diff options
author | kwiberg@webrtc.org <kwiberg@webrtc.org> | 2014-11-06 07:54:31 +0000 |
---|---|---|
committer | kwiberg@webrtc.org <kwiberg@webrtc.org> | 2014-11-06 07:54:31 +0000 |
commit | 3ebce78042e261e510a28ee4b2a64ea3148b6aa5 (patch) | |
tree | 62b6e85ecac56d7a38cf08bab999cd7d196a1bfe | |
parent | b831a9e3d5f9f0563d249b726cffa8a070e58aee (diff) | |
download | webrtc-3ebce78042e261e510a28ee4b2a64ea3148b6aa5.tar.gz |
Remove the state_ member from AudioDecoder
The subclasses that need a state pointer should declare them---with
the right type, not void*, to get rid of all those casts.
Two small but not quite trivial cleanups are included because they
blocked the state_ removal:
- AudioDecoderG722Stereo now inherits directly from AudioDecoder
instead of being a subclass of AudioDecoderG722.
- AudioDecoder now has a CngDecoderInstance member function, which
is implemented only by AudioDecoderCng. This replaces the previous
practice of calling AudioDecoder::state() and casting the result
to a CNG_dec_inst*. It still isn't pretty, but now the blemish is
plainly visible in the AudioDecoder class declaration.
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24169005
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7644 4adac7df-926f-26a2-2b94-8c16560cd09d
-rw-r--r-- | modules/audio_coding/main/acm2/acm_isac.cc | 1 | ||||
-rw-r--r-- | modules/audio_coding/neteq/audio_decoder.cc | 6 | ||||
-rw-r--r-- | modules/audio_coding/neteq/audio_decoder_impl.cc | 100 | ||||
-rw-r--r-- | modules/audio_coding/neteq/audio_decoder_impl.h | 30 | ||||
-rw-r--r-- | modules/audio_coding/neteq/comfort_noise.cc | 4 | ||||
-rw-r--r-- | modules/audio_coding/neteq/interface/audio_decoder.h | 9 | ||||
-rw-r--r-- | modules/audio_coding/neteq/normal.cc | 4 |
7 files changed, 86 insertions, 68 deletions
diff --git a/modules/audio_coding/main/acm2/acm_isac.cc b/modules/audio_coding/main/acm2/acm_isac.cc index bc20c961..8fa96e50 100644 --- a/modules/audio_coding/main/acm2/acm_isac.cc +++ b/modules/audio_coding/main/acm2/acm_isac.cc @@ -277,7 +277,6 @@ ACMISAC::ACMISAC(int16_t codec_id) return; } codec_inst_ptr_->inst = NULL; - state_ = codec_inst_ptr_; } ACMISAC::~ACMISAC() { diff --git a/modules/audio_coding/neteq/audio_decoder.cc b/modules/audio_coding/neteq/audio_decoder.cc index 04a74eef..d5a27628 100644 --- a/modules/audio_coding/neteq/audio_decoder.cc +++ b/modules/audio_coding/neteq/audio_decoder.cc @@ -12,6 +12,7 @@ #include <assert.h> +#include "webrtc/base/checks.h" #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h" namespace webrtc { @@ -51,6 +52,11 @@ bool AudioDecoder::PacketHasFec(const uint8_t* encoded, return false; } +CNG_dec_inst* AudioDecoder::CngDecoderInstance() { + FATAL() << "Not a CNG decoder"; + return NULL; +} + bool AudioDecoder::CodecSupported(NetEqDecoder codec_type) { switch (codec_type) { case kDecoderPCMu: diff --git a/modules/audio_coding/neteq/audio_decoder_impl.cc b/modules/audio_coding/neteq/audio_decoder_impl.cc index 07b1b4be..eb078234 100644 --- a/modules/audio_coding/neteq/audio_decoder_impl.cc +++ b/modules/audio_coding/neteq/audio_decoder_impl.cc @@ -103,17 +103,17 @@ AudioDecoderPcm16BMultiCh::AudioDecoderPcm16BMultiCh(int num_channels) { // iLBC #ifdef WEBRTC_CODEC_ILBC AudioDecoderIlbc::AudioDecoderIlbc() { - WebRtcIlbcfix_DecoderCreate(reinterpret_cast<iLBC_decinst_t**>(&state_)); + WebRtcIlbcfix_DecoderCreate(&dec_state_); } AudioDecoderIlbc::~AudioDecoderIlbc() { - WebRtcIlbcfix_DecoderFree(static_cast<iLBC_decinst_t*>(state_)); + WebRtcIlbcfix_DecoderFree(dec_state_); } int AudioDecoderIlbc::Decode(const uint8_t* encoded, size_t encoded_len, int16_t* decoded, SpeechType* speech_type) { int16_t temp_type = 1; // Default is speech. - int16_t ret = WebRtcIlbcfix_Decode(static_cast<iLBC_decinst_t*>(state_), + int16_t ret = WebRtcIlbcfix_Decode(dec_state_, reinterpret_cast<const int16_t*>(encoded), static_cast<int16_t>(encoded_len), decoded, &temp_type); @@ -122,12 +122,11 @@ int AudioDecoderIlbc::Decode(const uint8_t* encoded, size_t encoded_len, } int AudioDecoderIlbc::DecodePlc(int num_frames, int16_t* decoded) { - return WebRtcIlbcfix_NetEqPlc(static_cast<iLBC_decinst_t*>(state_), - decoded, num_frames); + return WebRtcIlbcfix_NetEqPlc(dec_state_, decoded, num_frames); } int AudioDecoderIlbc::Init() { - return WebRtcIlbcfix_Decoderinit30Ms(static_cast<iLBC_decinst_t*>(state_)); + return WebRtcIlbcfix_Decoderinit30Ms(dec_state_); } #endif @@ -135,19 +134,18 @@ int AudioDecoderIlbc::Init() { #ifdef WEBRTC_CODEC_ISAC AudioDecoderIsac::AudioDecoderIsac(int decode_sample_rate_hz) { DCHECK(decode_sample_rate_hz == 16000 || decode_sample_rate_hz == 32000); - WebRtcIsac_Create(reinterpret_cast<ISACStruct**>(&state_)); - WebRtcIsac_SetDecSampRate(static_cast<ISACStruct*>(state_), - decode_sample_rate_hz); + WebRtcIsac_Create(&isac_state_); + WebRtcIsac_SetDecSampRate(isac_state_, decode_sample_rate_hz); } AudioDecoderIsac::~AudioDecoderIsac() { - WebRtcIsac_Free(static_cast<ISACStruct*>(state_)); + WebRtcIsac_Free(isac_state_); } int AudioDecoderIsac::Decode(const uint8_t* encoded, size_t encoded_len, int16_t* decoded, SpeechType* speech_type) { int16_t temp_type = 1; // Default is speech. - int16_t ret = WebRtcIsac_Decode(static_cast<ISACStruct*>(state_), + int16_t ret = WebRtcIsac_Decode(isac_state_, encoded, static_cast<int16_t>(encoded_len), decoded, &temp_type); @@ -159,7 +157,7 @@ int AudioDecoderIsac::DecodeRedundant(const uint8_t* encoded, size_t encoded_len, int16_t* decoded, SpeechType* speech_type) { int16_t temp_type = 1; // Default is speech. - int16_t ret = WebRtcIsac_DecodeRcu(static_cast<ISACStruct*>(state_), + int16_t ret = WebRtcIsac_DecodeRcu(isac_state_, encoded, static_cast<int16_t>(encoded_len), decoded, &temp_type); @@ -168,12 +166,11 @@ int AudioDecoderIsac::DecodeRedundant(const uint8_t* encoded, } int AudioDecoderIsac::DecodePlc(int num_frames, int16_t* decoded) { - return WebRtcIsac_DecodePlc(static_cast<ISACStruct*>(state_), - decoded, num_frames); + return WebRtcIsac_DecodePlc(isac_state_, decoded, num_frames); } int AudioDecoderIsac::Init() { - return WebRtcIsac_DecoderInit(static_cast<ISACStruct*>(state_)); + return WebRtcIsac_DecoderInit(isac_state_); } int AudioDecoderIsac::IncomingPacket(const uint8_t* payload, @@ -181,7 +178,7 @@ int AudioDecoderIsac::IncomingPacket(const uint8_t* payload, uint16_t rtp_sequence_number, uint32_t rtp_timestamp, uint32_t arrival_timestamp) { - return WebRtcIsac_UpdateBwEstimate(static_cast<ISACStruct*>(state_), + return WebRtcIsac_UpdateBwEstimate(isac_state_, payload, static_cast<int32_t>(payload_len), rtp_sequence_number, @@ -190,24 +187,24 @@ int AudioDecoderIsac::IncomingPacket(const uint8_t* payload, } int AudioDecoderIsac::ErrorCode() { - return WebRtcIsac_GetErrorCode(static_cast<ISACStruct*>(state_)); + return WebRtcIsac_GetErrorCode(isac_state_); } #endif // iSAC fix #ifdef WEBRTC_CODEC_ISACFX AudioDecoderIsacFix::AudioDecoderIsacFix() { - WebRtcIsacfix_Create(reinterpret_cast<ISACFIX_MainStruct**>(&state_)); + WebRtcIsacfix_Create(&isac_state_); } AudioDecoderIsacFix::~AudioDecoderIsacFix() { - WebRtcIsacfix_Free(static_cast<ISACFIX_MainStruct*>(state_)); + WebRtcIsacfix_Free(isac_state_); } int AudioDecoderIsacFix::Decode(const uint8_t* encoded, size_t encoded_len, int16_t* decoded, SpeechType* speech_type) { int16_t temp_type = 1; // Default is speech. - int16_t ret = WebRtcIsacfix_Decode(static_cast<ISACFIX_MainStruct*>(state_), + int16_t ret = WebRtcIsacfix_Decode(isac_state_, encoded, static_cast<int16_t>(encoded_len), decoded, &temp_type); @@ -216,7 +213,7 @@ int AudioDecoderIsacFix::Decode(const uint8_t* encoded, size_t encoded_len, } int AudioDecoderIsacFix::Init() { - return WebRtcIsacfix_DecoderInit(static_cast<ISACFIX_MainStruct*>(state_)); + return WebRtcIsacfix_DecoderInit(isac_state_); } int AudioDecoderIsacFix::IncomingPacket(const uint8_t* payload, @@ -225,32 +222,32 @@ int AudioDecoderIsacFix::IncomingPacket(const uint8_t* payload, uint32_t rtp_timestamp, uint32_t arrival_timestamp) { return WebRtcIsacfix_UpdateBwEstimate( - static_cast<ISACFIX_MainStruct*>(state_), + isac_state_, payload, static_cast<int32_t>(payload_len), rtp_sequence_number, rtp_timestamp, arrival_timestamp); } int AudioDecoderIsacFix::ErrorCode() { - return WebRtcIsacfix_GetErrorCode(static_cast<ISACFIX_MainStruct*>(state_)); + return WebRtcIsacfix_GetErrorCode(isac_state_); } #endif // G.722 #ifdef WEBRTC_CODEC_G722 AudioDecoderG722::AudioDecoderG722() { - WebRtcG722_CreateDecoder(reinterpret_cast<G722DecInst**>(&state_)); + WebRtcG722_CreateDecoder(&dec_state_); } AudioDecoderG722::~AudioDecoderG722() { - WebRtcG722_FreeDecoder(static_cast<G722DecInst*>(state_)); + WebRtcG722_FreeDecoder(dec_state_); } int AudioDecoderG722::Decode(const uint8_t* encoded, size_t encoded_len, int16_t* decoded, SpeechType* speech_type) { int16_t temp_type = 1; // Default is speech. int16_t ret = WebRtcG722_Decode( - static_cast<G722DecInst*>(state_), + dec_state_, const_cast<int16_t*>(reinterpret_cast<const int16_t*>(encoded)), static_cast<int16_t>(encoded_len), decoded, &temp_type); *speech_type = ConvertSpeechType(temp_type); @@ -258,7 +255,7 @@ int AudioDecoderG722::Decode(const uint8_t* encoded, size_t encoded_len, } int AudioDecoderG722::Init() { - return WebRtcG722_DecoderInit(static_cast<G722DecInst*>(state_)); + return WebRtcG722_DecoderInit(dec_state_); } int AudioDecoderG722::PacketDuration(const uint8_t* encoded, @@ -267,18 +264,15 @@ int AudioDecoderG722::PacketDuration(const uint8_t* encoded, return static_cast<int>(2 * encoded_len / channels_); } -AudioDecoderG722Stereo::AudioDecoderG722Stereo() - : AudioDecoderG722(), - state_left_(state_), // Base member |state_| is used for left channel. - state_right_(NULL) { +AudioDecoderG722Stereo::AudioDecoderG722Stereo() { channels_ = 2; - // |state_left_| already created by the base class AudioDecoderG722. - WebRtcG722_CreateDecoder(reinterpret_cast<G722DecInst**>(&state_right_)); + WebRtcG722_CreateDecoder(&dec_state_left_); + WebRtcG722_CreateDecoder(&dec_state_right_); } AudioDecoderG722Stereo::~AudioDecoderG722Stereo() { - // |state_left_| will be freed by the base class AudioDecoderG722. - WebRtcG722_FreeDecoder(static_cast<G722DecInst*>(state_right_)); + WebRtcG722_FreeDecoder(dec_state_left_); + WebRtcG722_FreeDecoder(dec_state_right_); } int AudioDecoderG722Stereo::Decode(const uint8_t* encoded, size_t encoded_len, @@ -289,13 +283,13 @@ int AudioDecoderG722Stereo::Decode(const uint8_t* encoded, size_t encoded_len, SplitStereoPacket(encoded, encoded_len, encoded_deinterleaved); // Decode left and right. int16_t ret = WebRtcG722_Decode( - static_cast<G722DecInst*>(state_left_), + dec_state_left_, reinterpret_cast<int16_t*>(encoded_deinterleaved), static_cast<int16_t>(encoded_len / 2), decoded, &temp_type); if (ret >= 0) { int decoded_len = ret; ret = WebRtcG722_Decode( - static_cast<G722DecInst*>(state_right_), + dec_state_right_, reinterpret_cast<int16_t*>(&encoded_deinterleaved[encoded_len / 2]), static_cast<int16_t>(encoded_len / 2), &decoded[decoded_len], &temp_type); if (ret == decoded_len) { @@ -317,11 +311,10 @@ int AudioDecoderG722Stereo::Decode(const uint8_t* encoded, size_t encoded_len, } int AudioDecoderG722Stereo::Init() { - int ret = WebRtcG722_DecoderInit(static_cast<G722DecInst*>(state_right_)); - if (ret != 0) { - return ret; - } - return AudioDecoderG722::Init(); + int r = WebRtcG722_DecoderInit(dec_state_left_); + if (r != 0) + return r; + return WebRtcG722_DecoderInit(dec_state_right_); } // Split the stereo packet and place left and right channel after each other @@ -401,18 +394,17 @@ int AudioDecoderCelt::DecodePlc(int num_frames, int16_t* decoded) { AudioDecoderOpus::AudioDecoderOpus(int num_channels) { DCHECK(num_channels == 1 || num_channels == 2); channels_ = num_channels; - WebRtcOpus_DecoderCreate(reinterpret_cast<OpusDecInst**>(&state_), - static_cast<int>(channels_)); + WebRtcOpus_DecoderCreate(&dec_state_, static_cast<int>(channels_)); } AudioDecoderOpus::~AudioDecoderOpus() { - WebRtcOpus_DecoderFree(static_cast<OpusDecInst*>(state_)); + WebRtcOpus_DecoderFree(dec_state_); } int AudioDecoderOpus::Decode(const uint8_t* encoded, size_t encoded_len, int16_t* decoded, SpeechType* speech_type) { int16_t temp_type = 1; // Default is speech. - int16_t ret = WebRtcOpus_DecodeNew(static_cast<OpusDecInst*>(state_), encoded, + int16_t ret = WebRtcOpus_DecodeNew(dec_state_, encoded, static_cast<int16_t>(encoded_len), decoded, &temp_type); if (ret > 0) @@ -425,7 +417,7 @@ int AudioDecoderOpus::DecodeRedundant(const uint8_t* encoded, size_t encoded_len, int16_t* decoded, SpeechType* speech_type) { int16_t temp_type = 1; // Default is speech. - int16_t ret = WebRtcOpus_DecodeFec(static_cast<OpusDecInst*>(state_), encoded, + int16_t ret = WebRtcOpus_DecodeFec(dec_state_, encoded, static_cast<int16_t>(encoded_len), decoded, &temp_type); if (ret > 0) @@ -435,12 +427,12 @@ int AudioDecoderOpus::DecodeRedundant(const uint8_t* encoded, } int AudioDecoderOpus::Init() { - return WebRtcOpus_DecoderInitNew(static_cast<OpusDecInst*>(state_)); + return WebRtcOpus_DecoderInitNew(dec_state_); } int AudioDecoderOpus::PacketDuration(const uint8_t* encoded, size_t encoded_len) { - return WebRtcOpus_DurationEst(static_cast<OpusDecInst*>(state_), + return WebRtcOpus_DurationEst(dec_state_, encoded, static_cast<int>(encoded_len)); } @@ -458,19 +450,15 @@ bool AudioDecoderOpus::PacketHasFec(const uint8_t* encoded, #endif AudioDecoderCng::AudioDecoderCng() { - WebRtcCng_CreateDec(reinterpret_cast<CNG_dec_inst**>(&state_)); - assert(state_); + CHECK_EQ(0, WebRtcCng_CreateDec(&dec_state_)); } AudioDecoderCng::~AudioDecoderCng() { - if (state_) { - WebRtcCng_FreeDec(static_cast<CNG_dec_inst*>(state_)); - } + WebRtcCng_FreeDec(dec_state_); } int AudioDecoderCng::Init() { - assert(state_); - return WebRtcCng_InitDec(static_cast<CNG_dec_inst*>(state_)); + return WebRtcCng_InitDec(dec_state_); } } // namespace webrtc diff --git a/modules/audio_coding/neteq/audio_decoder_impl.h b/modules/audio_coding/neteq/audio_decoder_impl.h index 214392e7..b30331f3 100644 --- a/modules/audio_coding/neteq/audio_decoder_impl.h +++ b/modules/audio_coding/neteq/audio_decoder_impl.h @@ -19,6 +19,22 @@ #include "webrtc/engine_configurations.h" #endif #include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h" +#ifdef WEBRTC_CODEC_G722 +#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h" +#endif +#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/isacfix.h" +#endif +#ifdef WEBRTC_CODEC_ISAC +#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h" +#endif +#ifdef WEBRTC_CODEC_OPUS +#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h" +#endif #include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h" #include "webrtc/typedefs.h" @@ -109,6 +125,7 @@ class AudioDecoderIlbc : public AudioDecoder { virtual int Init(); private: + iLBC_decinst_t* dec_state_; DISALLOW_COPY_AND_ASSIGN(AudioDecoderIlbc); }; #endif @@ -133,6 +150,7 @@ class AudioDecoderIsac : public AudioDecoder { virtual int ErrorCode(); private: + ISACStruct* isac_state_; DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsac); }; #endif @@ -153,6 +171,7 @@ class AudioDecoderIsacFix : public AudioDecoder { virtual int ErrorCode(); private: + ISACFIX_MainStruct* isac_state_; DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacFix); }; #endif @@ -169,10 +188,11 @@ class AudioDecoderG722 : public AudioDecoder { virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len); private: + G722DecInst* dec_state_; DISALLOW_COPY_AND_ASSIGN(AudioDecoderG722); }; -class AudioDecoderG722Stereo : public AudioDecoderG722 { +class AudioDecoderG722Stereo : public AudioDecoder { public: AudioDecoderG722Stereo(); virtual ~AudioDecoderG722Stereo(); @@ -189,8 +209,8 @@ class AudioDecoderG722Stereo : public AudioDecoderG722 { void SplitStereoPacket(const uint8_t* encoded, size_t encoded_len, uint8_t* encoded_deinterleaved); - void* const state_left_; - void* state_right_; + G722DecInst* dec_state_left_; + G722DecInst* dec_state_right_; DISALLOW_COPY_AND_ASSIGN(AudioDecoderG722Stereo); }; @@ -229,6 +249,7 @@ class AudioDecoderOpus : public AudioDecoder { virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const; private: + OpusDecInst* dec_state_; DISALLOW_COPY_AND_ASSIGN(AudioDecoderOpus); }; #endif @@ -252,7 +273,10 @@ class AudioDecoderCng : public AudioDecoder { uint32_t rtp_timestamp, uint32_t arrival_timestamp) { return -1; } + virtual CNG_dec_inst* CngDecoderInstance() OVERRIDE { return dec_state_; } + private: + CNG_dec_inst* dec_state_; DISALLOW_COPY_AND_ASSIGN(AudioDecoderCng); }; diff --git a/modules/audio_coding/neteq/comfort_noise.cc b/modules/audio_coding/neteq/comfort_noise.cc index 31bb40c9..e2be066e 100644 --- a/modules/audio_coding/neteq/comfort_noise.cc +++ b/modules/audio_coding/neteq/comfort_noise.cc @@ -36,7 +36,7 @@ int ComfortNoise::UpdateParameters(Packet* packet) { return kUnknownPayloadType; } decoder_database_->SetActiveCngDecoder(packet->header.payloadType); - CNG_dec_inst* cng_inst = static_cast<CNG_dec_inst*>(cng_decoder->state()); + CNG_dec_inst* cng_inst = cng_decoder->CngDecoderInstance(); int16_t ret = WebRtcCng_UpdateSid(cng_inst, packet->payload, packet->payload_length); @@ -72,7 +72,7 @@ int ComfortNoise::Generate(size_t requested_length, if (!cng_decoder) { return kUnknownPayloadType; } - CNG_dec_inst* cng_inst = static_cast<CNG_dec_inst*>(cng_decoder->state()); + CNG_dec_inst* cng_inst = cng_decoder->CngDecoderInstance(); // The expression &(*output)[0][0] is a pointer to the first element in // the first channel. if (WebRtcCng_Generate(cng_inst, &(*output)[0][0], diff --git a/modules/audio_coding/neteq/interface/audio_decoder.h b/modules/audio_coding/neteq/interface/audio_decoder.h index 16d78c9e..be85c4dd 100644 --- a/modules/audio_coding/neteq/interface/audio_decoder.h +++ b/modules/audio_coding/neteq/interface/audio_decoder.h @@ -14,6 +14,7 @@ #include <stdlib.h> // NULL #include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -63,7 +64,7 @@ class AudioDecoder { // Used by PacketDuration below. Save the value -1 for errors. enum { kNotImplemented = -2 }; - AudioDecoder() : channels_(1), state_(NULL) {} + AudioDecoder() : channels_(1) {} virtual ~AudioDecoder() {} // Decodes |encode_len| bytes from |encoded| and writes the result in @@ -114,8 +115,9 @@ class AudioDecoder { // Returns true if the packet has FEC and false otherwise. virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const; - // Returns the underlying decoder state. - void* state() { return state_; } + // If this is a CNG decoder, return the underlying CNG_dec_inst*. If this + // isn't a CNG decoder, don't call this method. + virtual CNG_dec_inst* CngDecoderInstance(); // Returns true if |codec_type| is supported. static bool CodecSupported(NetEqDecoder codec_type); @@ -134,7 +136,6 @@ class AudioDecoder { static SpeechType ConvertSpeechType(int16_t type); size_t channels_; - void* state_; private: DISALLOW_COPY_AND_ASSIGN(AudioDecoder); diff --git a/modules/audio_coding/neteq/normal.cc b/modules/audio_coding/neteq/normal.cc index 46d03fb8..ca2c1ee5 100644 --- a/modules/audio_coding/neteq/normal.cc +++ b/modules/audio_coding/neteq/normal.cc @@ -147,9 +147,9 @@ int Normal::Process(const int16_t* input, AudioDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder(); if (cng_decoder) { - CNG_dec_inst* cng_inst = static_cast<CNG_dec_inst*>(cng_decoder->state()); // Generate long enough for 32kHz. - if (WebRtcCng_Generate(cng_inst, cng_output, kCngLength, 0) < 0) { + if (WebRtcCng_Generate(cng_decoder->CngDecoderInstance(), cng_output, + kCngLength, 0) < 0) { // Error returned; set return vector to all zeros. memset(cng_output, 0, sizeof(cng_output)); } |