Remove dead code

webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.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/ilbc/audio_encoder_ilbc.h"
 
 #include <algorithm>
 #include <limits>
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
 #include "webrtc/rtc_base/checks.h"
 #include "webrtc/rtc_base/safe_conversions.h"
-#include "webrtc/rtc_base/string_to_number.h"
 
 namespace webrtc {
 
 namespace {
 
 const int kSampleRateHz = 8000;
 
 AudioEncoderIlbcConfig CreateConfig(const CodecInst& codec_inst) {
   AudioEncoderIlbcConfig config;
   config.frame_size_ms = codec_inst.pacsize / 8;
@@ skipping 10 matching lines @@
     case 60:
       // 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
       return 13333;
     default:
       FATAL();
   }
 }
 
 }  // namespace
 
-rtc::Optional<AudioEncoderIlbcConfig> AudioEncoderIlbcImpl::SdpToConfig(
-    const SdpAudioFormat& format) {
-  if (STR_CASE_CMP(format.name.c_str(), "ilbc") != 0 ||
-      format.clockrate_hz != 8000 || format.num_channels != 1) {
-    return rtc::Optional<AudioEncoderIlbcConfig>();
-  }
-
-  AudioEncoderIlbcConfig config;
-  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(20, std::min(whole_packets * 10, 60));
-    }
-  }
-  return config.IsOk() ? rtc::Optional<AudioEncoderIlbcConfig>(config)
-                       : rtc::Optional<AudioEncoderIlbcConfig>();
-}
-
 AudioEncoderIlbcImpl::AudioEncoderIlbcImpl(const AudioEncoderIlbcConfig& config,
                                            int payload_type)
     : frame_size_ms_(config.frame_size_ms),
       payload_type_(payload_type),
       num_10ms_frames_per_packet_(
           static_cast<size_t>(config.frame_size_ms / 10)),
       encoder_(nullptr) {
   RTC_CHECK(config.IsOk());
   Reset();
 }
 
 AudioEncoderIlbcImpl::AudioEncoderIlbcImpl(const CodecInst& codec_inst)
     : AudioEncoderIlbcImpl(CreateConfig(codec_inst), codec_inst.pltype) {}
 
-AudioEncoderIlbcImpl::AudioEncoderIlbcImpl(int payload_type,
-                                           const SdpAudioFormat& format)
-    : AudioEncoderIlbcImpl(*SdpToConfig(format), payload_type) {}
-
 AudioEncoderIlbcImpl::~AudioEncoderIlbcImpl() {
   RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
 }
 
-rtc::Optional<AudioCodecInfo> AudioEncoderIlbcImpl::QueryAudioEncoder(
-    const SdpAudioFormat& format) {
-  if (STR_CASE_CMP(format.name.c_str(), GetPayloadName()) == 0) {
-    const auto config_opt = SdpToConfig(format);
-    if (format.clockrate_hz == 8000 && format.num_channels == 1 &&
-        config_opt) {
-      RTC_DCHECK(config_opt->IsOk());
-      return rtc::Optional<AudioCodecInfo>(
-          {rtc::dchecked_cast<int>(kSampleRateHz), 1,
-           GetIlbcBitrate(config_opt->frame_size_ms)});
-    }
-  }
-  return rtc::Optional<AudioCodecInfo>();
-}
-
 int AudioEncoderIlbcImpl::SampleRateHz() const {
   return kSampleRateHz;
 }
 
 size_t AudioEncoderIlbcImpl::NumChannels() const {
   return 1;
 }
 
 size_t AudioEncoderIlbcImpl::Num10MsFramesInNextPacket() const {
   return num_10ms_frames_per_packet_;
@@ skipping 69 matching lines @@
   switch (num_10ms_frames_per_packet_) {
     case 2:   return 38;
     case 3:   return 50;
     case 4:   return 2 * 38;
     case 6:   return 2 * 50;
     default:  FATAL();
   }
 }
 
 }  // namespace webrtc