Remove dead code

webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.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/pcm16b/audio_encoder_pcm16b.h"
 
 #include <algorithm>
 
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "webrtc/rtc_base/checks.h"
 #include "webrtc/rtc_base/safe_conversions.h"
-#include "webrtc/rtc_base/string_to_number.h"
 
 namespace webrtc {
 
 size_t AudioEncoderPcm16B::EncodeCall(const int16_t* audio,
                                       size_t input_len,
                                       uint8_t* encoded) {
   return WebRtcPcm16b_Encode(audio, input_len, encoded);
 }
 
 size_t AudioEncoderPcm16B::BytesPerSample() const {
   return 2;
 }
 
 AudioEncoder::CodecType AudioEncoderPcm16B::GetCodecType() const {
   return CodecType::kOther;
 }
 
 namespace {
+
 AudioEncoderPcm16B::Config CreateConfig(const CodecInst& codec_inst) {
   AudioEncoderPcm16B::Config config;
   config.num_channels = codec_inst.channels;
   config.sample_rate_hz = codec_inst.plfreq;
   config.frame_size_ms = rtc::CheckedDivExact(
       codec_inst.pacsize, rtc::CheckedDivExact(config.sample_rate_hz, 1000));
   config.payload_type = codec_inst.pltype;
   return config;
 }
 
-AudioEncoderPcm16B::Config CreateConfig(int payload_type,
-                                        const SdpAudioFormat& format) {
-  AudioEncoderPcm16B::Config config;
-  config.num_channels = format.num_channels;
-  config.sample_rate_hz = format.clockrate_hz;
-  config.frame_size_ms = 10;
-  auto ptime_iter = format.parameters.find("ptime");
-  if (ptime_iter != format.parameters.end()) {
-    auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
-    if (ptime && *ptime > 0) {
-      const int whole_packets = *ptime / 10;
-      config.frame_size_ms = std::max(10, std::min(whole_packets * 10, 60));
-    }
-  }
-  config.payload_type = payload_type;
-  return config;
-}
 }  // namespace
 
 bool AudioEncoderPcm16B::Config::IsOk() const {
   if ((sample_rate_hz != 8000) && (sample_rate_hz != 16000) &&
       (sample_rate_hz != 32000) && (sample_rate_hz != 48000))
     return false;
   return AudioEncoderPcm::Config::IsOk();
 }
 
 AudioEncoderPcm16B::AudioEncoderPcm16B(const CodecInst& codec_inst)
     : AudioEncoderPcm16B(CreateConfig(codec_inst)) {}
 
-AudioEncoderPcm16B::AudioEncoderPcm16B(int payload_type,
-                                       const SdpAudioFormat& format)
-    : AudioEncoderPcm16B(CreateConfig(payload_type, format)) {}
-
-rtc::Optional<AudioCodecInfo> AudioEncoderPcm16B::QueryAudioEncoder(
-    const SdpAudioFormat& format) {
-  if (STR_CASE_CMP(format.name.c_str(), GetPayloadName()) == 0 &&
-      format.num_channels >= 1) {
-    Config config = CreateConfig(0, format);
-    if (config.IsOk()) {
-      constexpr int bits_per_sample = 16;
-      return rtc::Optional<AudioCodecInfo>(
-          {config.sample_rate_hz, config.num_channels,
-           config.sample_rate_hz * bits_per_sample *
-               rtc::dchecked_cast<int>(config.num_channels)});
-    }
-  }
-  return rtc::Optional<AudioCodecInfo>();
-}
-
 }  // namespace webrtc