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Changing non-const reference arguments to pointers, ACM

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Part of refactoring of ACM, and recent lint-warnings.
This CL changes non-const references in the ACM API to pointers.

BUG=issue1372

Review URL: https://webrtc-codereview.appspot.com/1103012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3543 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
tina.legrand@webrtc.org 2013-02-20 15:22:23 +00:00
parent 0b6293aaaa
commit 374aa49e1a
20 changed files with 214 additions and 207 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

28
webrtc/modules/audio_coding/main/interface/audio_coding_module.h
View File

@ -111,7 +111,7 @@ class AudioCodingModule: public Module {
// -1 if the list number (list_id) is invalid.
// 0 if succeeded.
//
static WebRtc_Word32 Codec(const WebRtc_UWord8 list_id, CodecInst& codec);
static WebRtc_Word32 Codec(WebRtc_UWord8 list_id, CodecInst* codec);
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 Codec()
@ -132,7 +132,7 @@ class AudioCodingModule: public Module {
// -1 if no codec matches the given parameters.
// 0 if succeeded.
//
static WebRtc_Word32 Codec(const char* payload_name, CodecInst& codec,
static WebRtc_Word32 Codec(const char* payload_name, CodecInst* codec,
int sampling_freq_hz, int channels);
///////////////////////////////////////////////////////////////////////////
@ -264,7 +264,7 @@ class AudioCodingModule: public Module {
// -1 if failed to get send codec,
// 0 if succeeded.
//
virtual WebRtc_Word32 SendCodec(CodecInst& current_send_codec) const = 0;
virtual WebRtc_Word32 SendCodec(CodecInst* current_send_codec) const = 0;
///////////////////////////////////////////////////////////////////////////
// int SecondarySendCodec()
@ -441,8 +441,8 @@ class AudioCodingModule: public Module {
// -1 if fails to retrieve the setting of DTX/VAD,
// 0 if succeeded.
//
virtual WebRtc_Word32 VAD(bool& dtx_enabled, bool& vad_enabled,
ACMVADMode& vad_mode) const = 0;
virtual WebRtc_Word32 VAD(bool* dtx_enabled, bool* vad_enabled,
ACMVADMode* vad_mode) const = 0;
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 ReplaceInternalDTXWithWebRtc()
@ -476,7 +476,7 @@ class AudioCodingModule: public Module {
// 0 if succeeded.
//
virtual WebRtc_Word32 IsInternalDTXReplacedWithWebRtc(
bool& uses_webrtc_dtx) = 0;
bool* uses_webrtc_dtx) = 0;
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 RegisterVADCallback()
@ -589,7 +589,7 @@ class AudioCodingModule: public Module {
// -1 if failed to retrieve the codec,
// 0 if the codec is successfully retrieved.
//
virtual WebRtc_Word32 ReceiveCodec(CodecInst& curr_receive_codec) const = 0;
virtual WebRtc_Word32 ReceiveCodec(CodecInst* curr_receive_codec) const = 0;
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 IncomingPacket()
@ -729,7 +729,8 @@ class AudioCodingModule: public Module {
// 0 if the output is a valid mode.
// -1 if ACM failed to output a valid mode.
//
virtual WebRtc_Word32 BackgroundNoiseMode(ACMBackgroundNoiseMode& mode) = 0;
// TODO(tlegrand): Change function to return the mode.
virtual WebRtc_Word32 BackgroundNoiseMode(ACMBackgroundNoiseMode* mode) = 0;
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 PlayoutTimestamp()
@ -744,8 +745,8 @@ class AudioCodingModule: public Module {
// 0 if the output is a correct timestamp.
// -1 if failed to output the correct timestamp.
//
//
virtual WebRtc_Word32 PlayoutTimestamp(WebRtc_UWord32& timestamp) = 0;
// TODO(tlegrand): Change function to return the timestamp.
virtual WebRtc_Word32 PlayoutTimestamp(WebRtc_UWord32* timestamp) = 0;
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 DecoderEstimatedBandwidth()
@ -817,9 +818,8 @@ class AudioCodingModule: public Module {
// -1 if the function fails,
// 0 if the function succeeds.
//
virtual WebRtc_Word32
PlayoutData10Ms(const WebRtc_Word32 desired_freq_hz,
AudioFrame &audio_frame) = 0;
virtual WebRtc_Word32 PlayoutData10Ms(WebRtc_Word32 desired_freq_hz,
AudioFrame* audio_frame) = 0;
///////////////////////////////////////////////////////////////////////////
// (CNG) Comfort Noise Generation
@ -939,7 +939,7 @@ class AudioCodingModule: public Module {
// 0 if statistics are set successfully.
//
virtual WebRtc_Word32 NetworkStatistics(
ACMNetworkStatistics& network_statistics) const = 0;
ACMNetworkStatistics* network_statistics) const = 0;
//
// Set an initial delay for playout.

22
webrtc/modules/audio_coding/main/source/audio_coding_module.cc
View File

@ -34,15 +34,15 @@ WebRtc_UWord8 AudioCodingModule::NumberOfCodecs() {
}
// Get supported codec param with id
WebRtc_Word32 AudioCodingModule::Codec(const WebRtc_UWord8 list_id,
CodecInst& codec) {
WebRtc_Word32 AudioCodingModule::Codec(WebRtc_UWord8 list_id,
CodecInst* codec) {
// Get the codec settings for the codec with the given list ID
return ACMCodecDB::Codec(list_id, &codec);
return ACMCodecDB::Codec(list_id, codec);
}
// Get supported codec Param with name, frequency and number of channels.
WebRtc_Word32 AudioCodingModule::Codec(const char* payload_name,
CodecInst& codec, int sampling_freq_hz,
CodecInst* codec, int sampling_freq_hz,
int channels) {
int codec_id;
@ -51,20 +51,20 @@ WebRtc_Word32 AudioCodingModule::Codec(const char* payload_name,
if (codec_id < 0) {
// We couldn't find a matching codec, set the parameterss to unacceptable
// values and return.
codec.plname[0] = '\0';
codec.pltype = -1;
codec.pacsize = 0;
codec.rate = 0;
codec.plfreq = 0;
codec->plname[0] = '\0';
codec->pltype = -1;
codec->pacsize = 0;
codec->rate = 0;
codec->plfreq = 0;
return -1;
}
// Get default codec settings.
ACMCodecDB::Codec(codec_id, &codec);
ACMCodecDB::Codec(codec_id, codec);
// Keep the number of channels from the function call. For most codecs it
// will be the same value as in defaul codec settings, but not for all.
codec.channels = channels;
codec->channels = channels;
return 0;
}

86
webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
View File

@ -1171,11 +1171,12 @@ WebRtc_Word32 AudioCodingModuleImpl::RegisterSendCodec(
// Get current send codec.
WebRtc_Word32 AudioCodingModuleImpl::SendCodec(
CodecInst& current_codec) const {
CodecInst* current_codec) const {
WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
"SendCodec()");
CriticalSectionScoped lock(acm_crit_sect_);
assert(current_codec);
if (!send_codec_registered_) {
WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
"SendCodec Failed, no codec is registered");
@ -1185,7 +1186,7 @@ WebRtc_Word32 AudioCodingModuleImpl::SendCodec(
WebRtcACMCodecParams encoder_param;
codecs_[current_send_codec_idx_]->EncoderParams(&encoder_param);
encoder_param.codec_inst.pltype = send_codec_inst_.pltype;
memcpy(&current_codec, &(encoder_param.codec_inst), sizeof(CodecInst));
memcpy(current_codec, &(encoder_param.codec_inst), sizeof(CodecInst));
return 0;
}
@ -1597,13 +1598,14 @@ int AudioCodingModuleImpl::SetVADSafe(bool enable_dtx,
}
// Get VAD/DTX settings.
WebRtc_Word32 AudioCodingModuleImpl::VAD(bool& dtx_enabled, bool& vad_enabled,
ACMVADMode& mode) const {
// TODO(tlegrand): Change this method to void.
WebRtc_Word32 AudioCodingModuleImpl::VAD(bool* dtx_enabled, bool* vad_enabled,
ACMVADMode* mode) const {
CriticalSectionScoped lock(acm_crit_sect_);
dtx_enabled = dtx_enabled_;
vad_enabled = vad_enabled_;
mode = vad_mode_;
*dtx_enabled = dtx_enabled_;
*vad_enabled = vad_enabled_;
*mode = vad_mode_;
return 0;
}
@ -1931,7 +1933,7 @@ WebRtc_Word32 AudioCodingModuleImpl::RegisterRecCodecMSSafe(
// Get current received codec.
WebRtc_Word32 AudioCodingModuleImpl::ReceiveCodec(
CodecInst& current_codec) const {
CodecInst* current_codec) const {
WebRtcACMCodecParams decoder_param;
CriticalSectionScoped lock(acm_crit_sect_);
@ -1940,7 +1942,7 @@ WebRtc_Word32 AudioCodingModuleImpl::ReceiveCodec(
if (codecs_[id]->DecoderInitialized()) {
if (codecs_[id]->DecoderParams(&decoder_param,
last_recv_audio_codec_pltype_)) {
memcpy(&current_codec, &decoder_param.codec_inst,
memcpy(current_codec, &decoder_param.codec_inst,
sizeof(CodecInst));
return 0;
}
@ -1950,7 +1952,7 @@ WebRtc_Word32 AudioCodingModuleImpl::ReceiveCodec(
// If we are here then we haven't found any codec. Set codec pltype to -1 to
// indicate that the structure is invalid and return -1.
current_codec.pltype = -1;
current_codec->pltype = -1;
return -1;
}
@ -2222,10 +2224,10 @@ AudioPlayoutMode AudioCodingModuleImpl::PlayoutMode() const {
// Get 10 milliseconds of raw audio data to play out.
// Automatic resample to the requested frequency.
WebRtc_Word32 AudioCodingModuleImpl::PlayoutData10Ms(
const WebRtc_Word32 desired_freq_hz, AudioFrame& audio_frame) {
WebRtc_Word32 desired_freq_hz, AudioFrame* audio_frame) {
bool stereo_mode;
if (GetSilence(desired_freq_hz, &audio_frame))
if (GetSilence(desired_freq_hz, audio_frame))
return 0; // Silence is generated, return.
// RecOut always returns 10 ms.
@ -2235,9 +2237,9 @@ WebRtc_Word32 AudioCodingModuleImpl::PlayoutData10Ms(
return -1;
}
audio_frame.num_channels_ = audio_frame_.num_channels_;
audio_frame.vad_activity_ = audio_frame_.vad_activity_;
audio_frame.speech_type_ = audio_frame_.speech_type_;
audio_frame->num_channels_ = audio_frame_.num_channels_;
audio_frame->vad_activity_ = audio_frame_.vad_activity_;
audio_frame->speech_type_ = audio_frame_.speech_type_;
stereo_mode = (audio_frame_.num_channels_ > 1);
// For stereo playout:
@ -2256,7 +2258,7 @@ WebRtc_Word32 AudioCodingModuleImpl::PlayoutData10Ms(
if ((receive_freq != desired_freq_hz) && (desired_freq_hz != -1)) {
// Resample payload_data.
WebRtc_Word16 temp_len = output_resampler_.Resample10Msec(
audio_frame_.data_, receive_freq, audio_frame.data_,
audio_frame_.data_, receive_freq, audio_frame->data_,
desired_freq_hz, audio_frame_.num_channels_);
if (temp_len < 0) {
@ -2266,40 +2268,40 @@ WebRtc_Word32 AudioCodingModuleImpl::PlayoutData10Ms(
}
// Set the payload data length from the resampler.
audio_frame.samples_per_channel_ = (WebRtc_UWord16) temp_len;
audio_frame->samples_per_channel_ = (WebRtc_UWord16) temp_len;
// Set the sampling frequency.
audio_frame.sample_rate_hz_ = desired_freq_hz;
audio_frame->sample_rate_hz_ = desired_freq_hz;
} else {
memcpy(audio_frame.data_, audio_frame_.data_,
audio_frame_.samples_per_channel_ * audio_frame.num_channels_
memcpy(audio_frame->data_, audio_frame_.data_,
audio_frame_.samples_per_channel_ * audio_frame->num_channels_
* sizeof(WebRtc_Word16));
// Set the payload length.
audio_frame.samples_per_channel_ =
audio_frame->samples_per_channel_ =
audio_frame_.samples_per_channel_;
// Set the sampling frequency.
audio_frame.sample_rate_hz_ = receive_freq;
audio_frame->sample_rate_hz_ = receive_freq;
}
// Tone detection done for master channel.
if (dtmf_detector_ != NULL) {
// Dtmf Detection.
if (audio_frame.sample_rate_hz_ == 8000) {
// Use audio_frame.data_ then Dtmf detector doesn't
if (audio_frame->sample_rate_hz_ == 8000) {
// Use audio_frame->data_ then Dtmf detector doesn't
// need resampling.
if (!stereo_mode) {
dtmf_detector_->Detect(audio_frame.data_,
audio_frame.samples_per_channel_,
audio_frame.sample_rate_hz_, tone_detected,
dtmf_detector_->Detect(audio_frame->data_,
audio_frame->samples_per_channel_,
audio_frame->sample_rate_hz_, tone_detected,
tone);
} else {
// We are in 8 kHz so the master channel needs only 80 samples.
WebRtc_Word16 master_channel[80];
for (int n = 0; n < 80; n++) {
master_channel[n] = audio_frame.data_[n << 1];
master_channel[n] = audio_frame->data_[n << 1];
}
dtmf_detector_->Detect(master_channel,
audio_frame.samples_per_channel_,
audio_frame.sample_rate_hz_, tone_detected,
audio_frame->samples_per_channel_,
audio_frame->sample_rate_hz_, tone_detected,
tone);
}
} else {
@ -2346,9 +2348,9 @@ WebRtc_Word32 AudioCodingModuleImpl::PlayoutData10Ms(
}
}
audio_frame.id_ = id_;
audio_frame.energy_ = -1;
audio_frame.timestamp_ = 0;
audio_frame->id_ = id_;
audio_frame->energy_ = -1;
audio_frame->timestamp_ = 0;
return 0;
}
@ -2373,9 +2375,9 @@ WebRtc_Word16 AudioCodingModuleImpl::SetReceiveVADMode(const ACMVADMode mode) {
//
WebRtc_Word32 AudioCodingModuleImpl::NetworkStatistics(
ACMNetworkStatistics& statistics) const {
ACMNetworkStatistics* statistics) const {
WebRtc_Word32 status;
status = neteq_.NetworkStatistics(&statistics);
status = neteq_.NetworkStatistics(statistics);
return status;
}
@ -2594,13 +2596,13 @@ WebRtc_Word32 AudioCodingModuleImpl::ReplaceInternalDTXWithWebRtc(
}
WebRtc_Word32 AudioCodingModuleImpl::IsInternalDTXReplacedWithWebRtc(
bool& uses_webrtc_dtx) {
bool* uses_webrtc_dtx) {
CriticalSectionScoped lock(acm_crit_sect_);
if (!HaveValidEncoder("IsInternalDTXReplacedWithWebRtc")) {
return -1;
}
if (codecs_[current_send_codec_idx_]->IsInternalDTXReplaced(&uses_webrtc_dtx)
if (codecs_[current_send_codec_idx_]->IsInternalDTXReplaced(uses_webrtc_dtx)
< 0) {
return -1;
}
@ -2655,19 +2657,19 @@ WebRtc_Word32 AudioCodingModuleImpl::SetBackgroundNoiseMode(
}
WebRtc_Word32 AudioCodingModuleImpl::BackgroundNoiseMode(
ACMBackgroundNoiseMode& mode) {
return neteq_.BackgroundNoiseMode(mode);
ACMBackgroundNoiseMode* mode) {
return neteq_.BackgroundNoiseMode(*mode);
}
WebRtc_Word32 AudioCodingModuleImpl::PlayoutTimestamp(
WebRtc_UWord32& timestamp) {
WebRtc_UWord32* timestamp) {
WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
"PlayoutTimestamp()");
if (track_neteq_buffer_) {
timestamp = playout_ts_;
*timestamp = playout_ts_;
return 0;
} else {
return neteq_.PlayoutTimestamp(timestamp);
return neteq_.PlayoutTimestamp(*timestamp);
}
}

20
webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h
View File

@ -68,7 +68,7 @@ class AudioCodingModuleImpl : public AudioCodingModule {
int SecondarySendCodec(CodecInst* secondary_codec) const;
// Get current send codec.
WebRtc_Word32 SendCodec(CodecInst& current_codec) const;
WebRtc_Word32 SendCodec(CodecInst* current_codec) const;
// Get current send frequency.
WebRtc_Word32 SendFrequency() const;
@ -99,7 +99,7 @@ class AudioCodingModuleImpl : public AudioCodingModule {
WebRtc_Word32 SetBackgroundNoiseMode(const ACMBackgroundNoiseMode mode);
// Get current background noise mode.
WebRtc_Word32 BackgroundNoiseMode(ACMBackgroundNoiseMode& mode);
WebRtc_Word32 BackgroundNoiseMode(ACMBackgroundNoiseMode* mode);
/////////////////////////////////////////
// (FEC) Forward Error Correction
@ -121,8 +121,8 @@ class AudioCodingModuleImpl : public AudioCodingModule {
const bool enable_vad = false,
const ACMVADMode mode = VADNormal);
WebRtc_Word32 VAD(bool& dtx_enabled, bool& vad_enabled,
ACMVADMode& mode) const;
WebRtc_Word32 VAD(bool* dtx_enabled, bool* vad_enabled,
ACMVADMode* mode) const;
WebRtc_Word32 RegisterVADCallback(ACMVADCallback* vad_callback);
@ -153,7 +153,7 @@ class AudioCodingModuleImpl : public AudioCodingModule {
WebRtc_Word32 RegisterReceiveCodec(const CodecInst& receive_codec);
// Get current received codec.
WebRtc_Word32 ReceiveCodec(CodecInst& current_codec) const;
WebRtc_Word32 ReceiveCodec(CodecInst* current_codec) const;
// Incoming packet from network parsed and ready for decode.
WebRtc_Word32 IncomingPacket(const WebRtc_UWord8* incoming_payload,
@ -189,18 +189,18 @@ class AudioCodingModuleImpl : public AudioCodingModule {
AudioPlayoutMode PlayoutMode() const;
// Get playout timestamp.
WebRtc_Word32 PlayoutTimestamp(WebRtc_UWord32& timestamp);
WebRtc_Word32 PlayoutTimestamp(WebRtc_UWord32* timestamp);
// Get 10 milliseconds of raw audio data to play out, and
// automatic resample to the requested frequency if > 0.
WebRtc_Word32 PlayoutData10Ms(const WebRtc_Word32 desired_freq_hz,
AudioFrame &audio_frame);
WebRtc_Word32 PlayoutData10Ms(WebRtc_Word32 desired_freq_hz,
AudioFrame* audio_frame);
/////////////////////////////////////////
// Statistics
//
WebRtc_Word32 NetworkStatistics(ACMNetworkStatistics& statistics) const;
WebRtc_Word32 NetworkStatistics(ACMNetworkStatistics* statistics) const;
void DestructEncoderInst(void* inst);
@ -221,7 +221,7 @@ class AudioCodingModuleImpl : public AudioCodingModule {
WebRtc_Word32 ReplaceInternalDTXWithWebRtc(const bool use_webrtc_dtx);
WebRtc_Word32 IsInternalDTXReplacedWithWebRtc(bool& uses_webrtc_dtx);
WebRtc_Word32 IsInternalDTXReplacedWithWebRtc(bool* uses_webrtc_dtx);
WebRtc_Word32 SetISACMaxRate(const WebRtc_UWord32 max_bit_per_sec);

42
webrtc/modules/audio_coding/main/test/APITest.cc
View File

@ -182,7 +182,7 @@ APITest::SetUp()
WebRtc_Word16 numCodecs = _acmA->NumberOfCodecs();
for(WebRtc_UWord8 n = 0; n < numCodecs; n++)
{
AudioCodingModule::Codec(n, dummyCodec);
AudioCodingModule::Codec(n, &dummyCodec);
if((STR_CASE_CMP(dummyCodec.plname, "CN") == 0) &&
(dummyCodec.plfreq == 32000))
{
@ -205,7 +205,7 @@ APITest::SetUp()
// test if re-registration works;
CodecInst nextCodec;
int currentPayloadType = dummyCodec.pltype;
AudioCodingModule::Codec(n + 1, nextCodec);
AudioCodingModule::Codec(n + 1, &nextCodec);
dummyCodec.pltype = nextCodec.pltype;
if(!FixedPayloadTypeCodec(nextCodec.plname))
{
@ -218,7 +218,7 @@ APITest::SetUp()
{
// test if un-registration works;
CodecInst nextCodec;
AudioCodingModule::Codec(n + 1, nextCodec);
AudioCodingModule::Codec(n + 1, &nextCodec);
nextCodec.pltype = dummyCodec.pltype;
if(!FixedPayloadTypeCodec(nextCodec.plname))
{
@ -248,11 +248,11 @@ APITest::SetUp()
_thereIsDecoderB = true;
// Register Send Codec
AudioCodingModule::Codec((WebRtc_UWord8)_codecCntrA, dummyCodec);
AudioCodingModule::Codec((WebRtc_UWord8)_codecCntrA, &dummyCodec);
CHECK_ERROR_MT(_acmA->RegisterSendCodec(dummyCodec));
_thereIsEncoderA = true;
//
AudioCodingModule::Codec((WebRtc_UWord8)_codecCntrB, dummyCodec);
AudioCodingModule::Codec((WebRtc_UWord8)_codecCntrB, &dummyCodec);
CHECK_ERROR_MT(_acmB->RegisterSendCodec(dummyCodec));
_thereIsEncoderB = true;
@ -410,7 +410,7 @@ APITest::PullAudioRunA()
{
_pullEventA->Wait(100);
AudioFrame audioFrame;
if(_acmA->PlayoutData10Ms(_outFreqHzA, audioFrame) < 0)
if(_acmA->PlayoutData10Ms(_outFreqHzA, &audioFrame) < 0)
{
bool thereIsDecoder;
{
@ -438,7 +438,7 @@ APITest::PullAudioRunB()
{
_pullEventB->Wait(100);
AudioFrame audioFrame;
if(_acmB->PlayoutData10Ms(_outFreqHzB, audioFrame) < 0)
if(_acmB->PlayoutData10Ms(_outFreqHzB, &audioFrame) < 0)
{
bool thereIsDecoder;
{
@ -794,7 +794,7 @@ APITest::CheckVADStatus(char side)
if(side == 'A')
{
_acmA->VAD(dtxEnabled, vadEnabled, vadMode);
_acmA->VAD(&dtxEnabled, &vadEnabled, &vadMode);
_acmA->RegisterVADCallback(NULL);
_vadCallbackA->Reset();
_acmA->RegisterVADCallback(_vadCallbackA);
@ -838,7 +838,7 @@ APITest::CheckVADStatus(char side)
}
else
{
_acmB->VAD(dtxEnabled, vadEnabled, vadMode);
_acmB->VAD(&dtxEnabled, &vadEnabled, &vadMode);
_acmB->RegisterVADCallback(NULL);
_vadCallbackB->Reset();
@ -920,7 +920,7 @@ APITest::TestDelay(char side)
inTimestamp = myChannel->LastInTimestamp();
CHECK_ERROR_MT(myACM->PlayoutTimestamp(outTimestamp));
CHECK_ERROR_MT(myACM->PlayoutTimestamp(&outTimestamp));
if(!_randomTest)
{
@ -932,7 +932,7 @@ APITest::TestDelay(char side)
myEvent->Wait(1000);
inTimestamp = myChannel->LastInTimestamp();
CHECK_ERROR_MT(myACM->PlayoutTimestamp(outTimestamp));
CHECK_ERROR_MT(myACM->PlayoutTimestamp(&outTimestamp));
//std::cout << outTimestamp << std::endl << std::flush;
estimDelay = (double)((WebRtc_UWord32)(inTimestamp - outTimestamp)) /
@ -968,7 +968,7 @@ APITest::TestDelay(char side)
*myMinDelay = (rand() % 1000) + 1;
ACMNetworkStatistics networkStat;
CHECK_ERROR_MT(myACM->NetworkStatistics(networkStat));
CHECK_ERROR_MT(myACM->NetworkStatistics(&networkStat));
if(!_randomTest)
{
@ -1039,9 +1039,9 @@ APITest::TestRegisteration(char sendSide)
}
CodecInst myCodec;
if(sendACM->SendCodec(myCodec) < 0)
if(sendACM->SendCodec(&myCodec) < 0)
{
AudioCodingModule::Codec(_codecCntrA, myCodec);
AudioCodingModule::Codec(_codecCntrA, &myCodec);
}
if(!_randomTest)
@ -1332,7 +1332,7 @@ APITest::TestSendVAD(char side)
if(side == 'A')
{
AudioCodingModule::Codec(_codecCntrA, myCodec);
AudioCodingModule::Codec(_codecCntrA, &myCodec);
vad = &_sendVADA;
dtx = &_sendDTXA;
mode = &_sendVADModeA;
@ -1341,7 +1341,7 @@ APITest::TestSendVAD(char side)
}
else
{
AudioCodingModule::Codec(_codecCntrB, myCodec);
AudioCodingModule::Codec(_codecCntrB, &myCodec);
vad = &_sendVADB;
dtx = &_sendDTXB;
mode = &_sendVADModeB;
@ -1408,11 +1408,11 @@ APITest::CurrentCodec(char side)
CodecInst myCodec;
if(side == 'A')
{
_acmA->SendCodec(myCodec);
_acmA->SendCodec(&myCodec);
}
else
{
_acmB->SendCodec(myCodec);
_acmB->SendCodec(&myCodec);
}
if(!_randomTest)
@ -1493,11 +1493,11 @@ APITest::ChangeCodec(char side)
Wait(1000);
// After Initialization CN is lost, re-register them
if(AudioCodingModule::Codec("CN", myCodec, 8000, 1) >= 0)
if(AudioCodingModule::Codec("CN", &myCodec, 8000, 1) >= 0)
{
CHECK_ERROR_MT(myACM->RegisterSendCodec(myCodec));
}
if(AudioCodingModule::Codec("CN", myCodec, 16000, 1) >= 0)
if(AudioCodingModule::Codec("CN", &myCodec, 16000, 1) >= 0)
{
CHECK_ERROR_MT(myACM->RegisterSendCodec(myCodec));
}
@ -1507,7 +1507,7 @@ APITest::ChangeCodec(char side)
_writeToFile = false;
}
AudioCodingModule::Codec(*codecCntr, myCodec);
AudioCodingModule::Codec(*codecCntr, &myCodec);
} while(!STR_CASE_CMP(myCodec.plname, "CN") ||
!STR_CASE_CMP(myCodec.plname, "telephone-event") ||
!STR_CASE_CMP(myCodec.plname, "RED"));

12
webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
View File

@ -73,14 +73,14 @@ void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream) {
// Choose codec on command line.
printf("List of supported codec.\n");
for (int n = 0; n < noOfCodecs; n++) {
acm->Codec(n, sendCodec);
acm->Codec(n, &sendCodec);
printf("%d %s\n", n, sendCodec.plname);
}
printf("Choose your codec:");
ASSERT_GT(scanf("%d", &codecNo), 0);
}
acm->Codec(codecNo, sendCodec);
acm->Codec(codecNo, &sendCodec);
if (!strcmp(sendCodec.plname, "CELT")) {
sendCodec.channels = 1;
}
@ -144,7 +144,7 @@ void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream) {
noOfCodecs = acm->NumberOfCodecs();
for (int i = 0; i < noOfCodecs; i++) {
acm->Codec((WebRtc_UWord8) i, recvCodec);
acm->Codec((WebRtc_UWord8) i, &recvCodec);
if (acm->RegisterReceiveCodec(recvCodec) != 0) {
printf("Unable to register codec: for run: codecId: %d\n", codeId);
exit(1);
@ -224,7 +224,7 @@ bool Receiver::IncomingPacket() {
bool Receiver::PlayoutData() {
AudioFrame audioFrame;
if (_acm->PlayoutData10Ms(_frequency, audioFrame) != 0) {
if (_acm->PlayoutData10Ms(_frequency, &audioFrame) != 0) {
printf("Error when calling PlayoutData10Ms, for run: codecId: %d\n",
codeId);
exit(1);
@ -305,7 +305,7 @@ void EncodeDecodeTest::Perform() {
}
if (_testMode != 2) {
for (int n = 0; n < numCodecs; n++) {
acm->Codec(n, sendCodecTmp);
acm->Codec(n, &sendCodecTmp);
if (STR_CASE_CMP(sendCodecTmp.plname, "telephone-event") == 0) {
numPars[n] = 0;
} else if (STR_CASE_CMP(sendCodecTmp.plname, "cn") == 0) {
@ -381,7 +381,7 @@ void EncodeDecodeTest::EncodeToFile(int fileType, int codeId, int* codePars,
_sender.Setup(acm, &rtpFile);
struct CodecInst sendCodecInst;
if (acm->SendCodec(sendCodecInst) >= 0) {
if (acm->SendCodec(&sendCodecInst) >= 0) {
_sender.Run();
}
_sender.Teardown();

18
webrtc/modules/audio_coding/main/test/SpatialAudio.cc
View File

@ -82,7 +82,7 @@ SpatialAudio::Setup()
WebRtc_UWord8 num_encoders = _acmReceiver->NumberOfCodecs();
// Register all available codes as receiving codecs once more.
for (WebRtc_UWord8 n = 0; n < num_encoders; n++) {
status = _acmReceiver->Codec(n, codecInst);
status = _acmReceiver->Codec(n, &codecInst);
if (status < 0) {
printf("Error in Codec(), no matching codec found");
}
@ -109,7 +109,7 @@ SpatialAudio::Perform()
Setup();
CodecInst codecInst;
_acmLeft->Codec((WebRtc_UWord8)1, codecInst);
_acmLeft->Codec((WebRtc_UWord8)1, &codecInst);
CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
EncodeDecode();
@ -122,7 +122,7 @@ SpatialAudio::Perform()
while((pannCntr + 1) < NUM_PANN_COEFFS)
{
_acmLeft->Codec((WebRtc_UWord8)0, codecInst);
_acmLeft->Codec((WebRtc_UWord8)0, &codecInst);
codecInst.pacsize = 480;
CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
CHECK_ERROR(_acmRight->RegisterSendCodec(codecInst));
@ -131,7 +131,7 @@ SpatialAudio::Perform()
pannCntr++;
// Change codec
_acmLeft->Codec((WebRtc_UWord8)3, codecInst);
_acmLeft->Codec((WebRtc_UWord8)3, &codecInst);
codecInst.pacsize = 320;
CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
CHECK_ERROR(_acmRight->RegisterSendCodec(codecInst));
@ -144,11 +144,11 @@ SpatialAudio::Perform()
}
}
_acmLeft->Codec((WebRtc_UWord8)4, codecInst);
_acmLeft->Codec((WebRtc_UWord8)4, &codecInst);
CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
EncodeDecode();
_acmLeft->Codec((WebRtc_UWord8)0, codecInst);
_acmLeft->Codec((WebRtc_UWord8)0, &codecInst);
codecInst.pacsize = 480;
CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
CHECK_ERROR(_acmRight->RegisterSendCodec(codecInst));
@ -200,7 +200,8 @@ SpatialAudio::EncodeDecode(
CHECK_ERROR(_acmLeft->Process());
CHECK_ERROR(_acmRight->Process());
CHECK_ERROR(_acmReceiver->PlayoutData10Ms(outFileSampFreq, audioFrame));
CHECK_ERROR(_acmReceiver->PlayoutData10Ms(outFileSampFreq,
&audioFrame));
_outFile.Write10MsData(audioFrame);
}
_inFile.Rewind();
@ -221,7 +222,8 @@ SpatialAudio::EncodeDecode()
CHECK_ERROR(_acmLeft->Process());
CHECK_ERROR(_acmReceiver->PlayoutData10Ms(outFileSampFreq, audioFrame));
CHECK_ERROR(_acmReceiver->PlayoutData10Ms(outFileSampFreq,
&audioFrame));
_outFile.Write10MsData(audioFrame);
}
_inFile.Rewind();

10
webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
View File

@ -145,7 +145,7 @@ void TestAllCodecs::Perform() {
uint8_t num_encoders = acm_a_->NumberOfCodecs();
CodecInst my_codec_param;
for (uint8_t n = 0; n < num_encoders; n++) {
acm_b_->Codec(n, my_codec_param);
acm_b_->Codec(n, &my_codec_param);
if (!strcmp(my_codec_param.plname, "opus")) {
my_codec_param.channels = 1;
}
@ -752,7 +752,7 @@ void TestAllCodecs::RegisterSendCodec(char side, char* codec_name,
// Get all codec parameters before registering
CodecInst my_codec_param;
CHECK_ERROR(AudioCodingModule::Codec(codec_name, my_codec_param,
CHECK_ERROR(AudioCodingModule::Codec(codec_name, &my_codec_param,
sampling_freq_hz, 1));
my_codec_param.rate = rate;
my_codec_param.pacsize = packet_size;
@ -795,7 +795,7 @@ void TestAllCodecs::Run(TestPack* channel) {
}
// Run received side of ACM.
CHECK_ERROR(acm_b_->PlayoutData10Ms(out_freq_hz, audio_frame));
CHECK_ERROR(acm_b_->PlayoutData10Ms(out_freq_hz, &audio_frame));
// Write output speech to file.
outfile_b_.Write10MsData(audio_frame.data_,
@ -824,9 +824,9 @@ void TestAllCodecs::OpenOutFile(int test_number) {
void TestAllCodecs::DisplaySendReceiveCodec() {
CodecInst my_codec_param;
acm_a_->SendCodec(my_codec_param);
acm_a_->SendCodec(&my_codec_param);
printf("%s -> ", my_codec_param.plname);
acm_b_->ReceiveCodec(my_codec_param);
acm_b_->ReceiveCodec(&my_codec_param);
printf("%s\n", my_codec_param.plname);
}

10
webrtc/modules/audio_coding/main/test/TestFEC.cc
View File

@ -79,7 +79,7 @@ void TestFEC::Perform()
}
for(WebRtc_UWord8 n = 0; n < numEncoders; n++)
{
_acmB->Codec(n, myCodecParam);
_acmB->Codec(n, &myCodecParam);
if(_testMode != 0)
{
printf("%s\n", myCodecParam.plname);
@ -553,7 +553,7 @@ WebRtc_Word16 TestFEC::RegisterSendCodec(char side, char* codecName, WebRtc_Word
}
CodecInst myCodecParam;
CHECK_ERROR(AudioCodingModule::Codec(codecName, myCodecParam,
CHECK_ERROR(AudioCodingModule::Codec(codecName, &myCodecParam,
samplingFreqHz, 1));
CHECK_ERROR(myACM->RegisterSendCodec(myCodecParam));
@ -575,7 +575,7 @@ void TestFEC::Run()
_inFileA.Read10MsData(audioFrame);
CHECK_ERROR(_acmA->Add10MsData(audioFrame));
CHECK_ERROR(_acmA->Process());
CHECK_ERROR(_acmB->PlayoutData10Ms(outFreqHzB, audioFrame));
CHECK_ERROR(_acmB->PlayoutData10Ms(outFreqHzB, &audioFrame));
_outFileB.Write10MsData(audioFrame.data_, audioFrame.samples_per_channel_);
msecPassed += 10;
if(msecPassed >= 1000)
@ -616,9 +616,9 @@ void TestFEC::OpenOutFile(WebRtc_Word16 test_number) {
void TestFEC::DisplaySendReceiveCodec()
{
CodecInst myCodecParam;
_acmA->SendCodec(myCodecParam);
_acmA->SendCodec(&myCodecParam);
printf("%s -> ", myCodecParam.plname);
_acmB->ReceiveCodec(myCodecParam);
_acmB->ReceiveCodec(&myCodecParam);
printf("%s\n", myCodecParam.plname);
}

30
webrtc/modules/audio_coding/main/test/TestStereo.cc
View File

@ -182,19 +182,19 @@ void TestStereo::Perform() {
WebRtc_UWord8 num_encoders = acm_a_->NumberOfCodecs();
CodecInst my_codec_param;
for (WebRtc_UWord8 n = 0; n < num_encoders; n++) {
EXPECT_EQ(0, acm_b_->Codec(n, my_codec_param));
EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param));
EXPECT_EQ(0, acm_b_->RegisterReceiveCodec(my_codec_param));
}
// Test that unregister all receive codecs works.
for (WebRtc_UWord8 n = 0; n < num_encoders; n++) {
EXPECT_EQ(0, acm_b_->Codec(n, my_codec_param));
EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param));
EXPECT_EQ(0, acm_b_->UnregisterReceiveCodec(my_codec_param.pltype));
}
// Register all available codes as receiving codecs once more.
for (WebRtc_UWord8 n = 0; n < num_encoders; n++) {
EXPECT_EQ(0, acm_b_->Codec(n, my_codec_param));
EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param));
EXPECT_EQ(0, acm_b_->RegisterReceiveCodec(my_codec_param));
}
@ -222,12 +222,12 @@ void TestStereo::Perform() {
// Continue with setting a stereo codec as send codec and verify that
// VAD/DTX gets turned off.
EXPECT_EQ(0, acm_a_->SetVAD(true, true, VADNormal));
EXPECT_EQ(0, acm_a_->VAD(dtx, vad, vad_mode));
EXPECT_EQ(0, acm_a_->VAD(&dtx, &vad, &vad_mode));
EXPECT_TRUE(dtx);
EXPECT_TRUE(vad);
char codec_pcma_temp[] = "PCMA";
RegisterSendCodec('A', codec_pcma_temp, 8000, 64000, 80, 2, pcma_pltype_);
EXPECT_EQ(0, acm_a_->VAD(dtx, vad, vad_mode));
EXPECT_EQ(0, acm_a_->VAD(&dtx, &vad, &vad_mode));
EXPECT_FALSE(dtx);
EXPECT_FALSE(vad);
if (test_mode_ != 0) {
@ -366,19 +366,19 @@ void TestStereo::Perform() {
// Test that VAD/DTX cannot be turned on while sending stereo.
EXPECT_EQ(-1, acm_a_->SetVAD(true, true, VADNormal));
EXPECT_EQ(0, acm_a_->VAD(dtx, vad, vad_mode));
EXPECT_EQ(0, acm_a_->VAD(&dtx, &vad, &vad_mode));
EXPECT_FALSE(dtx);
EXPECT_FALSE(vad);
EXPECT_EQ(-1, acm_a_->SetVAD(true, false, VADNormal));
EXPECT_EQ(0, acm_a_->VAD(dtx, vad, vad_mode));
EXPECT_EQ(0, acm_a_->VAD(&dtx, &vad, &vad_mode));
EXPECT_FALSE(dtx);
EXPECT_FALSE(vad);
EXPECT_EQ(-1, acm_a_->SetVAD(false, true, VADNormal));
EXPECT_EQ(0, acm_a_->VAD(dtx, vad, vad_mode));
EXPECT_EQ(0, acm_a_->VAD(&dtx, &vad, &vad_mode));
EXPECT_FALSE(dtx);
EXPECT_FALSE(vad);
EXPECT_EQ(0, acm_a_->SetVAD(false, false, VADNormal));
EXPECT_EQ(0, acm_a_->VAD(dtx, vad, vad_mode));
EXPECT_EQ(0, acm_a_->VAD(&dtx, &vad, &vad_mode));
EXPECT_FALSE(dtx);
EXPECT_FALSE(vad);
@ -603,7 +603,7 @@ void TestStereo::Perform() {
// Make sure it is possible to set VAD/CNG, now that we are sending mono
// again.
EXPECT_EQ(0, acm_a_->SetVAD(true, true, VADNormal));
EXPECT_EQ(0, acm_a_->VAD(dtx, vad, vad_mode));
EXPECT_EQ(0, acm_a_->VAD(&dtx, &vad, &vad_mode));
EXPECT_TRUE(dtx);
EXPECT_TRUE(vad);
EXPECT_EQ(0, acm_a_->SetVAD(false, false, VADNormal));
@ -687,7 +687,7 @@ void TestStereo::Perform() {
opus_pltype_);
CodecInst opus_codec_param;
for (WebRtc_UWord8 n = 0; n < num_encoders; n++) {
EXPECT_EQ(0, acm_b_->Codec(n, opus_codec_param));
EXPECT_EQ(0, acm_b_->Codec(n, &opus_codec_param));
if (!strcmp(opus_codec_param.plname, "opus")) {
opus_codec_param.channels = 1;
EXPECT_EQ(0, acm_b_->RegisterReceiveCodec(opus_codec_param));
@ -821,7 +821,7 @@ void TestStereo::RegisterSendCodec(char side, char* codec_name,
CodecInst my_codec_param;
// Get all codec parameters before registering
CHECK_ERROR(AudioCodingModule::Codec(codec_name, my_codec_param,
CHECK_ERROR(AudioCodingModule::Codec(codec_name, &my_codec_param,
sampling_freq_hz, channels));
my_codec_param.rate = rate;
my_codec_param.pacsize = pack_size;
@ -888,7 +888,7 @@ void TestStereo::Run(TestPackStereo* channel, int in_channels, int out_channels,
}
// Run received side of ACM
CHECK_ERROR(acm_b_->PlayoutData10Ms(out_freq_hz_b, audio_frame));
CHECK_ERROR(acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
// Write output speech to file
out_file_.Write10MsData(
@ -919,11 +919,11 @@ void TestStereo::OpenOutFile(WebRtc_Word16 test_number) {
void TestStereo::DisplaySendReceiveCodec() {
CodecInst my_codec_param;
acm_a_->SendCodec(my_codec_param);
acm_a_->SendCodec(&my_codec_param);
if (test_mode_ != 0) {
printf("%s -> ", my_codec_param.plname);
}
acm_b_->ReceiveCodec(my_codec_param);
acm_b_->ReceiveCodec(&my_codec_param);
if (test_mode_ != 0) {
printf("%s\n", my_codec_param.plname);
}

16
webrtc/modules/audio_coding/main/test/TestVADDTX.cc
View File

@ -78,7 +78,7 @@ void TestVADDTX::Perform()
}
for(WebRtc_UWord8 n = 0; n < numEncoders; n++)
{
_acmB->Codec(n, myCodecParam);
_acmB->Codec(n, &myCodecParam);
if(_testMode != 0)
{
printf("%s\n", myCodecParam.plname);
@ -174,7 +174,7 @@ void TestVADDTX::runTestCases()
if(_testMode != 0)
{
CodecInst myCodecParam;
_acmA->SendCodec(myCodecParam);
_acmA->SendCodec(&myCodecParam);
printf("%s\n", myCodecParam.plname);
}
else
@ -239,7 +239,7 @@ void TestVADDTX::SetVAD(bool statusDTX, bool statusVAD, WebRtc_Word16 vadMode)
if (_acmA->SetVAD(statusDTX, statusVAD, (ACMVADMode) vadMode) < 0) {
assert(false);
}
if (_acmA->VAD(dtxEnabled, vadEnabled, vadModeSet) < 0) {
if (_acmA->VAD(&dtxEnabled, &vadEnabled, &vadModeSet) < 0) {
assert(false);
}
@ -282,7 +282,7 @@ VADDTXstruct TestVADDTX::GetVAD()
bool dtxEnabled, vadEnabled;
ACMVADMode vadModeSet;
if (_acmA->VAD(dtxEnabled, vadEnabled, vadModeSet) < 0) {
if (_acmA->VAD(&dtxEnabled, &vadEnabled, &vadModeSet) < 0) {
assert(false);
}
@ -328,7 +328,7 @@ WebRtc_Word16 TestVADDTX::RegisterSendCodec(char side,
for(WebRtc_Word16 codecCntr = 0; codecCntr < myACM->NumberOfCodecs();
codecCntr++)
{
CHECK_ERROR(myACM->Codec((WebRtc_UWord8)codecCntr, myCodecParam));
CHECK_ERROR(myACM->Codec((WebRtc_UWord8)codecCntr, &myCodecParam));
if(!STR_CASE_CMP(myCodecParam.plname, codecName))
{
if((samplingFreqHz == -1) || (myCodecParam.plfreq == samplingFreqHz))
@ -366,7 +366,7 @@ void TestVADDTX::Run()
CHECK_ERROR(_acmA->Process());
CHECK_ERROR(_acmB->PlayoutData10Ms(outFreqHzB, audioFrame));
CHECK_ERROR(_acmB->PlayoutData10Ms(outFreqHzB, &audioFrame));
_outFileB.Write10MsData(audioFrame.data_, audioFrame.samples_per_channel_);
}
#ifdef PRINT_STAT
@ -399,7 +399,7 @@ WebRtc_Word16 TestVADDTX::VerifyTest()
WebRtc_UWord8 vadPattern = 0;
WebRtc_UWord8 emptyFramePattern[6];
CodecInst myCodecParam;
_acmA->SendCodec(myCodecParam);
_acmA->SendCodec(&myCodecParam);
bool dtxInUse = true;
bool isReplaced = false;
if ((STR_CASE_CMP(myCodecParam.plname,"G729") == 0) ||
@ -408,7 +408,7 @@ WebRtc_Word16 TestVADDTX::VerifyTest()
(STR_CASE_CMP(myCodecParam.plname,"AMR-wb") == 0) ||
(STR_CASE_CMP(myCodecParam.plname,"speex") == 0))
{
_acmA->IsInternalDTXReplacedWithWebRtc(isReplaced);
_acmA->IsInternalDTXReplacedWithWebRtc(&isReplaced);
if (!isReplaced)
{
dtxInUse = false;

26
webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc
View File

@ -78,7 +78,7 @@ TwoWayCommunication::ChooseCodec(WebRtc_UWord8* codecID_A,
printf("========================\n");
for(WebRtc_UWord8 codecCntr = 0; codecCntr < noCodec; codecCntr++)
{
tmpACM->Codec(codecCntr, codecInst);
tmpACM->Codec(codecCntr, &codecInst);
printf("%d- %s\n", codecCntr, codecInst.plname);
}
printf("\nChoose a send codec for side A [0]: ");
@ -110,10 +110,10 @@ WebRtc_Word16 TwoWayCommunication::SetUp()
CodecInst codecInst_A;
CodecInst codecInst_B;
CodecInst dummyCodec;
_acmA->Codec(codecID_A, codecInst_A);
_acmB->Codec(codecID_B, codecInst_B);
_acmA->Codec(codecID_A, &codecInst_A);
_acmB->Codec(codecID_B, &codecInst_B);
_acmA->Codec(6, dummyCodec);
_acmA->Codec(6, &dummyCodec);
//--- Set A codecs
CHECK_ERROR(_acmA->RegisterSendCodec(codecInst_A));
@ -214,9 +214,9 @@ WebRtc_Word16 TwoWayCommunication::SetUpAutotest()
CodecInst codecInst_B;
CodecInst dummyCodec;
_acmA->Codec("ISAC", codecInst_A, 16000, 1);
_acmB->Codec("L16", codecInst_B, 8000, 1);
_acmA->Codec(6, dummyCodec);
_acmA->Codec("ISAC", &codecInst_A, 16000, 1);
_acmB->Codec("L16", &codecInst_B, 8000, 1);
_acmA->Codec(6, &dummyCodec);
//--- Set A codecs
CHECK_ERROR(_acmA->RegisterSendCodec(codecInst_A));
@ -320,7 +320,7 @@ TwoWayCommunication::Perform()
CodecInst codecInst_B;
CodecInst dummy;
_acmB->SendCodec(codecInst_B);
_acmB->SendCodec(&codecInst_B);
if(_testMode != 0)
{
@ -345,16 +345,16 @@ TwoWayCommunication::Perform()
_acmRefA->Process();
_acmRefB->Process();
_acmA->PlayoutData10Ms(outFreqHzA, audioFrame);
_acmA->PlayoutData10Ms(outFreqHzA, &audioFrame);
_outFileA.Write10MsData(audioFrame);
_acmRefA->PlayoutData10Ms(outFreqHzA, audioFrame);
_acmRefA->PlayoutData10Ms(outFreqHzA, &audioFrame);
_outFileRefA.Write10MsData(audioFrame);
_acmB->PlayoutData10Ms(outFreqHzB, audioFrame);
_acmB->PlayoutData10Ms(outFreqHzB, &audioFrame);
_outFileB.Write10MsData(audioFrame);
_acmRefB->PlayoutData10Ms(outFreqHzB, audioFrame);
_acmRefB->PlayoutData10Ms(outFreqHzB, &audioFrame);
_outFileRefB.Write10MsData(audioFrame);
msecPassed += 10;
@ -398,7 +398,7 @@ TwoWayCommunication::Perform()
printf("Register Send Codec (audio back in side A)\n");
}
CHECK_ERROR(_acmB->RegisterSendCodec(codecInst_B));
CHECK_ERROR(_acmB->SendCodec(dummy));
CHECK_ERROR(_acmB->SendCodec(&dummy));
}
if(((secPassed%7) == 6) && (msecPassed == 0))
{

10
webrtc/modules/audio_coding/main/test/delay_test.cc
View File

@ -108,7 +108,7 @@ class DelayTest {
WebRtc_UWord8 num_encoders = acm_a_->NumberOfCodecs();
CodecInst my_codec_param;
for(int n = 0; n < num_encoders; n++) {
acm_b_->Codec(n, my_codec_param);
acm_b_->Codec(n, &my_codec_param);
if (STR_CASE_CMP(my_codec_param.plname, "opus") == 0)
my_codec_param.channels = 1;
else if (my_codec_param.channels > 1)
@ -155,7 +155,7 @@ class DelayTest {
void SendCodec(const CodecConfig& config) {
CodecInst my_codec_param;
ASSERT_EQ(0, AudioCodingModule::Codec(config.name, my_codec_param,
ASSERT_EQ(0, AudioCodingModule::Codec(config.name, &my_codec_param,
config.sample_rate_hz,
config.num_channels));
encoding_sample_rate_hz_ = my_codec_param.plfreq;
@ -201,7 +201,7 @@ class DelayTest {
// Print delay information every 16 frame
if ((num_frames & 0x3F) == 0x3F) {
ACMNetworkStatistics statistics;
acm_b_->NetworkStatistics(statistics);
acm_b_->NetworkStatistics(&statistics);
fprintf(stdout, "delay: min=%3d max=%3d mean=%3d median=%3d"
" ts-based average = %6.3f, "
"curr buff-lev = %4u opt buff-lev = %4u \n",
@ -218,11 +218,11 @@ class DelayTest {
in_file_a_.Read10MsData(audio_frame);
ASSERT_EQ(0, acm_a_->Add10MsData(audio_frame));
ASSERT_LE(0, acm_a_->Process());
ASSERT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, audio_frame));
ASSERT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
out_file_b_.Write10MsData(audio_frame.data_,
audio_frame.samples_per_channel_ *
audio_frame.num_channels_);
acm_b_->PlayoutTimestamp(playout_ts);
acm_b_->PlayoutTimestamp(&playout_ts);
received_ts = channel_a2b_->LastInTimestamp();
inst_delay_sec = static_cast<uint32_t>(received_ts - playout_ts) /
static_cast<double>(encoding_sample_rate_hz_);

8
webrtc/modules/audio_coding/main/test/dual_stream_unittest.cc
View File

@ -111,7 +111,7 @@ void DualStreamTest::PopulateCodecInstances(int frame_size_primary_ms,
red_encoder_.pltype = -1;
for (int n = 0; n < AudioCodingModule::NumberOfCodecs(); n++) {
AudioCodingModule::Codec(n, my_codec);
AudioCodingModule::Codec(n, &my_codec);
if (strcmp(my_codec.plname, "ISAC") == 0 &&
my_codec.plfreq == sampling_rate) {
my_codec.rate = 32000;
@ -480,7 +480,7 @@ TEST_F(DualStreamTest, Api) {
bool vad_status;
bool dtx_status;
ACMVADMode vad_mode;
EXPECT_EQ(0, acm_dual_stream_->VAD(vad_status, dtx_status, vad_mode));
EXPECT_EQ(0, acm_dual_stream_->VAD(&vad_status, &dtx_status, &vad_mode));
EXPECT_TRUE(vad_status);
EXPECT_TRUE(dtx_status);
EXPECT_EQ(VADNormal, vad_mode);
@ -492,7 +492,7 @@ TEST_F(DualStreamTest, Api) {
ASSERT_EQ(0, memcmp(&my_codec, &secondary_encoder_, sizeof(my_codec)));
// Test if VAD get disabled after registering secondary codec.
EXPECT_EQ(0, acm_dual_stream_->VAD(vad_status, dtx_status, vad_mode));
EXPECT_EQ(0, acm_dual_stream_->VAD(&vad_status, &dtx_status, &vad_mode));
EXPECT_FALSE(vad_status);
EXPECT_FALSE(dtx_status);
@ -506,7 +506,7 @@ TEST_F(DualStreamTest, Api) {
ASSERT_EQ(0, acm_dual_stream_->SetVAD(true, true, VADVeryAggr));
// Make sure VAD is activated.
EXPECT_EQ(0, acm_dual_stream_->VAD(vad_status, dtx_status, vad_mode));
EXPECT_EQ(0, acm_dual_stream_->VAD(&vad_status, &dtx_status, &vad_mode));
EXPECT_TRUE(vad_status);
EXPECT_TRUE(dtx_status);
EXPECT_EQ(VADVeryAggr, vad_mode);

16
webrtc/modules/audio_coding/main/test/iSACTest.cc
View File

@ -55,7 +55,7 @@ WebRtc_Word16 SetISAConfig(
(isacConfig.currentFrameSizeMsec != 0))
{
CodecInst sendCodec;
acm->SendCodec(sendCodec);
acm->SendCodec(&sendCodec);
if(isacConfig.currentRateBitPerSec < 0)
{
sendCodec.rate = -1;
@ -155,7 +155,7 @@ ISACTest::Setup()
for(codecCntr = 0; codecCntr < AudioCodingModule::NumberOfCodecs(); codecCntr++)
{
AudioCodingModule::Codec(codecCntr, codecParam);
AudioCodingModule::Codec(codecCntr, &codecParam);
if(!STR_CASE_CMP(codecParam.plname, "ISAC") && codecParam.plfreq == 16000)
{
memcpy(&_paramISAC16kHz, &codecParam, sizeof(CodecInst));
@ -210,14 +210,14 @@ ISACTest::Setup()
Run10ms();
}
CodecInst receiveCodec;
CHECK_ERROR(_acmA->ReceiveCodec(receiveCodec));
CHECK_ERROR(_acmA->ReceiveCodec(&receiveCodec));
if(_testMode != 0)
{
printf("Side A Receive Codec\n");
printf("%s %d\n", receiveCodec.plname, receiveCodec.plfreq);
}
CHECK_ERROR(_acmB->ReceiveCodec(receiveCodec));
CHECK_ERROR(_acmB->ReceiveCodec(&receiveCodec));
if(_testMode != 0)
{
printf("Side B Receive Codec\n");
@ -357,10 +357,10 @@ ISACTest::Run10ms()
CHECK_ERROR(_acmA->Process());
CHECK_ERROR(_acmB->Process());
CHECK_ERROR(_acmA->PlayoutData10Ms(32000, audioFrame));
CHECK_ERROR(_acmA->PlayoutData10Ms(32000, &audioFrame));
_outFileA.Write10MsData(audioFrame);
CHECK_ERROR(_acmB->PlayoutData10Ms(32000, audioFrame));
CHECK_ERROR(_acmB->PlayoutData10Ms(32000, &audioFrame));
_outFileB.Write10MsData(audioFrame);
}
@ -444,9 +444,9 @@ ISACTest::EncodeDecode(
{
myEvent->Wait(5000);
_acmA->SendCodec(sendCodec);
_acmA->SendCodec(&sendCodec);
if(_testMode == 2) printf("[%d] ", sendCodec.rate);
_acmB->SendCodec(sendCodec);
_acmB->SendCodec(&sendCodec);
if(_testMode == 2) printf("[%d] ", sendCodec.rate);
}
}

16
webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc
View File

@ -80,7 +80,7 @@ class InitialPlayoutDelayTest : public ::testing::Test {
const int kChannels[2] = {1, 2};
for (int n = 0; n < 3; ++n) {
for (int k = 0; k < 2; ++k) {
AudioCodingModule::Codec("L16", codec, kFsHz[n], kChannels[k]);
AudioCodingModule::Codec("L16", &codec, kFsHz[n], kChannels[k]);
acm_b_->RegisterReceiveCodec(codec);
}
}
@ -114,7 +114,7 @@ class InitialPlayoutDelayTest : public ::testing::Test {
timestamp += in_audio_frame.samples_per_channel_;
ASSERT_EQ(0, acm_a_->Add10MsData(in_audio_frame));
ASSERT_LE(0, acm_a_->Process());
ASSERT_EQ(0, acm_b_->PlayoutData10Ms(codec.plfreq, out_audio_frame));
ASSERT_EQ(0, acm_b_->PlayoutData10Ms(codec.plfreq, &out_audio_frame));
rms = FrameRms(out_audio_frame);
++num_frames;
}
@ -131,38 +131,38 @@ class InitialPlayoutDelayTest : public ::testing::Test {
TEST_F( InitialPlayoutDelayTest, NbMono) {
CodecInst codec;
AudioCodingModule::Codec("L16", codec, 8000, 1);
AudioCodingModule::Codec("L16", &codec, 8000, 1);
Run(codec, 3000);
}
TEST_F( InitialPlayoutDelayTest, WbMono) {
CodecInst codec;
AudioCodingModule::Codec("L16", codec, 16000, 1);
AudioCodingModule::Codec("L16", &codec, 16000, 1);
Run(codec, 3000);
}
TEST_F( InitialPlayoutDelayTest, SwbMono) {
CodecInst codec;
AudioCodingModule::Codec("L16", codec, 32000, 1);
AudioCodingModule::Codec("L16", &codec, 32000, 1);
Run(codec, 2000); // NetEq buffer is not sufficiently large for 3 sec of
// PCM16 super-wideband.
}
TEST_F( InitialPlayoutDelayTest, NbStereo) {
CodecInst codec;
AudioCodingModule::Codec("L16", codec, 8000, 2);
AudioCodingModule::Codec("L16", &codec, 8000, 2);
Run(codec, 3000);
}
TEST_F( InitialPlayoutDelayTest, WbStereo) {
CodecInst codec;
AudioCodingModule::Codec("L16", codec, 16000, 2);
AudioCodingModule::Codec("L16", &codec, 16000, 2);
Run(codec, 3000);
}
TEST_F( InitialPlayoutDelayTest, SwbStereo) {
CodecInst codec;
AudioCodingModule::Codec("L16", codec, 32000, 2);
AudioCodingModule::Codec("L16", &codec, 32000, 2);
Run(codec, 2000); // NetEq buffer is not sufficiently large for 3 sec of
// PCM16 super-wideband.
}

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

@ -138,7 +138,7 @@ ChooseCodec(
}
} while(outOfRange);
CHECK_ERROR(AudioCodingModule::Codec((WebRtc_UWord8)codecID, codecInst));
CHECK_ERROR(AudioCodingModule::Codec((WebRtc_UWord8)codecID, &codecInst));
return 0;
}
@ -151,7 +151,7 @@ PrintCodecs()
printf("No Name [Hz] [bps]\n");
for(WebRtc_UWord8 codecCntr = 0; codecCntr < noCodec; codecCntr++)
{
AudioCodingModule::Codec(codecCntr, codecInst);
AudioCodingModule::Codec(codecCntr, &codecInst);
printf("%2d- %-18s %5d %6d\n",
codecCntr, codecInst.plname, codecInst.plfreq, codecInst.rate);
}

5
webrtc/modules/utility/source/coder.cc
View File

@ -76,14 +76,13 @@ WebRtc_Word32 AudioCoder::Decode(AudioFrame& decodedAudio,
return -1;
}
}
return _acm->PlayoutData10Ms((WebRtc_UWord16)sampFreqHz,
(AudioFrame&)decodedAudio);
return _acm->PlayoutData10Ms((WebRtc_UWord16)sampFreqHz, &decodedAudio);
}
WebRtc_Word32 AudioCoder::PlayoutData(AudioFrame& decodedAudio,
WebRtc_UWord16& sampFreqHz)
{
return _acm->PlayoutData10Ms(sampFreqHz, (AudioFrame&)decodedAudio);
return _acm->PlayoutData10Ms(sampFreqHz, &decodedAudio);
}
WebRtc_Word32 AudioCoder::Encode(const AudioFrame& audio,

40
webrtc/voice_engine/channel.cc
View File

@ -661,7 +661,7 @@ Channel::OnInitializeDecoder(
receiveCodec.rate = rate;
strncpy(receiveCodec.plname, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
_audioCodingModule.Codec(payloadName, dummyCodec, frequency, channels);
_audioCodingModule.Codec(payloadName, &dummyCodec, frequency, channels);
receiveCodec.pacsize = dummyCodec.pacsize;
// Register the new codec to the ACM
@ -839,7 +839,7 @@ WebRtc_Word32 Channel::GetAudioFrame(const WebRtc_Word32 id,
// Get 10ms raw PCM data from the ACM (mixer limits output frequency)
if (_audioCodingModule.PlayoutData10Ms(audioFrame.sample_rate_hz_,
audioFrame) == -1)
&audioFrame) == -1)
{
WEBRTC_TRACE(kTraceError, kTraceVoice,
VoEId(_instanceId,_channelId),
@ -1413,7 +1413,7 @@ Channel::Init()
for (int idx = 0; idx < nSupportedCodecs; idx++)
{
// Open up the RTP/RTCP receiver for all supported codecs
if ((_audioCodingModule.Codec(idx, codec) == -1) ||
if ((_audioCodingModule.Codec(idx, &codec) == -1) ||
(_rtpRtcpModule->RegisterReceivePayload(codec) == -1))
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
@ -2254,7 +2254,7 @@ WebRtc_Word32
Channel::GetNetEQBGNMode(NetEqBgnModes& mode)
{
ACMBackgroundNoiseMode noiseMode(On);
_audioCodingModule.BackgroundNoiseMode(noiseMode);
_audioCodingModule.BackgroundNoiseMode(&noiseMode);
switch (noiseMode)
{
case On:
@ -2275,13 +2275,13 @@ Channel::GetNetEQBGNMode(NetEqBgnModes& mode)
WebRtc_Word32
Channel::GetSendCodec(CodecInst& codec)
{
return (_audioCodingModule.SendCodec(codec));
return (_audioCodingModule.SendCodec(&codec));
}
WebRtc_Word32
Channel::GetRecCodec(CodecInst& codec)
{
return (_audioCodingModule.ReceiveCodec(codec));
return (_audioCodingModule.ReceiveCodec(&codec));
}
WebRtc_Word32
@ -2342,7 +2342,7 @@ Channel::GetVADStatus(bool& enabledVAD, ACMVADMode& mode, bool& disabledDTX)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetVADStatus");
if (_audioCodingModule.VAD(disabledDTX, enabledVAD, mode) != 0)
if (_audioCodingModule.VAD(&disabledDTX, &enabledVAD, &mode) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
@ -2504,7 +2504,7 @@ Channel::SetSendCNPayloadType(int type, PayloadFrequencies frequency)
else if (frequency == kFreq16000Hz)
samplingFreqHz = 16000;
if (_audioCodingModule.Codec("CN", codec, samplingFreqHz, kMono) == -1)
if (_audioCodingModule.Codec("CN", &codec, samplingFreqHz, kMono) == -1)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
@ -2546,7 +2546,7 @@ Channel::SetISACInitTargetRate(int rateBps, bool useFixedFrameSize)
"Channel::SetISACInitTargetRate()");
CodecInst sendCodec;
if (_audioCodingModule.SendCodec(sendCodec) == -1)
if (_audioCodingModule.SendCodec(&sendCodec) == -1)
{
_engineStatisticsPtr->SetLastError(
VE_CODEC_ERROR, kTraceError,
@ -2614,7 +2614,7 @@ Channel::SetISACMaxRate(int rateBps)
"Channel::SetISACMaxRate()");
CodecInst sendCodec;
if (_audioCodingModule.SendCodec(sendCodec) == -1)
if (_audioCodingModule.SendCodec(&sendCodec) == -1)
{
_engineStatisticsPtr->SetLastError(
VE_CODEC_ERROR, kTraceError,
@ -2678,7 +2678,7 @@ Channel::SetISACMaxPayloadSize(int sizeBytes)
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetISACMaxPayloadSize()");
CodecInst sendCodec;
if (_audioCodingModule.SendCodec(sendCodec) == -1)
if (_audioCodingModule.SendCodec(&sendCodec) == -1)
{
_engineStatisticsPtr->SetLastError(
VE_CODEC_ERROR, kTraceError,
@ -6082,8 +6082,12 @@ Channel::GetNetworkStatistics(NetworkStatistics& stats)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetNetworkStatistics()");
return _audioCodingModule.NetworkStatistics(
(ACMNetworkStatistics &)stats);
ACMNetworkStatistics acm_stats;
int return_value = _audioCodingModule.NetworkStatistics(&acm_stats);
if (return_value > 0) {
memcpy(&stats, &acm_stats, sizeof(NetworkStatistics));
}
return return_value;
}
int
@ -6416,7 +6420,7 @@ Channel::GetPlayoutTimeStamp(WebRtc_UWord32& playoutTimestamp)
WebRtc_UWord32 timestamp(0);
CodecInst currRecCodec;
if (_audioCodingModule.PlayoutTimestamp(timestamp) == -1)
if (_audioCodingModule.PlayoutTimestamp(&timestamp) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetPlayoutTimeStamp() failed to read playout"
@ -6434,7 +6438,7 @@ Channel::GetPlayoutTimeStamp(WebRtc_UWord32& playoutTimestamp)
}
WebRtc_Word32 playoutFrequency = _audioCodingModule.PlayoutFrequency();
if (_audioCodingModule.ReceiveCodec(currRecCodec) == 0) {
if (_audioCodingModule.ReceiveCodec(&currRecCodec) == 0) {
if (STR_CASE_CMP("G722", currRecCodec.plname) == 0) {
playoutFrequency = 8000;
} else if (STR_CASE_CMP("opus", currRecCodec.plname) == 0) {
@ -6513,7 +6517,7 @@ Channel::UpdatePacketDelay(const WebRtc_UWord32 timestamp,
rtpReceiveFrequency = _audioCodingModule.ReceiveFrequency();
CodecInst currRecCodec;
if (_audioCodingModule.ReceiveCodec(currRecCodec) == 0) {
if (_audioCodingModule.ReceiveCodec(&currRecCodec) == 0) {
if (STR_CASE_CMP("G722", currRecCodec.plname) == 0) {
// Even though the actual sampling rate for G.722 audio is
// 16,000 Hz, the RTP clock rate for the G722 payload format is
@ -6618,7 +6622,7 @@ Channel::RegisterReceiveCodecsToRTPModule()
for (int idx = 0; idx < nSupportedCodecs; idx++)
{
// Open up the RTP/RTCP receiver for all supported codecs
if ((_audioCodingModule.Codec(idx, codec) == -1) ||
if ((_audioCodingModule.Codec(idx, &codec) == -1) ||
(_rtpRtcpModule->RegisterReceivePayload(codec) == -1))
{
WEBRTC_TRACE(
@ -6710,7 +6714,7 @@ int Channel::SetRedPayloadType(int red_payload_type) {
// Get default RED settings from the ACM database
const int num_codecs = AudioCodingModule::NumberOfCodecs();
for (int idx = 0; idx < num_codecs; idx++) {
_audioCodingModule.Codec(idx, codec);
_audioCodingModule.Codec(idx, &codec);
if (!STR_CASE_CMP(codec.plname, "RED")) {
found_red = true;
break;

2
webrtc/voice_engine/voe_codec_impl.cc
View File

@ -68,7 +68,7 @@ int VoECodecImpl::GetCodec(int index, CodecInst& codec)
WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
"GetCodec(index=%d, codec=?)", index);
CodecInst acmCodec;
if (AudioCodingModule::Codec(index, (CodecInst&) acmCodec)
if (AudioCodingModule::Codec(index, &acmCodec)
== -1)
{
_shared->SetLastError(VE_INVALID_LISTNR, kTraceError,