Google Git
Sign in
webrtc / src / becf9c897c41eea3f021f99d87889c32c78b0de9 / . / webrtc / modules / rtp_rtcp / source / rtp_sender.cc
blob: 991f12ad69eb7ff34c7cd3f0073455e663912635 [file] [log] [blame]
/*
* 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_sender.h"
#include <cstdlib> // srand
#include "webrtc/modules/pacing/include/paced_sender.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/trace.h"
namespace webrtc {
RTPSender::RTPSender(const WebRtc_Word32 id, const bool audio, Clock *clock,
Transport *transport, RtpAudioFeedback *audio_feedback,
PacedSender *paced_sender)
: Bitrate(clock), id_(id), audio_configured_(audio), audio_(NULL),
video_(NULL), paced_sender_(paced_sender),
send_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
transport_(transport), sending_media_(true), // Default to sending media.
max_payload_length_(IP_PACKET_SIZE - 28), // Default is IP-v4/UDP.
target_send_bitrate_(0), packet_over_head_(28), payload_type_(-1),
payload_type_map_(), rtp_header_extension_map_(),
transmission_time_offset_(0),
// NACK.
nack_byte_count_times_(), nack_byte_count_(), nack_bitrate_(clock),
packet_history_(new RTPPacketHistory(clock)),
// Statistics
packets_sent_(0), payload_bytes_sent_(0), start_time_stamp_forced_(false),
start_time_stamp_(0), ssrc_db_(*SSRCDatabase::GetSSRCDatabase()),
remote_ssrc_(0), sequence_number_forced_(false), sequence_number_(0),
sequence_number_rtx_(0), ssrc_forced_(false), ssrc_(0), time_stamp_(0),
csrcs_(0), csrc_(), include_csrcs_(true), rtx_(false), ssrc_rtx_(0) {
memset(nack_byte_count_times_, 0, sizeof(nack_byte_count_times_));
memset(nack_byte_count_, 0, sizeof(nack_byte_count_));
memset(csrc_, 0, sizeof(csrc_));
// We need to seed the random generator.
srand(static_cast<WebRtc_UWord32>(clock_->TimeInMilliseconds()));
ssrc_ = ssrc_db_.CreateSSRC(); // Can't be 0.
if (audio) {
audio_ = new RTPSenderAudio(id, clock_, this);
audio_->RegisterAudioCallback(audio_feedback);
} else {
video_ = new RTPSenderVideo(id, clock_, this);
}
WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
}
RTPSender::~RTPSender() {
if (remote_ssrc_ != 0) {
ssrc_db_.ReturnSSRC(remote_ssrc_);
}
ssrc_db_.ReturnSSRC(ssrc_);
SSRCDatabase::ReturnSSRCDatabase();
delete send_critsect_;
while (!payload_type_map_.empty()) {
std::map<WebRtc_Word8, ModuleRTPUtility::Payload *>::iterator it =
payload_type_map_.begin();
delete it->second;
payload_type_map_.erase(it);
}
delete packet_history_;
delete audio_;
delete video_;
WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id_, "%s deleted", __FUNCTION__);
}
void RTPSender::SetTargetSendBitrate(const WebRtc_UWord32 bits) {
target_send_bitrate_ = static_cast<uint16_t>(bits / 1000);
}
WebRtc_UWord16 RTPSender::ActualSendBitrateKbit() const {
return (WebRtc_UWord16)(Bitrate::BitrateNow() / 1000);
}
WebRtc_UWord32 RTPSender::VideoBitrateSent() const {
if (video_) {
return video_->VideoBitrateSent();
}
return 0;
}
WebRtc_UWord32 RTPSender::FecOverheadRate() const {
if (video_) {
return video_->FecOverheadRate();
}
return 0;
}
WebRtc_UWord32 RTPSender::NackOverheadRate() const {
return nack_bitrate_.BitrateLast();
}
WebRtc_Word32 RTPSender::SetTransmissionTimeOffset(
const WebRtc_Word32 transmission_time_offset) {
if (transmission_time_offset > (0x800000 - 1) ||
transmission_time_offset < -(0x800000 - 1)) { // Word24.
return -1;
}
CriticalSectionScoped cs(send_critsect_);
transmission_time_offset_ = transmission_time_offset;
return 0;
}
WebRtc_Word32 RTPSender::RegisterRtpHeaderExtension(const RTPExtensionType type,
const WebRtc_UWord8 id) {
CriticalSectionScoped cs(send_critsect_);
return rtp_header_extension_map_.Register(type, id);
}
WebRtc_Word32 RTPSender::DeregisterRtpHeaderExtension(
const RTPExtensionType type) {
CriticalSectionScoped cs(send_critsect_);
return rtp_header_extension_map_.Deregister(type);
}
WebRtc_UWord16 RTPSender::RtpHeaderExtensionTotalLength() const {
CriticalSectionScoped cs(send_critsect_);
return rtp_header_extension_map_.GetTotalLengthInBytes();
}
WebRtc_Word32 RTPSender::RegisterPayload(
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const WebRtc_Word8 payload_number, const WebRtc_UWord32 frequency,
const WebRtc_UWord8 channels, const WebRtc_UWord32 rate) {
assert(payload_name);
CriticalSectionScoped cs(send_critsect_);
std::map<WebRtc_Word8, ModuleRTPUtility::Payload *>::iterator it =
payload_type_map_.find(payload_number);
if (payload_type_map_.end() != it) {
// We already use this payload type.
ModuleRTPUtility::Payload *payload = it->second;
assert(payload);
// Check if it's the same as we already have.
if (ModuleRTPUtility::StringCompare(payload->name, payload_name,
RTP_PAYLOAD_NAME_SIZE - 1)) {
if (audio_configured_ && payload->audio &&
payload->typeSpecific.Audio.frequency == frequency &&
(payload->typeSpecific.Audio.rate == rate ||
payload->typeSpecific.Audio.rate == 0 || rate == 0)) {
payload->typeSpecific.Audio.rate = rate;
// Ensure that we update the rate if new or old is zero.
return 0;
}
if (!audio_configured_ && !payload->audio) {
return 0;
}
}
return -1;
}
WebRtc_Word32 ret_val = -1;
ModuleRTPUtility::Payload *payload = NULL;
if (audio_configured_) {
ret_val = audio_->RegisterAudioPayload(payload_name, payload_number,
frequency, channels, rate, payload);
} else {
ret_val = video_->RegisterVideoPayload(payload_name, payload_number, rate,
payload);
}
if (payload) {
payload_type_map_[payload_number] = payload;
}
return ret_val;
}
WebRtc_Word32 RTPSender::DeRegisterSendPayload(
const WebRtc_Word8 payload_type) {
CriticalSectionScoped lock(send_critsect_);
std::map<WebRtc_Word8, ModuleRTPUtility::Payload *>::iterator it =
payload_type_map_.find(payload_type);
if (payload_type_map_.end() == it) {
return -1;
}
ModuleRTPUtility::Payload *payload = it->second;
delete payload;
payload_type_map_.erase(it);
return 0;
}
WebRtc_Word8 RTPSender::SendPayloadType() const { return payload_type_; }
int RTPSender::SendPayloadFrequency() const { return audio_->AudioFrequency(); }
WebRtc_Word32 RTPSender::SetMaxPayloadLength(
const WebRtc_UWord16 max_payload_length,
const WebRtc_UWord16 packet_over_head) {
// Sanity check.
if (max_payload_length < 100 || max_payload_length > IP_PACKET_SIZE) {
WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_, "%s invalid argument",
__FUNCTION__);
return -1;
}
CriticalSectionScoped cs(send_critsect_);
max_payload_length_ = max_payload_length;
packet_over_head_ = packet_over_head;
WEBRTC_TRACE(kTraceInfo, kTraceRtpRtcp, id_, "SetMaxPayloadLength to %d.",
max_payload_length);
return 0;
}
WebRtc_UWord16 RTPSender::MaxDataPayloadLength() const {
if (audio_configured_) {
return max_payload_length_ - RTPHeaderLength();
} else {
return max_payload_length_ - RTPHeaderLength() -
video_->FECPacketOverhead() - ((rtx_) ? 2 : 0);
// Include the FEC/ULP/RED overhead.
}
}
WebRtc_UWord16 RTPSender::MaxPayloadLength() const {
return max_payload_length_;
}
WebRtc_UWord16 RTPSender::PacketOverHead() const { return packet_over_head_; }
void RTPSender::SetRTXStatus(const bool enable, const bool set_ssrc,
const WebRtc_UWord32 ssrc) {
CriticalSectionScoped cs(send_critsect_);
rtx_ = enable;
if (enable) {
if (set_ssrc) {
ssrc_rtx_ = ssrc;
} else {
ssrc_rtx_ = ssrc_db_.CreateSSRC(); // Can't be 0.
}
}
}
void RTPSender::RTXStatus(bool *enable, WebRtc_UWord32 *SSRC) const {
CriticalSectionScoped cs(send_critsect_);
*enable = rtx_;
*SSRC = ssrc_rtx_;
}
WebRtc_Word32 RTPSender::CheckPayloadType(const WebRtc_Word8 payload_type,
RtpVideoCodecTypes *video_type) {
CriticalSectionScoped cs(send_critsect_);
if (payload_type < 0) {
WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_, "\tinvalid payload_type (%d)",
payload_type);
return -1;
}
if (audio_configured_) {
WebRtc_Word8 red_pl_type = -1;
if (audio_->RED(red_pl_type) == 0) {
// We have configured RED.
if (red_pl_type == payload_type) {
// And it's a match...
return 0;
}
}
}
if (payload_type_ == payload_type) {
if (!audio_configured_) {
*video_type = video_->VideoCodecType();
}
return 0;
}
std::map<WebRtc_Word8, ModuleRTPUtility::Payload *>::iterator it =
payload_type_map_.find(payload_type);
if (it == payload_type_map_.end()) {
WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
"\tpayloadType:%d not registered", payload_type);
return -1;
}
payload_type_ = payload_type;
ModuleRTPUtility::Payload *payload = it->second;
assert(payload);
if (!payload->audio && !audio_configured_) {
video_->SetVideoCodecType(payload->typeSpecific.Video.videoCodecType);
*video_type = payload->typeSpecific.Video.videoCodecType;
video_->SetMaxConfiguredBitrateVideo(payload->typeSpecific.Video.maxRate);
}
return 0;
}
WebRtc_Word32 RTPSender::SendOutgoingData(
const FrameType frame_type, const WebRtc_Word8 payload_type,
const WebRtc_UWord32 capture_timestamp, int64_t capture_time_ms,
const WebRtc_UWord8 *payload_data, const WebRtc_UWord32 payload_size,
const RTPFragmentationHeader *fragmentation,
VideoCodecInformation *codec_info, const RTPVideoTypeHeader *rtp_type_hdr) {
{
// Drop this packet if we're not sending media packets.
CriticalSectionScoped cs(send_critsect_);
if (!sending_media_) {
return 0;
}
}
RtpVideoCodecTypes video_type = kRtpNoVideo;
if (CheckPayloadType(payload_type, &video_type) != 0) {
WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
"%s invalid argument failed to find payload_type:%d",
__FUNCTION__, payload_type);
return -1;
}
if (audio_configured_) {
assert(frame_type == kAudioFrameSpeech || frame_type == kAudioFrameCN ||
frame_type == kFrameEmpty);
return audio_->SendAudio(frame_type, payload_type, capture_timestamp,
payload_data, payload_size, fragmentation);
} else {
assert(frame_type != kAudioFrameSpeech && frame_type != kAudioFrameCN);
if (frame_type == kFrameEmpty) {
return SendPaddingAccordingToBitrate(payload_type, capture_timestamp,
capture_time_ms);
}
return video_->SendVideo(video_type, frame_type, payload_type,
capture_timestamp, capture_time_ms, payload_data,
payload_size, fragmentation, codec_info,
rtp_type_hdr);
}
}
WebRtc_Word32 RTPSender::SendPaddingAccordingToBitrate(
WebRtc_Word8 payload_type, WebRtc_UWord32 capture_timestamp,
int64_t capture_time_ms) {
// Current bitrate since last estimate(1 second) averaged with the
// estimate since then, to get the most up to date bitrate.
uint32_t current_bitrate = BitrateNow();
int bitrate_diff = target_send_bitrate_ * 1000 - current_bitrate;
if (bitrate_diff <= 0) {
return 0;
}
int bytes = 0;
if (current_bitrate == 0) {
// Start up phase. Send one 33.3 ms batch to start with.
bytes = (bitrate_diff / 8) / 30;
} else {
bytes = (bitrate_diff / 8);
// Cap at 200 ms of target send data.
int bytes_cap = target_send_bitrate_ * 25; // 1000 / 8 / 5.
if (bytes > bytes_cap) {
bytes = bytes_cap;
}
}
return SendPadData(payload_type, capture_timestamp, capture_time_ms, bytes);
}
WebRtc_Word32 RTPSender::SendPadData(
WebRtc_Word8 payload_type, WebRtc_UWord32 capture_timestamp,
int64_t capture_time_ms, WebRtc_Word32 bytes) {
// Drop this packet if we're not sending media packets.
if (!sending_media_) {
return 0;
}
// Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
int max_length = 224;
WebRtc_UWord8 data_buffer[IP_PACKET_SIZE];
for (; bytes > 0; bytes -= max_length) {
int padding_bytes_in_packet = max_length;
if (bytes < max_length) {
padding_bytes_in_packet = (bytes + 16) & 0xffe0; // Keep our modulus 32.
}
if (padding_bytes_in_packet < 32) {
// Sanity don't send empty packets.
break;
}
// Correct seq num, timestamp and payload type.
int header_length = BuildRTPheader(
data_buffer, payload_type, false, // No markerbit.
capture_timestamp, true, // Timestamp provided.
true); // Increment sequence number.
data_buffer[0] |= 0x20; // Set padding bit.
WebRtc_Word32 *data =
reinterpret_cast<WebRtc_Word32 *>(&(data_buffer[header_length]));
// Fill data buffer with random data.
for (int j = 0; j < (padding_bytes_in_packet >> 2); ++j) {
data[j] = rand(); // NOLINT
}
// Set number of padding bytes in the last byte of the packet.
data_buffer[header_length + padding_bytes_in_packet - 1] =
padding_bytes_in_packet;
// Send the packet.
if (0 > SendToNetwork(data_buffer, padding_bytes_in_packet, header_length,
capture_time_ms, kDontRetransmit)) {
// Error sending the packet.
break;
}
}
if (bytes > 31) { // 31 due to our modulus 32.
// We did not manage to send all bytes.
return -1;
}
return 0;
}
void RTPSender::SetStorePacketsStatus(const bool enable,
const WebRtc_UWord16 number_to_store) {
packet_history_->SetStorePacketsStatus(enable, number_to_store);
}
bool RTPSender::StorePackets() const { return packet_history_->StorePackets(); }
WebRtc_Word32 RTPSender::ReSendPacket(WebRtc_UWord16 packet_id,
WebRtc_UWord32 min_resend_time) {
WebRtc_UWord16 length = IP_PACKET_SIZE;
WebRtc_UWord8 data_buffer[IP_PACKET_SIZE];
WebRtc_UWord8 *buffer_to_send_ptr = data_buffer;
int64_t stored_time_in_ms;
StorageType type;
bool found = packet_history_->GetRTPPacket(packet_id, min_resend_time,
data_buffer, &length,
&stored_time_in_ms, &type);
if (!found) {
// Packet not found.
return 0;
}
if (length == 0 || type == kDontRetransmit) {
// No bytes copied (packet recently resent, skip resending) or
// packet should not be retransmitted.
return 0;
}
WebRtc_UWord8 data_buffer_rtx[IP_PACKET_SIZE];
if (rtx_) {
buffer_to_send_ptr = data_buffer_rtx;
CriticalSectionScoped cs(send_critsect_);
// Add RTX header.
ModuleRTPUtility::RTPHeaderParser rtp_parser(
reinterpret_cast<const WebRtc_UWord8 *>(data_buffer), length);
WebRtcRTPHeader rtp_header;
rtp_parser.Parse(rtp_header);
// Add original RTP header.
memcpy(data_buffer_rtx, data_buffer, rtp_header.header.headerLength);
// Replace sequence number.
WebRtc_UWord8 *ptr = data_buffer_rtx + 2;
ModuleRTPUtility::AssignUWord16ToBuffer(ptr, sequence_number_rtx_++);
// Replace SSRC.
ptr += 6;
ModuleRTPUtility::AssignUWord32ToBuffer(ptr, ssrc_rtx_);
// Add OSN (original sequence number).
ptr = data_buffer_rtx + rtp_header.header.headerLength;
ModuleRTPUtility::AssignUWord16ToBuffer(ptr,
rtp_header.header.sequenceNumber);
ptr += 2;
// Add original payload data.
memcpy(ptr, data_buffer + rtp_header.header.headerLength,
length - rtp_header.header.headerLength);
length += 2;
}
WebRtc_Word32 bytes_sent = ReSendToNetwork(buffer_to_send_ptr, length);
if (bytes_sent <= 0) {
WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_,
"Transport failed to resend packet_id %u", packet_id);
return -1;
}
// Store the time when the packet was last resent.
packet_history_->UpdateResendTime(packet_id);
return bytes_sent;
}
WebRtc_Word32 RTPSender::ReSendToNetwork(const WebRtc_UWord8 *packet,
const WebRtc_UWord32 size) {
WebRtc_Word32 bytes_sent = -1;
if (transport_) {
bytes_sent = transport_->SendPacket(id_, packet, size);
}
if (bytes_sent <= 0) {
return -1;
}
// Update send statistics.
CriticalSectionScoped cs(send_critsect_);
Bitrate::Update(bytes_sent);
packets_sent_++;
// We on purpose don't add to payload_bytes_sent_ since this is a
// re-transmit and not new payload data.
return bytes_sent;
}
int RTPSender::SelectiveRetransmissions() const {
if (!video_)
return -1;
return video_->SelectiveRetransmissions();
}
int RTPSender::SetSelectiveRetransmissions(uint8_t settings) {
if (!video_)
return -1;
return video_->SetSelectiveRetransmissions(settings);
}
void RTPSender::OnReceivedNACK(
const std::list<uint16_t>& nack_sequence_numbers,
const WebRtc_UWord16 avg_rtt) {
const WebRtc_Word64 now = clock_->TimeInMilliseconds();
WebRtc_UWord32 bytes_re_sent = 0;
// Enough bandwidth to send NACK?
if (!ProcessNACKBitRate(now)) {
WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
"NACK bitrate reached. Skip sending NACK response. Target %d",
target_send_bitrate_);
return;
}
for (std::list<uint16_t>::const_iterator it = nack_sequence_numbers.begin();
it != nack_sequence_numbers.end(); ++it) {
const WebRtc_Word32 bytes_sent = ReSendPacket(*it, 5 + avg_rtt);
if (bytes_sent > 0) {
bytes_re_sent += bytes_sent;
} else if (bytes_sent == 0) {
// The packet has previously been resent.
// Try resending next packet in the list.
continue;
} else if (bytes_sent < 0) {
// Failed to send one Sequence number. Give up the rest in this nack.
WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_,
"Failed resending RTP packet %d, Discard rest of packets",
*it);
break;
}
// Delay bandwidth estimate (RTT * BW).
if (target_send_bitrate_ != 0 && avg_rtt) {
// kbits/s * ms = bits => bits/8 = bytes
WebRtc_UWord32 target_bytes =
(static_cast<WebRtc_UWord32>(target_send_bitrate_) * avg_rtt) >> 3;
if (bytes_re_sent > target_bytes) {
break; // Ignore the rest of the packets in the list.
}
}
}
if (bytes_re_sent > 0) {
// TODO(pwestin) consolidate these two methods.
UpdateNACKBitRate(bytes_re_sent, now);
nack_bitrate_.Update(bytes_re_sent);
}
}
bool RTPSender::ProcessNACKBitRate(const WebRtc_UWord32 now) {
WebRtc_UWord32 num = 0;
WebRtc_Word32 byte_count = 0;
const WebRtc_UWord32 avg_interval = 1000;
CriticalSectionScoped cs(send_critsect_);
if (target_send_bitrate_ == 0) {
return true;
}
for (num = 0; num < NACK_BYTECOUNT_SIZE; ++num) {
if ((now - nack_byte_count_times_[num]) > avg_interval) {
// Don't use data older than 1sec.
break;
} else {
byte_count += nack_byte_count_[num];
}
}
WebRtc_Word32 time_interval = avg_interval;
if (num == NACK_BYTECOUNT_SIZE) {
// More than NACK_BYTECOUNT_SIZE nack messages has been received
// during the last msg_interval.
time_interval = now - nack_byte_count_times_[num - 1];
if (time_interval < 0) {
time_interval = avg_interval;
}
}
return (byte_count * 8) < (target_send_bitrate_ * time_interval);
}
void RTPSender::UpdateNACKBitRate(const WebRtc_UWord32 bytes,
const WebRtc_UWord32 now) {
CriticalSectionScoped cs(send_critsect_);
// Save bitrate statistics.
if (bytes > 0) {
if (now == 0) {
// Add padding length.
nack_byte_count_[0] += bytes;
} else {
if (nack_byte_count_times_[0] == 0) {
// First no shift.
} else {
// Shift.
for (int i = (NACK_BYTECOUNT_SIZE - 2); i >= 0; i--) {
nack_byte_count_[i + 1] = nack_byte_count_[i];
nack_byte_count_times_[i + 1] = nack_byte_count_times_[i];
}
}
nack_byte_count_[0] = bytes;
nack_byte_count_times_[0] = now;
}
}
}
void RTPSender::TimeToSendPacket(uint16_t sequence_number,
int64_t capture_time_ms) {
StorageType type;
uint16_t length = IP_PACKET_SIZE;
uint8_t data_buffer[IP_PACKET_SIZE];
int64_t stored_time_ms; // TODO(pwestin) can we deprecate this?
if (packet_history_ == NULL) {
return;
}
if (!packet_history_->GetRTPPacket(sequence_number, 0, data_buffer, &length,
&stored_time_ms, &type)) {
assert(false);
return;
}
assert(length > 0);
ModuleRTPUtility::RTPHeaderParser rtp_parser(data_buffer, length);
WebRtcRTPHeader rtp_header;
rtp_parser.Parse(rtp_header);
int64_t diff_ms = clock_->TimeInMilliseconds() - capture_time_ms;
if (UpdateTransmissionTimeOffset(data_buffer, length, rtp_header, diff_ms)) {
// Update stored packet in case of receiving a re-transmission request.
packet_history_->ReplaceRTPHeader(data_buffer,
rtp_header.header.sequenceNumber,
rtp_header.header.headerLength);
}
int bytes_sent = -1;
if (transport_) {
bytes_sent = transport_->SendPacket(id_, data_buffer, length);
}
if (bytes_sent <= 0) {
return;
}
// Update send statistics.
CriticalSectionScoped cs(send_critsect_);
Bitrate::Update(bytes_sent);
packets_sent_++;
if (bytes_sent > rtp_header.header.headerLength) {
payload_bytes_sent_ += bytes_sent - rtp_header.header.headerLength;
}
}
// TODO(pwestin): send in the RTPHeaderParser to avoid parsing it again.
WebRtc_Word32 RTPSender::SendToNetwork(
uint8_t *buffer, int payload_length, int rtp_header_length,
int64_t capture_time_ms, StorageType storage) {
ModuleRTPUtility::RTPHeaderParser rtp_parser(
buffer, payload_length + rtp_header_length);
WebRtcRTPHeader rtp_header;
rtp_parser.Parse(rtp_header);
// |capture_time_ms| <= 0 is considered invalid.
// TODO(holmer): This should be changed all over Video Engine so that negative
// time is consider invalid, while 0 is considered a valid time.
if (capture_time_ms > 0) {
int64_t time_now = clock_->TimeInMilliseconds();
UpdateTransmissionTimeOffset(buffer, payload_length + rtp_header_length,
rtp_header, time_now - capture_time_ms);
}
// Used for NACK and to spread out the transmission of packets.
if (packet_history_->PutRTPPacket(buffer, rtp_header_length + payload_length,
max_payload_length_, capture_time_ms,
storage) != 0) {
return -1;
}
if (paced_sender_) {
if (!paced_sender_->SendPacket(
PacedSender::kNormalPriority, rtp_header.header.ssrc,
rtp_header.header.sequenceNumber, capture_time_ms,
payload_length + rtp_header_length)) {
// We can't send the packet right now.
// We will be called when it is time.
return payload_length + rtp_header_length;
}
}
// Send packet.
WebRtc_Word32 bytes_sent = -1;
if (transport_) {
bytes_sent = transport_->SendPacket(id_, buffer,
payload_length + rtp_header_length);
}
if (bytes_sent <= 0) {
return -1;
}
// Update send statistics.
CriticalSectionScoped cs(send_critsect_);
Bitrate::Update(bytes_sent);
packets_sent_++;
if (bytes_sent > rtp_header_length) {
payload_bytes_sent_ += bytes_sent - rtp_header_length;
}
return 0;
}
void RTPSender::ProcessBitrate() {
CriticalSectionScoped cs(send_critsect_);
Bitrate::Process();
nack_bitrate_.Process();
if (audio_configured_) {
return;
}
video_->ProcessBitrate();
}
WebRtc_UWord16 RTPSender::RTPHeaderLength() const {
WebRtc_UWord16 rtp_header_length = 12;
if (include_csrcs_) {
rtp_header_length += sizeof(WebRtc_UWord32) * csrcs_;
}
rtp_header_length += RtpHeaderExtensionTotalLength();
return rtp_header_length;
}
WebRtc_UWord16 RTPSender::IncrementSequenceNumber() {
CriticalSectionScoped cs(send_critsect_);
return sequence_number_++;
}
void RTPSender::ResetDataCounters() {
packets_sent_ = 0;
payload_bytes_sent_ = 0;
}
WebRtc_UWord32 RTPSender::Packets() const {
// Don't use critsect to avoid potential deadlock.
return packets_sent_;
}
// Number of sent RTP bytes.
// Don't use critsect to avoid potental deadlock.
WebRtc_UWord32 RTPSender::Bytes() const {
return payload_bytes_sent_;
}
WebRtc_Word32 RTPSender::BuildRTPheader(
WebRtc_UWord8 *data_buffer, const WebRtc_Word8 payload_type,
const bool marker_bit, const WebRtc_UWord32 capture_time_stamp,
const bool time_stamp_provided, const bool inc_sequence_number) {
assert(payload_type >= 0);
CriticalSectionScoped cs(send_critsect_);
data_buffer[0] = static_cast<WebRtc_UWord8>(0x80); // version 2.
data_buffer[1] = static_cast<WebRtc_UWord8>(payload_type);
if (marker_bit) {
data_buffer[1] |= kRtpMarkerBitMask; // Marker bit is set.
}
if (time_stamp_provided) {
time_stamp_ = start_time_stamp_ + capture_time_stamp;
} else {
// Make a unique time stamp.
// We can't inc by the actual time, since then we increase the risk of back
// timing.
time_stamp_++;
}
ModuleRTPUtility::AssignUWord16ToBuffer(data_buffer + 2, sequence_number_);
ModuleRTPUtility::AssignUWord32ToBuffer(data_buffer + 4, time_stamp_);
ModuleRTPUtility::AssignUWord32ToBuffer(data_buffer + 8, ssrc_);
WebRtc_Word32 rtp_header_length = 12;
// Add the CSRCs if any.
if (include_csrcs_ && csrcs_ > 0) {
if (csrcs_ > kRtpCsrcSize) {
// error
assert(false);
return -1;
}
WebRtc_UWord8 *ptr = &data_buffer[rtp_header_length];
for (WebRtc_UWord32 i = 0; i < csrcs_; ++i) {
ModuleRTPUtility::AssignUWord32ToBuffer(ptr, csrc_[i]);
ptr += 4;
}
data_buffer[0] = (data_buffer[0] & 0xf0) | csrcs_;
// Update length of header.
rtp_header_length += sizeof(WebRtc_UWord32) * csrcs_;
}
sequence_number_++; // Prepare for next packet.
WebRtc_UWord16 len = BuildRTPHeaderExtension(data_buffer + rtp_header_length);
if (len) {
data_buffer[0] |= 0x10; // Set extension bit.
rtp_header_length += len;
}
return rtp_header_length;
}
WebRtc_UWord16 RTPSender::BuildRTPHeaderExtension(
WebRtc_UWord8 *data_buffer) const {
if (rtp_header_extension_map_.Size() <= 0) {
return 0;
}
// RTP header extension, RFC 3550.
// 0 1 2 3
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | defined by profile | length |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | header extension |
// | .... |
//
const WebRtc_UWord32 kPosLength = 2;
const WebRtc_UWord32 kHeaderLength = RTP_ONE_BYTE_HEADER_LENGTH_IN_BYTES;
// Add extension ID (0xBEDE).
ModuleRTPUtility::AssignUWord16ToBuffer(data_buffer,
RTP_ONE_BYTE_HEADER_EXTENSION);
// Add extensions.
WebRtc_UWord16 total_block_length = 0;
RTPExtensionType type = rtp_header_extension_map_.First();
while (type != kRtpExtensionNone) {
WebRtc_UWord8 block_length = 0;
if (type == kRtpExtensionTransmissionTimeOffset) {
block_length = BuildTransmissionTimeOffsetExtension(
data_buffer + kHeaderLength + total_block_length);
}
total_block_length += block_length;
type = rtp_header_extension_map_.Next(type);
}
if (total_block_length == 0) {
// No extension added.
return 0;
}
// Set header length (in number of Word32, header excluded).
assert(total_block_length % 4 == 0);
ModuleRTPUtility::AssignUWord16ToBuffer(data_buffer + kPosLength,
total_block_length / 4);
// Total added length.
return kHeaderLength + total_block_length;
}
WebRtc_UWord8 RTPSender::BuildTransmissionTimeOffsetExtension(
WebRtc_UWord8* data_buffer) const {
// From RFC 5450: Transmission Time Offsets in RTP Streams.
//
// The transmission time is signaled to the receiver in-band using the
// general mechanism for RTP header extensions [RFC5285]. The payload
// of this extension (the transmitted value) is a 24-bit signed integer.
// When added to the RTP timestamp of the packet, it represents the
// "effective" RTP transmission time of the packet, on the RTP
// timescale.
//
// The form of the transmission offset extension block:
//
// 0 1 2 3
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | ID | len=2 | transmission offset |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// Get id defined by user.
WebRtc_UWord8 id;
if (rtp_header_extension_map_.GetId(kRtpExtensionTransmissionTimeOffset,
&id) != 0) {
// Not registered.
return 0;
}
int pos = 0;
const WebRtc_UWord8 len = 2;
data_buffer[pos++] = (id << 4) + len;
ModuleRTPUtility::AssignUWord24ToBuffer(data_buffer + pos,
transmission_time_offset_);
pos += 3;
assert(pos == TRANSMISSION_TIME_OFFSET_LENGTH_IN_BYTES);
return TRANSMISSION_TIME_OFFSET_LENGTH_IN_BYTES;
}
bool RTPSender::UpdateTransmissionTimeOffset(
WebRtc_UWord8 *rtp_packet, const WebRtc_UWord16 rtp_packet_length,
const WebRtcRTPHeader &rtp_header, const WebRtc_Word64 time_diff_ms) const {
CriticalSectionScoped cs(send_critsect_);
// Get length until start of transmission block.
int transmission_block_pos =
rtp_header_extension_map_.GetLengthUntilBlockStartInBytes(
kRtpExtensionTransmissionTimeOffset);
if (transmission_block_pos < 0) {
WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
"Failed to update transmission time offset, not registered.");
return false;
}
int block_pos = 12 + rtp_header.header.numCSRCs + transmission_block_pos;
if (rtp_packet_length < block_pos + 4 ||
rtp_header.header.headerLength < block_pos + 4) {
WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
"Failed to update transmission time offset, invalid length.");
return false;
}
// Verify that header contains extension.
if (!((rtp_packet[12 + rtp_header.header.numCSRCs] == 0xBE) &&
(rtp_packet[12 + rtp_header.header.numCSRCs + 1] == 0xDE))) {
WEBRTC_TRACE(
kTraceStream, kTraceRtpRtcp, id_,
"Failed to update transmission time offset, hdr extension not found.");
return false;
}
// Get id.
WebRtc_UWord8 id = 0;
if (rtp_header_extension_map_.GetId(kRtpExtensionTransmissionTimeOffset,
&id) != 0) {
WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
"Failed to update transmission time offset, no id.");
return false;
}
// Verify first byte in block.
const WebRtc_UWord8 first_block_byte = (id << 4) + 2;
if (rtp_packet[block_pos] != first_block_byte) {
WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
"Failed to update transmission time offset.");
return false;
}
// Update transmission offset field.
ModuleRTPUtility::AssignUWord24ToBuffer(rtp_packet + block_pos + 1,
time_diff_ms * 90); // RTP timestamp.
return true;
}
void RTPSender::SetSendingStatus(const bool enabled) {
if (enabled) {
WebRtc_UWord32 frequency_hz;
if (audio_configured_) {
WebRtc_UWord32 frequency = audio_->AudioFrequency();
// sanity
switch (frequency) {
case 8000:
case 12000:
case 16000:
case 24000:
case 32000:
break;
default:
assert(false);
return;
}
frequency_hz = frequency;
} else {
frequency_hz = kDefaultVideoFrequency;
}
WebRtc_UWord32 RTPtime = ModuleRTPUtility::GetCurrentRTP(clock_,
frequency_hz);
// Will be ignored if it's already configured via API.
SetStartTimestamp(RTPtime, false);
} else {
if (!ssrc_forced_) {
// Generate a new SSRC.
ssrc_db_.ReturnSSRC(ssrc_);
ssrc_ = ssrc_db_.CreateSSRC(); // Can't be 0.
}
// Don't initialize seq number if SSRC passed externally.
if (!sequence_number_forced_ && !ssrc_forced_) {
// Generate a new sequence number.
sequence_number_ =
rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER); // NOLINT
}
}
}
void RTPSender::SetSendingMediaStatus(const bool enabled) {
CriticalSectionScoped cs(send_critsect_);
sending_media_ = enabled;
}
bool RTPSender::SendingMedia() const {
CriticalSectionScoped cs(send_critsect_);
return sending_media_;
}
WebRtc_UWord32 RTPSender::Timestamp() const {
CriticalSectionScoped cs(send_critsect_);
return time_stamp_;
}
void RTPSender::SetStartTimestamp(WebRtc_UWord32 timestamp, bool force) {
CriticalSectionScoped cs(send_critsect_);
if (force) {
start_time_stamp_forced_ = force;
start_time_stamp_ = timestamp;
} else {
if (!start_time_stamp_forced_) {
start_time_stamp_ = timestamp;
}
}
}
WebRtc_UWord32 RTPSender::StartTimestamp() const {
CriticalSectionScoped cs(send_critsect_);
return start_time_stamp_;
}
WebRtc_UWord32 RTPSender::GenerateNewSSRC() {
// If configured via API, return 0.
CriticalSectionScoped cs(send_critsect_);
if (ssrc_forced_) {
return 0;
}
ssrc_ = ssrc_db_.CreateSSRC(); // Can't be 0.
return ssrc_;
}
void RTPSender::SetSSRC(WebRtc_UWord32 ssrc) {
// This is configured via the API.
CriticalSectionScoped cs(send_critsect_);
if (ssrc_ == ssrc && ssrc_forced_) {
return; // Since it's same ssrc, don't reset anything.
}
ssrc_forced_ = true;
ssrc_db_.ReturnSSRC(ssrc_);
ssrc_db_.RegisterSSRC(ssrc);
ssrc_ = ssrc;
if (!sequence_number_forced_) {
sequence_number_ =
rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER); // NOLINT
}
}
WebRtc_UWord32 RTPSender::SSRC() const {
CriticalSectionScoped cs(send_critsect_);
return ssrc_;
}
void RTPSender::SetCSRCStatus(const bool include) {
include_csrcs_ = include;
}
void RTPSender::SetCSRCs(const WebRtc_UWord32 arr_of_csrc[kRtpCsrcSize],
const WebRtc_UWord8 arr_length) {
assert(arr_length <= kRtpCsrcSize);
CriticalSectionScoped cs(send_critsect_);
for (int i = 0; i < arr_length; i++) {
csrc_[i] = arr_of_csrc[i];
}
csrcs_ = arr_length;
}
WebRtc_Word32 RTPSender::CSRCs(WebRtc_UWord32 arr_of_csrc[kRtpCsrcSize]) const {
assert(arr_of_csrc);
CriticalSectionScoped cs(send_critsect_);
for (int i = 0; i < csrcs_ && i < kRtpCsrcSize; i++) {
arr_of_csrc[i] = csrc_[i];
}
return csrcs_;
}
void RTPSender::SetSequenceNumber(WebRtc_UWord16 seq) {
CriticalSectionScoped cs(send_critsect_);
sequence_number_forced_ = true;
sequence_number_ = seq;
}
WebRtc_UWord16 RTPSender::SequenceNumber() const {
CriticalSectionScoped cs(send_critsect_);
return sequence_number_;
}
// Audio.
WebRtc_Word32 RTPSender::SendTelephoneEvent(const WebRtc_UWord8 key,
const WebRtc_UWord16 time_ms,
const WebRtc_UWord8 level) {
if (!audio_configured_) {
return -1;
}
return audio_->SendTelephoneEvent(key, time_ms, level);
}
bool RTPSender::SendTelephoneEventActive(WebRtc_Word8 *telephone_event) const {
if (!audio_configured_) {
return false;
}
return audio_->SendTelephoneEventActive(*telephone_event);
}
WebRtc_Word32 RTPSender::SetAudioPacketSize(
const WebRtc_UWord16 packet_size_samples) {
if (!audio_configured_) {
return -1;
}
return audio_->SetAudioPacketSize(packet_size_samples);
}
WebRtc_Word32 RTPSender::SetAudioLevelIndicationStatus(const bool enable,
const WebRtc_UWord8 ID) {
if (!audio_configured_) {
return -1;
}
return audio_->SetAudioLevelIndicationStatus(enable, ID);
}
WebRtc_Word32 RTPSender::AudioLevelIndicationStatus(bool *enable,
WebRtc_UWord8* id) const {
return audio_->AudioLevelIndicationStatus(*enable, *id);
}
WebRtc_Word32 RTPSender::SetAudioLevel(const WebRtc_UWord8 level_d_bov) {
return audio_->SetAudioLevel(level_d_bov);
}
WebRtc_Word32 RTPSender::SetRED(const WebRtc_Word8 payload_type) {
if (!audio_configured_) {
return -1;
}
return audio_->SetRED(payload_type);
}
WebRtc_Word32 RTPSender::RED(WebRtc_Word8 *payload_type) const {
if (!audio_configured_) {
return -1;
}
return audio_->RED(*payload_type);
}
// Video
VideoCodecInformation *RTPSender::CodecInformationVideo() {
if (audio_configured_) {
return NULL;
}
return video_->CodecInformationVideo();
}
RtpVideoCodecTypes RTPSender::VideoCodecType() const {
if (audio_configured_) {
return kRtpNoVideo;
}
return video_->VideoCodecType();
}
WebRtc_UWord32 RTPSender::MaxConfiguredBitrateVideo() const {
if (audio_configured_) {
return 0;
}
return video_->MaxConfiguredBitrateVideo();
}
WebRtc_Word32 RTPSender::SendRTPIntraRequest() {
if (audio_configured_) {
return -1;
}
return video_->SendRTPIntraRequest();
}
WebRtc_Word32 RTPSender::SetGenericFECStatus(
const bool enable, const WebRtc_UWord8 payload_type_red,
const WebRtc_UWord8 payload_type_fec) {
if (audio_configured_) {
return -1;
}
return video_->SetGenericFECStatus(enable, payload_type_red,
payload_type_fec);
}
WebRtc_Word32 RTPSender::GenericFECStatus(
bool *enable, WebRtc_UWord8 *payload_type_red,
WebRtc_UWord8 *payload_type_fec) const {
if (audio_configured_) {
return -1;
}
return video_->GenericFECStatus(
*enable, *payload_type_red, *payload_type_fec);
}
WebRtc_Word32 RTPSender::SetFecParameters(
const FecProtectionParams *delta_params,
const FecProtectionParams *key_params) {
if (audio_configured_) {
return -1;
}
return video_->SetFecParameters(delta_params, key_params);
}
} // namespace webrtc