Audio codec usage statistics

webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc

 /*
  *  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.
  */
 
 #include "webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h"
 
 #include <assert.h>
 #include <stdlib.h>
 #include <vector>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
 #include "webrtc/modules/audio_coding/acm2/call_statistics.h"
 #include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 #include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/typedefs.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 
 namespace webrtc {
 
+namespace {
+
+// Adds a codec usage sample to the histogram.
+void UpdateCodecTypeHistogram(size_t codec_type) {
+  RTC_HISTOGRAM_ENUMERATION(
+      "WebRTC.Audio.Encoder.CodecType", codec_type,
+      static_cast<size_t>(
+          webrtc::AudioEncoder::CodecType::kMaxLoggedAudioCodecNames));
+}
+}  // namespace
+
 namespace acm2 {
 
 struct EncoderFactory {
   AudioEncoder* external_speech_encoder = nullptr;
   CodecManager codec_manager;
   RentACodec rent_a_codec;
 };
 
 namespace {
 
@@ skipping 136 matching lines @@
       expected_in_ts_(0xD87F3F9F),
       receiver_(config),
       bitrate_logger_("WebRTC.Audio.TargetBitrateInKbps"),
       encoder_factory_(new EncoderFactory),
       encoder_stack_(nullptr),
       previous_pltype_(255),
       receiver_initialized_(false),
       first_10ms_data_(false),
       first_frame_(true),
       packetization_callback_(NULL),
-      vad_callback_(NULL) {
+      vad_callback_(NULL),
+      codec_histogram_bins_log_(),
+      number_of_consecutive_empty_packets_(0) {
   if (InitializeReceiverSafe() < 0) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                  "Cannot initialize receiver");
   }
   WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceAudioCoding, id_, "Created");
 }
 
 AudioCodingModuleImpl::~AudioCodingModuleImpl() = default;
 
 int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
@@ skipping 25 matching lines @@
                                                input_data.length_per_channel),
       &encode_buffer_);
 
   bitrate_logger_.MaybeLog(encoder_stack_->GetTargetBitrate() / 1000);
   if (encode_buffer_.size() == 0 && !encoded_info.send_even_if_empty) {
     // Not enough data.
     return 0;
   }
   previous_pltype = previous_pltype_;  // Read it while we have the critsect.
 
+  // Log codec type to histogram once every 500 packets.
+  if (encoded_info.encoded_bytes == 0) {
+    ++number_of_consecutive_empty_packets_;
+  } else {
+    size_t codec_type = static_cast<size_t>(encoded_info.encoder_type);
+    codec_histogram_bins_log_[codec_type] +=
+        number_of_consecutive_empty_packets_;

[ossu, 2016/05/12 15:18:46]

Well, now, this is clearly wrong. If we're not getting any empty packets, we'll
be adding zero here each time. Actually, we'll always be adding one less than
we'd want to.

[aleloi, 2016/05/12 15:30:14]

Acknowledged.

+    number_of_consecutive_empty_packets_ = 0;
+    if (codec_histogram_bins_log_[codec_type] >= 500) {
+      codec_histogram_bins_log_[codec_type] -= 500;
+      UpdateCodecTypeHistogram(codec_type);
+    }
+  }
+
   RTPFragmentationHeader my_fragmentation;
   ConvertEncodedInfoToFragmentationHeader(encoded_info, &my_fragmentation);
   FrameType frame_type;
   if (encode_buffer_.size() == 0 && encoded_info.send_even_if_empty) {
     frame_type = kEmptyFrame;
     encoded_info.payload_type = previous_pltype;
   } else {
     RTC_DCHECK_GT(encode_buffer_.size(), 0u);
     frame_type = encoded_info.speech ? kAudioFrameSpeech : kAudioFrameCN;
   }
@@ skipping 694 matching lines @@
   return receiver_.LeastRequiredDelayMs();
 }
 
 void AudioCodingModuleImpl::GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const {
   receiver_.GetDecodingCallStatistics(call_stats);
 }
 
 }  // namespace acm2
 }  // namespace webrtc