Remove dead code

webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.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/g711/audio_encoder_pcm.h"
 
 #include <algorithm>
 #include <limits>
 
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/g711/g711_interface.h"
 #include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/string_to_number.h"
 
 namespace webrtc {
 
 namespace {
 
 template <typename T>
 typename T::Config CreateConfig(const CodecInst& codec_inst) {
   typename T::Config config;
   config.frame_size_ms = codec_inst.pacsize / 8;
   config.num_channels = codec_inst.channels;
   config.payload_type = codec_inst.pltype;
   return config;
 }
 
-template <typename T>
-typename T::Config CreateConfig(int payload_type,
-                                const SdpAudioFormat& format) {
-  typename T::Config config;
-  config.frame_size_ms = 20;
-  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.num_channels = format.num_channels;
-  config.payload_type = payload_type;
-  return config;
-}
-
-template <typename T>
-rtc::Optional<AudioCodecInfo> QueryAudioEncoderImpl(
-    const SdpAudioFormat& format) {
-  if (STR_CASE_CMP(format.name.c_str(), T::GetPayloadName()) == 0 &&
-      format.clockrate_hz == 8000 && format.num_channels >= 1 &&
-      CreateConfig<T>(0, format).IsOk()) {
-    return rtc::Optional<AudioCodecInfo>({8000, format.num_channels, 64000});
-  }
-  return rtc::Optional<AudioCodecInfo>();
-}
-
 }  // namespace
 
 bool AudioEncoderPcm::Config::IsOk() const {
   return (frame_size_ms % 10 == 0) && (num_channels >= 1);
 }
 
 AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz)
     : sample_rate_hz_(sample_rate_hz),
       num_channels_(config.num_channels),
       payload_type_(config.payload_type),
@@ skipping 58 matching lines @@
   return info;
 }
 
 void AudioEncoderPcm::Reset() {
   speech_buffer_.clear();
 }
 
 AudioEncoderPcmA::AudioEncoderPcmA(const CodecInst& codec_inst)
     : AudioEncoderPcmA(CreateConfig<AudioEncoderPcmA>(codec_inst)) {}
 
-AudioEncoderPcmA::AudioEncoderPcmA(int payload_type,
-                                   const SdpAudioFormat& format)
-    : AudioEncoderPcmA(CreateConfig<AudioEncoderPcmA>(payload_type, format)) {}
-
-rtc::Optional<AudioCodecInfo> AudioEncoderPcmA::QueryAudioEncoder(
-    const SdpAudioFormat& format) {
-  return QueryAudioEncoderImpl<AudioEncoderPcmA>(format);
-}
-
 size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
                                     size_t input_len,
                                     uint8_t* encoded) {
   return WebRtcG711_EncodeA(audio, input_len, encoded);
 }
 
 size_t AudioEncoderPcmA::BytesPerSample() const {
   return 1;
 }
 
 AudioEncoder::CodecType AudioEncoderPcmA::GetCodecType() const {
   return AudioEncoder::CodecType::kPcmA;
 }
 
 AudioEncoderPcmU::AudioEncoderPcmU(const CodecInst& codec_inst)
     : AudioEncoderPcmU(CreateConfig<AudioEncoderPcmU>(codec_inst)) {}
 
-AudioEncoderPcmU::AudioEncoderPcmU(int payload_type,
-                                   const SdpAudioFormat& format)
-    : AudioEncoderPcmU(CreateConfig<AudioEncoderPcmU>(payload_type, format)) {}
-
-rtc::Optional<AudioCodecInfo> AudioEncoderPcmU::QueryAudioEncoder(
-    const SdpAudioFormat& format) {
-  return QueryAudioEncoderImpl<AudioEncoderPcmU>(format);
-}
-
 size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
                                     size_t input_len,
                                     uint8_t* encoded) {
   return WebRtcG711_EncodeU(audio, input_len, encoded);
 }
 
 size_t AudioEncoderPcmU::BytesPerSample() const {
   return 1;
 }
 
 AudioEncoder::CodecType AudioEncoderPcmU::GetCodecType() const {
   return AudioEncoder::CodecType::kPcmU;
 }
 
 }  // namespace webrtc