num_csrcs <= kRtpCsrcSize)) {
// Copy new.
memcpy(current_remote_csrc_,
rtp_header->header.arrOfCSRCs,
num_csrcs * sizeof(WebRtc_UWord32));
}
if (num_csrcs > 0 || old_num_csrcs > 0) {
num_csrcs_diff = num_csrcs - old_num_csrcs;
num_csrcs_ = num_csrcs; // Update stored CSRCs.
} else {
// No change.
return;
}
} // End critsect.
bool have_called_callback = false;
// Search for new CSRC in old array.
for (WebRtc_UWord8 i = 0; i < rtp_header->header.numCSRCs; ++i) {
const WebRtc_UWord32 csrc = rtp_header->header.arrOfCSRCs[i];
bool found_match = false;
for (WebRtc_UWord8 j = 0; j < old_num_csrcs; ++j) {
if (csrc == old_remote_csrc[j]) { // old list
found_match = true;
break;
}
}
if (!found_match && csrc) {
// Didn't find it, report it as new.
have_called_callback = true;
cb_rtp_feedback_->OnIncomingCSRCChanged(id_, csrc, true);
}
}
// Search for old CSRC in new array.
for (WebRtc_UWord8 i = 0; i < old_num_csrcs; ++i) {
const WebRtc_UWord32 csrc = old_remote_csrc[i];
bool found_match = false;
for (WebRtc_UWord8 j = 0; j < rtp_header->header.numCSRCs; ++j) {
if (csrc == rtp_header->header.arrOfCSRCs[j]) {
found_match = true;
break;
}
}
if (!found_match && csrc) {
// Did not find it, report as removed.
have_called_callback = true;
cb_rtp_feedback_->OnIncomingCSRCChanged(id_, csrc, false);
}
}
if (!have_called_callback) {
// If the CSRC list contain non-unique entries we will end up here.
// Using CSRC 0 to signal this event, not interop safe, other
// implementations might have CSRC 0 as a valid value.
if (num_csrcs_diff > 0) {
cb_rtp_feedback_->OnIncomingCSRCChanged(id_, 0, true);
} else if (num_csrcs_diff < 0) {
cb_rtp_feedback_->OnIncomingCSRCChanged(id_, 0, false);
}
}
}
WebRtc_Word32 RTPReceiver::ResetStatistics() {
CriticalSectionScoped lock(critical_section_rtp_receiver_);
last_report_inorder_packets_ = 0;
last_report_old_packets_ = 0;
last_report_seq_max_ = 0;
last_report_fraction_lost_ = 0;
last_report_cumulative_lost_ = 0;
last_report_extended_high_seq_num_ = 0;
last_report_jitter_ = 0;
last_report_jitter_transmission_time_offset_ = 0;
jitter_q4_ = 0;
jitter_max_q4_ = 0;
cumulative_loss_ = 0;
jitter_q4_transmission_time_offset_ = 0;
received_seq_wraps_ = 0;
received_seq_max_ = 0;
received_seq_first_ = 0;
received_byte_count_ = 0;
received_old_packet_count_ = 0;
received_inorder_packet_count_ = 0;
return 0;
}
WebRtc_Word32 RTPReceiver::ResetDataCounters() {
CriticalSectionScoped lock(critical_section_rtp_receiver_);
received_byte_count_ = 0;
received_old_packet_count_ = 0;
received_inorder_packet_count_ = 0;
last_report_inorder_packets_ = 0;
return 0;
}
WebRtc_Word32 RTPReceiver::Statistics(
WebRtc_UWord8* fraction_lost,
WebRtc_UWord32* cum_lost,
WebRtc_UWord32* ext_max,
WebRtc_UWord32* jitter,
WebRtc_UWord32* max_jitter,
WebRtc_UWord32* jitter_transmission_time_offset,
bool reset) const {
WebRtc_Word32 missing;
return Statistics(fraction_lost,
cum_lost,
ext_max,
jitter,
max_jitter,
jitter_transmission_time_offset,
&missing,
reset);
}
WebRtc_Word32 RTPReceiver::Statistics(
WebRtc_UWord8* fraction_lost,
WebRtc_UWord32* cum_lost,
WebRtc_UWord32* ext_max,
WebRtc_UWord32* jitter,
WebRtc_UWord32* max_jitter,
WebRtc_UWord32* jitter_transmission_time_offset,
WebRtc_Word32* missing,
bool reset) const {
CriticalSectionScoped lock(critical_section_rtp_receiver_);
if (missing == NULL) {
return -1;
}
if (received_seq_first_ == 0 && received_byte_count_ == 0) {
// We have not received anything. -1 required by RTCP sender.
return -1;
}
if (!reset) {
if (last_report_inorder_packets_ == 0) {
// No report.
return -1;
}
// Just get last report.
if (fraction_lost) {
*fraction_lost = last_report_fraction_lost_;
}
if (cum_lost) {
*cum_lost = last_report_cumulative_lost_; // 24 bits valid.
}
if (ext_max) {
*ext_max = last_report_extended_high_seq_num_;
}
if (jitter) {
*jitter = last_report_jitter_;
}
if (max_jitter) {
// Note: internal jitter value is in Q4 and needs to be scaled by 1/16.
*max_jitter = (jitter_max_q4_ >> 4);
}
if (jitter_transmission_time_offset) {
*jitter_transmission_time_offset =
last_report_jitter_transmission_time_offset_;
}
return 0;
}
if (last_report_inorder_packets_ == 0) {
// First time we send a report.
last_report_seq_max_ = received_seq_first_ - 1;
}
// Calculate fraction lost.
WebRtc_UWord16 exp_since_last = (received_seq_max_ - last_report_seq_max_);
if (last_report_seq_max_ > received_seq_max_) {
// Can we assume that the seq_num can't go decrease over a full RTCP period?
exp_since_last = 0;
}
// Number of received RTP packets since last report, counts all packets but
// not re-transmissions.
WebRtc_UWord32 rec_since_last =
received_inorder_packet_count_ - last_report_inorder_packets_;
if (nack_method_ == kNackOff) {
// This is needed for re-ordered packets.
WebRtc_UWord32 old_packets =
received_old_packet_count_ - last_report_old_packets_;
rec_since_last += old_packets;
} else {
// With NACK we don't know the expected retransmitions during the last
// second. We know how many "old" packets we have received. We just count
// the number of old received to estimate the loss, but it still does not
// guarantee an exact number since we run this based on time triggered by
// sending of a RTP packet. This should have a minimum effect.
// With NACK we don't count old packets as received since they are
// re-transmitted. We use RTT to decide if a packet is re-ordered or
// re-transmitted.
}
*missing = 0;
if (exp_since_last > rec_since_last) {
*missing = (exp_since_last - rec_since_last);
}
WebRtc_UWord8 local_fraction_lost = 0;
if (exp_since_last) {
// Scale 0 to 255, where 255 is 100% loss.
local_fraction_lost = (WebRtc_UWord8)((255 * (*missing)) / exp_since_last);
}
if (fraction_lost) {
*fraction_lost = local_fraction_lost;
}
// We need a counter for cumulative loss too.
cumulative_loss_ += *missing;
if (jitter_q4_ > jitter_max_q4_) {
jitter_max_q4_ = jitter_q4_;
}
if (cum_lost) {
*cum_lost = cumulative_loss_;
}
if (ext_max) {
*ext_max = (received_seq_wraps_ << 16) + received_seq_max_;
}
// Note: internal jitter value is in Q4 and needs to be scaled by 1/16.
if (jitter) {
*jitter = (jitter_q4_ >> 4);
}
if (max_jitter) {
*max_jitter = (jitter_max_q4_ >> 4);
}
if (jitter_transmission_time_offset) {
*jitter_transmission_time_offset =
(jitter_q4_transmission_time_offset_ >> 4);
}
if (reset) {
// Store this report.
last_report_fraction_lost_ = local_fraction_lost;
last_report_cumulative_lost_ = cumulative_loss_; // 24 bits valid.
last_report_extended_high_seq_num_ =
(received_seq_wraps_ << 16) + received_seq_max_;
last_report_jitter_ = (jitter_q4_ >> 4);
last_report_jitter_transmission_time_offset_ =
(jitter_q4_transmission_time_offset_ >> 4);
// Only for report blocks in RTCP SR and RR.
last_report_inorder_packets_ = received_inorder_packet_count_;
last_report_old_packets_ = received_old_packet_count_;
last_report_seq_max_ = received_seq_max_;
}
return 0;
}
WebRtc_Word32 RTPReceiver::DataCounters(
WebRtc_UWord32* bytes_received,
WebRtc_UWord32* packets_received) const {
CriticalSectionScoped lock(critical_section_rtp_receiver_);
if (bytes_received) {
*bytes_received = received_byte_count_;
}
if (packets_received) {
*packets_received =
received_old_packet_count_ + received_inorder_packet_count_;
}
return 0;
}
void RTPReceiver::ProcessBitrate() {
CriticalSectionScoped cs(critical_section_rtp_receiver_);
Bitrate::Process();
}
} // namespace webrtc
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