Remove dead code

webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.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/g722/audio_encoder_g722.h"
 
 #include <algorithm>
 
 #include <limits>
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/g722/g722_interface.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 size_t kSampleRateHz = 16000;
 
 AudioEncoderG722Config CreateConfig(const CodecInst& codec_inst) {
   AudioEncoderG722Config config;
   config.num_channels = rtc::dchecked_cast<int>(codec_inst.channels);
   config.frame_size_ms = codec_inst.pacsize / 16;
   return config;
 }
 
 }  // namespace
 
-rtc::Optional<AudioEncoderG722Config> AudioEncoderG722Impl::SdpToConfig(
-    const SdpAudioFormat& 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::dchecked_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 = std::max(10, std::min(whole_packets * 10, 60));
-    }
-  }
-  return config.IsOk() ? rtc::Optional<AudioEncoderG722Config>(config)
-                       : rtc::Optional<AudioEncoderG722Config>();
-}
-
 AudioEncoderG722Impl::AudioEncoderG722Impl(const AudioEncoderG722Config& config,
                                            int payload_type)
     : num_channels_(config.num_channels),
       payload_type_(payload_type),
       num_10ms_frames_per_packet_(
           static_cast<size_t>(config.frame_size_ms / 10)),
       num_10ms_frames_buffered_(0),
       first_timestamp_in_buffer_(0),
       encoders_(new EncoderState[num_channels_]),
       interleave_buffer_(2 * num_channels_) {
   RTC_CHECK(config.IsOk());
   const size_t samples_per_channel =
       kSampleRateHz / 100 * num_10ms_frames_per_packet_;
   for (size_t i = 0; i < num_channels_; ++i) {
     encoders_[i].speech_buffer.reset(new int16_t[samples_per_channel]);
     encoders_[i].encoded_buffer.SetSize(samples_per_channel / 2);
   }
   Reset();
 }
 
 AudioEncoderG722Impl::AudioEncoderG722Impl(const CodecInst& codec_inst)
     : AudioEncoderG722Impl(CreateConfig(codec_inst), codec_inst.pltype) {}
 
-AudioEncoderG722Impl::AudioEncoderG722Impl(int payload_type,
-                                           const SdpAudioFormat& format)
-    : AudioEncoderG722Impl(*SdpToConfig(format), payload_type) {}
-
 AudioEncoderG722Impl::~AudioEncoderG722Impl() = default;
 
-rtc::Optional<AudioCodecInfo> AudioEncoderG722Impl::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 && config_opt) {
-      RTC_DCHECK(config_opt->IsOk());
-      return rtc::Optional<AudioCodecInfo>(
-          {rtc::dchecked_cast<int>(kSampleRateHz),
-           rtc::dchecked_cast<size_t>(config_opt->num_channels), 64000});
-    }
-  }
-  return rtc::Optional<AudioCodecInfo>();
-}
-
 int AudioEncoderG722Impl::SampleRateHz() const {
   return kSampleRateHz;
 }
 
 size_t AudioEncoderG722Impl::NumChannels() const {
   return num_channels_;
 }
 
 int AudioEncoderG722Impl::RtpTimestampRateHz() const {
   // The RTP timestamp rate for G.722 is 8000 Hz, even though it is a 16 kHz
@@ skipping 82 matching lines @@
 
 AudioEncoderG722Impl::EncoderState::~EncoderState() {
   RTC_CHECK_EQ(0, WebRtcG722_FreeEncoder(encoder));
 }
 
 size_t AudioEncoderG722Impl::SamplesPerChannel() const {
   return kSampleRateHz / 100 * num_10ms_frames_per_packet_;
 }
 
 }  // namespace webrtc