From 24ea822dcbb66532d8fee73500cf4a6a62cf0024 Mon Sep 17 00:00:00 2001 From: Jonas Olsson Date: Thu, 25 Jan 2018 10:14:29 +0100 Subject: [PATCH] Remove logging in audio/* from release builds. This makes the binary around 8000 bytes smaller. Bug: webrtc:8529 Change-Id: Ic59b16e300dd4dd5471e1079103982300cb5d660 Reviewed-on: https://webrtc-review.googlesource.com/43300 Reviewed-by: Henrik Andreassson Reviewed-by: Fredrik Solenberg Commit-Queue: Jonas Olsson Cr-Commit-Position: refs/heads/master@{#21762} --- audio/audio_receive_stream.cc | 7 +++-- audio/audio_send_stream.cc | 25 +++++++++-------- audio/audio_state.cc | 2 +- audio/channel.cc | 53 ++++++++++++++++++----------------- rtc_base/logging.h | 14 ++++----- 5 files changed, 52 insertions(+), 49 deletions(-) diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc index 45ffe34f14..c5e2b16c20 100644 --- a/audio/audio_receive_stream.cc +++ b/audio/audio_receive_stream.cc @@ -102,7 +102,7 @@ AudioReceiveStream::AudioReceiveStream( std::unique_ptr channel_proxy) : audio_state_(audio_state), channel_proxy_(std::move(channel_proxy)) { - RTC_LOG(LS_INFO) << "AudioReceiveStream: " << config.ToString(); + RTC_LOG(LS_INFO) << "AudioReceiveStream: " << config.rtp.remote_ssrc; RTC_DCHECK(receiver_controller); RTC_DCHECK(packet_router); RTC_DCHECK(config.decoder_factory); @@ -127,7 +127,7 @@ AudioReceiveStream::AudioReceiveStream( AudioReceiveStream::~AudioReceiveStream() { RTC_DCHECK_RUN_ON(&worker_thread_checker_); - RTC_LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString(); + RTC_LOG(LS_INFO) << "~AudioReceiveStream: " << config_.rtp.remote_ssrc; Stop(); channel_proxy_->DisassociateSendChannel(); channel_proxy_->RegisterTransport(nullptr); @@ -138,7 +138,6 @@ AudioReceiveStream::~AudioReceiveStream() { void AudioReceiveStream::Reconfigure( const webrtc::AudioReceiveStream::Config& config) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - RTC_LOG(LS_INFO) << "AudioReceiveStream::Reconfigure: " << config_.ToString(); ConfigureStream(this, config, false); } @@ -348,6 +347,8 @@ internal::AudioState* AudioReceiveStream::audio_state() const { void AudioReceiveStream::ConfigureStream(AudioReceiveStream* stream, const Config& new_config, bool first_time) { + RTC_LOG(LS_INFO) << "AudioReceiveStream::ConfigureStream: " + << new_config.ToString(); RTC_DCHECK(stream); const auto& channel_proxy = stream->channel_proxy_; const auto& old_config = stream->config_; diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc index 80c2c6b435..b39f9cd600 100644 --- a/audio/audio_send_stream.cc +++ b/audio/audio_send_stream.cc @@ -129,7 +129,7 @@ AudioSendStream::AudioSendStream( kRecoverablePacketLossRateMinNumAckedPairs), rtp_rtcp_module_(nullptr), suspended_rtp_state_(suspended_rtp_state) { - RTC_LOG(LS_INFO) << "AudioSendStream: " << config.ToString(); + RTC_LOG(LS_INFO) << "AudioSendStream: " << config.rtp.ssrc; RTC_DCHECK(worker_queue_); RTC_DCHECK(audio_state_); RTC_DCHECK(channel_proxy_); @@ -153,7 +153,7 @@ AudioSendStream::AudioSendStream( AudioSendStream::~AudioSendStream() { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - RTC_LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString(); + RTC_LOG(LS_INFO) << "~AudioSendStream: " << config_.rtp.ssrc; RTC_DCHECK(!sending_); transport_->send_side_cc()->DeRegisterPacketFeedbackObserver(this); channel_proxy_->RegisterTransport(nullptr); @@ -190,7 +190,8 @@ void AudioSendStream::ConfigureStream( webrtc::internal::AudioSendStream* stream, const webrtc::AudioSendStream::Config& new_config, bool first_time) { - RTC_LOG(LS_INFO) << "AudioSendStream::Configuring: " << new_config.ToString(); + RTC_LOG(LS_INFO) << "AudioSendStream::ConfigureStream: " + << new_config.ToString(); const auto& channel_proxy = stream->channel_proxy_; const auto& old_config = stream->config_; @@ -501,7 +502,7 @@ bool AudioSendStream::SetupSendCodec(AudioSendStream* stream, spec.format); if (!encoder) { - RTC_LOG(LS_ERROR) << "Unable to create encoder for " << spec.format; + RTC_DLOG(LS_ERROR) << "Unable to create encoder for " << spec.format; return false; } // If a bitrate has been specified for the codec, use it over the @@ -514,8 +515,8 @@ bool AudioSendStream::SetupSendCodec(AudioSendStream* stream, if (new_config.audio_network_adaptor_config) { if (encoder->EnableAudioNetworkAdaptor( *new_config.audio_network_adaptor_config, stream->event_log_)) { - RTC_LOG(LS_INFO) << "Audio network adaptor enabled on SSRC " - << new_config.rtp.ssrc; + RTC_DLOG(LS_INFO) << "Audio network adaptor enabled on SSRC " + << new_config.rtp.ssrc; } else { RTC_NOTREACHED(); } @@ -597,8 +598,8 @@ void AudioSendStream::ReconfigureANA(AudioSendStream* stream, CallEncoder(stream->channel_proxy_, [&](AudioEncoder* encoder) { if (encoder->EnableAudioNetworkAdaptor( *new_config.audio_network_adaptor_config, stream->event_log_)) { - RTC_LOG(LS_INFO) << "Audio network adaptor enabled on SSRC " - << new_config.rtp.ssrc; + RTC_DLOG(LS_INFO) << "Audio network adaptor enabled on SSRC " + << new_config.rtp.ssrc; } else { RTC_NOTREACHED(); } @@ -607,8 +608,8 @@ void AudioSendStream::ReconfigureANA(AudioSendStream* stream, CallEncoder(stream->channel_proxy_, [&](AudioEncoder* encoder) { encoder->DisableAudioNetworkAdaptor(); }); - RTC_LOG(LS_INFO) << "Audio network adaptor disabled on SSRC " - << new_config.rtp.ssrc; + RTC_DLOG(LS_INFO) << "Audio network adaptor disabled on SSRC " + << new_config.rtp.ssrc; } } @@ -719,8 +720,8 @@ void AudioSendStream::RegisterCngPayloadType(int payload_type, if (rtp_rtcp_module_->RegisterSendPayload(codec) != 0) { rtp_rtcp_module_->DeRegisterSendPayload(codec.pltype); if (rtp_rtcp_module_->RegisterSendPayload(codec) != 0) { - RTC_LOG(LS_ERROR) << "RegisterCngPayloadType() failed to register CN to " - "RTP/RTCP module"; + RTC_DLOG(LS_ERROR) << "RegisterCngPayloadType() failed to register CN to " + "RTP/RTCP module"; } } } diff --git a/audio/audio_state.cc b/audio/audio_state.cc index 06a6863310..3d38c8934c 100644 --- a/audio/audio_state.cc +++ b/audio/audio_state.cc @@ -51,7 +51,7 @@ void AudioState::AddReceivingStream(webrtc::AudioReceiveStream* stream) { receiving_streams_.insert(stream); if (!config_.audio_mixer->AddSource( static_cast(stream))) { - RTC_LOG(LS_ERROR) << "Failed to add source to mixer."; + RTC_DLOG(LS_ERROR) << "Failed to add source to mixer."; } // Make sure playback is initialized; start playing if enabled. diff --git a/audio/channel.cc b/audio/channel.cc index 1799e7aae1..ac4b917e16 100644 --- a/audio/channel.cc +++ b/audio/channel.cc @@ -345,7 +345,7 @@ int32_t Channel::SendData(FrameType frameType, // received from the capture device as // undefined for voice for now. -1, payloadData, payloadSize, fragmentation, nullptr, nullptr)) { - RTC_LOG(LS_ERROR) + RTC_DLOG(LS_ERROR) << "Channel::SendData() failed to send data to RTP/RTCP module"; return -1; } @@ -359,14 +359,14 @@ bool Channel::SendRtp(const uint8_t* data, rtc::CritScope cs(&_callbackCritSect); if (_transportPtr == NULL) { - RTC_LOG(LS_ERROR) + RTC_DLOG(LS_ERROR) << "Channel::SendPacket() failed to send RTP packet due to" << " invalid transport object"; return false; } if (!_transportPtr->SendRtp(data, len, options)) { - RTC_LOG(LS_ERROR) << "Channel::SendPacket() RTP transmission failed"; + RTC_DLOG(LS_ERROR) << "Channel::SendPacket() RTP transmission failed"; return false; } return true; @@ -375,14 +375,15 @@ bool Channel::SendRtp(const uint8_t* data, bool Channel::SendRtcp(const uint8_t* data, size_t len) { rtc::CritScope cs(&_callbackCritSect); if (_transportPtr == NULL) { - RTC_LOG(LS_ERROR) << "Channel::SendRtcp() failed to send RTCP packet due to" - << " invalid transport object"; + RTC_DLOG(LS_ERROR) + << "Channel::SendRtcp() failed to send RTCP packet due to" + << " invalid transport object"; return false; } int n = _transportPtr->SendRtcp(data, len); if (n < 0) { - RTC_LOG(LS_ERROR) << "Channel::SendRtcp() transmission failed"; + RTC_DLOG(LS_ERROR) << "Channel::SendRtcp() transmission failed"; return false; } return true; @@ -401,9 +402,9 @@ int32_t Channel::OnInitializeDecoder(int payload_type, const SdpAudioFormat& audio_format, uint32_t rate) { if (!audio_coding_->RegisterReceiveCodec(payload_type, audio_format)) { - RTC_LOG(LS_WARNING) << "Channel::OnInitializeDecoder() invalid codec (pt=" - << payload_type << ", " << audio_format - << ") received -1"; + RTC_DLOG(LS_WARNING) << "Channel::OnInitializeDecoder() invalid codec (pt=" + << payload_type << ", " << audio_format + << ") received -1"; return -1; } @@ -422,7 +423,7 @@ int32_t Channel::OnReceivedPayloadData(const uint8_t* payloadData, // Push the incoming payload (parsed and ready for decoding) into the ACM if (audio_coding_->IncomingPacket(payloadData, payloadSize, *rtpHeader) != 0) { - RTC_LOG(LS_ERROR) + RTC_DLOG(LS_ERROR) << "Channel::OnReceivedPayloadData() unable to push data to the ACM"; return -1; } @@ -452,7 +453,7 @@ AudioMixer::Source::AudioFrameInfo Channel::GetAudioFrameWithInfo( bool muted; if (audio_coding_->PlayoutData10Ms(audio_frame->sample_rate_hz_, audio_frame, &muted) == -1) { - RTC_LOG(LS_ERROR) << "Channel::GetAudioFrame() PlayoutData10Ms() failed!"; + RTC_DLOG(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 @@ -736,7 +737,7 @@ int32_t Channel::StartSend() { } _rtpRtcpModule->SetSendingMediaStatus(true); if (_rtpRtcpModule->SetSendingStatus(true) != 0) { - RTC_LOG(LS_ERROR) << "StartSend() RTP/RTCP failed to start sending"; + RTC_DLOG(LS_ERROR) << "StartSend() RTP/RTCP failed to start sending"; _rtpRtcpModule->SetSendingMediaStatus(false); rtc::CritScope cs(&_callbackCritSect); channel_state_.SetSending(false); @@ -785,7 +786,7 @@ void Channel::StopSend() { // Reset sending SSRC and sequence number and triggers direct transmission // of RTCP BYE if (_rtpRtcpModule->SetSendingStatus(false) == -1) { - RTC_LOG(LS_ERROR) << "StartSend() RTP/RTCP failed to stop sending"; + RTC_DLOG(LS_ERROR) << "StartSend() RTP/RTCP failed to stop sending"; } _rtpRtcpModule->SetSendingMediaStatus(false); } @@ -816,7 +817,7 @@ bool Channel::SetEncoder(int payload_type, if (_rtpRtcpModule->RegisterSendPayload(rtp_codec) != 0) { _rtpRtcpModule->DeRegisterSendPayload(payload_type); if (_rtpRtcpModule->RegisterSendPayload(rtp_codec) != 0) { - RTC_LOG(LS_ERROR) + RTC_DLOG(LS_ERROR) << "SetEncoder() failed to register codec to RTP/RTCP module"; return false; } @@ -1051,7 +1052,7 @@ int Channel::SendTelephoneEventOutband(int event, int duration_ms) { } if (_rtpRtcpModule->SendTelephoneEventOutband( event, duration_ms, kTelephoneEventAttenuationdB) != 0) { - RTC_LOG(LS_ERROR) << "SendTelephoneEventOutband() failed to send event"; + RTC_DLOG(LS_ERROR) << "SendTelephoneEventOutband() failed to send event"; return -1; } return 0; @@ -1068,7 +1069,7 @@ int Channel::SetSendTelephoneEventPayloadType(int payload_type, if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) { _rtpRtcpModule->DeRegisterSendPayload(codec.pltype); if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) { - RTC_LOG(LS_ERROR) + RTC_DLOG(LS_ERROR) << "SetSendTelephoneEventPayloadType() failed to register " "send payload type"; return -1; @@ -1079,7 +1080,7 @@ int Channel::SetSendTelephoneEventPayloadType(int payload_type, int Channel::SetLocalSSRC(unsigned int ssrc) { if (channel_state_.Get().sending) { - RTC_LOG(LS_ERROR) << "SetLocalSSRC() already sending"; + RTC_DLOG(LS_ERROR) << "SetLocalSSRC() already sending"; return -1; } _rtpRtcpModule->SetSSRC(ssrc); @@ -1157,7 +1158,7 @@ void Channel::SetRTCPStatus(bool enable) { int Channel::SetRTCP_CNAME(const char cName[256]) { if (_rtpRtcpModule->SetCNAME(cName) != 0) { - RTC_LOG(LS_ERROR) << "SetRTCP_CNAME() failed to set RTCP CNAME"; + RTC_DLOG(LS_ERROR) << "SetRTCP_CNAME() failed to set RTCP CNAME"; return -1; } return 0; @@ -1166,7 +1167,7 @@ int Channel::SetRTCP_CNAME(const char cName[256]) { int Channel::GetRemoteRTCPReportBlocks( std::vector* report_blocks) { if (report_blocks == NULL) { - RTC_LOG(LS_ERROR) << "GetRemoteRTCPReportBlock()s invalid report_blocks."; + RTC_DLOG(LS_ERROR) << "GetRemoteRTCPReportBlock()s invalid report_blocks."; return -1; } @@ -1231,7 +1232,7 @@ int Channel::GetRTPStatistics(CallStatistics& stats) { } if (_rtpRtcpModule->DataCountersRTP(&bytesSent, &packetsSent) != 0) { - RTC_LOG(LS_WARNING) + RTC_DLOG(LS_WARNING) << "GetRTPStatistics() failed to retrieve RTP datacounters" << " => output will not be complete"; } @@ -1314,7 +1315,7 @@ void Channel::ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input) { // is done and payload is ready for packetization and transmission. // Otherwise, it will return without invoking the callback. if (audio_coding_->Add10MsData(*audio_input) < 0) { - RTC_LOG(LS_ERROR) << "ACM::Add10MsData() failed."; + RTC_DLOG(LS_ERROR) << "ACM::Add10MsData() failed."; return; } @@ -1378,11 +1379,11 @@ uint32_t Channel::GetDelayEstimate() const { int Channel::SetMinimumPlayoutDelay(int delayMs) { if ((delayMs < kVoiceEngineMinMinPlayoutDelayMs) || (delayMs > kVoiceEngineMaxMinPlayoutDelayMs)) { - RTC_LOG(LS_ERROR) << "SetMinimumPlayoutDelay() invalid min delay"; + RTC_DLOG(LS_ERROR) << "SetMinimumPlayoutDelay() invalid min delay"; return -1; } if (audio_coding_->SetMinimumPlayoutDelay(delayMs) != 0) { - RTC_LOG(LS_ERROR) + RTC_DLOG(LS_ERROR) << "SetMinimumPlayoutDelay() failed to set min playout delay"; return -1; } @@ -1396,7 +1397,7 @@ int Channel::GetPlayoutTimestamp(unsigned int& timestamp) { playout_timestamp_rtp = playout_timestamp_rtp_; } if (playout_timestamp_rtp == 0) { - RTC_LOG(LS_ERROR) << "GetPlayoutTimestamp() failed to retrieve timestamp"; + RTC_DLOG(LS_ERROR) << "GetPlayoutTimestamp() failed to retrieve timestamp"; return -1; } timestamp = playout_timestamp_rtp; @@ -1421,8 +1422,8 @@ void Channel::UpdatePlayoutTimestamp(bool rtcp) { uint16_t delay_ms = 0; if (_audioDeviceModulePtr->PlayoutDelay(&delay_ms) == -1) { - RTC_LOG(LS_WARNING) << "Channel::UpdatePlayoutTimestamp() failed to read" - << " playout delay from the ADM"; + RTC_DLOG(LS_WARNING) << "Channel::UpdatePlayoutTimestamp() failed to read" + << " playout delay from the ADM"; return; } diff --git a/rtc_base/logging.h b/rtc_base/logging.h index e3879fc05b..91b71a63e6 100644 --- a/rtc_base/logging.h +++ b/rtc_base/logging.h @@ -279,7 +279,7 @@ class LogMessageVoidify { rtc::LogMessage(__FILE__, __LINE__, rtc::sev).stream() // The _V version is for when a variable is passed in. It doesn't do the -// namespace concatination. +// namespace concatenation. #define RTC_LOG_V(sev) \ RTC_LOG_SEVERITY_PRECONDITION(sev) \ rtc::LogMessage(__FILE__, __LINE__, sev).stream() @@ -354,13 +354,13 @@ inline bool LogCheckLevel(LoggingSeverity sev) { #define RTC_DLOG_F(sev) RTC_LOG_F(sev) #else #define RTC_DLOG_EAT_STREAM_PARAMS(sev) \ - (true ? true : ((void)(rtc::sev), true)) \ - ? static_cast(0) \ - : rtc::LogMessageVoidify() & \ - rtc::LogMessage(__FILE__, __LINE__, rtc::sev).stream() -#define RTC_DLOG(sev) RTC_DLOG_EAT_STREAM_PARAMS(sev) + (true ? true : ((void)(sev), true)) \ + ? static_cast(0) \ + : rtc::LogMessageVoidify() & \ + rtc::LogMessage(__FILE__, __LINE__, sev).stream() +#define RTC_DLOG(sev) RTC_DLOG_EAT_STREAM_PARAMS(rtc::sev) #define RTC_DLOG_V(sev) RTC_DLOG_EAT_STREAM_PARAMS(sev) -#define RTC_DLOG_F(sev) RTC_DLOG_EAT_STREAM_PARAMS(sev) +#define RTC_DLOG_F(sev) RTC_DLOG_EAT_STREAM_PARAMS(rtc::sev) #endif } // namespace rtc