Audio codec usage statistics

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

 /*
  *  Copyright (c) 2014 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/codecs/opus/audio_encoder_opus.h"
 
 #include <algorithm>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"

[hlundin-webrtc, 2016/05/13 09:54:23]

Not needed.

 
 namespace webrtc {
 
 namespace {
 
 const int kSampleRateHz = 48000;
 const int kMinBitrateBps = 500;
 const int kMaxBitrateBps = 512000;
 
 AudioEncoderOpus::Config CreateConfig(const CodecInst& codec_inst) {
@@ skipping 176 matching lines @@
             RTC_CHECK_GE(status, 0);  // Fails only if fed invalid data.
 
             return static_cast<size_t>(status);
           });
   input_buffer_.clear();
 
   info.encoded_timestamp = first_timestamp_in_buffer_;
   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 = AudioEncoder::CodecType::kOpus;
   return info;
 }
 
 size_t AudioEncoderOpus::Num10msFramesPerPacket() const {
   return static_cast<size_t>(rtc::CheckedDivExact(config_.frame_size_ms, 10));
 }
 
 size_t AudioEncoderOpus::SamplesPer10msFrame() const {
   return rtc::CheckedDivExact(kSampleRateHz, 100) * config_.num_channels;
 }
@@ skipping 35 matching lines @@
     RTC_CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_));
   }
   RTC_CHECK_EQ(0,
                WebRtcOpus_SetPacketLossRate(
                    inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5)));
   config_ = config;
   return true;
 }
 
 }  // namespace webrtc