Audio codec usage statistics

webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.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/g722/audio_encoder_g722.h"
 
 #include <limits>
 #include "webrtc/base/checks.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"

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

You don't have to include audio_encoder.h. From the style guide: "any includes
present in the related header do not need to be included again in the related
cc."
(https://google.github.io/styleguide/cppguide.html#Names_and_Order_of_Includes).

[aleloi, 2016/05/13 10:56:07]

Done.

 
 namespace webrtc {
 
 namespace {
 
 const size_t kSampleRateHz = 16000;
 
 AudioEncoderG722::Config CreateConfig(const CodecInst& codec_inst) {
   AudioEncoderG722::Config config;
   config.num_channels = codec_inst.channels;
@@ skipping 111 matching lines @@
           for (size_t j = 0; j < num_channels_; ++j)
             encoded[i * num_channels_ + j] =
                 interleave_buffer_.data()[2 * j] << 4 |
                 interleave_buffer_.data()[2 * j + 1];
         }
 
         return bytes_to_encode;
       });
   info.encoded_timestamp = first_timestamp_in_buffer_;
   info.payload_type = payload_type_;
+  info.encoder_type = CodecType::kG722;
   return info;
 }
 
 AudioEncoderG722::EncoderState::EncoderState() {
   RTC_CHECK_EQ(0, WebRtcG722_CreateEncoder(&encoder));
 }
 
 AudioEncoderG722::EncoderState::~EncoderState() {
   RTC_CHECK_EQ(0, WebRtcG722_FreeEncoder(encoder));
 }
 
 size_t AudioEncoderG722::SamplesPerChannel() const {
   return kSampleRateHz / 100 * num_10ms_frames_per_packet_;
 }
 
 }  // namespace webrtc