Move AudioDecoderOpus next to AudioEncoderOpus

webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc

 /*
  *  Copyright (c) 2012 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.
  */
 
@@ skipping 11 matching lines @@
 #ifdef WEBRTC_CODEC_ILBC
 #include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
 #endif
 #ifdef WEBRTC_CODEC_ISACFX
 #include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h"
 #endif
 #ifdef WEBRTC_CODEC_ISAC
 #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"
 #endif
 
 namespace webrtc {
 
 // PCMu
 
 void AudioDecoderPcmU::Reset() {
@@ skipping 249 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;
   }
 }
 #endif
 
-// Opus
-#ifdef WEBRTC_CODEC_OPUS

[hlundin-webrtc, 2015/09/15 10:30:45]

Did you just lose this conditional compilation of the code? Or is it conditioned
elsewhere?

[kwiberg-webrtc, 2015/09/15 10:45:07]

The entire file I moved it to is conditioned on include_opus (gyp) or
rtc_include_opus (gn), the same thing that says whether WEBRTC_CODEC_OPUS should
be defined.

However, when I do the same operation for other decoders that won't be the case,
since Opus is the only one that's currently conditionally excluded (see
https://code.google.com/p/webrtc/issues/detail?id=4557). My plan there is to
accept a temporary increase in code that should be but isn't conditionally
excluded; shortly thereafter, the final fix for bug 4557 will set things
straight again, but I'm pretty sure it's easiest to do things in the order I
described.

[hlundin-webrtc, 2015/09/15 10:49:13]

Acknowledged.

-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_);
 }
 
 AudioDecoderCng::~AudioDecoderCng() {
   WebRtcCng_FreeDec(dec_state_);
 }
 
 void AudioDecoderCng::Reset() {
@@ skipping 198 matching lines @@
     case kDecoderRED:
     case kDecoderAVT:
     case kDecoderArbitrary:
     default: {
       return NULL;
     }
   }
 }
 
 }  // namespace webrtc