Adding time profiling support to AudioFrame

webrtc/modules/include/module_common_types.h

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #ifndef WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
 #define WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
 
 #include <assert.h>
 #include <string.h>  // memcpy
 
 #include <algorithm>
 #include <limits>
 
 #include "webrtc/api/optional.h"
 #include "webrtc/api/video/video_rotation.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/video_coding/codecs/h264/include/h264_globals.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8_globals.h"
 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9_globals.h"
 #include "webrtc/rtc_base/constructormagic.h"
 #include "webrtc/rtc_base/deprecation.h"
 #include "webrtc/rtc_base/safe_conversions.h"
+#include "webrtc/rtc_base/timeutils.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 struct RTPAudioHeader {
   uint8_t numEnergy;                  // number of valid entries in arrOfEnergy
   uint8_t arrOfEnergy[kRtpCsrcSize];  // one energy byte (0-9) per channel
   bool isCNG;                         // is this CNG
   size_t channel;                     // number of channels 2 = stereo
 };
@@ skipping 290 matching lines @@
   // ResetWithoutMuting() to skip this wasteful zeroing.
   void ResetWithoutMuting();
 
   void UpdateFrame(int id, uint32_t timestamp, const int16_t* data,
                    size_t samples_per_channel, int sample_rate_hz,
                    SpeechType speech_type, VADActivity vad_activity,
                    size_t num_channels = 1);
 
   void CopyFrom(const AudioFrame& src);
 
+  // Sets a wall-time clock timestamp in milliseconds to be used for profiling
+  // of time between two points in the audio chain.
+  // Example:
+  //   t0: UpdateProfileTime()
+  //   t1: TimeSinceLastProfile() => t1 - t0 [msec]
+  void UpdateProfileTime();

[hlundin-webrtc, 2017/09/14 13:34:33]

Suggest UpdateProfileTimestamp to match variable name.

[henrika_webrtc, 2017/09/15 13:33:57]

Done.

+  // Returns the time difference between now and when UpdateProfileTime() was
+  // last called. Returns -1 if UpdateProfileTime() has not yet been called.
+  int64_t TimeSinceLastProfile() const;

[hlundin-webrtc, 2017/09/14 13:34:34]

Suggest ElapsedProfileTimeMs().

[henrika_webrtc, 2017/09/15 13:33:57]

Done.

+
   // data() returns a zeroed static buffer if the frame is muted.
   // mutable_frame() always returns a non-static buffer; the first call to
   // mutable_frame() zeros the non-static buffer and marks the frame unmuted.
   const int16_t* data() const;
   int16_t* mutable_data();
 
   // Prefer to mute frames using AudioFrameOperations::Mute.
   void Mute();
   // Frame is muted by default.
   bool muted() const;
@@ skipping 12 matching lines @@
   // -1 represents an uninitialized value.
   int64_t elapsed_time_ms_ = -1;
   // NTP time of the estimated capture time in local timebase in milliseconds.
   // -1 represents an uninitialized value.
   int64_t ntp_time_ms_ = -1;
   size_t samples_per_channel_ = 0;
   int sample_rate_hz_ = 0;
   size_t num_channels_ = 0;
   SpeechType speech_type_ = kUndefined;
   VADActivity vad_activity_ = kVadUnknown;
+  // Monotonically increasing timestamp intended for profiling of audio frames.
+  // Typically used for measuring elapsed time between two different points in
+  // the audio path. No lock is used to save resources and we are thread safe
+  // by design. Also, rtc::Optional is not used since it will cause a "complex

[hlundin-webrtc, 2017/09/14 13:45:39]

+kwiberg@, would you consider this reason enough to create a corresponding .cc
file, or is it too much work?

[henrika_webrtc, 2017/09/15 13:33:57]

IMHO, adding a cc-file for this functionality only feels like overkill.

[kwiberg-webrtc, 2017/09/15 17:52:29]

If build targets are set up properly (as they should be, and almost always are
nowadays), adding a .cc file is very little work. So if using Optional is the
right thing to do, don't avoid it just because you'll need a .cc file.

+  // class/struct needs an explicit out-of-line destructor" build error.
+  int64_t profile_time_stamp_ms_ = 0;

[hlundin-webrtc, 2017/09/14 13:34:33]

The convention in this file is to write timestamp as one word. Thus,
profile_timestamp_ms_.

[henrika_webrtc, 2017/09/15 13:33:57]

Done.

 
  private:
   // A permamently zeroed out buffer to represent muted frames. This is a
   // header-only class, so the only way to avoid creating a separate empty
   // buffer per translation unit is to wrap a static in an inline function.
   static const int16_t* empty_data() {
     static const int16_t kEmptyData[kMaxDataSizeSamples] = {0};
     static_assert(sizeof(kEmptyData) == kMaxDataSizeBytes, "kMaxDataSizeBytes");
     return kEmptyData;
   }
@@ skipping 19 matching lines @@
   // TODO(wu): Zero is a valid value for |timestamp_|. We should initialize
   // to an invalid value, or add a new member to indicate invalidity.
   timestamp_ = 0;
   elapsed_time_ms_ = -1;
   ntp_time_ms_ = -1;
   samples_per_channel_ = 0;
   sample_rate_hz_ = 0;
   num_channels_ = 0;
   speech_type_ = kUndefined;
   vad_activity_ = kVadUnknown;
+  profile_time_stamp_ms_ = 0;
 }
 
 inline void AudioFrame::UpdateFrame(int id,
                                     uint32_t timestamp,
                                     const int16_t* data,
                                     size_t samples_per_channel,
                                     int sample_rate_hz,
                                     SpeechType speech_type,
                                     VADActivity vad_activity,
                                     size_t num_channels) {
@@ skipping 30 matching lines @@
   num_channels_ = src.num_channels_;
 
   const size_t length = samples_per_channel_ * num_channels_;
   assert(length <= kMaxDataSizeSamples);
   if (!src.muted()) {
     memcpy(data_, src.data(), sizeof(int16_t) * length);
     muted_ = false;
   }
 }
 
+inline void AudioFrame::UpdateProfileTime() {
+  {

[hlundin-webrtc, 2017/09/14 13:34:33]

Why the extra braces?

[henrika_webrtc, 2017/09/15 13:33:57]

My bad

+    profile_time_stamp_ms_ = rtc::TimeMillis()  ;

[hlundin-webrtc, 2017/09/14 13:34:33]

Delete space before ;

[henrika_webrtc, 2017/09/15 13:33:57]

Done.

+  }
+}
+
+inline int64_t AudioFrame::TimeSinceLastProfile() const {
+  if (profile_time_stamp_ms_ == 0) {
+    // Profiling has not been activated.
+    return -1;
+  }
+  return rtc::TimeSince(profile_time_stamp_ms_);
+}
+
 inline const int16_t* AudioFrame::data() const {
   return muted_ ? empty_data() : data_;
 }
 
 // TODO(henrik.lundin) Can we skip zeroing the buffer?
 // See https://bugs.chromium.org/p/webrtc/issues/detail?id=5647.
 inline int16_t* AudioFrame::mutable_data() {
   if (muted_) {
     memset(data_, 0, kMaxDataSizeBytes);
     muted_ = false;
@@ skipping 168 matching lines @@
   static constexpr int kNotAProbe = -1;
   int send_bitrate_bps = -1;
   int probe_cluster_id = kNotAProbe;
   int probe_cluster_min_probes = -1;
   int probe_cluster_min_bytes = -1;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_