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AudioDecoder: Replace Init() with Reset()

Browse Source 
The Init() method was previously used to initialize and reset
decoders, and returned an error code. The new Reset() method is used
for reset only; the constructor is now responsible for fully
initializing the AudioDecoder.

Reset() doesn't return an error code; it turned out that none of the
functions it ended up calling could actually fail, so this CL removes
their error return codes as well.

R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1319683002 .

Cr-Commit-Position: refs/heads/master@{#9798}
This commit is contained in:
Karl Wiberg
2015-08-27 15:22:11 +02:00
parent 1c3dd38cb8
commit 4376648df0
38 changed files with 105 additions and 208 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
webrtc/modules/audio_coding/codecs/audio_decoder.h
View File

@ -64,8 +64,8 @@ class AudioDecoder {
// one or several lost packets.
virtual size_t DecodePlc(size_t num_frames, int16_t* decoded);
// Initializes the decoder.
virtual int Init() = 0;
// Resets the decoder state (empty buffers etc.).
virtual void Reset() = 0;
// Notifies the decoder of an incoming packet to NetEQ.
virtual int IncomingPacket(const uint8_t* payload,

12
webrtc/modules/audio_coding/codecs/cng/cng_unittest.cc
View File

@ -194,7 +194,7 @@ TEST_F(CngTest, CngUpdateSid) {
EXPECT_EQ(0, WebRtcCng_CreateDec(&cng_dec_inst_));
EXPECT_EQ(0, WebRtcCng_InitEnc(cng_enc_inst_, 16000, kSidNormalIntervalUpdate,
kCNGNumParamsNormal));
EXPECT_EQ(0, WebRtcCng_InitDec(cng_dec_inst_));
WebRtcCng_InitDec(cng_dec_inst_);
// Run normal Encode and UpdateSid.
EXPECT_EQ(kCNGNumParamsNormal + 1, WebRtcCng_Encode(
@ -205,7 +205,7 @@ TEST_F(CngTest, CngUpdateSid) {
// Reinit with new length.
EXPECT_EQ(0, WebRtcCng_InitEnc(cng_enc_inst_, 16000, kSidNormalIntervalUpdate,
kCNGNumParamsHigh));
EXPECT_EQ(0, WebRtcCng_InitDec(cng_dec_inst_));
WebRtcCng_InitDec(cng_dec_inst_);
// Expect 0 because of unstable parameters after switching length.
EXPECT_EQ(0, WebRtcCng_Encode(cng_enc_inst_, speech_data_, 160, sid_data,
@ -242,7 +242,7 @@ TEST_F(CngTest, CngUpdateSidErroneous) {
EXPECT_EQ(6220, WebRtcCng_GetErrorCodeDec(cng_dec_inst_));
// Initialize decoder.
EXPECT_EQ(0, WebRtcCng_InitDec(cng_dec_inst_));
WebRtcCng_InitDec(cng_dec_inst_);
// First run with valid parameters, then with too many CNG parameters.
// The function will operate correctly by only reading the maximum number of
@ -268,7 +268,7 @@ TEST_F(CngTest, CngGenerate) {
EXPECT_EQ(0, WebRtcCng_CreateDec(&cng_dec_inst_));
EXPECT_EQ(0, WebRtcCng_InitEnc(cng_enc_inst_, 16000, kSidNormalIntervalUpdate,
kCNGNumParamsNormal));
EXPECT_EQ(0, WebRtcCng_InitDec(cng_dec_inst_));
WebRtcCng_InitDec(cng_dec_inst_);
// Normal Encode.
EXPECT_EQ(kCNGNumParamsNormal + 1, WebRtcCng_Encode(
@ -301,7 +301,7 @@ TEST_F(CngTest, CngAutoSid) {
EXPECT_EQ(0, WebRtcCng_CreateDec(&cng_dec_inst_));
EXPECT_EQ(0, WebRtcCng_InitEnc(cng_enc_inst_, 16000, kSidNormalIntervalUpdate,
kCNGNumParamsNormal));
EXPECT_EQ(0, WebRtcCng_InitDec(cng_dec_inst_));
WebRtcCng_InitDec(cng_dec_inst_);
// Normal Encode, 100 msec, where no SID data should be generated.
for (int i = 0; i < 10; i++) {
@ -328,7 +328,7 @@ TEST_F(CngTest, CngAutoSidShort) {
EXPECT_EQ(0, WebRtcCng_CreateDec(&cng_dec_inst_));
EXPECT_EQ(0, WebRtcCng_InitEnc(cng_enc_inst_, 16000, kSidShortIntervalUpdate,
kCNGNumParamsNormal));
EXPECT_EQ(0, WebRtcCng_InitDec(cng_dec_inst_));
WebRtcCng_InitDec(cng_dec_inst_);
// First call will never generate SID, unless forced to.
EXPECT_EQ(0, WebRtcCng_Encode(cng_enc_inst_, speech_data_, 160, sid_data,

2
webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h
View File

@ -70,7 +70,7 @@ int16_t WebRtcCng_CreateDec(CNG_dec_inst** cng_inst);
int WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, int fs, int16_t interval,
int16_t quality);
int16_t WebRtcCng_InitDec(CNG_dec_inst* cng_inst);
void WebRtcCng_InitDec(CNG_dec_inst* cng_inst);
/****************************************************************************
* WebRtcCng_FreeEnc/Dec(...)

4
webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c
View File

@ -169,7 +169,7 @@ int WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, int fs, int16_t interval,
return 0;
}
int16_t WebRtcCng_InitDec(CNG_dec_inst* cng_inst) {
void WebRtcCng_InitDec(CNG_dec_inst* cng_inst) {
int i;
WebRtcCngDecoder* inst = (WebRtcCngDecoder*) cng_inst;
@ -188,8 +188,6 @@ int16_t WebRtcCng_InitDec(CNG_dec_inst* cng_inst) {
inst->dec_used_reflCoefs[0] = 0;
inst->dec_used_energy = 0;
inst->initflag = 1;
return 0;
}
/****************************************************************************

6
webrtc/modules/audio_coding/codecs/g722/g722_decode.c
View File

@ -157,11 +157,7 @@ static void block4(G722DecoderState *s, int band, int d)
G722DecoderState* WebRtc_g722_decode_init(G722DecoderState* s,
int rate,
int options) {
if (s == NULL)
{
if ((s = (G722DecoderState *) malloc(sizeof(*s))) == NULL)
return NULL;
}
s = s ? s : malloc(sizeof(*s));
memset(s, 0, sizeof(*s));
if (rate == 48000)
s->bits_per_sample = 6;

11
webrtc/modules/audio_coding/codecs/g722/g722_interface.c
View File

@ -66,17 +66,10 @@ int16_t WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst)
}
}
int16_t WebRtcG722_DecoderInit(G722DecInst *G722dec_inst)
{
void WebRtcG722_DecoderInit(G722DecInst* inst) {
// Create and/or reset the G.722 decoder
// Bitrate 64 kbps and wideband mode (2)
G722dec_inst = (G722DecInst *) WebRtc_g722_decode_init(
(G722DecoderState*) G722dec_inst, 64000, 2);
if (G722dec_inst == NULL) {
return -1;
} else {
return 0;
}
WebRtc_g722_decode_init((G722DecoderState*)inst, 64000, 2);
}
int WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)

12
webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h
View File

@ -113,22 +113,16 @@ size_t WebRtcG722_Encode(G722EncInst* G722enc_inst,
*/
int16_t WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst);
/****************************************************************************
* WebRtcG722_DecoderInit(...)
*
* This function initializes a G729 instance
* This function initializes a G722 instance
*
* Input:
* - G729_decinst_t : G729 instance, i.e. the user that should receive
* be initialized
*
* Return value : 0 - Ok
* -1 - Error
* - inst : G722 instance
*/
int16_t WebRtcG722_DecoderInit(G722DecInst *G722dec_inst);
void WebRtcG722_DecoderInit(G722DecInst* inst);
/****************************************************************************
* WebRtcG722_FreeDecoder(...)

6
webrtc/modules/audio_coding/codecs/ilbc/ilbc.c
View File

@ -131,13 +131,11 @@ int16_t WebRtcIlbcfix_DecoderInit(IlbcDecoderInstance* iLBCdec_inst,
return(-1);
}
}
int16_t WebRtcIlbcfix_DecoderInit20Ms(IlbcDecoderInstance *iLBCdec_inst) {
void WebRtcIlbcfix_DecoderInit20Ms(IlbcDecoderInstance* iLBCdec_inst) {
WebRtcIlbcfix_InitDecode((IlbcDecoder*) iLBCdec_inst, 20, 1);
return(0);
}
int16_t WebRtcIlbcfix_Decoderinit30Ms(IlbcDecoderInstance *iLBCdec_inst) {
void WebRtcIlbcfix_Decoderinit30Ms(IlbcDecoderInstance* iLBCdec_inst) {
WebRtcIlbcfix_InitDecode((IlbcDecoder*) iLBCdec_inst, 30, 1);
return(0);
}

4
webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h
View File

@ -159,8 +159,8 @@ extern "C" {
int16_t WebRtcIlbcfix_DecoderInit(IlbcDecoderInstance *iLBCdec_inst,
int16_t frameLen);
int16_t WebRtcIlbcfix_DecoderInit20Ms(IlbcDecoderInstance *iLBCdec_inst);
int16_t WebRtcIlbcfix_Decoderinit30Ms(IlbcDecoderInstance *iLBCdec_inst);
void WebRtcIlbcfix_DecoderInit20Ms(IlbcDecoderInstance* iLBCdec_inst);
void WebRtcIlbcfix_Decoderinit30Ms(IlbcDecoderInstance* iLBCdec_inst);
/****************************************************************************
* WebRtcIlbcfix_Decode(...)

2
webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
View File

@ -95,7 +95,7 @@ class AudioDecoderIsacT final : public AudioDecoder {
bool HasDecodePlc() const override;
size_t DecodePlc(size_t num_frames, int16_t* decoded) override;
int Init() override;
void Reset() override;
int IncomingPacket(const uint8_t* payload,
size_t payload_len,
uint16_t rtp_sequence_number,

6
webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
View File

@ -185,7 +185,7 @@ template <typename T>
AudioDecoderIsacT<T>::AudioDecoderIsacT(LockedIsacBandwidthInfo* bwinfo)
: bwinfo_(bwinfo), decoder_sample_rate_hz_(-1) {
CHECK_EQ(0, T::Create(&isac_state_));
CHECK_EQ(0, T::DecoderInit(isac_state_));
T::DecoderInit(isac_state_);
if (bwinfo_) {
IsacBandwidthInfo bwinfo;
T::GetBandwidthInfo(isac_state_, &bwinfo);
@ -232,8 +232,8 @@ size_t AudioDecoderIsacT<T>::DecodePlc(size_t num_frames, int16_t* decoded) {
}
template <typename T>
int AudioDecoderIsacT<T>::Init() {
return T::DecoderInit(isac_state_);
void AudioDecoderIsacT<T>::Reset() {
T::DecoderInit(isac_state_);
}
template <typename T>

4
webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h
View File

@ -50,8 +50,8 @@ struct IsacFix {
size_t num_lost_frames) {
return WebRtcIsacfix_DecodePlc(inst, decoded, num_lost_frames);
}
static inline int16_t DecoderInit(instance_type* inst) {
return WebRtcIsacfix_DecoderInit(inst);
static inline void DecoderInit(instance_type* inst) {
WebRtcIsacfix_DecoderInit(inst);
}
static inline int Encode(instance_type* inst,
const int16_t* speech_in,

7
webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h
View File

@ -174,14 +174,9 @@ extern "C" {
*
* Input:
* - ISAC_main_inst : ISAC instance.
*
* Return value
* : 0 - Ok
* -1 - Error
*/
int16_t WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct *ISAC_main_inst);
void WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct* ISAC_main_inst);
/****************************************************************************
* WebRtcIsacfix_UpdateBwEstimate1(...)

8
webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
View File

@ -568,13 +568,9 @@ int16_t WebRtcIsacfix_GetNewBitStream(ISACFIX_MainStruct *ISAC_main_inst,
*
* Input:
* - ISAC_main_inst : ISAC instance.
*
* Return value
* : 0 - Ok
* -1 - Error
*/
int16_t WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct *ISAC_main_inst)
void WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct *ISAC_main_inst)
{
ISACFIX_SubStruct *ISAC_inst;
@ -597,8 +593,6 @@ int16_t WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct *ISAC_main_inst)
#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
WebRtcIsacfix_InitPreFilterbank(&ISAC_inst->ISACdec_obj.decimatorstr_obj);
#endif
return 0;
}

2
webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
View File

@ -48,7 +48,7 @@ void IsacSpeedTest::SetUp() {
// Create encoder memory.
EXPECT_EQ(0, WebRtcIsacfix_Create(&ISACFIX_main_inst_));
EXPECT_EQ(0, WebRtcIsacfix_EncoderInit(ISACFIX_main_inst_, 1));
EXPECT_EQ(0, WebRtcIsacfix_DecoderInit(ISACFIX_main_inst_));
WebRtcIsacfix_DecoderInit(ISACFIX_main_inst_);
// Set bitrate and block length.
EXPECT_EQ(0, WebRtcIsacfix_Control(ISACFIX_main_inst_, bit_rate_,
block_duration_ms_));

7
webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
View File

@ -539,12 +539,7 @@ int main(int argc, char* argv[])
printf("\n\n Error in encoderinit: %d.\n\n", errtype);
}
err = WebRtcIsacfix_DecoderInit(ISAC_main_inst);
/* Error check */
if (err < 0) {
errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
printf("\n\n Error in decoderinit: %d.\n\n", errtype);
}
WebRtcIsacfix_DecoderInit(ISAC_main_inst);
}

4
webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h
View File

@ -50,8 +50,8 @@ struct IsacFloat {
return WebRtcIsac_DecodePlc(inst, decoded, num_lost_frames);
}
static inline int16_t DecoderInit(instance_type* inst) {
return WebRtcIsac_DecoderInit(inst);
static inline void DecoderInit(instance_type* inst) {
WebRtcIsac_DecoderInit(inst);
}
static inline int Encode(instance_type* inst,
const int16_t* speech_in,

8
webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
View File

@ -157,15 +157,9 @@ extern "C" {
*
* Input:
* - ISAC_main_inst : ISAC instance.
*
* Return value
* : 0 - Ok
* -1 - Error
*/
int16_t WebRtcIsac_DecoderInit(
ISACStruct* ISAC_main_inst);
void WebRtcIsac_DecoderInit(ISACStruct* ISAC_main_inst);
/******************************************************************************
* WebRtcIsac_UpdateBwEstimate(...)

26
webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
View File

@ -924,12 +924,8 @@ int16_t WebRtcIsac_GetNewBitStream(ISACStruct* ISAC_main_inst,
*
* Input:
* - ISAC_main_inst : ISAC instance.
*
* Return value
* : 0 - Ok
* -1 - Error
*/
static int16_t DecoderInitLb(ISACLBStruct* instISAC) {
static void DecoderInitLb(ISACLBStruct* instISAC) {
int i;
/* Initialize stream vector to zero. */
for (i = 0; i < STREAM_SIZE_MAX_60; i++) {
@ -940,10 +936,9 @@ static int16_t DecoderInitLb(ISACLBStruct* instISAC) {
WebRtcIsac_InitPostFilterbank(
&instISAC->ISACdecLB_obj.postfiltbankstr_obj);
WebRtcIsac_InitPitchFilter(&instISAC->ISACdecLB_obj.pitchfiltstr_obj);
return 0;
}
static int16_t DecoderInitUb(ISACUBStruct* instISAC) {
static void DecoderInitUb(ISACUBStruct* instISAC) {
int i;
/* Init stream vector to zero */
for (i = 0; i < STREAM_SIZE_MAX_60; i++) {
@ -953,24 +948,18 @@ static int16_t DecoderInitUb(ISACUBStruct* instISAC) {
WebRtcIsac_InitMasking(&instISAC->ISACdecUB_obj.maskfiltstr_obj);
WebRtcIsac_InitPostFilterbank(
&instISAC->ISACdecUB_obj.postfiltbankstr_obj);
return (0);
}
int16_t WebRtcIsac_DecoderInit(ISACStruct* ISAC_main_inst) {
void WebRtcIsac_DecoderInit(ISACStruct* ISAC_main_inst) {
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
if (DecoderInitLb(&instISAC->instLB) < 0) {
return -1;
}
DecoderInitLb(&instISAC->instLB);
if (instISAC->decoderSamplingRateKHz == kIsacSuperWideband) {
memset(instISAC->synthesisFBState1, 0,
FB_STATE_SIZE_WORD32 * sizeof(int32_t));
memset(instISAC->synthesisFBState2, 0,
FB_STATE_SIZE_WORD32 * sizeof(int32_t));
if (DecoderInitUb(&(instISAC->instUB)) < 0) {
return -1;
}
DecoderInitUb(&(instISAC->instUB));
}
if ((instISAC->initFlag & BIT_MASK_ENC_INIT) != BIT_MASK_ENC_INIT) {
WebRtcIsac_InitBandwidthEstimator(&instISAC->bwestimator_obj,
@ -979,7 +968,6 @@ int16_t WebRtcIsac_DecoderInit(ISACStruct* ISAC_main_inst) {
}
instISAC->initFlag |= BIT_MASK_DEC_INIT;
instISAC->resetFlag_8kHz = 0;
return 0;
}
@ -2353,9 +2341,7 @@ int16_t WebRtcIsac_SetDecSampRate(ISACStruct* ISAC_main_inst,
memset(instISAC->synthesisFBState2, 0,
FB_STATE_SIZE_WORD32 * sizeof(int32_t));
if (DecoderInitUb(&(instISAC->instUB)) < 0) {
return -1;
}
DecoderInitUb(&instISAC->instUB);
}
instISAC->decoderSamplingRateKHz = decoder_operational_rate;
return 0;

17
webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
View File

@ -499,13 +499,8 @@ int main(int argc, char* argv[]) {
return 0;
}
}
if (testNum != 2) {
if (WebRtcIsac_DecoderInit(ISAC_main_inst) < 0) {
printf("Error could not initialize the decoder \n");
cout << flush;
return 0;
}
}
if (testNum != 2)
WebRtcIsac_DecoderInit(ISAC_main_inst);
if (CodingMode == 1) {
err = WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize);
if (err < 0) {
@ -570,13 +565,7 @@ int main(int argc, char* argv[]) {
cout << flush;
}
err = WebRtcIsac_DecoderInit(ISAC_main_inst);
/* Error check */
if (err < 0) {
errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst);
printf("\n\n Error in decoderinit: %d.\n\n", errtype);
cout << flush;
}
WebRtcIsac_DecoderInit(ISAC_main_inst);
}
cur_framesmpls = 0;

8
webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
View File

@ -166,13 +166,7 @@ int main(int argc, char* argv[])
return -1;
}
// Initialize Decoder
if(WebRtcIsac_DecoderInit(codecInstance[clientCntr]) < 0)
{
printf("Could not initialize decoder of client %d\n",
clientCntr + 1);
return -1;
}
WebRtcIsac_DecoderInit(codecInstance[clientCntr]);
// setup Rate if in Instantaneous mode
if(codingMode != 0)

5
webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
View File

@ -253,10 +253,7 @@ int main(int argc, char* argv[]) {
printf("cannot initialize encoder\n");
return -1;
}
if (WebRtcIsac_DecoderInit(ISAC_main_inst) < 0) {
printf("cannot initialize decoder\n");
return -1;
}
WebRtcIsac_DecoderInit(ISAC_main_inst);
// {
// int32_t b1, b2;

4
webrtc/modules/audio_coding/codecs/isac/unittest.cc
View File

@ -111,7 +111,7 @@ void TestGetSetBandwidthInfo(const int16_t* speech_data,
typename T::instance_type* encdec;
ASSERT_EQ(0, T::Create(&encdec));
ASSERT_EQ(0, T::EncoderInit(encdec, adaptive ? 0 : 1));
ASSERT_EQ(0, T::DecoderInit(encdec));
T::DecoderInit(encdec);
ASSERT_EQ(0, T::SetEncSampRate(encdec, sample_rate_hz));
if (adaptive)
ASSERT_EQ(0, T::ControlBwe(encdec, bit_rate, frame_size_ms, false));
@ -129,7 +129,7 @@ void TestGetSetBandwidthInfo(const int16_t* speech_data,
ASSERT_EQ(0, T::Control(enc, bit_rate, frame_size_ms));
typename T::instance_type* dec;
ASSERT_EQ(0, T::Create(&dec));
ASSERT_EQ(0, T::DecoderInit(dec));
T::DecoderInit(dec);
T::SetInitialBweBottleneck(dec, bit_rate);
T::SetEncSampRateInDecoder(dec, sample_rate_hz);

5
webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h
View File

@ -212,11 +212,8 @@ int WebRtcOpus_DecoderChannels(OpusDecInst* inst);
*
* Input:
* - inst : Decoder context
*
* Return value : 0 - Success
* -1 - Error
*/
int16_t WebRtcOpus_DecoderInit(OpusDecInst* inst);
void WebRtcOpus_DecoderInit(OpusDecInst* inst);
/****************************************************************************
* WebRtcOpus_Decode(...)

8
webrtc/modules/audio_coding/codecs/opus/opus_interface.c
View File

@ -250,13 +250,9 @@ int WebRtcOpus_DecoderChannels(OpusDecInst* inst) {
return inst->channels;
}
int16_t WebRtcOpus_DecoderInit(OpusDecInst* inst) {
int error = opus_decoder_ctl(inst->decoder, OPUS_RESET_STATE);
if (error == OPUS_OK) {
void WebRtcOpus_DecoderInit(OpusDecInst* inst) {
opus_decoder_ctl(inst->decoder, OPUS_RESET_STATE);
inst->in_dtx_mode = 0;
return 0;
}
return -1;
}
/* For decoder to determine if it is to output speech or comfort noise. */

2
webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
View File

@ -376,7 +376,7 @@ TEST_P(OpusTest, OpusDecodeInit) {
kOpus20msFrameSamples, opus_decoder_, output_data_decode,
&audio_type)));
EXPECT_EQ(0, WebRtcOpus_DecoderInit(opus_decoder_));
WebRtcOpus_DecoderInit(opus_decoder_);
EXPECT_EQ(kOpus20msFrameSamples,
static_cast<size_t>(WebRtcOpus_Decode(

3
webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
View File

@ -988,9 +988,6 @@ TEST_F(AcmReceiverBitExactnessOldApi, MAYBE_48kHzOutputExternalDecoder) {
MockAudioDecoder mock_decoder;
// Set expectations on the mock decoder and also delegate the calls to the
// real decoder.
EXPECT_CALL(mock_decoder, Init())
.Times(AtLeast(1))
.WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::Init));
EXPECT_CALL(mock_decoder, IncomingPacket(_, _, _, _, _))
.Times(AtLeast(1))
.WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::IncomingPacket));

4
webrtc/modules/audio_coding/main/test/opus_test.cc
View File

@ -84,8 +84,8 @@ void OpusTest::Perform() {
// Create Opus decoders for mono and stereo for stand-alone testing of Opus.
ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_mono_decoder_, 1), -1);
ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_stereo_decoder_, 2), -1);
ASSERT_GT(WebRtcOpus_DecoderInit(opus_mono_decoder_), -1);
ASSERT_GT(WebRtcOpus_DecoderInit(opus_stereo_decoder_), -1);
WebRtcOpus_DecoderInit(opus_mono_decoder_);
WebRtcOpus_DecoderInit(opus_stereo_decoder_);
ASSERT_TRUE(acm_receiver_.get() != NULL);
EXPECT_EQ(0, acm_receiver_->InitializeReceiver());

39
webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
View File

@ -39,8 +39,7 @@ namespace webrtc {
// PCMu
int AudioDecoderPcmU::Init() {
return 0;
void AudioDecoderPcmU::Reset() {
}
size_t AudioDecoderPcmU::Channels() const {
return 1;
@ -70,8 +69,7 @@ size_t AudioDecoderPcmUMultiCh::Channels() const {
// PCMa
int AudioDecoderPcmA::Init() {
return 0;
void AudioDecoderPcmA::Reset() {
}
size_t AudioDecoderPcmA::Channels() const {
return 1;
@ -103,8 +101,7 @@ size_t AudioDecoderPcmAMultiCh::Channels() const {
#ifdef WEBRTC_CODEC_PCM16
AudioDecoderPcm16B::AudioDecoderPcm16B() {}
int AudioDecoderPcm16B::Init() {
return 0;
void AudioDecoderPcm16B::Reset() {
}
size_t AudioDecoderPcm16B::Channels() const {
return 1;
@ -143,6 +140,7 @@ size_t AudioDecoderPcm16BMultiCh::Channels() const {
#ifdef WEBRTC_CODEC_ILBC
AudioDecoderIlbc::AudioDecoderIlbc() {
WebRtcIlbcfix_DecoderCreate(&dec_state_);
WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
}
AudioDecoderIlbc::~AudioDecoderIlbc() {
@ -170,8 +168,8 @@ size_t AudioDecoderIlbc::DecodePlc(size_t num_frames, int16_t* decoded) {
return WebRtcIlbcfix_NetEqPlc(dec_state_, decoded, num_frames);
}
int AudioDecoderIlbc::Init() {
return WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
void AudioDecoderIlbc::Reset() {
WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
}
size_t AudioDecoderIlbc::Channels() const {
@ -183,6 +181,7 @@ size_t AudioDecoderIlbc::Channels() const {
#ifdef WEBRTC_CODEC_G722
AudioDecoderG722::AudioDecoderG722() {
WebRtcG722_CreateDecoder(&dec_state_);
WebRtcG722_DecoderInit(dec_state_);
}
AudioDecoderG722::~AudioDecoderG722() {
@ -206,8 +205,8 @@ int AudioDecoderG722::DecodeInternal(const uint8_t* encoded,
return static_cast<int>(ret);
}
int AudioDecoderG722::Init() {
return WebRtcG722_DecoderInit(dec_state_);
void AudioDecoderG722::Reset() {
WebRtcG722_DecoderInit(dec_state_);
}
int AudioDecoderG722::PacketDuration(const uint8_t* encoded,
@ -223,6 +222,8 @@ size_t AudioDecoderG722::Channels() const {
AudioDecoderG722Stereo::AudioDecoderG722Stereo() {
WebRtcG722_CreateDecoder(&dec_state_left_);
WebRtcG722_CreateDecoder(&dec_state_right_);
WebRtcG722_DecoderInit(dec_state_left_);
WebRtcG722_DecoderInit(dec_state_right_);
}
AudioDecoderG722Stereo::~AudioDecoderG722Stereo() {
@ -265,11 +266,9 @@ size_t AudioDecoderG722Stereo::Channels() const {
return 2;
}
int AudioDecoderG722Stereo::Init() {
int r = WebRtcG722_DecoderInit(dec_state_left_);
if (r != 0)
return r;
return WebRtcG722_DecoderInit(dec_state_right_);
void AudioDecoderG722Stereo::Reset() {
WebRtcG722_DecoderInit(dec_state_left_);
WebRtcG722_DecoderInit(dec_state_right_);
}
// Split the stereo packet and place left and right channel after each other
@ -306,6 +305,7 @@ 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() {
@ -348,8 +348,8 @@ int AudioDecoderOpus::DecodeRedundantInternal(const uint8_t* encoded,
return ret;
}
int AudioDecoderOpus::Init() {
return WebRtcOpus_DecoderInit(dec_state_);
void AudioDecoderOpus::Reset() {
WebRtcOpus_DecoderInit(dec_state_);
}
int AudioDecoderOpus::PacketDuration(const uint8_t* encoded,
@ -381,14 +381,15 @@ size_t AudioDecoderOpus::Channels() const {
AudioDecoderCng::AudioDecoderCng() {
CHECK_EQ(0, WebRtcCng_CreateDec(&dec_state_));
WebRtcCng_InitDec(dec_state_);
}
AudioDecoderCng::~AudioDecoderCng() {
WebRtcCng_FreeDec(dec_state_);
}
int AudioDecoderCng::Init() {
return WebRtcCng_InitDec(dec_state_);
void AudioDecoderCng::Reset() {
WebRtcCng_InitDec(dec_state_);
}
int AudioDecoderCng::IncomingPacket(const uint8_t* payload,

16
webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
View File

@ -37,7 +37,7 @@ namespace webrtc {
class AudioDecoderPcmU : public AudioDecoder {
public:
AudioDecoderPcmU() {}
int Init() override;
void Reset() override;
int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
size_t Channels() const override;
@ -55,7 +55,7 @@ class AudioDecoderPcmU : public AudioDecoder {
class AudioDecoderPcmA : public AudioDecoder {
public:
AudioDecoderPcmA() {}
int Init() override;
void Reset() override;
int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
size_t Channels() const override;
@ -102,7 +102,7 @@ class AudioDecoderPcmAMultiCh : public AudioDecoderPcmA {
class AudioDecoderPcm16B : public AudioDecoder {
public:
AudioDecoderPcm16B();
int Init() override;
void Reset() override;
int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
size_t Channels() const override;
@ -138,7 +138,7 @@ class AudioDecoderIlbc : public AudioDecoder {
~AudioDecoderIlbc() override;
bool HasDecodePlc() const override;
size_t DecodePlc(size_t num_frames, int16_t* decoded) override;
int Init() override;
void Reset() override;
size_t Channels() const override;
protected:
@ -160,7 +160,7 @@ class AudioDecoderG722 : public AudioDecoder {
AudioDecoderG722();
~AudioDecoderG722() override;
bool HasDecodePlc() const override;
int Init() override;
void Reset() override;
int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
size_t Channels() const override;
@ -180,7 +180,7 @@ class AudioDecoderG722Stereo : public AudioDecoder {
public:
AudioDecoderG722Stereo();
~AudioDecoderG722Stereo() override;
int Init() override;
void Reset() override;
protected:
int DecodeInternal(const uint8_t* encoded,
@ -212,7 +212,7 @@ class AudioDecoderOpus : public AudioDecoder {
explicit AudioDecoderOpus(size_t num_channels);
~AudioDecoderOpus() override;
int Init() 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;
@ -248,7 +248,7 @@ class AudioDecoderCng : public AudioDecoder {
public:
explicit AudioDecoderCng();
~AudioDecoderCng() override;
int Init() override;
void Reset() override;
int IncomingPacket(const uint8_t* payload,
size_t payload_len,
uint16_t rtp_sequence_number,

9
webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
View File

@ -171,7 +171,6 @@ class AudioDecoderTest : public ::testing::Test {
size_t processed_samples = 0u;
encoded_bytes_ = 0u;
InitEncoder();
EXPECT_EQ(0, decoder_->Init());
std::vector<int16_t> input;
std::vector<int16_t> decoded;
while (processed_samples + frame_size_ <= data_length_) {
@ -220,7 +219,7 @@ class AudioDecoderTest : public ::testing::Test {
size_t enc_len = EncodeFrame(input.get(), frame_size_, encoded_);
size_t dec_len;
AudioDecoder::SpeechType speech_type1, speech_type2;
EXPECT_EQ(0, decoder_->Init());
decoder_->Reset();
rtc::scoped_ptr<int16_t[]> output1(new int16_t[frame_size_ * channels_]);
dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,
frame_size_ * channels_ * sizeof(int16_t),
@ -228,7 +227,7 @@ class AudioDecoderTest : public ::testing::Test {
ASSERT_LE(dec_len, frame_size_ * channels_);
EXPECT_EQ(frame_size_ * channels_, dec_len);
// Re-init decoder and decode again.
EXPECT_EQ(0, decoder_->Init());
decoder_->Reset();
rtc::scoped_ptr<int16_t[]> output2(new int16_t[frame_size_ * channels_]);
dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,
frame_size_ * channels_ * sizeof(int16_t),
@ -249,7 +248,7 @@ class AudioDecoderTest : public ::testing::Test {
input_audio_.Read(frame_size_, codec_input_rate_hz_, input.get()));
size_t enc_len = EncodeFrame(input.get(), frame_size_, encoded_);
AudioDecoder::SpeechType speech_type;
EXPECT_EQ(0, decoder_->Init());
decoder_->Reset();
rtc::scoped_ptr<int16_t[]> output(new int16_t[frame_size_ * channels_]);
size_t dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,
frame_size_ * channels_ * sizeof(int16_t),
@ -341,7 +340,7 @@ class AudioDecoderIlbcTest : public AudioDecoderTest {
input_audio_.Read(frame_size_, codec_input_rate_hz_, input.get()));
size_t enc_len = EncodeFrame(input.get(), frame_size_, encoded_);
AudioDecoder::SpeechType speech_type;
EXPECT_EQ(0, decoder_->Init());
decoder_->Reset();
rtc::scoped_ptr<int16_t[]> output(new int16_t[frame_size_ * channels_]);
size_t dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,
frame_size_ * channels_ * sizeof(int16_t),

2
webrtc/modules/audio_coding/neteq/decoder_database.cc
View File

@ -72,7 +72,6 @@ int DecoderDatabase::InsertExternal(uint8_t rtp_payload_type,
if (!decoder) {
return kInvalidPointer;
}
decoder->Init();
std::pair<DecoderMap::iterator, bool> ret;
DecoderInfo info(codec_type, fs_hz, decoder, true);
ret = decoders_.insert(std::make_pair(rtp_payload_type, info));
@ -136,7 +135,6 @@ AudioDecoder* DecoderDatabase::GetDecoder(uint8_t rtp_payload_type) {
AudioDecoder* decoder = CreateAudioDecoder(info->codec_type);
assert(decoder); // Should not be able to have an unsupported codec here.
info->decoder = decoder;
info->decoder->Init();
}
return info->decoder;
}

2
webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h
View File

@ -27,7 +27,7 @@ class MockAudioDecoder : public AudioDecoder {
int(const uint8_t*, size_t, int, size_t, int16_t*, SpeechType*));
MOCK_CONST_METHOD0(HasDecodePlc, bool());
MOCK_METHOD2(DecodePlc, size_t(size_t, int16_t*));
MOCK_METHOD0(Init, int());
MOCK_METHOD0(Reset, void());
MOCK_METHOD5(IncomingPacket, int(const uint8_t*, size_t, uint16_t, uint32_t,
uint32_t));
MOCK_METHOD0(ErrorCode, int());

9
webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
View File

@ -28,7 +28,7 @@ using ::testing::Invoke;
class ExternalPcm16B : public AudioDecoder {
public:
ExternalPcm16B() {}
virtual int Init() { return 0; }
void Reset() override {}
protected:
int DecodeInternal(const uint8_t* encoded,
@ -58,8 +58,8 @@ class MockExternalPcm16B : public ExternalPcm16B {
.WillByDefault(Invoke(&real_, &ExternalPcm16B::HasDecodePlc));
ON_CALL(*this, DecodePlc(_, _))
.WillByDefault(Invoke(&real_, &ExternalPcm16B::DecodePlc));
ON_CALL(*this, Init())
.WillByDefault(Invoke(&real_, &ExternalPcm16B::Init));
ON_CALL(*this, Reset())
.WillByDefault(Invoke(&real_, &ExternalPcm16B::Reset));
ON_CALL(*this, IncomingPacket(_, _, _, _, _))
.WillByDefault(Invoke(&real_, &ExternalPcm16B::IncomingPacket));
ON_CALL(*this, ErrorCode())
@ -79,8 +79,7 @@ class MockExternalPcm16B : public ExternalPcm16B {
bool());
MOCK_METHOD2(DecodePlc,
size_t(size_t num_frames, int16_t* decoded));
MOCK_METHOD0(Init,
int());
MOCK_METHOD0(Reset, void());
MOCK_METHOD5(IncomingPacket,
int(const uint8_t* payload, size_t payload_len,
uint16_t rtp_sequence_number, uint32_t rtp_timestamp,

2
webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
View File

@ -40,8 +40,6 @@ class NetEqExternalDecoderUnitTest : public test::NetEqExternalDecoderTest {
payload_size_bytes_(0),
last_send_time_(0),
last_arrival_time_(0) {
// Init() will trigger external_decoder_->Init().
EXPECT_CALL(*external_decoder_, Init());
// NetEq is not allowed to delete the external decoder (hence Times(0)).
EXPECT_CALL(*external_decoder_, Die()).Times(0);
Init();

19
webrtc/modules/audio_coding/neteq/neteq_impl.cc
View File

@ -1178,15 +1178,14 @@ int NetEqImpl::Decode(PacketList* packet_list, Operations* operation,
if (reset_decoder_) {
// TODO(hlundin): Write test for this.
// Reset decoder.
if (decoder) {
decoder->Init();
}
if (decoder)
decoder->Reset();
// Reset comfort noise decoder.
AudioDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
if (cng_decoder) {
cng_decoder->Init();
}
if (cng_decoder)
cng_decoder->Reset();
reset_decoder_ = false;
}
@ -1896,11 +1895,9 @@ void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) {
mute_factor_array_[i] = 16384; // 1.0 in Q14.
}
// Reset comfort noise decoder, if there is one active.
AudioDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
if (cng_decoder) {
cng_decoder->Init();
}
if (cng_decoder)
cng_decoder->Reset();
// Reinit post-decode VAD with new sample rate.
assert(vad_.get()); // Cannot be NULL here.

13
webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
View File

@ -444,10 +444,7 @@ TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
return encoded_len;
}
virtual int Init() {
next_value_ = 1;
return 0;
}
void Reset() override { next_value_ = 1; }
size_t Channels() const override { return 1; }
@ -524,7 +521,7 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
// Create a mock decoder object.
MockAudioDecoder mock_decoder;
EXPECT_CALL(mock_decoder, Init()).WillRepeatedly(Return(0));
EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
EXPECT_CALL(mock_decoder, IncomingPacket(_, kPayloadLengthBytes, _, _, _))
.WillRepeatedly(Return(0));
@ -690,7 +687,7 @@ TEST_F(NetEqImplTest, CodecInternalCng) {
// Create a mock decoder object.
MockAudioDecoder mock_decoder;
EXPECT_CALL(mock_decoder, Init()).WillRepeatedly(Return(0));
EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
EXPECT_CALL(mock_decoder, IncomingPacket(_, kPayloadLengthBytes, _, _, _))
.WillRepeatedly(Return(0));
@ -829,9 +826,7 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) {
class MockAudioDecoder : public AudioDecoder {
public:
int Init() override {
return 0;
}
void Reset() override {}
MOCK_CONST_METHOD2(PacketDuration, int(const uint8_t*, size_t));
MOCK_METHOD5(DecodeInternal, int(const uint8_t*, size_t, int, int16_t*,
SpeechType*));

5
webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
View File

@ -33,7 +33,7 @@ class MockAudioDecoderOpus : public AudioDecoderOpus {
virtual ~MockAudioDecoderOpus() { Die(); }
MOCK_METHOD0(Die, void());
MOCK_METHOD0(Init, int());
MOCK_METHOD0(Reset, void());
int PacketDuration(const uint8_t* encoded,
size_t encoded_len) const override {
@ -271,7 +271,6 @@ struct NetEqNetworkStatsCheck {
TEST(NetEqNetworkStatsTest, OpusDecodeFec) {
MockAudioDecoderOpus decoder(1);
EXPECT_CALL(decoder, Init());
NetEqNetworkStatsTest test(kDecoderOpus, &decoder);
test.DecodeFecTest();
EXPECT_CALL(decoder, Die()).Times(1);
@ -279,7 +278,6 @@ TEST(NetEqNetworkStatsTest, OpusDecodeFec) {
TEST(NetEqNetworkStatsTest, StereoOpusDecodeFec) {
MockAudioDecoderOpus decoder(2);
EXPECT_CALL(decoder, Init());
NetEqNetworkStatsTest test(kDecoderOpus, &decoder);
test.DecodeFecTest();
EXPECT_CALL(decoder, Die()).Times(1);
@ -287,7 +285,6 @@ TEST(NetEqNetworkStatsTest, StereoOpusDecodeFec) {
TEST(NetEqNetworkStatsTest, NoiseExpansionTest) {
MockAudioDecoderOpus decoder(1);
EXPECT_CALL(decoder, Init());
NetEqNetworkStatsTest test(kDecoderOpus, &decoder);
test.NoiseExpansionTest();
EXPECT_CALL(decoder, Die()).Times(1);