From 6e3ef255fd7a5ed705e77394194f03a76f1f9864 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Rafa=C3=ABl=20Carr=C3=A9?= <rafael.carre@savoirfairelinux.com> Date: Wed, 24 Aug 2011 17:23:16 -0400 Subject: [PATCH] _debug* -> _debug Don't use a 8k buffer to format message, let v* functions do the work Use yellow for warnings Use either syslog either console output --- daemon/src/audio/alsa/alsalayer.cpp | 62 +++++++++---------- daemon/src/audio/audiortp/AudioRtpFactory.cpp | 6 +- .../src/audio/audiortp/AudioZrtpSession.cpp | 2 +- daemon/src/logger.cpp | 54 +++++++--------- daemon/src/logger.h | 4 -- daemon/src/managerimpl.cpp | 12 ++-- 6 files changed, 62 insertions(+), 78 deletions(-) diff --git a/daemon/src/audio/alsa/alsalayer.cpp b/daemon/src/audio/alsa/alsalayer.cpp index 7f310dd39c..3393b6e1f8 100644 --- a/daemon/src/audio/alsa/alsalayer.cpp +++ b/daemon/src/audio/alsa/alsalayer.cpp @@ -111,7 +111,7 @@ AlsaLayer::~AlsaLayer (void) void AlsaLayer::closeLayer() { - _debugAlsa ("Audio: Close ALSA streams"); + _debug ("Audio: Close ALSA streams"); try { /* Stop the audio thread first */ @@ -121,7 +121,7 @@ AlsaLayer::closeLayer() audioThread_ = NULL; } } catch (...) { - _debugException ("Audio: Exception: when stopping audiortp"); + _debug ("Audio: Exception: when stopping audiortp"); throw; } @@ -155,10 +155,10 @@ AlsaLayer::openDevice (int indexIn, int indexOut, int indexRing, int sampleRate, audioPlugin_ = plugin; - _debugAlsa (" Setting AlsaLayer: device in=%2d, out=%2d, ring=%2d", indexIn_, indexOut_, indexRing_); - _debugAlsa (" : alsa plugin=%s", audioPlugin_.c_str()); - _debugAlsa (" : nb channel in=%2d, out=%2d", inChannel_, outChannel_); - _debugAlsa (" : sample rate=%5d, format=%s", audioSampleRate_, SFLDataFormatString); + _debug (" Setting AlsaLayer: device in=%2d, out=%2d, ring=%2d", indexIn_, indexOut_, indexRing_); + _debug (" : alsa plugin=%s", audioPlugin_.c_str()); + _debug (" : nb channel in=%2d, out=%2d", inChannel_, outChannel_); + _debug (" : sample rate=%5d, format=%s", audioSampleRate_, SFLDataFormatString); audioThread_ = NULL; @@ -218,7 +218,7 @@ AlsaLayer::startStream (void) audioThread_ = new AlsaThread (this); audioThread_->start(); } catch (...) { - _debugException ("Fail to start audio thread"); + _debug ("Fail to start audio thread"); } } @@ -240,7 +240,7 @@ AlsaLayer::stopStream (void) audioThread_ = NULL; } } catch (...) { - _debugException ("Audio: Exception: when stopping audiortp"); + _debug ("Audio: Exception: when stopping audiortp"); throw; } @@ -404,12 +404,12 @@ bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type) int err; if ((err = snd_pcm_hw_params_any (pcm_handle, hwparams)) < 0) { - _debugAlsa ("Audio: Error: Cannot initialize hardware parameter structure (%s)", snd_strerror (err)); + _debug ("Audio: Error: Cannot initialize hardware parameter structure (%s)", snd_strerror (err)); return false; } if ((err = snd_pcm_hw_params_set_access (pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { - _debugAlsa ("Audio: Error: Cannot set access type (%s)", snd_strerror (err)); + _debug ("Audio: Error: Cannot set access type (%s)", snd_strerror (err)); return false; } @@ -417,7 +417,7 @@ bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type) format = SND_PCM_FORMAT_S16_LE; if ((err = snd_pcm_hw_params_set_format (pcm_handle, hwparams, (snd_pcm_format_t) format)) < 0) { - _debugAlsa ("Audio: Error: Cannot set sample format (%s)", snd_strerror (err)); + _debug ("Audio: Error: Cannot set sample format (%s)", snd_strerror (err)); return false; } @@ -427,19 +427,19 @@ bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type) unsigned int exact_ivalue = audioSampleRate_; if ((err = snd_pcm_hw_params_set_rate_near (pcm_handle, hwparams, &exact_ivalue, &dir) < 0)) { - _debugAlsa ("Audio: Error: Cannot set sample rate (%s)", snd_strerror (err)); + _debug ("Audio: Error: Cannot set sample rate (%s)", snd_strerror (err)); return false; } else _debug ("Audio: Set audio rate to %d", audioSampleRate_); if (dir != 0) { - _debugAlsa ("Audio: Error: (%i) The chosen rate %d Hz is not supported by your hardware.Using %d Hz instead. ", type , audioSampleRate_, exact_ivalue); + _debug ("Audio: Error: (%i) The chosen rate %d Hz is not supported by your hardware.Using %d Hz instead. ", type , audioSampleRate_, exact_ivalue); audioSampleRate_ = exact_ivalue; } /* Set the number of channels */ if ((err = snd_pcm_hw_params_set_channels (pcm_handle, hwparams, 1)) < 0) { - _debugAlsa ("Audio: Error: Cannot set channel count (%s)", snd_strerror (err)); + _debug ("Audio: Error: Cannot set channel count (%s)", snd_strerror (err)); return false; } @@ -449,12 +449,12 @@ bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type) dir = 0; if ((err = snd_pcm_hw_params_set_period_size_near (pcm_handle, hwparams, &exact_lvalue, &dir)) < 0) { - _debugAlsa ("Audio: Error: Cannot set period time (%s)", snd_strerror (err)); + _debug ("Audio: Error: Cannot set period time (%s)", snd_strerror (err)); return false; } if (dir != 0) - _debugAlsa ("Audio: Warning: (%i) The chosen period size %lu bytes is not supported by your hardware.Using %lu instead. ", type, periodsize, exact_lvalue); + _debug ("Audio: Warning: (%i) The chosen period size %lu bytes is not supported by your hardware.Using %lu instead. ", type, periodsize, exact_lvalue); periodSize_ = exact_lvalue; /* Set the number of fragments */ @@ -462,19 +462,19 @@ bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type) dir = 0; if ((err = snd_pcm_hw_params_set_periods_near (pcm_handle, hwparams, &exact_ivalue, &dir)) < 0) { - _debugAlsa ("Audio: Error: Cannot set periods number (%s)", snd_strerror (err)); + _debug ("Audio: Error: Cannot set periods number (%s)", snd_strerror (err)); return false; } if (dir != 0) - _debugAlsa ("Audio: Warning: The chosen period number %i bytes is not supported by your hardware.Using %i instead. ", periods, exact_ivalue); + _debug ("Audio: Warning: The chosen period number %i bytes is not supported by your hardware.Using %i instead. ", periods, exact_ivalue); periods = exact_ivalue; /* Set the hw parameters */ if ((err = snd_pcm_hw_params (pcm_handle, hwparams)) < 0) { - _debugAlsa ("Audio: Error: Cannot set hw parameters (%s)", snd_strerror (err)); + _debug ("Audio: Error: Cannot set hw parameters (%s)", snd_strerror (err)); return false; } @@ -487,12 +487,12 @@ bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type) /* Set the start threshold */ if ((err = snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams, periodSize_ * 2)) < 0) { - _debugAlsa ("Audio: Error: Cannot set start threshold (%s)", snd_strerror (err)); + _debug ("Audio: Error: Cannot set start threshold (%s)", snd_strerror (err)); return false; } if ((err = snd_pcm_sw_params (pcm_handle, swparams)) < 0) { - _debugAlsa ("Audio: Error: Cannot set sw parameters (%s)", snd_strerror (err)); + _debug ("Audio: Error: Cannot set sw parameters (%s)", snd_strerror (err)); return false; } @@ -583,14 +583,14 @@ AlsaLayer::write (void* buffer, int length, snd_pcm_t * handle) handle_xrun_playback (handle); if (snd_pcm_writei (handle, buffer , frames) < 0) - _debugAlsa ("Audio: XRUN handling failed"); + _debug ("Audio: XRUN handling failed"); trigger_request_ = true; break; default: - _debugAlsa ("Audio: Write error unknown - dropping frames: %s", snd_strerror (err)); + _debug ("Audio: Write error unknown - dropping frames: %s", snd_strerror (err)); stopPlaybackStream (); break; } @@ -616,12 +616,12 @@ AlsaLayer::read (void* buffer, int toCopy) case -EPIPE: case -ESTRPIPE: case -EIO: - _debugAlsa ("Audio: XRUN capture ignored (%s)", snd_strerror (err)); + _debug ("Audio: XRUN capture ignored (%s)", snd_strerror (err)); handle_xrun_capture(); break; case EPERM: - _debugAlsa ("Audio: Capture EPERM (%s)", snd_strerror (err)); + _debug ("Audio: Capture EPERM (%s)", snd_strerror (err)); prepareCaptureStream (); startCaptureStream (); break; @@ -639,7 +639,7 @@ AlsaLayer::read (void* buffer, int toCopy) void AlsaLayer::handle_xrun_capture (void) { - _debugAlsa ("Audio: Handle xrun capture"); + _debug ("Audio: Handle xrun capture"); snd_pcm_status_t* status; snd_pcm_status_alloca (&status); @@ -651,20 +651,20 @@ AlsaLayer::handle_xrun_capture (void) startCaptureStream (); } } else - _debugAlsa ("Audio: Get status failed"); + _debug ("Audio: Get status failed"); } void AlsaLayer::handle_xrun_playback (snd_pcm_t *handle) { - _debugAlsa ("Audio: Handle xrun playback"); + _debug ("Audio: Handle xrun playback"); snd_pcm_status_t* status; snd_pcm_status_alloca (&status); int state; if ((state = snd_pcm_status (handle, status)) < 0) - _debugAlsa ("Audio: Error: Cannot get playback handle status (%s)" , snd_strerror (state)); + _debug ("Audio: Error: Cannot get playback handle status (%s)" , snd_strerror (state)); else { int state = snd_pcm_status_get_state (status); @@ -733,9 +733,9 @@ AlsaLayer::getSoundCardsInfo (int stream) snd_pcm_info_set_device (pcminfo , 0); snd_pcm_info_set_stream (pcminfo, (stream == SFL_PCM_CAPTURE) ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK); - if (snd_ctl_pcm_info (handle ,pcminfo) < 0) _debugAlsa (" Cannot get info"); + if (snd_ctl_pcm_info (handle ,pcminfo) < 0) _debug (" Cannot get info"); else { - _debugAlsa ("card %i : %s [%s]", + _debug ("card %i : %s [%s]", numCard, snd_ctl_card_info_get_id (info), snd_ctl_card_info_get_name (info)); diff --git a/daemon/src/audio/audiortp/AudioRtpFactory.cpp b/daemon/src/audio/audiortp/AudioRtpFactory.cpp index e7fa28e240..09c9cb7b78 100644 --- a/daemon/src/audio/audiortp/AudioRtpFactory.cpp +++ b/daemon/src/audio/audiortp/AudioRtpFactory.cpp @@ -58,7 +58,7 @@ AudioRtpFactory::~AudioRtpFactory() void AudioRtpFactory::initAudioRtpConfig () { if (_rtpSession != NULL) { - _debugException ("An audio rtp thread was already created but not" \ + _debug ("An audio rtp thread was already created but not" \ "destroyed. Forcing it before continuing."); stop(); } @@ -161,7 +161,7 @@ void AudioRtpFactory::stop (void) _info ("AudioRtpFactory: Stopping audio rtp session"); if (_rtpSession == NULL) { - _debugException ("AudioRtpFactory: Rtp session already deleted"); + _debug ("AudioRtpFactory: Rtp session already deleted"); return; } @@ -174,7 +174,7 @@ void AudioRtpFactory::stop (void) delete _rtpSession; _rtpSession = NULL; } catch (...) { - _debugException ("AudioRtpFactory: Error: Exception caught when stopping the audio rtp session"); + _debug ("AudioRtpFactory: Error: Exception caught when stopping the audio rtp session"); throw AudioRtpFactoryException ("AudioRtpFactory: Error: caught exception in AudioRtpFactory::stop"); } } diff --git a/daemon/src/audio/audiortp/AudioZrtpSession.cpp b/daemon/src/audio/audiortp/AudioZrtpSession.cpp index 05c6b260a3..0a7ef0fd6b 100644 --- a/daemon/src/audio/audiortp/AudioZrtpSession.cpp +++ b/daemon/src/audio/audiortp/AudioZrtpSession.cpp @@ -73,7 +73,7 @@ AudioZrtpSession::~AudioZrtpSession() try { terminate(); } catch (...) { - _debugException ("AudioZrtpSession: Thread destructor didn't terminate correctly"); + _debug ("AudioZrtpSession: Thread destructor didn't terminate correctly"); throw; } diff --git a/daemon/src/logger.cpp b/daemon/src/logger.cpp index 3bf25bae3a..3379af8e30 100644 --- a/daemon/src/logger.cpp +++ b/daemon/src/logger.cpp @@ -45,52 +45,40 @@ void log (const int level, const char* format, ...) return; va_list ap; - const char *prefix = "<> "; - const char *color_prefix = ""; - switch (level) { - case LOG_ERR: { - prefix = "<error> "; - color_prefix = RED; - break; + if (consoleLog) { + const char *color_prefix = ""; + switch (level) { + case LOG_ERR: + color_prefix = RED; + break; + case LOG_WARNING: + color_prefix = YELLOW; + break; } - case LOG_WARNING: { - prefix = "<warning> "; - color_prefix = LIGHT_RED; - break; - } - case LOG_INFO: { - prefix = "<info> "; - color_prefix = ""; - break; - } - case LOG_DEBUG: { - prefix = "<debug> "; - color_prefix = ""; - break; - } - } - char buffer[8192]; - va_start (ap, format); - vsnprintf (buffer, sizeof buffer, format, ap); - va_end (ap); + fputs(color_prefix, stderr); - if (consoleLog) - fprintf(stderr, "%s%s"END_COLOR"\n", color_prefix, buffer); + va_start (ap, format); + vfprintf(stderr, format, ap); + va_end (ap); - syslog (level, "%s%s", prefix, buffer); + fputs(END_COLOR"\n", stderr); + } else { + va_start (ap, format); + vsyslog (level, format, ap); + va_end (ap); + } } void setConsoleLog (bool c) { - Logger::consoleLog = c; + consoleLog = c; } void setDebugMode (bool d) { - Logger::debugMode = d; + debugMode = d; } } - diff --git a/daemon/src/logger.h b/daemon/src/logger.h index 8ea3909581..af5df66e20 100644 --- a/daemon/src/logger.h +++ b/daemon/src/logger.h @@ -46,10 +46,6 @@ void setDebugMode (bool); #define _info(...) Logger::log(LOG_INFO, __VA_ARGS__) #define _debug(...) Logger::log(LOG_DEBUG, __VA_ARGS__) -#define _debugException(...) Logger::log(LOG_DEBUG, __VA_ARGS__) -#define _debugInit(...) Logger::log(LOG_DEBUG, __VA_ARGS__) -#define _debugAlsa(...) Logger::log(LOG_DEBUG, __VA_ARGS__) - #define BLACK "\033[22;30m" #define RED "\033[22;31m" #define GREEN "\033[22;32m" diff --git a/daemon/src/managerimpl.cpp b/daemon/src/managerimpl.cpp index 04eaeea4bb..9c570ef5d6 100644 --- a/daemon/src/managerimpl.cpp +++ b/daemon/src/managerimpl.cpp @@ -123,11 +123,11 @@ void ManagerImpl::init (std::string config_file) if (_audiodriver) { unsigned int sampleRate = _audiodriver->getSampleRate(); - _debugInit ("Manager: Load telephone tone"); + _debug ("Manager: Load telephone tone"); std::string country(preferences.getZoneToneChoice()); _telephoneTone = new TelephoneTone (country, sampleRate); - _debugInit ("Manager: Loading DTMF key (%d)", sampleRate); + _debug ("Manager: Loading DTMF key (%d)", sampleRate); sampleRate = 8000; @@ -2708,7 +2708,7 @@ void ManagerImpl::setEchoCancelDelay(int delay) */ bool ManagerImpl::initAudioDriver (void) { - _debugInit ("Manager: AudioLayer Creation"); + _debug ("Manager: AudioLayer Creation"); audioLayerMutexLock(); @@ -2945,12 +2945,12 @@ void ManagerImpl::audioSamplingRateChanged (int samplerate) unsigned int sampleRate = _audiodriver->getSampleRate(); delete _telephoneTone; - _debugInit ("Manager: Load telephone tone"); + _debug ("Manager: Load telephone tone"); std::string country = preferences.getZoneToneChoice(); _telephoneTone = new TelephoneTone (country, sampleRate); delete _dtmfKey; - _debugInit ("Manager: Loading DTMF key with sample rate %d", sampleRate); + _debug ("Manager: Loading DTMF key with sample rate %d", sampleRate); _dtmfKey = new DTMF (sampleRate); // Restart audio layer if it was active @@ -2966,7 +2966,7 @@ void ManagerImpl::audioSamplingRateChanged (int samplerate) */ void ManagerImpl::initVolume () { - _debugInit ("Initiate Volume"); + _debug ("Initiate Volume"); setSpkrVolume (audioPreference.getVolumespkr()); setMicVolume (audioPreference.getVolumemic()); } -- GitLab