Move AudioDecoderOpus next to AudioEncoderOpus

webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc

Index: webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
new file mode 100644
index 0000000000000000000000000000000000000000..e78fc04452d4b4cc8b4741cfe119b2c8aa20bea6
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
@@ -0,0 +1,94 @@
+/*
+ *  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/opus/interface/audio_decoder_opus.h"
+
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+
+AudioDecoderOpus::AudioDecoderOpus(size_t num_channels)
+    : channels_(num_channels) {
+  DCHECK(num_channels == 1 || num_channels == 2);
+  WebRtcOpus_DecoderCreate(&dec_state_, static_cast<int>(channels_));
+  WebRtcOpus_DecoderInit(dec_state_);
+}
+
+AudioDecoderOpus::~AudioDecoderOpus() {
+  WebRtcOpus_DecoderFree(dec_state_);
+}
+
+int AudioDecoderOpus::DecodeInternal(const uint8_t* encoded,
+                                     size_t encoded_len,
+                                     int sample_rate_hz,
+                                     int16_t* decoded,
+                                     SpeechType* speech_type) {
+  DCHECK_EQ(sample_rate_hz, 48000);
+  int16_t temp_type = 1;  // Default is speech.
+  int ret =
+      WebRtcOpus_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
+  if (ret > 0)
+    ret *= static_cast<int>(channels_);  // Return total number of samples.
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+int AudioDecoderOpus::DecodeRedundantInternal(const uint8_t* encoded,
+                                              size_t encoded_len,
+                                              int sample_rate_hz,
+                                              int16_t* decoded,
+                                              SpeechType* speech_type) {
+  if (!PacketHasFec(encoded, encoded_len)) {
+    // This packet is a RED packet.
+    return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
+                          speech_type);
+  }
+
+  DCHECK_EQ(sample_rate_hz, 48000);
+  int16_t temp_type = 1;  // Default is speech.
+  int ret = WebRtcOpus_DecodeFec(dec_state_, encoded, encoded_len, decoded,
+                                 &temp_type);
+  if (ret > 0)
+    ret *= static_cast<int>(channels_);  // Return total number of samples.
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+void AudioDecoderOpus::Reset() {
+  WebRtcOpus_DecoderInit(dec_state_);
+}
+
+int AudioDecoderOpus::PacketDuration(const uint8_t* encoded,
+                                     size_t encoded_len) const {
+  return WebRtcOpus_DurationEst(dec_state_, encoded, encoded_len);
+}
+
+int AudioDecoderOpus::PacketDurationRedundant(const uint8_t* encoded,
+                                              size_t encoded_len) const {
+  if (!PacketHasFec(encoded, encoded_len)) {
+    // This packet is a RED packet.
+    return PacketDuration(encoded, encoded_len);
+  }
+
+  return WebRtcOpus_FecDurationEst(encoded, encoded_len);
+}
+
+bool AudioDecoderOpus::PacketHasFec(const uint8_t* encoded,
+                                    size_t encoded_len) const {
+  int fec;
+  fec = WebRtcOpus_PacketHasFec(encoded, encoded_len);
+  return (fec == 1);
+}
+
+size_t AudioDecoderOpus::Channels() const {
+  return channels_;
+}
+
+}  // namespace webrtc