admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Change to LOG(...) logging in most of voice_engine/channel.cc

Browse Source 
First patch set runs a script to directly convert log statements.
Second patch sets manually fixes smaller errors.

Due to the size of this change, the remaining WEBRTC_TRACE statements will
be handled in a different CL.

Bug: webrtc:5118
Change-Id: Ic39c3a6310a2b461b47a7b4757210d98637e8acd
Reviewed-on: https://webrtc-review.googlesource.com/1228
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20091}
This commit is contained in:
Sam Zackrisson 2017-10-02 14:32:33 +02:00 committed by Commit Bot
parent 978b876fd2
commit ecc51e96db
1 changed files with 27 additions and 134 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

161
voice_engine/channel.cc
View File

@ -42,7 +42,6 @@
#include "rtc_base/timeutils.h"
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/metrics.h"
#include "system_wrappers/include/trace.h"
#include "voice_engine/utility.h"
namespace webrtc {
@ -431,11 +430,6 @@ int32_t Channel::SendData(FrameType frameType,
size_t payloadSize,
const RTPFragmentationHeader* fragmentation) {
RTC_DCHECK_RUN_ON(encoder_queue_);
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SendData(frameType=%u, payloadType=%u, timeStamp=%u,"
" payloadSize=%" PRIuS ", fragmentation=0x%x)",
frameType, payloadType, timeStamp, payloadSize, fragmentation);
if (_includeAudioLevelIndication) {
// Store current audio level in the RTP/RTCP module.
// The level will be used in combination with voice-activity state
@ -463,15 +457,11 @@ int32_t Channel::SendData(FrameType frameType,
bool Channel::SendRtp(const uint8_t* data,
size_t len,
const PacketOptions& options) {
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SendPacket(channel=%d, len=%" PRIuS ")", len);
rtc::CritScope cs(&_callbackCritSect);
if (_transportPtr == NULL) {
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SendPacket() failed to send RTP packet due to"
" invalid transport object");
LOG(LS_ERROR) << "Channel::SendPacket() failed to send RTP packet due to"
<< " invalid transport object";
return false;
}
@ -486,14 +476,10 @@ bool Channel::SendRtp(const uint8_t* data,
}
bool Channel::SendRtcp(const uint8_t* data, size_t len) {
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SendRtcp(len=%" PRIuS ")", len);
rtc::CritScope cs(&_callbackCritSect);
if (_transportPtr == NULL) {
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SendRtcp() failed to send RTCP packet"
" due to invalid transport object");
LOG(LS_ERROR) << "Channel::SendRtcp() failed to send RTCP packet due to"
<< " invalid transport object";
return false;
}
@ -509,17 +495,12 @@ bool Channel::SendRtcp(const uint8_t* data, size_t len) {
}
void Channel::OnIncomingSSRCChanged(uint32_t ssrc) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::OnIncomingSSRCChanged(SSRC=%d)", ssrc);
// Update ssrc so that NTP for AV sync can be updated.
_rtpRtcpModule->SetRemoteSSRC(ssrc);
}
void Channel::OnIncomingCSRCChanged(uint32_t CSRC, bool added) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::OnIncomingCSRCChanged(CSRC=%d, added=%d)", CSRC,
added);
// TODO(saza): remove.
}
int32_t Channel::OnInitializeDecoder(
@ -528,11 +509,6 @@ int32_t Channel::OnInitializeDecoder(
int frequency,
size_t channels,
uint32_t rate) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::OnInitializeDecoder(payloadType=%d, "
"payloadName=%s, frequency=%u, channels=%" PRIuS ", rate=%u)",
payloadType, payloadName, frequency, channels, rate);
CodecInst receiveCodec = {0};
CodecInst dummyCodec = {0};
@ -548,10 +524,9 @@ int32_t Channel::OnInitializeDecoder(
// Register the new codec to the ACM
if (!audio_coding_->RegisterReceiveCodec(receiveCodec.pltype,
CodecInstToSdp(receiveCodec))) {
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::OnInitializeDecoder() invalid codec ("
"pt=%d, name=%s) received - 1",
payloadType, payloadName);
LOG(LS_WARNING) << "Channel::OnInitializeDecoder() invalid codec (pt="
<< payloadType << ", name=" << payloadName
<< ") received - 1";
return -1;
}
@ -561,19 +536,9 @@ int32_t Channel::OnInitializeDecoder(
int32_t Channel::OnReceivedPayloadData(const uint8_t* payloadData,
size_t payloadSize,
const WebRtcRTPHeader* rtpHeader) {
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::OnReceivedPayloadData(payloadSize=%" PRIuS
","
" payloadType=%u, audioChannel=%" PRIuS ")",
payloadSize, rtpHeader->header.payloadType,
rtpHeader->type.Audio.channel);
if (!channel_state_.Get().playing) {
// Avoid inserting into NetEQ when we are not playing. Count the
// packet as discarded.
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
"received packet is discarded since playing is not"
" activated");
return 0;
}
@ -602,8 +567,7 @@ bool Channel::OnRecoveredPacket(const uint8_t* rtp_packet,
size_t rtp_packet_length) {
RTPHeader header;
if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, &header)) {
WEBRTC_TRACE(kTraceDebug, webrtc::kTraceVoice, _channelId,
"IncomingPacket invalid RTP header");
LOG(LS_WARNING) << "IncomingPacket invalid RTP header";
return false;
}
header.payload_type_frequency =
@ -625,8 +589,7 @@ AudioMixer::Source::AudioFrameInfo Channel::GetAudioFrameWithInfo(
bool muted;
if (audio_coding_->PlayoutData10Ms(audio_frame->sample_rate_hz_, audio_frame,
&muted) == -1) {
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::GetAudioFrame() PlayoutData10Ms() failed!");
LOG(LS_ERROR) << "Channel::GetAudioFrame() PlayoutData10Ms() failed!";
// In all likelihood, the audio in this frame is garbage. We return an
// error so that the audio mixer module doesn't add it to the mix. As
// a result, it won't be played out and the actions skipped here are
@ -721,15 +684,9 @@ int32_t Channel::CreateChannel(Channel*& channel,
int32_t channelId,
uint32_t instanceId,
const VoEBase::ChannelConfig& config) {
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, channelId),
"Channel::CreateChannel(channelId=%d, instanceId=%d)", channelId,
instanceId);
channel = new Channel(channelId, instanceId, config);
if (channel == NULL) {
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, channelId),
"Channel::CreateChannel() unable to allocate memory for"
" channel");
LOG(LS_ERROR) << "unable to allocate memory for new channel";
return -1;
}
return 0;
@ -783,8 +740,6 @@ Channel::Channel(int32_t channelId,
decoder_factory_(config.acm_config.decoder_factory),
use_twcc_plr_for_ana_(
webrtc::field_trial::FindFullName("UseTwccPlrForAna") == "Enabled") {
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::Channel() - ctor");
AudioCodingModule::Config acm_config(config.acm_config);
acm_config.neteq_config.enable_muted_state = true;
audio_coding_.reset(AudioCodingModule::Create(acm_config));
@ -819,16 +774,13 @@ Channel::~Channel() {
int32_t Channel::Init() {
RTC_DCHECK(construction_thread_.CalledOnValidThread());
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::Init()");
channel_state_.Reset();
// --- Initial sanity
if (_moduleProcessThreadPtr == NULL) {
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::Init() must call SetEngineInformation() first");
LOG(LS_ERROR) << "Channel::Init() must call SetEngineInformation() first";
return -1;
}
@ -865,9 +817,6 @@ int32_t Channel::Init() {
void Channel::Terminate() {
RTC_DCHECK(construction_thread_.CalledOnValidThread());
// Must be called on the same thread as Init().
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::Terminate");
rtp_receive_statistics_->RegisterRtcpStatisticsCallback(NULL);
StopSend();
@ -878,9 +827,8 @@ void Channel::Terminate() {
// 2. De-register modules in process thread
// 3. Destroy modules
if (audio_coding_->RegisterTransportCallback(NULL) == -1) {
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
"Terminate() failed to de-register transport callback"
" (Audio coding module)");
LOG(LS_WARNING) << "Terminate() failed to de-register transport callback"
<< " (Audio coding module)";
}
// De-register modules in process thread
@ -895,8 +843,6 @@ int32_t Channel::SetEngineInformation(ProcessThread& moduleProcessThread,
rtc::TaskQueue* encoder_queue) {
RTC_DCHECK(encoder_queue);
RTC_DCHECK(!encoder_queue_);
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SetEngineInformation()");
_moduleProcessThreadPtr = &moduleProcessThread;
_audioDeviceModulePtr = &audioDeviceModule;
encoder_queue_ = encoder_queue;
@ -914,8 +860,6 @@ Channel::GetAudioDecoderFactory() const {
}
int32_t Channel::StartPlayout() {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::StartPlayout()");
if (channel_state_.Get().playing) {
return 0;
}
@ -926,8 +870,6 @@ int32_t Channel::StartPlayout() {
}
int32_t Channel::StopPlayout() {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::StopPlayout()");
if (!channel_state_.Get().playing) {
return 0;
}
@ -939,8 +881,6 @@ int32_t Channel::StopPlayout() {
}
int32_t Channel::StartSend() {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::StartSend()");
if (channel_state_.Get().sending) {
return 0;
}
@ -968,8 +908,6 @@ int32_t Channel::StartSend() {
}
void Channel::StopSend() {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::StopSend()");
if (!channel_state_.Get().sending) {
return;
}
@ -1041,9 +979,8 @@ bool Channel::SetEncoder(int payload_type,
if (_rtpRtcpModule->RegisterSendPayload(rtp_codec) != 0) {
_rtpRtcpModule->DeRegisterSendPayload(payload_type);
if (_rtpRtcpModule->RegisterSendPayload(rtp_codec) != 0) {
WEBRTC_TRACE(
kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
"SetEncoder() failed to register codec to RTP/RTCP module");
LOG(LS_ERROR)
<< "SetEncoder() failed to register codec to RTP/RTCP module";
return false;
}
}
@ -1077,22 +1014,17 @@ int32_t Channel::GetRecCodec(CodecInst& codec) {
}
int32_t Channel::SetSendCodec(const CodecInst& codec) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SetSendCodec()");
if (!codec_manager_.RegisterEncoder(codec) ||
!codec_manager_.MakeEncoder(&rent_a_codec_, audio_coding_.get())) {
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
"SetSendCodec() failed to register codec to ACM");
LOG(LS_ERROR) << "SetSendCodec() failed to register codec to ACM";
return -1;
}
if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) {
_rtpRtcpModule->DeRegisterSendPayload(codec.pltype);
if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) {
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
"SetSendCodec() failed to register codec to"
" RTP/RTCP module");
LOG(LS_ERROR)
<< "SetSendCodec() failed to register codec to RTP/RTCP module";
return -1;
}
}
@ -1103,8 +1035,6 @@ int32_t Channel::SetSendCodec(const CodecInst& codec) {
}
void Channel::SetBitRate(int bitrate_bps, int64_t probing_interval_ms) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SetBitRate(bitrate_bps=%d)", bitrate_bps);
audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
if (*encoder) {
(*encoder)->OnReceivedUplinkBandwidth(
@ -1183,9 +1113,6 @@ void Channel::RegisterTransport(Transport* transport) {
}
void Channel::OnRtpPacket(const RtpPacketReceived& packet) {
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::OnRtpPacket()");
RTPHeader header;
packet.GetHeader(&header);
@ -1241,8 +1168,6 @@ bool Channel::IsPacketRetransmitted(const RTPHeader& header,
}
int32_t Channel::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::ReceivedRTCPPacket()");
// Store playout timestamp for the received RTCP packet
UpdatePlayoutTimestamp(true);
@ -1318,8 +1243,6 @@ void Channel::SetChannelOutputVolumeScaling(float scaling) {
}
int Channel::SendTelephoneEventOutband(int event, int duration_ms) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SendTelephoneEventOutband(...)");
RTC_DCHECK_LE(0, event);
RTC_DCHECK_GE(255, event);
RTC_DCHECK_LE(0, duration_ms);
@ -1337,8 +1260,6 @@ int Channel::SendTelephoneEventOutband(int event, int duration_ms) {
int Channel::SetSendTelephoneEventPayloadType(int payload_type,
int payload_frequency) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SetSendTelephoneEventPayloadType()");
RTC_DCHECK_LE(0, payload_type);
RTC_DCHECK_GE(127, payload_type);
CodecInst codec = {0};
@ -1357,8 +1278,6 @@ int Channel::SetSendTelephoneEventPayloadType(int payload_type,
}
int Channel::SetLocalSSRC(unsigned int ssrc) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SetLocalSSRC()");
if (channel_state_.Get().sending) {
LOG(LS_ERROR) << "SetLocalSSRC() already sending";
return -1;
@ -1452,14 +1371,10 @@ void Channel::ResetReceiverCongestionControlObjects() {
}
void Channel::SetRTCPStatus(bool enable) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SetRTCPStatus()");
_rtpRtcpModule->SetRTCPStatus(enable ? RtcpMode::kCompound : RtcpMode::kOff);
}
int Channel::SetRTCP_CNAME(const char cName[256]) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SetRTCP_CNAME()");
if (_rtpRtcpModule->SetCNAME(cName) != 0) {
LOG(LS_ERROR) << "SetRTCP_CNAME() failed to set RTCP CNAME";
return -1;
@ -1535,9 +1450,8 @@ int Channel::GetRTPStatistics(CallStatistics& stats) {
}
if (_rtpRtcpModule->DataCountersRTP(&bytesSent, &packetsSent) != 0) {
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
"GetRTPStatistics() failed to retrieve RTP datacounters =>"
" output will not be complete");
LOG(LS_WARNING) << "GetRTPStatistics() failed to retrieve RTP datacounters"
<< " => output will not be complete";
}
stats.bytesSent = bytesSent;
@ -1724,8 +1638,6 @@ uint32_t Channel::GetDelayEstimate() const {
}
int Channel::SetMinimumPlayoutDelay(int delayMs) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SetMinimumPlayoutDelay()");
if ((delayMs < kVoiceEngineMinMinPlayoutDelayMs) ||
(delayMs > kVoiceEngineMaxMinPlayoutDelayMs)) {
LOG(LS_ERROR) << "SetMinimumPlayoutDelay() invalid min delay";
@ -1770,9 +1682,8 @@ void Channel::UpdatePlayoutTimestamp(bool rtcp) {
uint16_t delay_ms = 0;
if (_audioDeviceModulePtr->PlayoutDelay(&delay_ms) == -1) {
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::UpdatePlayoutTimestamp() failed to read playout"
" delay from the ADM");
LOG(LS_WARNING) << "Channel::UpdatePlayoutTimestamp() failed to read"
<< " playout delay from the ADM";
return;
}
@ -1782,10 +1693,6 @@ void Channel::UpdatePlayoutTimestamp(bool rtcp) {
// Remove the playout delay.
playout_timestamp -= (delay_ms * (GetRtpTimestampRateHz() / 1000));
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::UpdatePlayoutTimestamp() => playoutTimestamp = %lu",
playout_timestamp);
{
rtc::CritScope lock(&video_sync_lock_);
if (!rtcp) {
@ -1796,9 +1703,6 @@ void Channel::UpdatePlayoutTimestamp(bool rtcp) {
}
void Channel::RegisterReceiveCodecsToRTPModule() {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::RegisterReceiveCodecsToRTPModule()");
CodecInst codec;
const uint8_t nSupportedCodecs = AudioCodingModule::NumberOfCodecs();
@ -1806,21 +1710,10 @@ void Channel::RegisterReceiveCodecsToRTPModule() {
// Open up the RTP/RTCP receiver for all supported codecs
if ((audio_coding_->Codec(idx, &codec) == -1) ||
(rtp_receiver_->RegisterReceivePayload(codec) == -1)) {
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::RegisterReceiveCodecsToRTPModule() unable"
" to register %s (%d/%d/%" PRIuS
"/%d) to RTP/RTCP "
"receiver",
codec.plname, codec.pltype, codec.plfreq, codec.channels,
codec.rate);
} else {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::RegisterReceiveCodecsToRTPModule() %s "
"(%d/%d/%" PRIuS
"/%d) has been added to the RTP/RTCP "
"receiver",
codec.plname, codec.pltype, codec.plfreq, codec.channels,
codec.rate);
LOG(LS_WARNING) << "Channel::RegisterReceiveCodecsToRTPModule() unable"
<< " to register " << codec.plname << " (" << codec.pltype
<< "/" << codec.plfreq << "/" << codec.channels << "/"
<< codec.rate << ") to RTP/RTCP receiver";
}
}
}