Move AudioDecoderOpus next to AudioEncoderOpus

All AudioDecoder subclasses have historically lived in NetEq, but they
fit better with the codec they wrap.

BUG=webrtc:4557
R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1342933005 .

Cr-Commit-Position: refs/heads/master@{#9944}

webrtc/modules/audio_coding/BUILD.gn

@@ -662,7 +662,9 @@ config("opus_config") {
 
 source_set("webrtc_opus") {
   sources = [
+    "codecs/opus/audio_decoder_opus.cc",
     "codecs/opus/audio_encoder_opus.cc",
+    "codecs/opus/interface/audio_decoder_opus.h",
     "codecs/opus/interface/audio_encoder_opus.h",
     "codecs/opus/interface/opus_interface.h",
     "codecs/opus/opus_inst.h",

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

webrtc/modules/audio_coding/codecs/opus/interface/audio_decoder_opus.h

@@ -0,0 +1,51 @@
+/*
+ *  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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_DECODER_OPUS_H
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_DECODER_OPUS_H
+
+#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
+
+namespace webrtc {
+
+class AudioDecoderOpus : public AudioDecoder {
+ public:
+  explicit AudioDecoderOpus(size_t num_channels);
+  ~AudioDecoderOpus() override;
+
+  void Reset() override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int PacketDurationRedundant(const uint8_t* encoded,
+                              size_t encoded_len) const override;
+  bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+  int DecodeRedundantInternal(const uint8_t* encoded,
+                              size_t encoded_len,
+                              int sample_rate_hz,
+                              int16_t* decoded,
+                              SpeechType* speech_type) override;
+
+ private:
+  OpusDecInst* dec_state_;
+  const size_t channels_;
+  DISALLOW_COPY_AND_ASSIGN(AudioDecoderOpus);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_DECODER_OPUS_H

webrtc/modules/audio_coding/codecs/opus/opus.gypi

@@ -43,7 +43,9 @@
         '<(webrtc_root)',
       ],
       'sources': [
+        'audio_decoder_opus.cc',
         'audio_encoder_opus.cc',
+        'interface/audio_decoder_opus.h',
         'interface/audio_encoder_opus.h',
         'interface/opus_interface.h',
         'opus_inst.h',

webrtc/modules/audio_coding/main/test/opus_test.h

@@ -14,6 +14,7 @@
 #include <math.h>
 
 #include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
 #include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
 #include "webrtc/modules/audio_coding/main/test/ACMTest.h"
 #include "webrtc/modules/audio_coding/main/test/Channel.h"

webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc

@@ -29,7 +29,7 @@
 #include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h"
 #endif
 #ifdef WEBRTC_CODEC_OPUS
-#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
+#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_decoder_opus.h"
 #endif
 #ifdef WEBRTC_CODEC_PCM16
 #include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
@@ -299,86 +299,6 @@ void AudioDecoderG722Stereo::SplitStereoPacket(const uint8_t* encoded,
 }
 #endif
 
-// Opus
-#ifdef WEBRTC_CODEC_OPUS
-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_;
-}
-#endif
-
 AudioDecoderCng::AudioDecoderCng() {
   CHECK_EQ(0, WebRtcCng_CreateDec(&dec_state_));
   WebRtcCng_InitDec(dec_state_);

webrtc/modules/audio_coding/neteq/audio_decoder_impl.h

@@ -27,9 +27,6 @@
 #ifdef WEBRTC_CODEC_ILBC
 #include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
 #endif
-#ifdef WEBRTC_CODEC_OPUS
-#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
-#endif
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -206,38 +203,6 @@ class AudioDecoderG722Stereo : public AudioDecoder {
 };
 #endif
 
-#ifdef WEBRTC_CODEC_OPUS
-class AudioDecoderOpus : public AudioDecoder {
- public:
-  explicit AudioDecoderOpus(size_t num_channels);
-  ~AudioDecoderOpus() override;
-
-  void Reset() override;
-  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
-  int PacketDurationRedundant(const uint8_t* encoded,
-                              size_t encoded_len) const override;
-  bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override;
-  size_t Channels() const override;
-
- protected:
-  int DecodeInternal(const uint8_t* encoded,
-                     size_t encoded_len,
-                     int sample_rate_hz,
-                     int16_t* decoded,
-                     SpeechType* speech_type) override;
-  int DecodeRedundantInternal(const uint8_t* encoded,
-                              size_t encoded_len,
-                              int sample_rate_hz,
-                              int16_t* decoded,
-                              SpeechType* speech_type) override;
-
- private:
-  OpusDecInst* dec_state_;
-  const size_t channels_;
-  DISALLOW_COPY_AND_ASSIGN(AudioDecoderOpus);
-};
-#endif
-
 // AudioDecoderCng is a special type of AudioDecoder. It inherits from
 // AudioDecoder just to fit in the DecoderDatabase. None of the class methods
 // should be used, except constructor, destructor, and accessors.

webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc

@@ -23,6 +23,7 @@
 #include "webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h"
+#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_decoder_opus.h"
 #include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h"
 #include "webrtc/modules/audio_coding/codecs/pcm16b/include/audio_encoder_pcm16b.h"
 #include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"

webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc

@@ -10,7 +10,7 @@
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
+#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_decoder_opus.h"
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"