Revert of Forward output glitch information from stream WebRTC log

media/audio/mac/audio_low_latency_input_mac.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "media/audio/mac/audio_low_latency_input_mac.h"
 #include <CoreServices/CoreServices.h>
 #include <mach/mach.h>
 #include <string>
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/mac/mac_logging.h"
 #include "base/metrics/histogram_macros.h"
 #include "base/metrics/sparse_histogram.h"
-#include "base/strings/stringprintf.h"
 #include "base/sys_info.h"
 #include "base/time/time.h"
 #include "media/audio/mac/audio_manager_mac.h"
 #include "media/base/audio_bus.h"
 #include "media/base/data_buffer.h"
 
 namespace media {
 
 // Number of blocks of buffers used in the |fifo_|.
 const int kNumberOfBlocksBufferInFifo = 2;
@@ skipping 204 matching lines @@
   DVLOG(1) << "logical processors: " << base::SysInfo::NumberOfProcessors();
   DVLOG(1) << "physical processors: " << NumberOfPhysicalProcessors();
   DVLOG(1) << "battery power: " << is_on_battery;
   DVLOG(1) << "resume events: " << num_resumes;
 }
 
 // See "Technical Note TN2091 - Device input using the HAL Output Audio Unit"
 // http://developer.apple.com/library/mac/#technotes/tn2091/_index.html
 // for more details and background regarding this implementation.
 
-AUAudioInputStream::AUAudioInputStream(
-    AudioManagerMac* manager,
-    const AudioParameters& input_params,
-    AudioDeviceID audio_device_id,
-    const AudioManager::LogCallback& log_callback)
+AUAudioInputStream::AUAudioInputStream(AudioManagerMac* manager,
+                                       const AudioParameters& input_params,
+                                       AudioDeviceID audio_device_id)
     : manager_(manager),
       number_of_frames_(input_params.frames_per_buffer()),
       number_of_frames_provided_(0),
       io_buffer_frame_size_(0),
       sink_(nullptr),
       audio_unit_(0),
       input_device_id_(audio_device_id),
       hardware_latency_frames_(0),
       number_of_channels_in_frame_(0),
       fifo_(input_params.channels(),
             number_of_frames_,
             kNumberOfBlocksBufferInFifo),
       input_callback_is_active_(false),
       start_was_deferred_(false),
       buffer_size_was_changed_(false),
       audio_unit_render_has_worked_(false),
       device_listener_is_active_(false),
       last_sample_time_(0.0),
       last_number_of_frames_(0),
       total_lost_frames_(0),
       largest_glitch_frames_(0),
       glitches_detected_(0),
       number_of_restart_indications_(0),
       number_of_restart_attempts_(0),
       total_number_of_restart_attempts_(0),
-      log_callback_(log_callback),
       weak_factory_(this) {
   DCHECK(manager_);
 
   // Set up the desired (output) format specified by the client.
   format_.mSampleRate = input_params.sample_rate();
   format_.mFormatID = kAudioFormatLinearPCM;
   format_.mFormatFlags = kLinearPCMFormatFlagIsPacked |
                          kLinearPCMFormatFlagIsSignedInteger;
   DCHECK(FormatIsInterleaved(format_.mFormatFlags));
   format_.mBitsPerChannel = input_params.bits_per_sample();
@@ skipping 1361 matching lines @@
   if (last_sample_time_ == 0)
     return;  // No stats gathered to report.
 
   // A value of 0 indicates that we got the buffer size we asked for.
   UMA_HISTOGRAM_COUNTS_10000("Media.Audio.Capture.FramesProvided",
                              number_of_frames_provided_);
   // Even if there aren't any glitches, we want to record it to get a feel for
   // how often we get no glitches vs the alternative.
   UMA_HISTOGRAM_COUNTS("Media.Audio.Capture.Glitches", glitches_detected_);
 
-  auto lost_frames_ms = (total_lost_frames_ * 1000) / format_.mSampleRate;
-  std::string log_message = base::StringPrintf(
-      "AU in: Total glitches=%d. Total frames lost=%d (%.0lf ms).",
-      glitches_detected_, total_lost_frames_, lost_frames_ms);
-  log_callback_.Run(log_message);
-
   if (glitches_detected_ != 0) {
+    auto lost_frames_ms = (total_lost_frames_ * 1000) / format_.mSampleRate;
     UMA_HISTOGRAM_LONG_TIMES("Media.Audio.Capture.LostFramesInMs",
                              base::TimeDelta::FromMilliseconds(lost_frames_ms));
     auto largest_glitch_ms =
         (largest_glitch_frames_ * 1000) / format_.mSampleRate;
     UMA_HISTOGRAM_CUSTOM_TIMES(
         "Media.Audio.Capture.LargestGlitchMs",
         base::TimeDelta::FromMilliseconds(largest_glitch_ms),
         base::TimeDelta::FromMilliseconds(1), base::TimeDelta::FromMinutes(1),
         50);
-    DLOG(WARNING) << log_message;
+    DLOG(WARNING) << "Total glitches=" << glitches_detected_
+                  << ". Total frames lost=" << total_lost_frames_ << " ("
+                  << lost_frames_ms << ")";
   }
 
   number_of_frames_provided_ = 0;
   glitches_detected_ = 0;
   last_sample_time_ = 0;
   last_number_of_frames_ = 0;
   total_lost_frames_ = 0;
   largest_glitch_frames_ = 0;
 }
 
 }  // namespace media