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Fix AudioFrame types.

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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
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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.volume_ = -1;
audio_frame.energy_ = -1;
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)
{
stats->participant = frame.id_;
stats->level = frame.volume_;
stats->level = 0; // TODO(andrew): to what should this be set?
}
} // namespace
@@ -1005,9 +1005,9 @@ void AudioConferenceMixerImpl::UpdateVADPositiveParticipants(
_scratchVadPositiveParticipants[
_scratchVadPositiveParticipantsAmount].participant =
audioFrame->id_;
// TODO(andrew): to what should this be set?
_scratchVadPositiveParticipants[
_scratchVadPositiveParticipantsAmount].level =
audioFrame->volume_;
_scratchVadPositiveParticipantsAmount].level = 0;
_scratchVadPositiveParticipantsAmount++;
}
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
* audio. It allows for adding and subtracting frames
* while keeping track of the resulting states.
* - Stereo data is interleaved starting with the left channel.
*
* Note
* - The +operator assume that you would never add
* 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.
*
*************************************************/
* - The +operator assume that you would never add exactly opposite frames when
* deciding the resulting state. To do this use the -operator.
*/
class AudioFrame
{
public:
enum { kMaxDataSizeSamples = 3840 }; // stereo 32KHz 60ms 2*32*60
enum { kMaxDataSizeSamples = 3840 }; // stereo, 32 kHz, 60ms (2*32*60)
enum VADActivity
{
@@ -742,39 +730,35 @@ public:
AudioFrame();
virtual ~AudioFrame();
WebRtc_Word32 UpdateFrame(
const WebRtc_Word32 id,
const WebRtc_UWord32 timeStamp,
const WebRtc_Word16* payloadData,
const WebRtc_UWord16 payloadDataLengthInSamples,
const int frequencyInHz,
const SpeechType speechType,
const VADActivity vadActivity,
const WebRtc_UWord8 audioChannel = 1,
const WebRtc_Word32 volume = -1,
const WebRtc_Word32 energy = -1);
int UpdateFrame(
int id,
uint32_t timestamp,
const int16_t* data,
int samples_per_channel,
int sample_rate_hz,
SpeechType speech_type,
VADActivity vad_activity,
int num_channels = 1,
uint32_t energy = -1);
AudioFrame& Append(const AudioFrame& rhs);
void Mute();
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);
// TODO(andrew): clean up types.
WebRtc_Word32 id_;
WebRtc_UWord32 timestamp_;
WebRtc_Word16 data_[kMaxDataSizeSamples];
WebRtc_UWord16 samples_per_channel_;
int id_;
uint32_t timestamp_;
int16_t data_[kMaxDataSizeSamples];
int samples_per_channel_;
int sample_rate_hz_;
WebRtc_UWord8 num_channels_;
int num_channels_;
SpeechType speech_type_;
VADActivity vad_activity_;
WebRtc_UWord32 energy_;
WebRtc_Word32 volume_; // TODO(andrew): investigate removing.
uint32_t energy_;
};
inline
@@ -788,8 +772,7 @@ AudioFrame::AudioFrame()
num_channels_(1),
speech_type_(kUndefined),
vad_activity_(kVadUnknown),
energy_(0xffffffff),
volume_(0xffffffff)
energy_(0xffffffff)
{
}
@@ -799,44 +782,42 @@ AudioFrame::~AudioFrame()
}
inline
WebRtc_Word32
int
AudioFrame::UpdateFrame(
const WebRtc_Word32 id,
const WebRtc_UWord32 timeStamp,
const WebRtc_Word16* payloadData,
const WebRtc_UWord16 payloadDataLengthInSamples,
const int frequencyInHz,
const SpeechType speechType,
const VADActivity vadActivity,
const WebRtc_UWord8 audioChannel,
const WebRtc_Word32 volume,
const WebRtc_Word32 energy)
int id,
uint32_t timestamp,
const int16_t* data,
int samples_per_channel,
int sample_rate_hz,
SpeechType speech_type,
VADActivity vad_activity,
int num_channels,
uint32_t energy)
{
id_ = id;
timestamp_ = timeStamp;
sample_rate_hz_ = frequencyInHz;
speech_type_ = speechType;
vad_activity_ = vadActivity;
volume_ = volume;
num_channels_ = audioChannel;
timestamp_ = timestamp;
sample_rate_hz_ = sample_rate_hz;
speech_type_ = speech_type;
vad_activity_ = vad_activity;
num_channels_ = num_channels;
energy_ = energy;
if((payloadDataLengthInSamples > kMaxDataSizeSamples) ||
(audioChannel > 2) || (audioChannel < 1))
if((samples_per_channel > kMaxDataSizeSamples) ||
(num_channels > 2) || (num_channels < 1))
{
samples_per_channel_ = 0;
return -1;
}
samples_per_channel_ = payloadDataLengthInSamples;
if(payloadData != NULL)
samples_per_channel_ = samples_per_channel;
if(data != NULL)
{
memcpy(data_, payloadData, sizeof(WebRtc_Word16) *
payloadDataLengthInSamples * num_channels_);
memcpy(data_, data, sizeof(int16_t) *
samples_per_channel * num_channels_);
}
else
{
memset(data_,0,sizeof(WebRtc_Word16) *
payloadDataLengthInSamples * num_channels_);
memset(data_,0,sizeof(int16_t) *
samples_per_channel * num_channels_);
}
return 0;
}
@@ -845,7 +826,7 @@ inline
void
AudioFrame::Mute()
{
memset(data_, 0, samples_per_channel_ * sizeof(WebRtc_Word16));
memset(data_, 0, samples_per_channel_ * num_channels_ * sizeof(int16_t));
}
inline
@@ -868,20 +849,19 @@ AudioFrame::operator=(const AudioFrame& rhs)
sample_rate_hz_ = rhs.sample_rate_hz_;
speech_type_ = rhs.speech_type_;
vad_activity_ = rhs.vad_activity_;
volume_ = rhs.volume_;
num_channels_ = rhs.num_channels_;
energy_ = rhs.energy_;
samples_per_channel_ = rhs.samples_per_channel_;
memcpy(data_, rhs.data_,
sizeof(WebRtc_Word16) * rhs.samples_per_channel_ * num_channels_);
sizeof(int16_t) * rhs.samples_per_channel_ * num_channels_);
return *this;
}
inline
AudioFrame&
AudioFrame::operator>>=(const WebRtc_Word32 rhs)
AudioFrame::operator>>=(const int rhs)
{
assert((num_channels_ > 0) && (num_channels_ < 3));
if((num_channels_ > 2) ||
@@ -889,9 +869,9 @@ AudioFrame::operator>>=(const WebRtc_Word32 rhs)
{
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;
}
@@ -926,8 +906,8 @@ AudioFrame::Append(const AudioFrame& rhs)
speech_type_ = kUndefined;
}
WebRtc_UWord16 offset = samples_per_channel_ * num_channels_;
for(WebRtc_UWord16 i = 0;
int offset = samples_per_channel_ * num_channels_;
for(int i = 0;
i < rhs.samples_per_channel_ * rhs.num_channels_;
i++)
{
@@ -986,14 +966,14 @@ AudioFrame::operator+=(const AudioFrame& rhs)
if(noPrevData)
{
memcpy(data_, rhs.data_,
sizeof(WebRtc_Word16) * rhs.samples_per_channel_ * num_channels_);
sizeof(int16_t) * rhs.samples_per_channel_ * num_channels_);
} else
{
// 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] +
(WebRtc_Word32)rhs.data_[i];
int32_t wrapGuard = static_cast<int32_t>(data_[i]) +
static_cast<int32_t>(rhs.data_[i]);
if(wrapGuard < -32768)
{
data_[i] = -32768;
@@ -1002,12 +982,11 @@ AudioFrame::operator+=(const AudioFrame& rhs)
data_[i] = 32767;
}else
{
data_[i] = (WebRtc_Word16)wrapGuard;
data_[i] = (int16_t)wrapGuard;
}
}
}
energy_ = 0xffffffff;
volume_ = 0xffffffff;
return *this;
}
@@ -1034,10 +1013,10 @@ AudioFrame::operator-=(const AudioFrame& rhs)
}
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] -
(WebRtc_Word32)rhs.data_[i];
int32_t wrapGuard = static_cast<int32_t>(data_[i]) -
static_cast<int32_t>(rhs.data_[i]);
if(wrapGuard < -32768)
{
data_[i] = -32768;
@@ -1048,11 +1027,10 @@ AudioFrame::operator-=(const AudioFrame& rhs)
}
else
{
data_[i] = (WebRtc_Word16)wrapGuard;
data_[i] = (int16_t)wrapGuard;
}
}
energy_ = 0xffffffff;
volume_ = 0xffffffff;
return *this;
}

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

@@ -33,10 +33,13 @@ public:
static void DestroyFilePlayer(FilePlayer* player);
virtual WebRtc_Word32 Get10msAudioFromFile(
WebRtc_Word16* decodedDataBuffer,
WebRtc_UWord32& decodedDataLengthInSamples,
const WebRtc_UWord32 frequencyInHz) = 0;
// Read 10 ms of audio at |frequencyInHz| to |outBuffer|. |lengthInSamples|
// will be set to the number of samples read (not the number of samples per
// channel).
virtual int Get10msAudioFromFile(
int16_t* outBuffer,
int& lengthInSamples,
int frequencyInHz) = 0;
// Register callback for receiving file playing notifications.
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_Word16* outBuffer,
WebRtc_UWord32& lengthInSamples,
WebRtc_UWord32 frequencyInHz)
int16_t* outBuffer,
int& lengthInSamples,
int frequencyInHz)
{
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();
// FilePlayer functions.
virtual WebRtc_Word32 Get10msAudioFromFile(
WebRtc_Word16* decodedDataBuffer,
WebRtc_UWord32& decodedDataLengthInSamples,
const WebRtc_UWord32 frequencyInHz);
virtual int Get10msAudioFromFile(
int16_t* outBuffer,
int& lengthInSamples,
int frequencyInHz);
virtual WebRtc_Word32 RegisterModuleFileCallback(FileCallback* callback);
virtual WebRtc_Word32 StartPlayingFile(
const char* fileName,

2
src/video_engine/vie_file_player.h
View File

@@ -123,7 +123,7 @@ class ViEFilePlayer
ThreadWrapper* decode_thread_;
EventWrapper* decode_event_;
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
// 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;
}
// TODO(andrew): refactor Mix functions here and in transmit_mixer.cc to use
// a shared helper.
WebRtc_Word32
Channel::MixOrReplaceAudioWithFile(const int mixingFrequency)
{
scoped_array<WebRtc_Word16> fileBuffer(new WebRtc_Word16[640]);
WebRtc_UWord32 fileSamples(0);
int fileSamples(0);
{
CriticalSectionScoped cs(&_fileCritSect);
@@ -6186,10 +6188,10 @@ Channel::MixOrReplaceAudioWithFile(const int mixingFrequency)
// Currently file stream is always mono.
// TODO(xians): Change the code when FilePlayer supports real stereo.
Utility::MixWithSat(_audioFrame.data_,
static_cast<int>(_audioFrame.num_channels_),
_audioFrame.num_channels_,
fileBuffer.get(),
1,
static_cast<int>(fileSamples));
fileSamples);
}
else
{
@@ -6199,7 +6201,7 @@ Channel::MixOrReplaceAudioWithFile(const int mixingFrequency)
_audioFrame.UpdateFrame(_channelId,
-1,
fileBuffer.get(),
static_cast<WebRtc_UWord16>(fileSamples),
fileSamples,
mixingFrequency,
AudioFrame::kNormalSpeech,
AudioFrame::kVadUnknown,
@@ -6216,7 +6218,7 @@ Channel::MixAudioWithFile(AudioFrame& audioFrame,
assert(mixingFrequency <= 32000);
scoped_array<WebRtc_Word16> fileBuffer(new WebRtc_Word16[640]);
WebRtc_UWord32 fileSamples(0);
int fileSamples(0);
{
CriticalSectionScoped cs(&_fileCritSect);
@@ -6246,10 +6248,10 @@ Channel::MixAudioWithFile(AudioFrame& audioFrame,
// Currently file stream is always mono.
// TODO(xians): Change the code when FilePlayer supports real stereo.
Utility::MixWithSat(audioFrame.data_,
static_cast<int>(audioFrame.num_channels_),
audioFrame.num_channels_,
fileBuffer.get(),
1,
static_cast<int>(fileSamples));
fileSamples);
}
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]);
WebRtc_UWord32 fileSamples(0);
int fileSamples(0);
{
CriticalSectionScoped cs(&_critSect);
if (_filePlayerPtr == NULL)
@@ -1260,10 +1259,10 @@ WebRtc_Word32 TransmitMixer::MixOrReplaceAudioWithFile(
// Currently file stream is always mono.
// TODO(xians): Change the code when FilePlayer supports real stereo.
Utility::MixWithSat(_audioFrame.data_,
static_cast<int>(_audioFrame.num_channels_),
_audioFrame.num_channels_,
fileBuffer.get(),
1,
static_cast<int>(fileSamples));
fileSamples);
} else
{
// Replace ACM audio with file.
@@ -1272,12 +1271,11 @@ WebRtc_Word32 TransmitMixer::MixOrReplaceAudioWithFile(
_audioFrame.UpdateFrame(-1,
-1,
fileBuffer.get(),
static_cast<WebRtc_UWord16>(fileSamples),
fileSamples,
mixingFrequency,
AudioFrame::kNormalSpeech,
AudioFrame::kVadUnknown,
1);
}
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,
_audioFrame);
assert(nSamples == _audioFrame.samples_per_channel_);
assert(static_cast<int>(nSamples) == _audioFrame.samples_per_channel_);
assert(samplesPerSec ==
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();
}
// 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,
const char* fileNameOutUTF8)
{
@@ -677,7 +679,7 @@ int VoEFileImpl::ConvertPCMToWAV(const char* fileNameInUTF8,
// Run throught the file
AudioFrame audioFrame;
WebRtc_Word16 decodedData[160];
WebRtc_UWord32 decLength=0;
int decLength=0;
const WebRtc_UWord32 frequency = 16000;
while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
@@ -767,7 +769,7 @@ int VoEFileImpl::ConvertPCMToWAV(InStream* streamIn, OutStream* streamOut)
// Run throught the file
AudioFrame audioFrame;
WebRtc_Word16 decodedData[160];
WebRtc_UWord32 decLength=0;
int decLength=0;
const WebRtc_UWord32 frequency = 16000;
while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
@@ -855,7 +857,7 @@ int VoEFileImpl::ConvertWAVToPCM(const char* fileNameInUTF8,
// Run throught the file
AudioFrame audioFrame;
WebRtc_Word16 decodedData[160];
WebRtc_UWord32 decLength=0;
int decLength=0;
const WebRtc_UWord32 frequency = 16000;
while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
@@ -947,7 +949,7 @@ int VoEFileImpl::ConvertWAVToPCM(InStream* streamIn, OutStream* streamOut)
// Run throught the file
AudioFrame audioFrame;
WebRtc_Word16 decodedData[160];
WebRtc_UWord32 decLength=0;
int decLength=0;
const WebRtc_UWord32 frequency = 16000;
while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
@@ -1033,7 +1035,7 @@ int VoEFileImpl::ConvertPCMToCompressed(const char* fileNameInUTF8,
// Run throught the file
AudioFrame audioFrame;
WebRtc_Word16 decodedData[160];
WebRtc_UWord32 decLength=0;
int decLength=0;
const WebRtc_UWord32 frequency = 16000;
while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
@@ -1125,7 +1127,7 @@ int VoEFileImpl::ConvertPCMToCompressed(InStream* streamIn,
// Run throught the file
AudioFrame audioFrame;
WebRtc_Word16 decodedData[160];
WebRtc_UWord32 decLength=0;
int decLength=0;
const WebRtc_UWord32 frequency = 16000;
while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
@@ -1215,7 +1217,7 @@ int VoEFileImpl::ConvertCompressedToPCM(const char* fileNameInUTF8,
// Run throught the file
AudioFrame audioFrame;
WebRtc_Word16 decodedData[160];
WebRtc_UWord32 decLength=0;
int decLength=0;
const WebRtc_UWord32 frequency = 16000;
while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
@@ -1312,7 +1314,7 @@ int VoEFileImpl::ConvertCompressedToPCM(InStream* streamIn,
// Run throught the file
AudioFrame audioFrame;
WebRtc_Word16 decodedData[160];
WebRtc_UWord32 decLength=0;
int decLength=0;
const WebRtc_UWord32 frequency = 16000;
while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))