Remove dead code

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

Index: webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
index bda379c8d256e0da8fcb064321c48f7e05d474a2..854f2eef0cfc059921c9ee0c0b38077268b0188d 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
@@ -13,14 +13,8 @@
 
 #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(
@@ -38,33 +32,6 @@ typename AudioEncoderIsacT<T>::Config CreateIsacConfig(
   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)
@@ -105,25 +72,6 @@ AudioEncoderIsacT<T>::AudioEncoderIsacT(
     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_));