Move AudioDecoderIsac* to its own files
Currently, it's sitting in AudioEncoderIsac*'s files, which is less
than obvious. This CL puts the encoder and decoder in separate files
together with the C implementation; CLs are afoot to make it so for
the other built-in codecs as well.
BUG=webrtc:4557
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1339253003 .
Cr-Commit-Position: refs/heads/master@{#10018}
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 5d85db3..5725ec0 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -412,12 +412,14 @@
source_set("isac") {
sources = [
+ "codecs/isac/main/interface/audio_decoder_isac.h",
"codecs/isac/main/interface/audio_encoder_isac.h",
"codecs/isac/main/interface/isac.h",
"codecs/isac/main/source/arith_routines.c",
"codecs/isac/main/source/arith_routines.h",
"codecs/isac/main/source/arith_routines_hist.c",
"codecs/isac/main/source/arith_routines_logist.c",
+ "codecs/isac/main/source/audio_decoder_isac.cc",
"codecs/isac/main/source/audio_encoder_isac.cc",
"codecs/isac/main/source/bandwidth_estimator.c",
"codecs/isac/main/source/bandwidth_estimator.h",
@@ -439,6 +441,7 @@
"codecs/isac/main/source/filterbanks.c",
"codecs/isac/main/source/intialize.c",
"codecs/isac/main/source/isac.c",
+ "codecs/isac/main/source/isac_float_type.h",
"codecs/isac/main/source/lattice.c",
"codecs/isac/main/source/lpc_analysis.c",
"codecs/isac/main/source/lpc_analysis.h",
@@ -493,12 +496,14 @@
source_set("isac_fix") {
sources = [
+ "codecs/isac/fix/interface/audio_decoder_isacfix.h",
"codecs/isac/fix/interface/audio_encoder_isacfix.h",
"codecs/isac/fix/interface/isacfix.h",
"codecs/isac/fix/source/arith_routines.c",
"codecs/isac/fix/source/arith_routines_hist.c",
"codecs/isac/fix/source/arith_routines_logist.c",
"codecs/isac/fix/source/arith_routins.h",
+ "codecs/isac/fix/source/audio_decoder_isacfix.cc",
"codecs/isac/fix/source/audio_encoder_isacfix.cc",
"codecs/isac/fix/source/bandwidth_estimator.c",
"codecs/isac/fix/source/bandwidth_estimator.h",
@@ -516,6 +521,7 @@
"codecs/isac/fix/source/filterbanks.c",
"codecs/isac/fix/source/filters.c",
"codecs/isac/fix/source/initialize.c",
+ "codecs/isac/fix/source/isac_fix_type.h",
"codecs/isac/fix/source/isacfix.c",
"codecs/isac/fix/source/lattice.c",
"codecs/isac/fix/source/lattice_c.c",
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h b/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
new file mode 100644
index 0000000..714fe2d
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
@@ -0,0 +1,53 @@
+/*
+ * 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_ISAC_AUDIO_DECODER_ISAC_T_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_
+
+#include <vector>
+
+#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+#include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
+
+namespace webrtc {
+
+template <typename T>
+class AudioDecoderIsacT final : public AudioDecoder {
+ public:
+ AudioDecoderIsacT();
+ explicit AudioDecoderIsacT(LockedIsacBandwidthInfo* bwinfo);
+ ~AudioDecoderIsacT() override;
+
+ bool HasDecodePlc() const override;
+ size_t DecodePlc(size_t num_frames, int16_t* decoded) override;
+ void Reset() override;
+ int IncomingPacket(const uint8_t* payload,
+ size_t payload_len,
+ uint16_t rtp_sequence_number,
+ uint32_t rtp_timestamp,
+ uint32_t arrival_timestamp) override;
+ int ErrorCode() override;
+ size_t Channels() const override;
+ int DecodeInternal(const uint8_t* encoded,
+ size_t encoded_len,
+ int sample_rate_hz,
+ int16_t* decoded,
+ SpeechType* speech_type) override;
+
+ private:
+ typename T::instance_type* isac_state_;
+ LockedIsacBandwidthInfo* bwinfo_;
+ int decoder_sample_rate_hz_;
+
+ RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacT);
+};
+
+} // namespace webrtc
+#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h b/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h
new file mode 100644
index 0000000..126c129
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h
@@ -0,0 +1,108 @@
+/*
+ * 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_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_
+
+#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_decoder_isac.h"
+
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+
+template <typename T>
+AudioDecoderIsacT<T>::AudioDecoderIsacT()
+ : AudioDecoderIsacT(nullptr) {}
+
+template <typename T>
+AudioDecoderIsacT<T>::AudioDecoderIsacT(LockedIsacBandwidthInfo* bwinfo)
+ : bwinfo_(bwinfo), decoder_sample_rate_hz_(-1) {
+ RTC_CHECK_EQ(0, T::Create(&isac_state_));
+ T::DecoderInit(isac_state_);
+ if (bwinfo_) {
+ IsacBandwidthInfo bi;
+ T::GetBandwidthInfo(isac_state_, &bi);
+ bwinfo_->Set(bi);
+ }
+}
+
+template <typename T>
+AudioDecoderIsacT<T>::~AudioDecoderIsacT() {
+ RTC_CHECK_EQ(0, T::Free(isac_state_));
+}
+
+template <typename T>
+int AudioDecoderIsacT<T>::DecodeInternal(const uint8_t* encoded,
+ size_t encoded_len,
+ int sample_rate_hz,
+ int16_t* decoded,
+ SpeechType* speech_type) {
+ // We want to crate the illusion that iSAC supports 48000 Hz decoding, while
+ // in fact it outputs 32000 Hz. This is the iSAC fullband mode.
+ if (sample_rate_hz == 48000)
+ sample_rate_hz = 32000;
+ RTC_CHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000)
+ << "Unsupported sample rate " << sample_rate_hz;
+ if (sample_rate_hz != decoder_sample_rate_hz_) {
+ RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, sample_rate_hz));
+ decoder_sample_rate_hz_ = sample_rate_hz;
+ }
+ int16_t temp_type = 1; // Default is speech.
+ int ret =
+ T::DecodeInternal(isac_state_, encoded, encoded_len, decoded, &temp_type);
+ *speech_type = ConvertSpeechType(temp_type);
+ return ret;
+}
+
+template <typename T>
+bool AudioDecoderIsacT<T>::HasDecodePlc() const {
+ return false;
+}
+
+template <typename T>
+size_t AudioDecoderIsacT<T>::DecodePlc(size_t num_frames, int16_t* decoded) {
+ return T::DecodePlc(isac_state_, decoded, num_frames);
+}
+
+template <typename T>
+void AudioDecoderIsacT<T>::Reset() {
+ T::DecoderInit(isac_state_);
+}
+
+template <typename T>
+int AudioDecoderIsacT<T>::IncomingPacket(const uint8_t* payload,
+ size_t payload_len,
+ uint16_t rtp_sequence_number,
+ uint32_t rtp_timestamp,
+ uint32_t arrival_timestamp) {
+ int ret = T::UpdateBwEstimate(isac_state_, payload, payload_len,
+ rtp_sequence_number, rtp_timestamp,
+ arrival_timestamp);
+ if (bwinfo_) {
+ IsacBandwidthInfo bwinfo;
+ T::GetBandwidthInfo(isac_state_, &bwinfo);
+ bwinfo_->Set(bwinfo);
+ }
+ return ret;
+}
+
+template <typename T>
+int AudioDecoderIsacT<T>::ErrorCode() {
+ return T::GetErrorCode(isac_state_);
+}
+
+template <typename T>
+size_t AudioDecoderIsacT<T>::Channels() const {
+ return 1;
+}
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
index 50d369b..6b361b4 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
@@ -13,7 +13,6 @@
#include <vector>
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
@@ -94,36 +93,5 @@
RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIsacT);
};
-template <typename T>
-class AudioDecoderIsacT final : public AudioDecoder {
- public:
- AudioDecoderIsacT();
- explicit AudioDecoderIsacT(LockedIsacBandwidthInfo* bwinfo);
- ~AudioDecoderIsacT() override;
-
- bool HasDecodePlc() const override;
- size_t DecodePlc(size_t num_frames, int16_t* decoded) override;
- void Reset() override;
- int IncomingPacket(const uint8_t* payload,
- size_t payload_len,
- uint16_t rtp_sequence_number,
- uint32_t rtp_timestamp,
- uint32_t arrival_timestamp) override;
- int ErrorCode() override;
- size_t Channels() const override;
- int DecodeInternal(const uint8_t* encoded,
- size_t encoded_len,
- int sample_rate_hz,
- int16_t* decoded,
- SpeechType* speech_type) override;
-
- private:
- typename T::instance_type* isac_state_;
- LockedIsacBandwidthInfo* bwinfo_;
- int decoder_sample_rate_hz_;
-
- RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacT);
-};
-
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
index 4122ee0..fbc1ba9 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
@@ -13,10 +13,7 @@
#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h"
-#include <algorithm>
-
#include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
namespace webrtc {
@@ -193,92 +190,6 @@
config_ = config;
}
-template <typename T>
-AudioDecoderIsacT<T>::AudioDecoderIsacT()
- : AudioDecoderIsacT(nullptr) {}
-
-template <typename T>
-AudioDecoderIsacT<T>::AudioDecoderIsacT(LockedIsacBandwidthInfo* bwinfo)
- : bwinfo_(bwinfo), decoder_sample_rate_hz_(-1) {
- RTC_CHECK_EQ(0, T::Create(&isac_state_));
- T::DecoderInit(isac_state_);
- if (bwinfo_) {
- IsacBandwidthInfo bi;
- T::GetBandwidthInfo(isac_state_, &bi);
- bwinfo_->Set(bi);
- }
-}
-
-template <typename T>
-AudioDecoderIsacT<T>::~AudioDecoderIsacT() {
- RTC_CHECK_EQ(0, T::Free(isac_state_));
-}
-
-template <typename T>
-int AudioDecoderIsacT<T>::DecodeInternal(const uint8_t* encoded,
- size_t encoded_len,
- int sample_rate_hz,
- int16_t* decoded,
- SpeechType* speech_type) {
- // We want to crate the illusion that iSAC supports 48000 Hz decoding, while
- // in fact it outputs 32000 Hz. This is the iSAC fullband mode.
- if (sample_rate_hz == 48000)
- sample_rate_hz = 32000;
- RTC_CHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000)
- << "Unsupported sample rate " << sample_rate_hz;
- if (sample_rate_hz != decoder_sample_rate_hz_) {
- RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, sample_rate_hz));
- decoder_sample_rate_hz_ = sample_rate_hz;
- }
- int16_t temp_type = 1; // Default is speech.
- int ret =
- T::DecodeInternal(isac_state_, encoded, encoded_len, decoded, &temp_type);
- *speech_type = ConvertSpeechType(temp_type);
- return ret;
-}
-
-template <typename T>
-bool AudioDecoderIsacT<T>::HasDecodePlc() const {
- return false;
-}
-
-template <typename T>
-size_t AudioDecoderIsacT<T>::DecodePlc(size_t num_frames, int16_t* decoded) {
- return T::DecodePlc(isac_state_, decoded, num_frames);
-}
-
-template <typename T>
-void AudioDecoderIsacT<T>::Reset() {
- T::DecoderInit(isac_state_);
-}
-
-template <typename T>
-int AudioDecoderIsacT<T>::IncomingPacket(const uint8_t* payload,
- size_t payload_len,
- uint16_t rtp_sequence_number,
- uint32_t rtp_timestamp,
- uint32_t arrival_timestamp) {
- int ret = T::UpdateBwEstimate(
- isac_state_, payload, payload_len,
- rtp_sequence_number, rtp_timestamp, arrival_timestamp);
- if (bwinfo_) {
- IsacBandwidthInfo bwinfo;
- T::GetBandwidthInfo(isac_state_, &bwinfo);
- bwinfo_->Set(bwinfo);
- }
- return ret;
-}
-
-template <typename T>
-int AudioDecoderIsacT<T>::ErrorCode() {
- return T::GetErrorCode(isac_state_);
-}
-
-template <typename T>
-size_t AudioDecoderIsacT<T>::Channels() const {
- return 1;
-}
-
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_decoder_isacfix.h b/webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_decoder_isacfix.h
new file mode 100644
index 0000000..ed162a1
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_decoder_isacfix.h
@@ -0,0 +1,22 @@
+/*
+ * 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_ISAC_FIX_INTERFACE_AUDIO_DECODER_ISACFIX_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_DECODER_ISACFIX_H_
+
+#include "webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h"
+
+namespace webrtc {
+
+using AudioDecoderIsacFix = AudioDecoderIsacT<IsacFix>;
+
+} // namespace webrtc
+#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_DECODER_ISACFIX_H_
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h b/webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h
index 5bca23e..00c0987 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h
@@ -11,116 +11,12 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_ENCODER_ISACFIX_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_ENCODER_ISACFIX_H_
-#include "webrtc/base/checks.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h"
namespace webrtc {
-struct IsacFix {
- typedef ISACFIX_MainStruct instance_type;
- static const bool has_swb = false;
- static const uint16_t kFixSampleRate = 16000;
- static inline int16_t Control(instance_type* inst,
- int32_t rate,
- int framesize) {
- return WebRtcIsacfix_Control(inst, rate, framesize);
- }
- static inline int16_t ControlBwe(instance_type* inst,
- int32_t rate_bps,
- int frame_size_ms,
- int16_t enforce_frame_size) {
- return WebRtcIsacfix_ControlBwe(inst, rate_bps, frame_size_ms,
- enforce_frame_size);
- }
- static inline int16_t Create(instance_type** inst) {
- return WebRtcIsacfix_Create(inst);
- }
- static inline int DecodeInternal(instance_type* inst,
- const uint8_t* encoded,
- size_t len,
- int16_t* decoded,
- int16_t* speech_type) {
- return WebRtcIsacfix_Decode(inst, encoded, len, decoded, speech_type);
- }
- static inline size_t DecodePlc(instance_type* inst,
- int16_t* decoded,
- size_t num_lost_frames) {
- return WebRtcIsacfix_DecodePlc(inst, decoded, num_lost_frames);
- }
- static inline void DecoderInit(instance_type* inst) {
- WebRtcIsacfix_DecoderInit(inst);
- }
- static inline int Encode(instance_type* inst,
- const int16_t* speech_in,
- uint8_t* encoded) {
- return WebRtcIsacfix_Encode(inst, speech_in, encoded);
- }
- static inline int16_t EncoderInit(instance_type* inst, int16_t coding_mode) {
- return WebRtcIsacfix_EncoderInit(inst, coding_mode);
- }
- static inline uint16_t EncSampRate(instance_type* inst) {
- return kFixSampleRate;
- }
-
- static inline int16_t Free(instance_type* inst) {
- return WebRtcIsacfix_Free(inst);
- }
- static inline void GetBandwidthInfo(instance_type* inst,
- IsacBandwidthInfo* bwinfo) {
- WebRtcIsacfix_GetBandwidthInfo(inst, bwinfo);
- }
- static inline int16_t GetErrorCode(instance_type* inst) {
- return WebRtcIsacfix_GetErrorCode(inst);
- }
-
- static inline int16_t GetNewFrameLen(instance_type* inst) {
- return WebRtcIsacfix_GetNewFrameLen(inst);
- }
- static inline void SetBandwidthInfo(instance_type* inst,
- const IsacBandwidthInfo* bwinfo) {
- WebRtcIsacfix_SetBandwidthInfo(inst, bwinfo);
- }
- static inline int16_t SetDecSampRate(instance_type* inst,
- uint16_t sample_rate_hz) {
- RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate);
- return 0;
- }
- static inline int16_t SetEncSampRate(instance_type* inst,
- uint16_t sample_rate_hz) {
- RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate);
- return 0;
- }
- static inline void SetEncSampRateInDecoder(instance_type* inst,
- uint16_t sample_rate_hz) {
- RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate);
- }
- static inline void SetInitialBweBottleneck(
- instance_type* inst,
- int bottleneck_bits_per_second) {
- WebRtcIsacfix_SetInitialBweBottleneck(inst, bottleneck_bits_per_second);
- }
- static inline int16_t UpdateBwEstimate(instance_type* inst,
- const uint8_t* encoded,
- size_t packet_size,
- uint16_t rtp_seq_number,
- uint32_t send_ts,
- uint32_t arr_ts) {
- return WebRtcIsacfix_UpdateBwEstimate(inst, encoded, packet_size,
- rtp_seq_number, send_ts, arr_ts);
- }
- static inline int16_t SetMaxPayloadSize(instance_type* inst,
- int16_t max_payload_size_bytes) {
- return WebRtcIsacfix_SetMaxPayloadSize(inst, max_payload_size_bytes);
- }
- static inline int16_t SetMaxRate(instance_type* inst, int32_t max_bit_rate) {
- return WebRtcIsacfix_SetMaxRate(inst, max_bit_rate);
- }
-};
-
using AudioEncoderIsacFix = AudioEncoderIsacT<IsacFix>;
-using AudioDecoderIsacFix = AudioDecoderIsacT<IsacFix>;
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_ENCODER_ISACFIX_H_
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/audio_decoder_isacfix.cc b/webrtc/modules/audio_coding/codecs/isac/fix/source/audio_decoder_isacfix.cc
new file mode 100644
index 0000000..8b4741b
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/audio_decoder_isacfix.cc
@@ -0,0 +1,20 @@
+/*
+ * 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/isac/fix/interface/audio_decoder_isacfix.h"
+
+#include "webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h"
+
+namespace webrtc {
+
+// Explicit instantiation:
+template class AudioDecoderIsacT<IsacFix>;
+
+} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/audio_encoder_isacfix.cc b/webrtc/modules/audio_coding/codecs/isac/fix/source/audio_encoder_isacfix.cc
index 9b2f317..ad75ea8 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/audio_encoder_isacfix.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/audio_encoder_isacfix.cc
@@ -14,10 +14,7 @@
namespace webrtc {
-const uint16_t IsacFix::kFixSampleRate;
-
// Explicit instantiation:
template class AudioEncoderIsacT<IsacFix>;
-template class AudioDecoderIsacT<IsacFix>;
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h b/webrtc/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h
new file mode 100644
index 0000000..69c73d6
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h
@@ -0,0 +1,123 @@
+/*
+ * 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_ISAC_FIX_SOURCE_ISAC_FIX_TYPE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ISAC_FIX_TYPE_H_
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
+
+namespace webrtc {
+
+class IsacFix {
+ public:
+ using instance_type = ISACFIX_MainStruct;
+ static const bool has_swb = false;
+ static inline int16_t Control(instance_type* inst,
+ int32_t rate,
+ int framesize) {
+ return WebRtcIsacfix_Control(inst, rate, framesize);
+ }
+ static inline int16_t ControlBwe(instance_type* inst,
+ int32_t rate_bps,
+ int frame_size_ms,
+ int16_t enforce_frame_size) {
+ return WebRtcIsacfix_ControlBwe(inst, rate_bps, frame_size_ms,
+ enforce_frame_size);
+ }
+ static inline int16_t Create(instance_type** inst) {
+ return WebRtcIsacfix_Create(inst);
+ }
+ static inline int DecodeInternal(instance_type* inst,
+ const uint8_t* encoded,
+ size_t len,
+ int16_t* decoded,
+ int16_t* speech_type) {
+ return WebRtcIsacfix_Decode(inst, encoded, len, decoded, speech_type);
+ }
+ static inline size_t DecodePlc(instance_type* inst,
+ int16_t* decoded,
+ size_t num_lost_frames) {
+ return WebRtcIsacfix_DecodePlc(inst, decoded, num_lost_frames);
+ }
+ static inline void DecoderInit(instance_type* inst) {
+ WebRtcIsacfix_DecoderInit(inst);
+ }
+ static inline int Encode(instance_type* inst,
+ const int16_t* speech_in,
+ uint8_t* encoded) {
+ return WebRtcIsacfix_Encode(inst, speech_in, encoded);
+ }
+ static inline int16_t EncoderInit(instance_type* inst, int16_t coding_mode) {
+ return WebRtcIsacfix_EncoderInit(inst, coding_mode);
+ }
+ static inline uint16_t EncSampRate(instance_type* inst) {
+ return kFixSampleRate;
+ }
+
+ static inline int16_t Free(instance_type* inst) {
+ return WebRtcIsacfix_Free(inst);
+ }
+ static inline void GetBandwidthInfo(instance_type* inst,
+ IsacBandwidthInfo* bwinfo) {
+ WebRtcIsacfix_GetBandwidthInfo(inst, bwinfo);
+ }
+ static inline int16_t GetErrorCode(instance_type* inst) {
+ return WebRtcIsacfix_GetErrorCode(inst);
+ }
+
+ static inline int16_t GetNewFrameLen(instance_type* inst) {
+ return WebRtcIsacfix_GetNewFrameLen(inst);
+ }
+ static inline void SetBandwidthInfo(instance_type* inst,
+ const IsacBandwidthInfo* bwinfo) {
+ WebRtcIsacfix_SetBandwidthInfo(inst, bwinfo);
+ }
+ static inline int16_t SetDecSampRate(instance_type* inst,
+ uint16_t sample_rate_hz) {
+ RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate);
+ return 0;
+ }
+ static inline int16_t SetEncSampRate(instance_type* inst,
+ uint16_t sample_rate_hz) {
+ RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate);
+ return 0;
+ }
+ static inline void SetEncSampRateInDecoder(instance_type* inst,
+ uint16_t sample_rate_hz) {
+ RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate);
+ }
+ static inline void SetInitialBweBottleneck(instance_type* inst,
+ int bottleneck_bits_per_second) {
+ WebRtcIsacfix_SetInitialBweBottleneck(inst, bottleneck_bits_per_second);
+ }
+ static inline int16_t UpdateBwEstimate(instance_type* inst,
+ const uint8_t* encoded,
+ size_t packet_size,
+ uint16_t rtp_seq_number,
+ uint32_t send_ts,
+ uint32_t arr_ts) {
+ return WebRtcIsacfix_UpdateBwEstimate(inst, encoded, packet_size,
+ rtp_seq_number, send_ts, arr_ts);
+ }
+ static inline int16_t SetMaxPayloadSize(instance_type* inst,
+ int16_t max_payload_size_bytes) {
+ return WebRtcIsacfix_SetMaxPayloadSize(inst, max_payload_size_bytes);
+ }
+ static inline int16_t SetMaxRate(instance_type* inst, int32_t max_bit_rate) {
+ return WebRtcIsacfix_SetMaxRate(inst, max_bit_rate);
+ }
+
+ private:
+ enum { kFixSampleRate = 16000 };
+};
+
+} // namespace webrtc
+#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ISAC_FIX_TYPE_H_
diff --git a/webrtc/modules/audio_coding/codecs/isac/isac.gypi b/webrtc/modules/audio_coding/codecs/isac/isac.gypi
index 8ecc2dc..354719f 100644
--- a/webrtc/modules/audio_coding/codecs/isac/isac.gypi
+++ b/webrtc/modules/audio_coding/codecs/isac/isac.gypi
@@ -28,11 +28,13 @@
],
},
'sources': [
+ 'main/interface/audio_decoder_isac.h',
'main/interface/audio_encoder_isac.h',
'main/interface/isac.h',
'main/source/arith_routines.c',
'main/source/arith_routines_hist.c',
'main/source/arith_routines_logist.c',
+ 'main/source/audio_decoder_isac.cc',
'main/source/audio_encoder_isac.cc',
'main/source/bandwidth_estimator.c',
'main/source/crc.c',
@@ -46,6 +48,7 @@
'main/source/filterbank_tables.c',
'main/source/intialize.c',
'main/source/isac.c',
+ 'main/source/isac_float_type.h',
'main/source/filterbanks.c',
'main/source/pitch_lag_tables.c',
'main/source/lattice.c',
diff --git a/webrtc/modules/audio_coding/codecs/isac/isacfix.gypi b/webrtc/modules/audio_coding/codecs/isac/isacfix.gypi
index 81b4375..4355fa9 100644
--- a/webrtc/modules/audio_coding/codecs/isac/isacfix.gypi
+++ b/webrtc/modules/audio_coding/codecs/isac/isacfix.gypi
@@ -27,11 +27,13 @@
],
},
'sources': [
+ 'fix/interface/audio_decoder_isacfix.h',
'fix/interface/audio_encoder_isacfix.h',
'fix/interface/isacfix.h',
'fix/source/arith_routines.c',
'fix/source/arith_routines_hist.c',
'fix/source/arith_routines_logist.c',
+ 'fix/source/audio_decoder_isacfix.cc',
'fix/source/audio_encoder_isacfix.cc',
'fix/source/bandwidth_estimator.c',
'fix/source/decode.c',
@@ -44,6 +46,7 @@
'fix/source/filterbanks.c',
'fix/source/filters.c',
'fix/source/initialize.c',
+ 'fix/source/isac_fix_type.h',
'fix/source/isacfix.c',
'fix/source/lattice.c',
'fix/source/lattice_c.c',
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_decoder_isac.h b/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_decoder_isac.h
new file mode 100644
index 0000000..b224296
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_decoder_isac.h
@@ -0,0 +1,22 @@
+/*
+ * 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_ISAC_MAIN_INTERFACE_AUDIO_DECODER_ISAC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_AUDIO_DECODER_ISAC_H_
+
+#include "webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/source/isac_float_type.h"
+
+namespace webrtc {
+
+using AudioDecoderIsac = AudioDecoderIsacT<IsacFloat>;
+
+} // namespace webrtc
+#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_AUDIO_ENCODER_ISAC_H_
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h b/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h
index da363d8..f6ef7c4 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h
+++ b/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h
@@ -11,114 +11,12 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_AUDIO_ENCODER_ISAC_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_AUDIO_ENCODER_ISAC_H_
-#include "webrtc/base/checks.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/source/isac_float_type.h"
namespace webrtc {
-struct IsacFloat {
- typedef ISACStruct instance_type;
- static const bool has_swb = true;
- static inline int16_t Control(instance_type* inst,
- int32_t rate,
- int framesize) {
- return WebRtcIsac_Control(inst, rate, framesize);
- }
- static inline int16_t ControlBwe(instance_type* inst,
- int32_t rate_bps,
- int frame_size_ms,
- int16_t enforce_frame_size) {
- return WebRtcIsac_ControlBwe(inst, rate_bps, frame_size_ms,
- enforce_frame_size);
- }
- static inline int16_t Create(instance_type** inst) {
- return WebRtcIsac_Create(inst);
- }
- static inline int DecodeInternal(instance_type* inst,
- const uint8_t* encoded,
- size_t len,
- int16_t* decoded,
- int16_t* speech_type) {
- return WebRtcIsac_Decode(inst, encoded, len, decoded, speech_type);
- }
- static inline size_t DecodePlc(instance_type* inst,
- int16_t* decoded,
- size_t num_lost_frames) {
- return WebRtcIsac_DecodePlc(inst, decoded, num_lost_frames);
- }
-
- static inline void DecoderInit(instance_type* inst) {
- WebRtcIsac_DecoderInit(inst);
- }
- static inline int Encode(instance_type* inst,
- const int16_t* speech_in,
- uint8_t* encoded) {
- return WebRtcIsac_Encode(inst, speech_in, encoded);
- }
- static inline int16_t EncoderInit(instance_type* inst, int16_t coding_mode) {
- return WebRtcIsac_EncoderInit(inst, coding_mode);
- }
- static inline uint16_t EncSampRate(instance_type* inst) {
- return WebRtcIsac_EncSampRate(inst);
- }
-
- static inline int16_t Free(instance_type* inst) {
- return WebRtcIsac_Free(inst);
- }
- static inline void GetBandwidthInfo(instance_type* inst,
- IsacBandwidthInfo* bwinfo) {
- WebRtcIsac_GetBandwidthInfo(inst, bwinfo);
- }
- static inline int16_t GetErrorCode(instance_type* inst) {
- return WebRtcIsac_GetErrorCode(inst);
- }
-
- static inline int16_t GetNewFrameLen(instance_type* inst) {
- return WebRtcIsac_GetNewFrameLen(inst);
- }
- static inline void SetBandwidthInfo(instance_type* inst,
- const IsacBandwidthInfo* bwinfo) {
- WebRtcIsac_SetBandwidthInfo(inst, bwinfo);
- }
- static inline int16_t SetDecSampRate(instance_type* inst,
- uint16_t sample_rate_hz) {
- return WebRtcIsac_SetDecSampRate(inst, sample_rate_hz);
- }
- static inline int16_t SetEncSampRate(instance_type* inst,
- uint16_t sample_rate_hz) {
- return WebRtcIsac_SetEncSampRate(inst, sample_rate_hz);
- }
- static inline void SetEncSampRateInDecoder(instance_type* inst,
- uint16_t sample_rate_hz) {
- WebRtcIsac_SetEncSampRateInDecoder(inst, sample_rate_hz);
- }
- static inline void SetInitialBweBottleneck(
- instance_type* inst,
- int bottleneck_bits_per_second) {
- WebRtcIsac_SetInitialBweBottleneck(inst, bottleneck_bits_per_second);
- }
- static inline int16_t UpdateBwEstimate(instance_type* inst,
- const uint8_t* encoded,
- size_t packet_size,
- uint16_t rtp_seq_number,
- uint32_t send_ts,
- uint32_t arr_ts) {
- return WebRtcIsac_UpdateBwEstimate(inst, encoded, packet_size,
- rtp_seq_number, send_ts, arr_ts);
- }
- static inline int16_t SetMaxPayloadSize(instance_type* inst,
- int16_t max_payload_size_bytes) {
- return WebRtcIsac_SetMaxPayloadSize(inst, max_payload_size_bytes);
- }
- static inline int16_t SetMaxRate(instance_type* inst, int32_t max_bit_rate) {
- return WebRtcIsac_SetMaxRate(inst, max_bit_rate);
- }
-};
-
using AudioEncoderIsac = AudioEncoderIsacT<IsacFloat>;
-using AudioDecoderIsac = AudioDecoderIsacT<IsacFloat>;
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_AUDIO_ENCODER_ISAC_H_
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/audio_decoder_isac.cc b/webrtc/modules/audio_coding/codecs/isac/main/source/audio_decoder_isac.cc
new file mode 100644
index 0000000..89879ad
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/audio_decoder_isac.cc
@@ -0,0 +1,20 @@
+/*
+ * 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/isac/main/interface/audio_decoder_isac.h"
+
+#include "webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h"
+
+namespace webrtc {
+
+// Explicit instantiation:
+template class AudioDecoderIsacT<IsacFloat>;
+
+} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac.cc b/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac.cc
index eac7cc7..c72af30 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac.cc
@@ -16,6 +16,5 @@
// Explicit instantiation:
template class AudioEncoderIsacT<IsacFloat>;
-template class AudioDecoderIsacT<IsacFloat>;
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/isac_float_type.h b/webrtc/modules/audio_coding/codecs/isac/main/source/isac_float_type.h
new file mode 100644
index 0000000..0335548
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/isac_float_type.h
@@ -0,0 +1,117 @@
+/*
+ * 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_ISAC_MAIN_SOURCE_ISAC_FLOAT_TYPE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_FLOAT_TYPE_H_
+
+#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
+
+namespace webrtc {
+
+struct IsacFloat {
+ using instance_type = ISACStruct;
+ static const bool has_swb = true;
+ static inline int16_t Control(instance_type* inst,
+ int32_t rate,
+ int framesize) {
+ return WebRtcIsac_Control(inst, rate, framesize);
+ }
+ static inline int16_t ControlBwe(instance_type* inst,
+ int32_t rate_bps,
+ int frame_size_ms,
+ int16_t enforce_frame_size) {
+ return WebRtcIsac_ControlBwe(inst, rate_bps, frame_size_ms,
+ enforce_frame_size);
+ }
+ static inline int16_t Create(instance_type** inst) {
+ return WebRtcIsac_Create(inst);
+ }
+ static inline int DecodeInternal(instance_type* inst,
+ const uint8_t* encoded,
+ size_t len,
+ int16_t* decoded,
+ int16_t* speech_type) {
+ return WebRtcIsac_Decode(inst, encoded, len, decoded, speech_type);
+ }
+ static inline size_t DecodePlc(instance_type* inst,
+ int16_t* decoded,
+ size_t num_lost_frames) {
+ return WebRtcIsac_DecodePlc(inst, decoded, num_lost_frames);
+ }
+
+ static inline void DecoderInit(instance_type* inst) {
+ WebRtcIsac_DecoderInit(inst);
+ }
+ static inline int Encode(instance_type* inst,
+ const int16_t* speech_in,
+ uint8_t* encoded) {
+ return WebRtcIsac_Encode(inst, speech_in, encoded);
+ }
+ static inline int16_t EncoderInit(instance_type* inst, int16_t coding_mode) {
+ return WebRtcIsac_EncoderInit(inst, coding_mode);
+ }
+ static inline uint16_t EncSampRate(instance_type* inst) {
+ return WebRtcIsac_EncSampRate(inst);
+ }
+
+ static inline int16_t Free(instance_type* inst) {
+ return WebRtcIsac_Free(inst);
+ }
+ static inline void GetBandwidthInfo(instance_type* inst,
+ IsacBandwidthInfo* bwinfo) {
+ WebRtcIsac_GetBandwidthInfo(inst, bwinfo);
+ }
+ static inline int16_t GetErrorCode(instance_type* inst) {
+ return WebRtcIsac_GetErrorCode(inst);
+ }
+
+ static inline int16_t GetNewFrameLen(instance_type* inst) {
+ return WebRtcIsac_GetNewFrameLen(inst);
+ }
+ static inline void SetBandwidthInfo(instance_type* inst,
+ const IsacBandwidthInfo* bwinfo) {
+ WebRtcIsac_SetBandwidthInfo(inst, bwinfo);
+ }
+ static inline int16_t SetDecSampRate(instance_type* inst,
+ uint16_t sample_rate_hz) {
+ return WebRtcIsac_SetDecSampRate(inst, sample_rate_hz);
+ }
+ static inline int16_t SetEncSampRate(instance_type* inst,
+ uint16_t sample_rate_hz) {
+ return WebRtcIsac_SetEncSampRate(inst, sample_rate_hz);
+ }
+ static inline void SetEncSampRateInDecoder(instance_type* inst,
+ uint16_t sample_rate_hz) {
+ WebRtcIsac_SetEncSampRateInDecoder(inst, sample_rate_hz);
+ }
+ static inline void SetInitialBweBottleneck(instance_type* inst,
+ int bottleneck_bits_per_second) {
+ WebRtcIsac_SetInitialBweBottleneck(inst, bottleneck_bits_per_second);
+ }
+ static inline int16_t UpdateBwEstimate(instance_type* inst,
+ const uint8_t* encoded,
+ size_t packet_size,
+ uint16_t rtp_seq_number,
+ uint32_t send_ts,
+ uint32_t arr_ts) {
+ return WebRtcIsac_UpdateBwEstimate(inst, encoded, packet_size,
+ rtp_seq_number, send_ts, arr_ts);
+ }
+ static inline int16_t SetMaxPayloadSize(instance_type* inst,
+ int16_t max_payload_size_bytes) {
+ return WebRtcIsac_SetMaxPayloadSize(inst, max_payload_size_bytes);
+ }
+ static inline int16_t SetMaxRate(instance_type* inst, int32_t max_bit_rate) {
+ return WebRtcIsac_SetMaxRate(inst, max_bit_rate);
+ }
+};
+
+} // namespace webrtc
+#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_FLOAT_TYPE_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/codec_owner.cc b/webrtc/modules/audio_coding/main/acm2/codec_owner.cc
index c07ecec..669eadb 100644
--- a/webrtc/modules/audio_coding/main/acm2/codec_owner.cc
+++ b/webrtc/modules/audio_coding/main/acm2/codec_owner.cc
@@ -16,7 +16,9 @@
#include "webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h"
#include "webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h"
#include "webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_decoder_isacfix.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_decoder_isac.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.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"
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
index 1428eac..52f327b 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
@@ -22,9 +22,11 @@
#include "webrtc/modules/audio_coding/codecs/ilbc/interface/audio_decoder_ilbc.h"
#endif
#ifdef WEBRTC_CODEC_ISACFX
+#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_decoder_isacfix.h"
#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_decoder_isac.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h"
#endif
#ifdef WEBRTC_CODEC_OPUS
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
index d7446a8..537f16c 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -24,7 +24,9 @@
#include "webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h"
#include "webrtc/modules/audio_coding/codecs/ilbc/interface/audio_decoder_ilbc.h"
#include "webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_decoder_isacfix.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_decoder_isac.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"
