Remove dead code

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/common_types.h"
 #include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/string_to_number.h"
 
 namespace webrtc {
-namespace {  // NOLINT (not a "regular" header file)
-int GetIsacMaxBitrate(int clockrate_hz) {
-  return (clockrate_hz == 32000) ? 56000 : 32000;
-}
-}  // namespace
 
 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;
   config.sample_rate_hz = codec_inst.plfreq;
   config.frame_size_ms =
       rtc::CheckedDivExact(1000 * codec_inst.pacsize, config.sample_rate_hz);
   config.adaptive_mode = (codec_inst.rate == -1);
   if (codec_inst.rate != -1)
     config.bit_rate = codec_inst.rate;
   return config;
 }
 
 template <typename T>
-typename AudioEncoderIsacT<T>::Config CreateIsacConfig(
-    int payload_type,
-    const SdpAudioFormat& format) {
-  typename AudioEncoderIsacT<T>::Config config;
-  config.payload_type = payload_type;
-  config.sample_rate_hz = format.clockrate_hz;
-
-  // We only support different frame sizes at 16000 Hz.
-  if (config.sample_rate_hz == 16000) {
-    auto ptime_iter = format.parameters.find("ptime");
-    if (ptime_iter != format.parameters.end()) {
-      auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
-      if (ptime && *ptime >= 60) {
-        config.frame_size_ms = 60;
-      } else {
-        config.frame_size_ms = 30;
-      }
-    }
-  }
-
-  // Set the default bitrate for ISAC to the maximum bitrate allowed at this
-  // clockrate. At this point, adaptive mode is not used by WebRTC.
-  config.bit_rate = GetIsacMaxBitrate(format.clockrate_hz);
-  return config;
-}
-
-template <typename T>
 bool AudioEncoderIsacT<T>::Config::IsOk() const {
   if (max_bit_rate < 32000 && max_bit_rate != -1)
     return false;
   if (max_payload_size_bytes < 120 && max_payload_size_bytes != -1)
     return false;
   if (adaptive_mode && !bwinfo)
     return false;
   switch (sample_rate_hz) {
     case 16000:
       if (max_bit_rate > 53400)
@@ skipping 20 matching lines @@
   RecreateEncoderInstance(config);
 }
 
 template <typename T>
 AudioEncoderIsacT<T>::AudioEncoderIsacT(
     const CodecInst& codec_inst,
     const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo)
     : AudioEncoderIsacT(CreateIsacConfig<T>(codec_inst, bwinfo)) {}
 
 template <typename T>
-AudioEncoderIsacT<T>::AudioEncoderIsacT(int payload_type,
-                                        const SdpAudioFormat& format)
-    : AudioEncoderIsacT(CreateIsacConfig<T>(payload_type, format)) {}
-
-template <typename T>
-rtc::Optional<AudioCodecInfo> AudioEncoderIsacT<T>::QueryAudioEncoder(
-    const SdpAudioFormat& format) {
-  if (STR_CASE_CMP(format.name.c_str(), GetPayloadName()) == 0) {
-    Config config = CreateIsacConfig<T>(0, format);
-    if (config.IsOk()) {
-      return rtc::Optional<AudioCodecInfo>(
-          {config.sample_rate_hz, 1, config.bit_rate, 10000,
-           GetIsacMaxBitrate(format.clockrate_hz)});
-    }
-  }
-  return rtc::Optional<AudioCodecInfo>();
-}
-
-template <typename T>
 AudioEncoderIsacT<T>::~AudioEncoderIsacT() {
   RTC_CHECK_EQ(0, T::Free(isac_state_));
 }
 
 template <typename T>
 int AudioEncoderIsacT<T>::SampleRateHz() const {
   return T::EncSampRate(isac_state_);
 }
 
 template <typename T>
@@ skipping 94 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_