generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

Fix AudioFrame types.

Browse Source 
volume_ is not set anywhere so I'm removing it.

BUG=
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/556004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2196 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org
2012-05-08 17:12:40 +00:00
parent f5fe1000de
commit e59a0aca6a
11 changed files with 110 additions and 129 deletions
Download Patch File Download Diff File Expand all files Collapse all files

1
src/modules/audio_coding/main/source/audio_coding_module_impl.cc
View File

@@ -1845,7 +1845,6 @@ WebRtc_Word32 AudioCodingModuleImpl::PlayoutData10Ms(
} }
audio_frame.id_ = _id; audio_frame.id_ = _id;
audio_frame.volume_ = -1;
audio_frame.energy_ = -1; audio_frame.energy_ = -1;
audio_frame.timestamp_ = 0; audio_frame.timestamp_ = 0;

6
src/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
View File

@@ -22,7 +22,7 @@ void SetParticipantStatistics(ParticipantStatistics* stats,
const AudioFrame& frame) const AudioFrame& frame)
{ {
stats->participant = frame.id_; stats->participant = frame.id_;
stats->level = frame.volume_; stats->level = 0; // TODO(andrew): to what should this be set?
} }
} // namespace } // namespace
@@ -1005,9 +1005,9 @@ void AudioConferenceMixerImpl::UpdateVADPositiveParticipants(
_scratchVadPositiveParticipants[ _scratchVadPositiveParticipants[
_scratchVadPositiveParticipantsAmount].participant = _scratchVadPositiveParticipantsAmount].participant =
audioFrame->id_; audioFrame->id_;
// TODO(andrew): to what should this be set?
_scratchVadPositiveParticipants[ _scratchVadPositiveParticipants[
_scratchVadPositiveParticipantsAmount].level = _scratchVadPositiveParticipantsAmount].level = 0;
audioFrame->volume_;
_scratchVadPositiveParticipantsAmount++; _scratchVadPositiveParticipantsAmount++;
} }
item = mixList.Next(item); item = mixList.Next(item);

158
src/modules/interface/module_common_types.h
View File

@@ -694,35 +694,23 @@ VideoFrame::Free()
} }
/************************************************* /* This class holds up to 60 ms of super-wideband (32 kHz) stereo audio. It
* allows for adding and subtracting frames while keeping track of the resulting
* states.
* *
* AudioFrame class * Notes
* - The total number of samples in |data_| is
* samples_per_channel_ * num_channels_
* *
* The AudioFrame class holds up to 60 ms wideband * - Stereo data is interleaved starting with the left channel.
* audio. It allows for adding and subtracting frames
* while keeping track of the resulting states.
* *
* Note * - The +operator assume that you would never add exactly opposite frames when
* - The +operator assume that you would never add * deciding the resulting state. To do this use the -operator.
* exact opposite frames when deciding the resulting */
* state. To do this use the -operator.
*
* - num_channels_ of 1 indicated mono, and 2
* indicates stereo.
*
* - samples_per_channel_ is the number of
* samples per channel. Therefore, the total
* number of samples in data_ is
* (samples_per_channel_ * num_channels_).
*
* - Stereo data is stored in interleaved fashion
* starting with the left channel.
*
*************************************************/
class AudioFrame class AudioFrame
{ {
public: public:
enum { kMaxDataSizeSamples = 3840 }; // stereo 32KHz 60ms 2*32*60 enum { kMaxDataSizeSamples = 3840 }; // stereo, 32 kHz, 60ms (2*32*60)
enum VADActivity enum VADActivity
{ {
@@ -742,39 +730,35 @@ public:
AudioFrame(); AudioFrame();
virtual ~AudioFrame(); virtual ~AudioFrame();
WebRtc_Word32 UpdateFrame( int UpdateFrame(
const WebRtc_Word32 id, int id,
const WebRtc_UWord32 timeStamp, uint32_t timestamp,
const WebRtc_Word16* payloadData, const int16_t* data,
const WebRtc_UWord16 payloadDataLengthInSamples, int samples_per_channel,
const int frequencyInHz, int sample_rate_hz,
const SpeechType speechType, SpeechType speech_type,
const VADActivity vadActivity, VADActivity vad_activity,
const WebRtc_UWord8 audioChannel = 1, int num_channels = 1,
const WebRtc_Word32 volume = -1, uint32_t energy = -1);
const WebRtc_Word32 energy = -1);
AudioFrame& Append(const AudioFrame& rhs); AudioFrame& Append(const AudioFrame& rhs);
void Mute(); void Mute();
AudioFrame& operator=(const AudioFrame& rhs); AudioFrame& operator=(const AudioFrame& rhs);
AudioFrame& operator>>=(const WebRtc_Word32 rhs); AudioFrame& operator>>=(const int rhs);
AudioFrame& operator+=(const AudioFrame& rhs); AudioFrame& operator+=(const AudioFrame& rhs);
AudioFrame& operator-=(const AudioFrame& rhs); AudioFrame& operator-=(const AudioFrame& rhs);
// TODO(andrew): clean up types. int id_;
WebRtc_Word32 id_; uint32_t timestamp_;
WebRtc_UWord32 timestamp_; int16_t data_[kMaxDataSizeSamples];
int samples_per_channel_;
WebRtc_Word16 data_[kMaxDataSizeSamples];
WebRtc_UWord16 samples_per_channel_;
int sample_rate_hz_; int sample_rate_hz_;
WebRtc_UWord8 num_channels_; int num_channels_;
SpeechType speech_type_; SpeechType speech_type_;
VADActivity vad_activity_; VADActivity vad_activity_;
WebRtc_UWord32 energy_; uint32_t energy_;
WebRtc_Word32 volume_; // TODO(andrew): investigate removing.
}; };
inline inline
@@ -788,8 +772,7 @@ AudioFrame::AudioFrame()
num_channels_(1), num_channels_(1),
speech_type_(kUndefined), speech_type_(kUndefined),
vad_activity_(kVadUnknown), vad_activity_(kVadUnknown),
energy_(0xffffffff), energy_(0xffffffff)
volume_(0xffffffff)
{ {
} }
@@ -799,44 +782,42 @@ AudioFrame::~AudioFrame()
} }
inline inline
WebRtc_Word32 int
AudioFrame::UpdateFrame( AudioFrame::UpdateFrame(
const WebRtc_Word32 id, int id,
const WebRtc_UWord32 timeStamp, uint32_t timestamp,
const WebRtc_Word16* payloadData, const int16_t* data,
const WebRtc_UWord16 payloadDataLengthInSamples, int samples_per_channel,
const int frequencyInHz, int sample_rate_hz,
const SpeechType speechType, SpeechType speech_type,
const VADActivity vadActivity, VADActivity vad_activity,
const WebRtc_UWord8 audioChannel, int num_channels,
const WebRtc_Word32 volume, uint32_t energy)
const WebRtc_Word32 energy)
{ {
id_ = id; id_ = id;
timestamp_ = timeStamp; timestamp_ = timestamp;
sample_rate_hz_ = frequencyInHz; sample_rate_hz_ = sample_rate_hz;
speech_type_ = speechType; speech_type_ = speech_type;
vad_activity_ = vadActivity; vad_activity_ = vad_activity;
volume_ = volume; num_channels_ = num_channels;
num_channels_ = audioChannel;
energy_ = energy; energy_ = energy;
if((payloadDataLengthInSamples > kMaxDataSizeSamples) || if((samples_per_channel > kMaxDataSizeSamples) ||
(audioChannel > 2) || (audioChannel < 1)) (num_channels > 2) || (num_channels < 1))
{ {
samples_per_channel_ = 0; samples_per_channel_ = 0;
return -1; return -1;
} }
samples_per_channel_ = payloadDataLengthInSamples; samples_per_channel_ = samples_per_channel;
if(payloadData != NULL) if(data != NULL)
{ {
memcpy(data_, payloadData, sizeof(WebRtc_Word16) * memcpy(data_, data, sizeof(int16_t) *
payloadDataLengthInSamples * num_channels_); samples_per_channel * num_channels_);
} }
else else
{ {
memset(data_,0,sizeof(WebRtc_Word16) * memset(data_,0,sizeof(int16_t) *
payloadDataLengthInSamples * num_channels_); samples_per_channel * num_channels_);
} }
return 0; return 0;
} }
@@ -845,7 +826,7 @@ inline
void void
AudioFrame::Mute() AudioFrame::Mute()
{ {
memset(data_, 0, samples_per_channel_ * sizeof(WebRtc_Word16)); memset(data_, 0, samples_per_channel_ * num_channels_ * sizeof(int16_t));
} }
inline inline
@@ -868,20 +849,19 @@ AudioFrame::operator=(const AudioFrame& rhs)
sample_rate_hz_ = rhs.sample_rate_hz_; sample_rate_hz_ = rhs.sample_rate_hz_;
speech_type_ = rhs.speech_type_; speech_type_ = rhs.speech_type_;
vad_activity_ = rhs.vad_activity_; vad_activity_ = rhs.vad_activity_;
volume_ = rhs.volume_;
num_channels_ = rhs.num_channels_; num_channels_ = rhs.num_channels_;
energy_ = rhs.energy_; energy_ = rhs.energy_;
samples_per_channel_ = rhs.samples_per_channel_; samples_per_channel_ = rhs.samples_per_channel_;
memcpy(data_, rhs.data_, memcpy(data_, rhs.data_,
sizeof(WebRtc_Word16) * rhs.samples_per_channel_ * num_channels_); sizeof(int16_t) * rhs.samples_per_channel_ * num_channels_);
return *this; return *this;
} }
inline inline
AudioFrame& AudioFrame&
AudioFrame::operator>>=(const WebRtc_Word32 rhs) AudioFrame::operator>>=(const int rhs)
{ {
assert((num_channels_ > 0) && (num_channels_ < 3)); assert((num_channels_ > 0) && (num_channels_ < 3));
if((num_channels_ > 2) || if((num_channels_ > 2) ||
@@ -889,9 +869,9 @@ AudioFrame::operator>>=(const WebRtc_Word32 rhs)
{ {
return *this; return *this;
} }
for(WebRtc_UWord16 i = 0; i < samples_per_channel_ * num_channels_; i++) for(int i = 0; i < samples_per_channel_ * num_channels_; i++)
{ {
data_[i] = WebRtc_Word16(data_[i] >> rhs); data_[i] = static_cast<int16_t>(data_[i] >> rhs);
} }
return *this; return *this;
} }
@@ -926,8 +906,8 @@ AudioFrame::Append(const AudioFrame& rhs)
speech_type_ = kUndefined; speech_type_ = kUndefined;
} }
WebRtc_UWord16 offset = samples_per_channel_ * num_channels_; int offset = samples_per_channel_ * num_channels_;
for(WebRtc_UWord16 i = 0; for(int i = 0;
i < rhs.samples_per_channel_ * rhs.num_channels_; i < rhs.samples_per_channel_ * rhs.num_channels_;
i++) i++)
{ {
@@ -986,14 +966,14 @@ AudioFrame::operator+=(const AudioFrame& rhs)
if(noPrevData) if(noPrevData)
{ {
memcpy(data_, rhs.data_, memcpy(data_, rhs.data_,
sizeof(WebRtc_Word16) * rhs.samples_per_channel_ * num_channels_); sizeof(int16_t) * rhs.samples_per_channel_ * num_channels_);
} else } else
{ {
// IMPROVEMENT this can be done very fast in assembly // IMPROVEMENT this can be done very fast in assembly
for(WebRtc_UWord16 i = 0; i < samples_per_channel_ * num_channels_; i++) for(int i = 0; i < samples_per_channel_ * num_channels_; i++)
{ {
WebRtc_Word32 wrapGuard = (WebRtc_Word32)data_[i] + int32_t wrapGuard = static_cast<int32_t>(data_[i]) +
(WebRtc_Word32)rhs.data_[i]; static_cast<int32_t>(rhs.data_[i]);
if(wrapGuard < -32768) if(wrapGuard < -32768)
{ {
data_[i] = -32768; data_[i] = -32768;
@@ -1002,12 +982,11 @@ AudioFrame::operator+=(const AudioFrame& rhs)
data_[i] = 32767; data_[i] = 32767;
}else }else
{ {
data_[i] = (WebRtc_Word16)wrapGuard; data_[i] = (int16_t)wrapGuard;
} }
} }
} }
energy_ = 0xffffffff; energy_ = 0xffffffff;
volume_ = 0xffffffff;
return *this; return *this;
} }
@@ -1034,10 +1013,10 @@ AudioFrame::operator-=(const AudioFrame& rhs)
} }
speech_type_ = kUndefined; speech_type_ = kUndefined;
for(WebRtc_UWord16 i = 0; i < samples_per_channel_ * num_channels_; i++) for(int i = 0; i < samples_per_channel_ * num_channels_; i++)
{ {
WebRtc_Word32 wrapGuard = (WebRtc_Word32)data_[i] - int32_t wrapGuard = static_cast<int32_t>(data_[i]) -
(WebRtc_Word32)rhs.data_[i]; static_cast<int32_t>(rhs.data_[i]);
if(wrapGuard < -32768) if(wrapGuard < -32768)
{ {
data_[i] = -32768; data_[i] = -32768;
@@ -1048,11 +1027,10 @@ AudioFrame::operator-=(const AudioFrame& rhs)
} }
else else
{ {
data_[i] = (WebRtc_Word16)wrapGuard; data_[i] = (int16_t)wrapGuard;
} }
} }
energy_ = 0xffffffff; energy_ = 0xffffffff;
volume_ = 0xffffffff;
return *this; return *this;
} }

11
src/modules/utility/interface/file_player.h
View File

@@ -33,10 +33,13 @@ public:
static void DestroyFilePlayer(FilePlayer* player); static void DestroyFilePlayer(FilePlayer* player);
virtual WebRtc_Word32 Get10msAudioFromFile( // Read 10 ms of audio at |frequencyInHz| to |outBuffer|. |lengthInSamples|
WebRtc_Word16* decodedDataBuffer, // will be set to the number of samples read (not the number of samples per
WebRtc_UWord32& decodedDataLengthInSamples, // channel).
const WebRtc_UWord32 frequencyInHz) = 0; virtual int Get10msAudioFromFile(
int16_t* outBuffer,
int& lengthInSamples,
int frequencyInHz) = 0;
// Register callback for receiving file playing notifications. // Register callback for receiving file playing notifications.
virtual WebRtc_Word32 RegisterModuleFileCallback( virtual WebRtc_Word32 RegisterModuleFileCallback(

6
src/modules/utility/source/file_player_impl.cc
View File

@@ -117,9 +117,9 @@ WebRtc_Word32 FilePlayerImpl::AudioCodec(CodecInst& audioCodec) const
} }
WebRtc_Word32 FilePlayerImpl::Get10msAudioFromFile( WebRtc_Word32 FilePlayerImpl::Get10msAudioFromFile(
WebRtc_Word16* outBuffer, int16_t* outBuffer,
WebRtc_UWord32& lengthInSamples, int& lengthInSamples,
WebRtc_UWord32 frequencyInHz) int frequencyInHz)
{ {
if(_codec.plfreq == 0) if(_codec.plfreq == 0)
{ {

9
src/modules/utility/source/file_player_impl.h
View File

@@ -32,11 +32,10 @@ public:
FilePlayerImpl(WebRtc_UWord32 instanceID, FileFormats fileFormat); FilePlayerImpl(WebRtc_UWord32 instanceID, FileFormats fileFormat);
~FilePlayerImpl(); ~FilePlayerImpl();
// FilePlayer functions. virtual int Get10msAudioFromFile(
virtual WebRtc_Word32 Get10msAudioFromFile( int16_t* outBuffer,
WebRtc_Word16* decodedDataBuffer, int& lengthInSamples,
WebRtc_UWord32& decodedDataLengthInSamples, int frequencyInHz);
const WebRtc_UWord32 frequencyInHz);
virtual WebRtc_Word32 RegisterModuleFileCallback(FileCallback* callback); virtual WebRtc_Word32 RegisterModuleFileCallback(FileCallback* callback);
virtual WebRtc_Word32 StartPlayingFile( virtual WebRtc_Word32 StartPlayingFile(
const char* fileName, const char* fileName,

2
src/video_engine/vie_file_player.h
View File

@@ -123,7 +123,7 @@ class ViEFilePlayer
ThreadWrapper* decode_thread_; ThreadWrapper* decode_thread_;
EventWrapper* decode_event_; EventWrapper* decode_event_;
WebRtc_Word16 decoded_audio_[kMaxDecodedAudioLength]; WebRtc_Word16 decoded_audio_[kMaxDecodedAudioLength];
WebRtc_UWord32 decoded_audio_length_; int decoded_audio_length_;
// Trick - list containing VoE buffer reading this file. Used if multiple // Trick - list containing VoE buffer reading this file. Used if multiple
// audio channels are sending. // audio channels are sending.

16
src/voice_engine/main/source/channel.cc
View File

@@ -6142,11 +6142,13 @@ Channel::GetRtpRtcp(RtpRtcp* &rtpRtcpModule) const
return 0; return 0;
} }
// TODO(andrew): refactor Mix functions here and in transmit_mixer.cc to use
// a shared helper.
WebRtc_Word32 WebRtc_Word32
Channel::MixOrReplaceAudioWithFile(const int mixingFrequency) Channel::MixOrReplaceAudioWithFile(const int mixingFrequency)
{ {
scoped_array<WebRtc_Word16> fileBuffer(new WebRtc_Word16[640]); scoped_array<WebRtc_Word16> fileBuffer(new WebRtc_Word16[640]);
WebRtc_UWord32 fileSamples(0); int fileSamples(0);
{ {
CriticalSectionScoped cs(&_fileCritSect); CriticalSectionScoped cs(&_fileCritSect);
@@ -6186,10 +6188,10 @@ Channel::MixOrReplaceAudioWithFile(const int mixingFrequency)
// Currently file stream is always mono. // Currently file stream is always mono.
// TODO(xians): Change the code when FilePlayer supports real stereo. // TODO(xians): Change the code when FilePlayer supports real stereo.
Utility::MixWithSat(_audioFrame.data_, Utility::MixWithSat(_audioFrame.data_,
static_cast<int>(_audioFrame.num_channels_), _audioFrame.num_channels_,
fileBuffer.get(), fileBuffer.get(),
1, 1,
static_cast<int>(fileSamples)); fileSamples);
} }
else else
{ {
@@ -6199,7 +6201,7 @@ Channel::MixOrReplaceAudioWithFile(const int mixingFrequency)
_audioFrame.UpdateFrame(_channelId, _audioFrame.UpdateFrame(_channelId,
-1, -1,
fileBuffer.get(), fileBuffer.get(),
static_cast<WebRtc_UWord16>(fileSamples), fileSamples,
mixingFrequency, mixingFrequency,
AudioFrame::kNormalSpeech, AudioFrame::kNormalSpeech,
AudioFrame::kVadUnknown, AudioFrame::kVadUnknown,
@@ -6216,7 +6218,7 @@ Channel::MixAudioWithFile(AudioFrame& audioFrame,
assert(mixingFrequency <= 32000); assert(mixingFrequency <= 32000);
scoped_array<WebRtc_Word16> fileBuffer(new WebRtc_Word16[640]); scoped_array<WebRtc_Word16> fileBuffer(new WebRtc_Word16[640]);
WebRtc_UWord32 fileSamples(0); int fileSamples(0);
{ {
CriticalSectionScoped cs(&_fileCritSect); CriticalSectionScoped cs(&_fileCritSect);
@@ -6246,10 +6248,10 @@ Channel::MixAudioWithFile(AudioFrame& audioFrame,
// Currently file stream is always mono. // Currently file stream is always mono.
// TODO(xians): Change the code when FilePlayer supports real stereo. // TODO(xians): Change the code when FilePlayer supports real stereo.
Utility::MixWithSat(audioFrame.data_, Utility::MixWithSat(audioFrame.data_,
static_cast<int>(audioFrame.num_channels_), audioFrame.num_channels_,
fileBuffer.get(), fileBuffer.get(),
1, 1,
static_cast<int>(fileSamples)); fileSamples);
} }
else else
{ {

10
src/voice_engine/main/source/transmit_mixer.cc
View File

@@ -1229,8 +1229,7 @@ WebRtc_Word32 TransmitMixer::MixOrReplaceAudioWithFile(
{ {
scoped_array<WebRtc_Word16> fileBuffer(new WebRtc_Word16[640]); scoped_array<WebRtc_Word16> fileBuffer(new WebRtc_Word16[640]);
WebRtc_UWord32 fileSamples(0); int fileSamples(0);
{ {
CriticalSectionScoped cs(&_critSect); CriticalSectionScoped cs(&_critSect);
if (_filePlayerPtr == NULL) if (_filePlayerPtr == NULL)
@@ -1260,10 +1259,10 @@ WebRtc_Word32 TransmitMixer::MixOrReplaceAudioWithFile(
// Currently file stream is always mono. // Currently file stream is always mono.
// TODO(xians): Change the code when FilePlayer supports real stereo. // TODO(xians): Change the code when FilePlayer supports real stereo.
Utility::MixWithSat(_audioFrame.data_, Utility::MixWithSat(_audioFrame.data_,
static_cast<int>(_audioFrame.num_channels_), _audioFrame.num_channels_,
fileBuffer.get(), fileBuffer.get(),
1, 1,
static_cast<int>(fileSamples)); fileSamples);
} else } else
{ {
// Replace ACM audio with file. // Replace ACM audio with file.
@@ -1272,12 +1271,11 @@ WebRtc_Word32 TransmitMixer::MixOrReplaceAudioWithFile(
_audioFrame.UpdateFrame(-1, _audioFrame.UpdateFrame(-1,
-1, -1,
fileBuffer.get(), fileBuffer.get(),
static_cast<WebRtc_UWord16>(fileSamples), fileSamples,
mixingFrequency, mixingFrequency,
AudioFrame::kNormalSpeech, AudioFrame::kNormalSpeech,
AudioFrame::kVadUnknown, AudioFrame::kVadUnknown,
1); 1);
} }
return 0; return 0;
} }

2
src/voice_engine/main/source/voe_base_impl.cc
View File

@@ -256,7 +256,7 @@ WebRtc_Word32 VoEBaseImpl::NeedMorePlayData(
_shared->output_mixer()->GetMixedAudio(samplesPerSec, nChannels, _shared->output_mixer()->GetMixedAudio(samplesPerSec, nChannels,
_audioFrame); _audioFrame);
assert(nSamples == _audioFrame.samples_per_channel_); assert(static_cast<int>(nSamples) == _audioFrame.samples_per_channel_);
assert(samplesPerSec == assert(samplesPerSec ==
static_cast<WebRtc_UWord32>(_audioFrame.sample_rate_hz_)); static_cast<WebRtc_UWord32>(_audioFrame.sample_rate_hz_));

18
src/voice_engine/main/source/voe_file_impl.cc
View File

@@ -628,6 +628,8 @@ int VoEFileImpl::StopRecordingMicrophone()
return _shared->transmit_mixer()->StopRecordingMicrophone(); return _shared->transmit_mixer()->StopRecordingMicrophone();
} }
// TODO(andrew): a cursory inspection suggests there's a large amount of
// overlap in these convert functions which could be refactored to a helper.
int VoEFileImpl::ConvertPCMToWAV(const char* fileNameInUTF8, int VoEFileImpl::ConvertPCMToWAV(const char* fileNameInUTF8,
const char* fileNameOutUTF8) const char* fileNameOutUTF8)
{ {
@@ -677,7 +679,7 @@ int VoEFileImpl::ConvertPCMToWAV(const char* fileNameInUTF8,
// Run throught the file // Run throught the file
AudioFrame audioFrame; AudioFrame audioFrame;
WebRtc_Word16 decodedData[160]; WebRtc_Word16 decodedData[160];
WebRtc_UWord32 decLength=0; int decLength=0;
const WebRtc_UWord32 frequency = 16000; const WebRtc_UWord32 frequency = 16000;
while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency)) while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
@@ -767,7 +769,7 @@ int VoEFileImpl::ConvertPCMToWAV(InStream* streamIn, OutStream* streamOut)
// Run throught the file // Run throught the file
AudioFrame audioFrame; AudioFrame audioFrame;
WebRtc_Word16 decodedData[160]; WebRtc_Word16 decodedData[160];
WebRtc_UWord32 decLength=0; int decLength=0;
const WebRtc_UWord32 frequency = 16000; const WebRtc_UWord32 frequency = 16000;
while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency)) while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
@@ -855,7 +857,7 @@ int VoEFileImpl::ConvertWAVToPCM(const char* fileNameInUTF8,
// Run throught the file // Run throught the file
AudioFrame audioFrame; AudioFrame audioFrame;
WebRtc_Word16 decodedData[160]; WebRtc_Word16 decodedData[160];
WebRtc_UWord32 decLength=0; int decLength=0;
const WebRtc_UWord32 frequency = 16000; const WebRtc_UWord32 frequency = 16000;
while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency)) while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
@@ -947,7 +949,7 @@ int VoEFileImpl::ConvertWAVToPCM(InStream* streamIn, OutStream* streamOut)
// Run throught the file // Run throught the file
AudioFrame audioFrame; AudioFrame audioFrame;
WebRtc_Word16 decodedData[160]; WebRtc_Word16 decodedData[160];
WebRtc_UWord32 decLength=0; int decLength=0;
const WebRtc_UWord32 frequency = 16000; const WebRtc_UWord32 frequency = 16000;
while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency)) while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
@@ -1033,7 +1035,7 @@ int VoEFileImpl::ConvertPCMToCompressed(const char* fileNameInUTF8,
// Run throught the file // Run throught the file
AudioFrame audioFrame; AudioFrame audioFrame;
WebRtc_Word16 decodedData[160]; WebRtc_Word16 decodedData[160];
WebRtc_UWord32 decLength=0; int decLength=0;
const WebRtc_UWord32 frequency = 16000; const WebRtc_UWord32 frequency = 16000;
while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency)) while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
@@ -1125,7 +1127,7 @@ int VoEFileImpl::ConvertPCMToCompressed(InStream* streamIn,
// Run throught the file // Run throught the file
AudioFrame audioFrame; AudioFrame audioFrame;
WebRtc_Word16 decodedData[160]; WebRtc_Word16 decodedData[160];
WebRtc_UWord32 decLength=0; int decLength=0;
const WebRtc_UWord32 frequency = 16000; const WebRtc_UWord32 frequency = 16000;
while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency)) while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
@@ -1215,7 +1217,7 @@ int VoEFileImpl::ConvertCompressedToPCM(const char* fileNameInUTF8,
// Run throught the file // Run throught the file
AudioFrame audioFrame; AudioFrame audioFrame;
WebRtc_Word16 decodedData[160]; WebRtc_Word16 decodedData[160];
WebRtc_UWord32 decLength=0; int decLength=0;
const WebRtc_UWord32 frequency = 16000; const WebRtc_UWord32 frequency = 16000;
while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency)) while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
@@ -1312,7 +1314,7 @@ int VoEFileImpl::ConvertCompressedToPCM(InStream* streamIn,
// Run throught the file // Run throught the file
AudioFrame audioFrame; AudioFrame audioFrame;
WebRtc_Word16 decodedData[160]; WebRtc_Word16 decodedData[160];
WebRtc_UWord32 decLength=0; int decLength=0;
const WebRtc_UWord32 frequency = 16000; const WebRtc_UWord32 frequency = 16000;
while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency)) while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))