G722 implementation of the AudioDecoderFactoryTemplate API

webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.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/g722/audio_decoder_g722.h"
 
 #include <string.h>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
 #include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
 
 namespace webrtc {
 
-AudioDecoderG722::AudioDecoderG722() {
+AudioDecoderG722Impl::AudioDecoderG722Impl() {
   WebRtcG722_CreateDecoder(&dec_state_);
   WebRtcG722_DecoderInit(dec_state_);
 }
 
-AudioDecoderG722::~AudioDecoderG722() {
+AudioDecoderG722Impl::~AudioDecoderG722Impl() {
   WebRtcG722_FreeDecoder(dec_state_);
 }
 
-bool AudioDecoderG722::HasDecodePlc() const {
+bool AudioDecoderG722Impl::HasDecodePlc() const {
   return false;
 }
 
-int AudioDecoderG722::DecodeInternal(const uint8_t* encoded,
-                                     size_t encoded_len,
-                                     int sample_rate_hz,
-                                     int16_t* decoded,
-                                     SpeechType* speech_type) {
+int AudioDecoderG722Impl::DecodeInternal(const uint8_t* encoded,
+                                         size_t encoded_len,
+                                         int sample_rate_hz,
+                                         int16_t* decoded,
+                                         SpeechType* speech_type) {
   RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
   int16_t temp_type = 1;  // Default is speech.
   size_t ret =
       WebRtcG722_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
   *speech_type = ConvertSpeechType(temp_type);
   return static_cast<int>(ret);
 }
 
-void AudioDecoderG722::Reset() {
+void AudioDecoderG722Impl::Reset() {
   WebRtcG722_DecoderInit(dec_state_);
 }
 
-std::vector<AudioDecoder::ParseResult> AudioDecoderG722::ParsePayload(
+std::vector<AudioDecoder::ParseResult> AudioDecoderG722Impl::ParsePayload(
     rtc::Buffer&& payload,
     uint32_t timestamp) {
   return LegacyEncodedAudioFrame::SplitBySamples(this, std::move(payload),
                                                  timestamp, 8, 16);
 }
 
-int AudioDecoderG722::PacketDuration(const uint8_t* encoded,
-                                     size_t encoded_len) const {
+int AudioDecoderG722Impl::PacketDuration(const uint8_t* encoded,
+                                         size_t encoded_len) const {
   // 1/2 encoded byte per sample per channel.
   return static_cast<int>(2 * encoded_len / Channels());
 }
 
-int AudioDecoderG722::SampleRateHz() const {
+int AudioDecoderG722Impl::SampleRateHz() const {
   return 16000;
 }
 
-size_t AudioDecoderG722::Channels() const {
+size_t AudioDecoderG722Impl::Channels() const {
   return 1;
 }
 
 AudioDecoderG722Stereo::AudioDecoderG722Stereo() {
   WebRtcG722_CreateDecoder(&dec_state_left_);
   WebRtcG722_CreateDecoder(&dec_state_right_);
   WebRtcG722_DecoderInit(dec_state_left_);
   WebRtcG722_DecoderInit(dec_state_right_);
 }
 
@@ skipping 73 matching lines @@
   // where N is the total number of samples.
   for (size_t i = 0; i < encoded_len / 2; i++) {
     uint8_t right_byte = encoded_deinterleaved[i + 1];
     memmove(&encoded_deinterleaved[i + 1], &encoded_deinterleaved[i + 2],
             encoded_len - i - 2);
     encoded_deinterleaved[encoded_len - 1] = right_byte;
   }
 }
 
 }  // namespace webrtc