Move AudioDecoderG722 next to AudioEncoderG722

webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc

Index: webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
index 274eec00c315578ec207f48fd56caf96313ca587..508129841a9c9edbf168901402d7cdb8d2153c17 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
@@ -11,13 +11,12 @@
 #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
 
 #include <assert.h>
-#include <string.h>  // memmove
 
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
 #include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
 #ifdef WEBRTC_CODEC_G722
-#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
+#include "webrtc/modules/audio_coding/codecs/g722/include/audio_decoder_g722.h"
 #endif
 #ifdef WEBRTC_CODEC_ILBC
 #include "webrtc/modules/audio_coding/codecs/ilbc/interface/audio_decoder_ilbc.h"
@@ -95,128 +94,6 @@ size_t AudioDecoderPcmAMultiCh::Channels() const {
   return channels_;
 }
 
-// G.722
-#ifdef WEBRTC_CODEC_G722
-AudioDecoderG722::AudioDecoderG722() {
-  WebRtcG722_CreateDecoder(&dec_state_);
-  WebRtcG722_DecoderInit(dec_state_);
-}
-
-AudioDecoderG722::~AudioDecoderG722() {
-  WebRtcG722_FreeDecoder(dec_state_);
-}
-
-bool AudioDecoderG722::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) {
-  RTC_DCHECK_EQ(sample_rate_hz, 16000);
-  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() {
-  WebRtcG722_DecoderInit(dec_state_);
-}
-
-int AudioDecoderG722::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());
-}
-
-size_t AudioDecoderG722::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_);
-}
-
-AudioDecoderG722Stereo::~AudioDecoderG722Stereo() {
-  WebRtcG722_FreeDecoder(dec_state_left_);
-  WebRtcG722_FreeDecoder(dec_state_right_);
-}
-
-int AudioDecoderG722Stereo::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, 16000);
-  int16_t temp_type = 1;  // Default is speech.
-  // De-interleave the bit-stream into two separate payloads.
-  uint8_t* encoded_deinterleaved = new uint8_t[encoded_len];
-  SplitStereoPacket(encoded, encoded_len, encoded_deinterleaved);
-  // Decode left and right.
-  size_t decoded_len = WebRtcG722_Decode(dec_state_left_, encoded_deinterleaved,
-                                         encoded_len / 2, decoded, &temp_type);
-  size_t ret = WebRtcG722_Decode(
-      dec_state_right_, &encoded_deinterleaved[encoded_len / 2],
-      encoded_len / 2, &decoded[decoded_len], &temp_type);
-  if (ret == decoded_len) {
-    ret += decoded_len;  // Return total number of samples.
-    // Interleave output.
-    for (size_t k = ret / 2; k < ret; k++) {
-        int16_t temp = decoded[k];
-        memmove(&decoded[2 * k - ret + 2], &decoded[2 * k - ret + 1],
-                (ret - k - 1) * sizeof(int16_t));
-        decoded[2 * k - ret + 1] = temp;
-    }
-  }
-  *speech_type = ConvertSpeechType(temp_type);
-  delete [] encoded_deinterleaved;
-  return static_cast<int>(ret);
-}
-
-size_t AudioDecoderG722Stereo::Channels() const {
-  return 2;
-}
-
-void AudioDecoderG722Stereo::Reset() {
-  WebRtcG722_DecoderInit(dec_state_left_);
-  WebRtcG722_DecoderInit(dec_state_right_);
-}
-
-// Split the stereo packet and place left and right channel after each other
-// in the output array.
-void AudioDecoderG722Stereo::SplitStereoPacket(const uint8_t* encoded,
-                                               size_t encoded_len,
-                                               uint8_t* encoded_deinterleaved) {
-  assert(encoded);
-  // Regroup the 4 bits/sample so |l1 l2| |r1 r2| |l3 l4| |r3 r4| ...,
-  // where "lx" is 4 bits representing left sample number x, and "rx" right
-  // sample. Two samples fit in one byte, represented with |...|.
-  for (size_t i = 0; i + 1 < encoded_len; i += 2) {
-    uint8_t right_byte = ((encoded[i] & 0x0F) << 4) + (encoded[i + 1] & 0x0F);
-    encoded_deinterleaved[i] = (encoded[i] & 0xF0) + (encoded[i + 1] >> 4);
-    encoded_deinterleaved[i + 1] = right_byte;
-  }
-
-  // Move one byte representing right channel each loop, and place it at the
-  // end of the bytestream vector. After looping the data is reordered to:
-  // |l1 l2| |l3 l4| ... |l(N-1) lN| |r1 r2| |r3 r4| ... |r(N-1) r(N)|,
-  // 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;
-  }
-}
-#endif
-
 AudioDecoderCng::AudioDecoderCng() {
   RTC_CHECK_EQ(0, WebRtcCng_CreateDec(&dec_state_));
   WebRtcCng_InitDec(dec_state_);