Move AudioDecoderG722 next to AudioEncoderG722

webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc

Index: webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
diff --git a/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc b/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
new file mode 100644
index 0000000000000000000000000000000000000000..55ebe7a315bac3a90290030826782f74082e221f
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
@@ -0,0 +1,138 @@
+/*
+ *  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/include/audio_decoder_g722.h"
+
+#include <string.h>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
+
+namespace webrtc {
+
+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) {
+  // 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;
+  }
+}
+
+}  // namespace webrtc