Expose ILBC codec in webrtc/api/audio_codecs/

webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc

 /*
  *  Copyright (c) 2015 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_decoder_ilbc.h"
 
 #include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
 #include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
 
 namespace webrtc {
 
-AudioDecoderIlbc::AudioDecoderIlbc() {
+AudioDecoderIlbcImpl::AudioDecoderIlbcImpl() {
   WebRtcIlbcfix_DecoderCreate(&dec_state_);
   WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
 }
 
-AudioDecoderIlbc::~AudioDecoderIlbc() {
+AudioDecoderIlbcImpl::~AudioDecoderIlbcImpl() {
   WebRtcIlbcfix_DecoderFree(dec_state_);
 }
 
-bool AudioDecoderIlbc::HasDecodePlc() const {
+bool AudioDecoderIlbcImpl::HasDecodePlc() const {
   return true;
 }
 
-int AudioDecoderIlbc::DecodeInternal(const uint8_t* encoded,
+int AudioDecoderIlbcImpl::DecodeInternal(const uint8_t* encoded,
                                      size_t encoded_len,
                                      int sample_rate_hz,
                                      int16_t* decoded,
                                      SpeechType* speech_type) {
   RTC_DCHECK_EQ(sample_rate_hz, 8000);
   int16_t temp_type = 1;  // Default is speech.
   int ret = WebRtcIlbcfix_Decode(dec_state_, encoded, encoded_len, decoded,
                                  &temp_type);
   *speech_type = ConvertSpeechType(temp_type);
   return ret;
 }
 
-size_t AudioDecoderIlbc::DecodePlc(size_t num_frames, int16_t* decoded) {
+size_t AudioDecoderIlbcImpl::DecodePlc(size_t num_frames, int16_t* decoded) {
   return WebRtcIlbcfix_NetEqPlc(dec_state_, decoded, num_frames);
 }
 
-void AudioDecoderIlbc::Reset() {
+void AudioDecoderIlbcImpl::Reset() {
   WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
 }
 
-std::vector<AudioDecoder::ParseResult> AudioDecoderIlbc::ParsePayload(
+std::vector<AudioDecoder::ParseResult> AudioDecoderIlbcImpl::ParsePayload(
     rtc::Buffer&& payload,
     uint32_t timestamp) {
   std::vector<ParseResult> results;
   size_t bytes_per_frame;
   int timestamps_per_frame;
   if (payload.size() >= 950) {
-    LOG(LS_WARNING) << "AudioDecoderIlbc::ParsePayload: Payload too large";
+    LOG(LS_WARNING) << "AudioDecoderIlbcImpl::ParsePayload: Payload too large";
     return results;
   }
   if (payload.size() % 38 == 0) {
     // 20 ms frames.
     bytes_per_frame = 38;
     timestamps_per_frame = 160;
   } else if (payload.size() % 50 == 0) {
     // 30 ms frames.
     bytes_per_frame = 50;
     timestamps_per_frame = 240;
   } else {
-    LOG(LS_WARNING) << "AudioDecoderIlbc::ParsePayload: Invalid payload";
+    LOG(LS_WARNING) << "AudioDecoderIlbcImpl::ParsePayload: Invalid payload";
     return results;
   }
 
   RTC_DCHECK_EQ(0, payload.size() % bytes_per_frame);
   if (payload.size() == bytes_per_frame) {
     std::unique_ptr<EncodedAudioFrame> frame(
         new LegacyEncodedAudioFrame(this, std::move(payload)));
     results.emplace_back(timestamp, 0, std::move(frame));
   } else {
     size_t byte_offset;
     uint32_t timestamp_offset;
     for (byte_offset = 0, timestamp_offset = 0;
          byte_offset < payload.size();
          byte_offset += bytes_per_frame,
              timestamp_offset += timestamps_per_frame) {
       std::unique_ptr<EncodedAudioFrame> frame(new LegacyEncodedAudioFrame(
           this, rtc::Buffer(payload.data() + byte_offset, bytes_per_frame)));
       results.emplace_back(timestamp + timestamp_offset, 0, std::move(frame));
     }
   }
 
   return results;
 }
 
-int AudioDecoderIlbc::SampleRateHz() const {
+int AudioDecoderIlbcImpl::SampleRateHz() const {
   return 8000;
 }
 
-size_t AudioDecoderIlbc::Channels() const {
+size_t AudioDecoderIlbcImpl::Channels() const {
   return 1;
 }
 
 }  // namespace webrtc