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Diffstat (limited to 'webrtc/modules/rtp_rtcp/source/rtp_sender.cc')
-rw-r--r--webrtc/modules/rtp_rtcp/source/rtp_sender.cc63
1 files changed, 33 insertions, 30 deletions
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 50f476829d..f4933afdd9 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -11,15 +11,17 @@
#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
#include <stdlib.h> // srand
+#include <algorithm>
#include <utility>
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/trace_event.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_cvo.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
+#include "webrtc/modules/rtp_rtcp/source/time_util.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/tick_util.h"
@@ -33,6 +35,7 @@ static const uint32_t kAbsSendTimeFraction = 18;
namespace {
const size_t kRtpHeaderLength = 12;
+const uint16_t kMaxInitRtpSeqNumber = 32767; // 2^15 -1.
const char* FrameTypeToString(FrameType frame_type) {
switch (frame_type) {
@@ -125,6 +128,7 @@ RTPSender::RTPSender(
// TickTime.
clock_delta_ms_(clock_->TimeInMilliseconds() -
TickTime::MillisecondTimestamp()),
+ random_(clock_->TimeInMicroseconds()),
bitrates_(new BitrateAggregator(bitrate_callback)),
total_bitrate_sent_(clock, bitrates_->total_bitrate_observer()),
audio_configured_(audio),
@@ -182,8 +186,8 @@ RTPSender::RTPSender(
ssrc_rtx_ = ssrc_db_.CreateSSRC(); // Can't be 0.
bitrates_->set_ssrc(ssrc_);
// Random start, 16 bits. Can't be 0.
- sequence_number_rtx_ = static_cast<uint16_t>(rand() + 1) & 0x7FFF;
- sequence_number_ = static_cast<uint16_t>(rand() + 1) & 0x7FFF;
+ sequence_number_rtx_ = random_.Rand(1, kMaxInitRtpSeqNumber);
+ sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
}
RTPSender::~RTPSender() {
@@ -292,7 +296,7 @@ int32_t RTPSender::RegisterPayload(
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
int8_t payload_number,
uint32_t frequency,
- uint8_t channels,
+ size_t channels,
uint32_t rate) {
assert(payload_name);
CriticalSectionScoped cs(send_critsect_.get());
@@ -323,11 +327,11 @@ int32_t RTPSender::RegisterPayload(
return -1;
}
int32_t ret_val = 0;
- RtpUtility::Payload* payload = NULL;
+ RtpUtility::Payload* payload = nullptr;
if (audio_configured_) {
// TODO(mflodman): Change to CreateAudioPayload and make static.
ret_val = audio_->RegisterAudioPayload(payload_name, payload_number,
- frequency, channels, rate, payload);
+ frequency, channels, rate, &payload);
} else {
payload = video_->CreateVideoPayload(payload_name, payload_number, rate);
}
@@ -452,7 +456,7 @@ int32_t RTPSender::CheckPayloadType(int8_t payload_type,
}
if (audio_configured_) {
int8_t red_pl_type = -1;
- if (audio_->RED(red_pl_type) == 0) {
+ if (audio_->RED(&red_pl_type) == 0) {
// We have configured RED.
if (red_pl_type == payload_type) {
// And it's a match...
@@ -469,7 +473,8 @@ int32_t RTPSender::CheckPayloadType(int8_t payload_type,
std::map<int8_t, RtpUtility::Payload*>::iterator it =
payload_type_map_.find(payload_type);
if (it == payload_type_map_.end()) {
- LOG(LS_WARNING) << "Payload type " << payload_type << " not registered.";
+ LOG(LS_WARNING) << "Payload type " << static_cast<int>(payload_type)
+ << " not registered.";
return -1;
}
SetSendPayloadType(payload_type);
@@ -512,7 +517,8 @@ int32_t RTPSender::SendOutgoingData(FrameType frame_type,
}
RtpVideoCodecTypes video_type = kRtpVideoGeneric;
if (CheckPayloadType(payload_type, &video_type) != 0) {
- LOG(LS_ERROR) << "Don't send data with unknown payload type.";
+ LOG(LS_ERROR) << "Don't send data with unknown payload type: "
+ << static_cast<int>(payload_type) << ".";
return -1;
}
@@ -573,7 +579,7 @@ size_t RTPSender::TrySendRedundantPayloads(size_t bytes_to_send) {
break;
RtpUtility::RtpHeaderParser rtp_parser(buffer, length);
RTPHeader rtp_header;
- rtp_parser.Parse(rtp_header);
+ rtp_parser.Parse(&rtp_header);
bytes_left -= static_cast<int>(length - rtp_header.headerLength);
}
return bytes_to_send - bytes_left;
@@ -583,8 +589,7 @@ void RTPSender::BuildPaddingPacket(uint8_t* packet,
size_t header_length,
size_t padding_length) {
packet[0] |= 0x20; // Set padding bit.
- int32_t *data =
- reinterpret_cast<int32_t *>(&(packet[header_length]));
+ int32_t* data = reinterpret_cast<int32_t*>(&(packet[header_length]));
// Fill data buffer with random data.
for (size_t j = 0; j < (padding_length >> 2); ++j) {
@@ -665,7 +670,7 @@ size_t RTPSender::SendPadData(size_t bytes,
RtpUtility::RtpHeaderParser rtp_parser(padding_packet, length);
RTPHeader rtp_header;
- rtp_parser.Parse(rtp_header);
+ rtp_parser.Parse(&rtp_header);
if (capture_time_ms > 0) {
UpdateTransmissionTimeOffset(
@@ -717,7 +722,7 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, int64_t min_resend_time) {
if (paced_sender_) {
RtpUtility::RtpHeaderParser rtp_parser(data_buffer, length);
RTPHeader header;
- if (!rtp_parser.Parse(header)) {
+ if (!rtp_parser.Parse(&header)) {
assert(false);
return -1;
}
@@ -725,7 +730,7 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, int64_t min_resend_time) {
// TickTime.
int64_t corrected_capture_tims_ms = capture_time_ms + clock_delta_ms_;
paced_sender_->InsertPacket(
- RtpPacketSender::kHighPriority, header.ssrc, header.sequenceNumber,
+ RtpPacketSender::kNormalPriority, header.ssrc, header.sequenceNumber,
corrected_capture_tims_ms, length - header.headerLength, true);
return length;
@@ -903,11 +908,11 @@ bool RTPSender::PrepareAndSendPacket(uint8_t* buffer,
int64_t capture_time_ms,
bool send_over_rtx,
bool is_retransmit) {
- uint8_t *buffer_to_send_ptr = buffer;
+ uint8_t* buffer_to_send_ptr = buffer;
RtpUtility::RtpHeaderParser rtp_parser(buffer, length);
RTPHeader rtp_header;
- rtp_parser.Parse(rtp_header);
+ rtp_parser.Parse(&rtp_header);
if (!is_retransmit && rtp_header.markerBit) {
TRACE_EVENT_ASYNC_END0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "PacedSend",
capture_time_ms);
@@ -996,14 +1001,14 @@ bool RTPSender::IsFecPacket(const uint8_t* buffer,
bool fec_enabled;
uint8_t pt_red;
uint8_t pt_fec;
- video_->GenericFECStatus(fec_enabled, pt_red, pt_fec);
+ video_->GenericFECStatus(&fec_enabled, &pt_red, &pt_fec);
return fec_enabled &&
header.payloadType == pt_red &&
buffer[header.headerLength] == pt_fec;
}
size_t RTPSender::TimeToSendPadding(size_t bytes) {
- if (bytes == 0)
+ if (audio_configured_ || bytes == 0)
return 0;
{
CriticalSectionScoped cs(send_critsect_.get());
@@ -1026,7 +1031,7 @@ int32_t RTPSender::SendToNetwork(uint8_t* buffer,
RtpUtility::RtpHeaderParser rtp_parser(buffer,
payload_length + rtp_header_length);
RTPHeader rtp_header;
- rtp_parser.Parse(rtp_header);
+ rtp_parser.Parse(&rtp_header);
int64_t now_ms = clock_->TimeInMilliseconds();
@@ -1169,7 +1174,7 @@ size_t RTPSender::CreateRtpHeader(uint8_t* header,
int32_t rtp_header_length = kRtpHeaderLength;
if (csrcs.size() > 0) {
- uint8_t *ptr = &header[rtp_header_length];
+ uint8_t* ptr = &header[rtp_header_length];
for (size_t i = 0; i < csrcs.size(); ++i) {
ByteWriter<uint32_t>::WriteBigEndian(ptr, csrcs[i]);
ptr += 4;
@@ -1638,7 +1643,7 @@ uint16_t RTPSender::UpdateTransportSequenceNumber(
void RTPSender::SetSendingStatus(bool enabled) {
if (enabled) {
uint32_t frequency_hz = SendPayloadFrequency();
- uint32_t RTPtime = RtpUtility::GetCurrentRTP(clock_, frequency_hz);
+ uint32_t RTPtime = CurrentRtp(*clock_, frequency_hz);
// Will be ignored if it's already configured via API.
SetStartTimestamp(RTPtime, false);
@@ -1653,8 +1658,7 @@ void RTPSender::SetSendingStatus(bool enabled) {
// 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
+ sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
}
}
}
@@ -1716,8 +1720,7 @@ void RTPSender::SetSSRC(uint32_t ssrc) {
ssrc_ = ssrc;
bitrates_->set_ssrc(ssrc_);
if (!sequence_number_forced_) {
- sequence_number_ =
- rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER); // NOLINT
+ sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
}
}
@@ -1775,7 +1778,7 @@ int32_t RTPSender::RED(int8_t *payload_type) const {
if (!audio_configured_) {
return -1;
}
- return audio_->RED(*payload_type);
+ return audio_->RED(payload_type);
}
RtpVideoCodecTypes RTPSender::VideoCodecType() const {
@@ -1801,7 +1804,7 @@ void RTPSender::GenericFECStatus(bool* enable,
uint8_t* payload_type_red,
uint8_t* payload_type_fec) const {
RTC_DCHECK(!audio_configured_);
- video_->GenericFECStatus(*enable, *payload_type_red, *payload_type_fec);
+ video_->GenericFECStatus(enable, payload_type_red, payload_type_fec);
}
int32_t RTPSender::SetFecParameters(
@@ -1823,7 +1826,7 @@ void RTPSender::BuildRtxPacket(uint8_t* buffer, size_t* length,
reinterpret_cast<const uint8_t*>(buffer), *length);
RTPHeader rtp_header;
- rtp_parser.Parse(rtp_header);
+ rtp_parser.Parse(&rtp_header);
// Add original RTP header.
memcpy(data_buffer_rtx, buffer, rtp_header.headerLength);
@@ -1836,7 +1839,7 @@ void RTPSender::BuildRtxPacket(uint8_t* buffer, size_t* length,
}
// Replace sequence number.
- uint8_t *ptr = data_buffer_rtx + 2;
+ uint8_t* ptr = data_buffer_rtx + 2;
ByteWriter<uint16_t>::WriteBigEndian(ptr, sequence_number_rtx_++);
// Replace SSRC.
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-13 16:25:32 +0000