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@7644 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_isac.cc b/webrtc/modules/audio_coding/main/acm2/acm_isac.cc
index bc20c96..8fa96e5 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_isac.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_isac.cc
@@ -277,7 +277,6 @@
return;
}
codec_inst_ptr_->inst = NULL;
- state_ = codec_inst_ptr_;
}
ACMISAC::~ACMISAC() {
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder.cc b/webrtc/modules/audio_coding/neteq/audio_decoder.cc
index 04a74ee..d5a2762 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder.cc
+++ b/webrtc/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 @@
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/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
index 07b1b4b..eb07823 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
@@ -103,17 +103,17 @@
// 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::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 @@
#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 @@
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::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 @@
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::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::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 @@
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::Init() {
- return WebRtcG722_DecoderInit(static_cast<G722DecInst*>(state_));
+ return WebRtcG722_DecoderInit(dec_state_);
}
int AudioDecoderG722::PacketDuration(const uint8_t* encoded,
@@ -267,18 +264,15 @@
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 @@
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::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 @@
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 @@
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::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 @@
#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/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
index 214392e..b30331f 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
+++ b/webrtc/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 @@
virtual int Init();
private:
+ iLBC_decinst_t* dec_state_;
DISALLOW_COPY_AND_ASSIGN(AudioDecoderIlbc);
};
#endif
@@ -133,6 +150,7 @@
virtual int ErrorCode();
private:
+ ISACStruct* isac_state_;
DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsac);
};
#endif
@@ -153,6 +171,7 @@
virtual int ErrorCode();
private:
+ ISACFIX_MainStruct* isac_state_;
DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacFix);
};
#endif
@@ -169,10 +188,11 @@
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 @@
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 @@
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 @@
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/webrtc/modules/audio_coding/neteq/comfort_noise.cc b/webrtc/modules/audio_coding/neteq/comfort_noise.cc
index 31bb40c..e2be066 100644
--- a/webrtc/modules/audio_coding/neteq/comfort_noise.cc
+++ b/webrtc/modules/audio_coding/neteq/comfort_noise.cc
@@ -36,7 +36,7 @@
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 @@
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/webrtc/modules/audio_coding/neteq/interface/audio_decoder.h b/webrtc/modules/audio_coding/neteq/interface/audio_decoder.h
index 16d78c9..be85c4d 100644
--- a/webrtc/modules/audio_coding/neteq/interface/audio_decoder.h
+++ b/webrtc/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 @@
// 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 @@
// 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 @@
static SpeechType ConvertSpeechType(int16_t type);
size_t channels_;
- void* state_;
private:
DISALLOW_COPY_AND_ASSIGN(AudioDecoder);
diff --git a/webrtc/modules/audio_coding/neteq/normal.cc b/webrtc/modules/audio_coding/neteq/normal.cc
index 46d03fb..ca2c1ee 100644
--- a/webrtc/modules/audio_coding/neteq/normal.cc
+++ b/webrtc/modules/audio_coding/neteq/normal.cc
@@ -147,9 +147,9 @@
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));
}