Remove dead code

webrtc/api/audio_codecs/g722/audio_encoder_g722.cc

 /*
  *  Copyright (c) 2017 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/api/audio_codecs/g722/audio_encoder_g722.h"
 
 #include <memory>
 #include <vector>
 
+#include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h"
 #include "webrtc/rtc_base/ptr_util.h"
 #include "webrtc/rtc_base/safe_conversions.h"
+#include "webrtc/rtc_base/safe_minmax.h"
+#include "webrtc/rtc_base/string_to_number.h"
 
 namespace webrtc {
 
 rtc::Optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig(
     const SdpAudioFormat& format) {
-  return AudioEncoderG722Impl::SdpToConfig(format);
+  if (STR_CASE_CMP(format.name.c_str(), "g722") != 0 ||
+      format.clockrate_hz != 8000) {
+    return rtc::Optional<AudioEncoderG722Config>();
+  }
+
+  AudioEncoderG722Config config;
+  config.num_channels = rtc::checked_cast<int>(format.num_channels);
+  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 = rtc::SafeClamp<int>(whole_packets * 10, 10, 60);
+    }
+  }
+  return config.IsOk() ? rtc::Optional<AudioEncoderG722Config>(config)
+                       : rtc::Optional<AudioEncoderG722Config>();
 }
 
 void AudioEncoderG722::AppendSupportedEncoders(
     std::vector<AudioCodecSpec>* specs) {
   const SdpAudioFormat fmt = {"g722", 8000, 1};
   const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
   specs->push_back({fmt, info});
 }
 
 AudioCodecInfo AudioEncoderG722::QueryAudioEncoder(
     const AudioEncoderG722Config& config) {
   RTC_DCHECK(config.IsOk());
   return {16000, rtc::dchecked_cast<size_t>(config.num_channels),
           64000 * config.num_channels};
 }
 
 std::unique_ptr<AudioEncoder> AudioEncoderG722::MakeAudioEncoder(
     const AudioEncoderG722Config& config,
     int payload_type) {
   RTC_DCHECK(config.IsOk());
   return rtc::MakeUnique<AudioEncoderG722Impl>(config, payload_type);
 }
 
 }  // namespace webrtc