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}
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 235eb3c..fd96219 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -662,7 +662,9 @@
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",
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 0000000..e78fc04
--- /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
diff --git a/webrtc/modules/audio_coding/codecs/opus/interface/audio_decoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/interface/audio_decoder_opus.h
new file mode 100644
index 0000000..9fa77b0
--- /dev/null
+++ b/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
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus.gypi b/webrtc/modules/audio_coding/codecs/opus/opus.gypi
index 4ae4340..5a420b4 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus.gypi
+++ b/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',
diff --git a/webrtc/modules/audio_coding/main/test/opus_test.h b/webrtc/modules/audio_coding/main/test/opus_test.h
index 4c3d8c1..63945cc 100644
--- a/webrtc/modules/audio_coding/main/test/opus_test.h
+++ b/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"
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
index 592f17b..2e05fe1 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
+++ b/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 @@
}
#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_);
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
index f2ca711..6be344b 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
+++ b/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 @@
};
#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.
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
index 392e3dc..54dcdf5 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/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"
diff --git a/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
index e6ac37c..e3a0cc1 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
+++ b/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"
