#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"
+#include "webrtc/system_wrappers/interface/logging.h"
#include "webrtc/system_wrappers/interface/trace_event.h"
namespace webrtc {
audio_ = new RTPSenderAudio(id, clock_, this);
audio_->RegisterAudioCallback(audio_feedback);
} else {
- video_ = new RTPSenderVideo(id, clock_, this);
+ video_ = new RTPSenderVideo(clock_, this);
}
- WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
}
RTPSender::~RTPSender() {
}
delete audio_;
delete video_;
-
- WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id_, "%s deleted", __FUNCTION__);
}
void RTPSender::SetTargetSendBitrate(const uint32_t bits) {
const uint16_t 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__);
+ LOG(LS_ERROR) << "Invalid max payload length: " << max_payload_length;
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;
}
CriticalSectionScoped cs(send_critsect_);
if (payload_type < 0) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_, "\tinvalid payload_type (%d)",
- payload_type);
+ LOG(LS_ERROR) << "Invalid payload_type " << payload_type;
return -1;
}
if (audio_configured_) {
std::map<int8_t, 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);
+ LOG(LS_WARNING) << "Payload type " << payload_type << " not registered.";
return -1;
}
payload_type_ = payload_type;
}
RtpVideoCodecTypes video_type = kRtpVideoGeneric;
if (CheckPayloadType(payload_type, &video_type) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
- "%s invalid argument failed to find payload_type:%d",
- __FUNCTION__, payload_type);
+ LOG(LS_ERROR) << "Don't send data with unknown payload type.";
return -1;
}
RTPHeader header;
if (!rtp_parser.Parse(header)) {
assert(false);
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
- "Failed to parse RTP header of packet to be retransmitted.");
return -1;
}
if (!paced_sender_->SendPacket(PacedSender::kHighPriority,
}
TRACE_EVENT_INSTANT2("webrtc_rtp", "RTPSender::SendPacketToNetwork",
"size", size, "sent", bytes_sent);
- // TODO(pwesin): Add a separate bitrate for sent bitrate after pacer.
+ // TODO(pwestin): Add a separate bitrate for sent bitrate after pacer.
if (bytes_sent <= 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_,
- "Transport failed to send packet");
+ LOG(LS_WARNING) << "Transport failed to send packet";
return false;
}
return true;
// Enough bandwidth to send NACK?
if (!ProcessNACKBitRate(now)) {
- WEBRTC_TRACE(kTraceStream,
- kTraceRtpRtcp,
- id_,
- "NACK bitrate reached. Skip sending NACK response. Target %d",
- target_bitrate_kbps);
+ LOG(LS_INFO) << "NACK bitrate reached. Skip sending NACK response. Target "
+ << target_bitrate_kbps;
return;
}
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);
+ LOG(LS_WARNING) << "Failed resending RTP packet " << *it
+ << ", Discard rest of packets";
break;
}
// Delay bandwidth estimate (RTT * BW).
uint8_t *rtp_packet, const uint16_t rtp_packet_length,
const RTPHeader &rtp_header, const int64_t time_diff_ms) const {
CriticalSectionScoped cs(send_critsect_);
-
+ // Get id.
+ uint8_t id = 0;
+ if (rtp_header_extension_map_.GetId(kRtpExtensionTransmissionTimeOffset,
+ &id) != 0) {
+ // Not registered.
+ return false;
+ }
// Get length until start of header extension block.
int extension_block_pos =
rtp_header_extension_map_.GetLengthUntilBlockStartInBytes(
kRtpExtensionTransmissionTimeOffset);
if (extension_block_pos < 0) {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
- "Failed to update transmission time offset, not registered.");
+ LOG(LS_WARNING)
+ << "Failed to update transmission time offset, not registered.";
return false;
}
int block_pos = 12 + rtp_header.numCSRCs + extension_block_pos;
if (rtp_packet_length < block_pos + kTransmissionTimeOffsetLength ||
rtp_header.headerLength <
block_pos + kTransmissionTimeOffsetLength) {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
- "Failed to update transmission time offset, invalid length.");
+ LOG(LS_WARNING)
+ << "Failed to update transmission time offset, invalid length.";
return false;
}
// Verify that header contains extension.
if (!((rtp_packet[12 + rtp_header.numCSRCs] == 0xBE) &&
(rtp_packet[12 + rtp_header.numCSRCs + 1] == 0xDE))) {
- WEBRTC_TRACE(
- kTraceStream, kTraceRtpRtcp, id_,
- "Failed to update transmission time offset, hdr extension not found.");
- return false;
- }
- // Get id.
- uint8_t id = 0;
- if (rtp_header_extension_map_.GetId(kRtpExtensionTransmissionTimeOffset,
- &id) != 0) {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
- "Failed to update transmission time offset, no id.");
+ LOG(LS_WARNING) << "Failed to update transmission time offset, hdr "
+ "extension not found.";
return false;
}
// Verify first byte in block.
const uint8_t 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.");
+ LOG(LS_WARNING) << "Failed to update transmission time offset.";
return false;
}
// Update transmission offset field (converting to a 90 kHz timestamp).
const uint8_t dBov) const {
CriticalSectionScoped cs(send_critsect_);
+ // Get id.
+ uint8_t id = 0;
+ if (rtp_header_extension_map_.GetId(kRtpExtensionAudioLevel, &id) != 0) {
+ // Not registered.
+ return false;
+ }
// Get length until start of header extension block.
int extension_block_pos =
rtp_header_extension_map_.GetLengthUntilBlockStartInBytes(
kRtpExtensionAudioLevel);
if (extension_block_pos < 0) {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
- "Failed to update audio level, not registered.");
+ // The feature is not enabled.
return false;
}
int block_pos = 12 + rtp_header.numCSRCs + extension_block_pos;
if (rtp_packet_length < block_pos + kAudioLevelLength ||
rtp_header.headerLength < block_pos + kAudioLevelLength) {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
- "Failed to update audio level, invalid length.");
+ LOG(LS_WARNING) << "Failed to update audio level, invalid length.";
return false;
}
// Verify that header contains extension.
if (!((rtp_packet[12 + rtp_header.numCSRCs] == 0xBE) &&
(rtp_packet[12 + rtp_header.numCSRCs + 1] == 0xDE))) {
- WEBRTC_TRACE(
- kTraceStream, kTraceRtpRtcp, id_,
- "Failed to update audio level, hdr extension not found.");
- return false;
- }
- // Get id.
- uint8_t id = 0;
- if (rtp_header_extension_map_.GetId(kRtpExtensionAudioLevel, &id) != 0) {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
- "Failed to update audio level, no id.");
+ LOG(LS_WARNING) << "Failed to update audio level, hdr extension not found.";
return false;
}
// Verify first byte in block.
const uint8_t first_block_byte = (id << 4) + 0;
if (rtp_packet[block_pos] != first_block_byte) {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
- "Failed to update audio level.");
+ LOG(LS_WARNING) << "Failed to update audio level.";
return false;
}
rtp_packet[block_pos + 1] = (is_voiced ? 0x80 : 0x00) + (dBov & 0x7f);
const RTPHeader &rtp_header, const int64_t now_ms) const {
CriticalSectionScoped cs(send_critsect_);
+ // Get id.
+ uint8_t id = 0;
+ if (rtp_header_extension_map_.GetId(kRtpExtensionAbsoluteSendTime,
+ &id) != 0) {
+ // Not registered.
+ return false;
+ }
// Get length until start of header extension block.
int extension_block_pos =
rtp_header_extension_map_.GetLengthUntilBlockStartInBytes(
kRtpExtensionAbsoluteSendTime);
if (extension_block_pos < 0) {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
- "Failed to update absolute send time, not registered.");
+ // The feature is not enabled.
return false;
}
int block_pos = 12 + rtp_header.numCSRCs + extension_block_pos;
if (rtp_packet_length < block_pos + kAbsoluteSendTimeLength ||
rtp_header.headerLength < block_pos + kAbsoluteSendTimeLength) {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
- "Failed to update absolute send time, invalid length.");
+ LOG(LS_WARNING) << "Failed to update absolute send time, invalid length.";
return false;
}
// Verify that header contains extension.
if (!((rtp_packet[12 + rtp_header.numCSRCs] == 0xBE) &&
(rtp_packet[12 + rtp_header.numCSRCs + 1] == 0xDE))) {
- WEBRTC_TRACE(
- kTraceStream, kTraceRtpRtcp, id_,
- "Failed to update absolute send time, hdr extension not found.");
- return false;
- }
- // Get id.
- uint8_t id = 0;
- if (rtp_header_extension_map_.GetId(kRtpExtensionAbsoluteSendTime,
- &id) != 0) {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
- "Failed to update absolute send time, no id.");
+ LOG(LS_WARNING)
+ << "Failed to update absolute send time, hdr extension not found.";
return false;
}
// Verify first byte in block.
const uint8_t first_block_byte = (id << 4) + 2;
if (rtp_packet[block_pos] != first_block_byte) {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
- "Failed to update absolute send time.");
+ LOG(LS_WARNING) << "Failed to update absolute send time.";
return false;
}
// Update absolute send time field (convert ms to 24-bit unsigned with 18 bit
projects / platform / framework / web / crosswalk.git / blobdiff