Audio codec usage statistics

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

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_
 
 #include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"

[hlundin-webrtc, 2016/05/13 11:18:19]

Still not needed... or?

[aleloi, 2016/05/16 08:02:24]

No, missed that one.

 
 namespace webrtc {
 
 template <typename T>
 typename AudioEncoderIsacT<T>::Config CreateIsacConfig(
     const CodecInst& codec_inst,
     const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo) {
   typename AudioEncoderIsacT<T>::Config config;
   config.bwinfo = bwinfo;
   config.payload_type = codec_inst.pltype;
@@ skipping 110 matching lines @@
   if (encoded_bytes == 0)
     return EncodedInfo();
 
   // Got enough input to produce a packet. Return the saved timestamp from
   // the first chunk of input that went into the packet.
   packet_in_progress_ = false;
   EncodedInfo info;
   info.encoded_bytes = encoded_bytes;
   info.encoded_timestamp = packet_timestamp_;
   info.payload_type = config_.payload_type;
+  info.encoder_type = AudioEncoder::CodecType::kIsac;
   return info;
 }
 
 template <typename T>
 void AudioEncoderIsacT<T>::Reset() {
   RecreateEncoderInstance(config_);
 }
 
 template <typename T>
 void AudioEncoderIsacT<T>::RecreateEncoderInstance(const Config& config) {
@@ skipping 23 matching lines @@
   // we get an encoding that isn't bit-for-bit identical with what a combined
   // encoder+decoder object produces.
   RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, config.sample_rate_hz));
 
   config_ = config;
 }
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_