Added UMA logging for audio codec usage. A histogram statistic named "WebRTC.Audio.Encoder.CodecType" has been created.

Review-Url: https://codereview.webrtc.org/1967503002
Cr-Commit-Position: refs/heads/master@{#12760}

webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc

@@ -29,6 +29,18 @@
 
 namespace webrtc {
 
+namespace {
+
+// Adds a codec usage sample to the histogram.
+void UpdateCodecTypeHistogram(size_t codec_type) {
+  RTC_HISTOGRAM_ENUMERATION(
+      "WebRTC.Audio.Encoder.CodecType", static_cast<int>(codec_type),
+      static_cast<int>(
+          webrtc::AudioEncoder::CodecType::kMaxLoggedAudioCodecTypes));
+}
+
+}  // namespace
+
 namespace acm2 {
 
 struct EncoderFactory {
@@ -185,7 +197,9 @@ AudioCodingModuleImpl::AudioCodingModuleImpl(
       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");
@@ -231,6 +245,20 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
   }
   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_ + 1;
+    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;

webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h

@@ -23,6 +23,7 @@
 #include "webrtc/modules/audio_coding/acm2/acm_receiver.h"
 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
 #include "webrtc/modules/audio_coding/acm2/codec_manager.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 
 namespace webrtc {
 
@@ -298,6 +299,10 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
   AudioPacketizationCallback* packetization_callback_
       GUARDED_BY(callback_crit_sect_);
   ACMVADCallback* vad_callback_ GUARDED_BY(callback_crit_sect_);
+
+  int codec_histogram_bins_log_[static_cast<size_t>(
+      AudioEncoder::CodecType::kMaxLoggedAudioCodecTypes)];
+  int number_of_consecutive_empty_packets_;
 };
 
 }  // namespace acm2

webrtc/modules/audio_coding/codecs/audio_encoder.h

@@ -25,12 +25,32 @@ namespace webrtc {
 // type must have an implementation of this class.
 class AudioEncoder {
  public:
+  // Used for UMA logging of codec usage. The same codecs, with the
+  // same values, must be listed in
+  // src/tools/metrics/histograms/histograms.xml in chromium to log
+  // correct values.
+  enum class CodecType {
+    kOther = 0,  // Codec not specified, and/or not listed in this enum
+    kOpus = 1,
+    kIsac = 2,
+    kPcmA = 3,
+    kPcmU = 4,
+    kG722 = 5,
+    kIlbc = 6,
+
+    // Number of histogram bins in the UMA logging of codec types. The
+    // total number of different codecs that are logged cannot exceed this
+    // number.
+    kMaxLoggedAudioCodecTypes
+  };
+
   struct EncodedInfoLeaf {
     size_t encoded_bytes = 0;
     uint32_t encoded_timestamp = 0;
     int payload_type = 0;
     bool send_even_if_empty = false;
     bool speech = true;
+    CodecType encoder_type = CodecType::kOther;
   };
 
   // This is the main struct for auxiliary encoding information. Each encoded

webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc

@@ -96,6 +96,7 @@ AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeImpl(
                                               encoded.data());
                           });
   speech_buffer_.clear();
+  info.encoder_type = GetCodecType();
   return info;
 }
 
@@ -116,6 +117,10 @@ size_t AudioEncoderPcmA::BytesPerSample() const {
   return 1;
 }
 
+AudioEncoder::CodecType AudioEncoderPcmA::GetCodecType() const {
+  return AudioEncoder::CodecType::kPcmA;
+}
+
 AudioEncoderPcmU::AudioEncoderPcmU(const CodecInst& codec_inst)
     : AudioEncoderPcmU(CreateConfig<AudioEncoderPcmU>(codec_inst)) {}
 
@@ -129,4 +134,8 @@ size_t AudioEncoderPcmU::BytesPerSample() const {
   return 1;
 }
 
+AudioEncoder::CodecType AudioEncoderPcmU::GetCodecType() const {
+  return AudioEncoder::CodecType::kPcmU;
+}
+
 }  // namespace webrtc

webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h

@@ -55,6 +55,10 @@ class AudioEncoderPcm : public AudioEncoder {
 
   virtual size_t BytesPerSample() const = 0;
 
+  // Used to set EncodedInfoLeaf::encoder_type in
+  // AudioEncoderPcm::EncodeImpl
+  virtual AudioEncoder::CodecType GetCodecType() const = 0;
+
  private:
   const int sample_rate_hz_;
   const size_t num_channels_;
@@ -84,6 +88,8 @@ class AudioEncoderPcmA final : public AudioEncoderPcm {
 
   size_t BytesPerSample() const override;
 
+  AudioEncoder::CodecType GetCodecType() const override;
+
  private:
   static const int kSampleRateHz = 8000;
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcmA);
@@ -106,6 +112,8 @@ class AudioEncoderPcmU final : public AudioEncoderPcm {
 
   size_t BytesPerSample() const override;
 
+  AudioEncoder::CodecType GetCodecType() const override;
+
  private:
   static const int kSampleRateHz = 8000;
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcmU);

webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc

@@ -145,6 +145,7 @@ AudioEncoder::EncodedInfo AudioEncoderG722::EncodeImpl(
       });
   info.encoded_timestamp = first_timestamp_in_buffer_;
   info.payload_type = payload_type_;
+  info.encoder_type = CodecType::kG722;
   return info;
 }
 

webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc

@@ -127,6 +127,7 @@ AudioEncoder::EncodedInfo AudioEncoderIlbc::EncodeImpl(
   info.encoded_bytes = encoded_bytes;
   info.encoded_timestamp = first_timestamp_in_buffer_;
   info.payload_type = config_.payload_type;
+  info.encoder_type = CodecType::kIlbc;
   return info;
 }
 

webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h

@@ -145,6 +145,7 @@ AudioEncoder::EncodedInfo AudioEncoderIsacT<T>::EncodeImpl(
   info.encoded_bytes = encoded_bytes;
   info.encoded_timestamp = packet_timestamp_;
   info.payload_type = config_.payload_type;
+  info.encoder_type = CodecType::kIsac;
   return info;
 }
 

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc

@@ -212,6 +212,7 @@ AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeImpl(
   info.payload_type = config_.payload_type;
   info.send_even_if_empty = true;  // Allows Opus to send empty packets.
   info.speech = (info.encoded_bytes > 0);
+  info.encoder_type = CodecType::kOpus;
   return info;
 }
 

webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc

@@ -26,6 +26,10 @@ size_t AudioEncoderPcm16B::BytesPerSample() const {
   return 2;
 }
 
+AudioEncoder::CodecType AudioEncoderPcm16B::GetCodecType() const {
+  return CodecType::kOther;
+}
+
 namespace {
 AudioEncoderPcm16B::Config CreateConfig(const CodecInst& codec_inst) {
   AudioEncoderPcm16B::Config config;

webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h

@@ -39,6 +39,8 @@ class AudioEncoderPcm16B final : public AudioEncoderPcm {
 
   size_t BytesPerSample() const override;
 
+  AudioEncoder::CodecType GetCodecType() const override;
+
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcm16B);
 };