UltraCoder/speex-webrtc-test
Archived
1
0
Fork 0
Code Issues Pull requests Projects Releases Packages Wiki Activity

Calculate frame size in AudioEffect

Browse source
This commit is contained in:
Kirill Kirilenko 2020-11-11 17:32:27 +03:00
parent 60b927a412
commit ff3d7a969f
7 changed files with 32 additions and 25 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

8
src/AudioEffect.cpp
View file

@ -2,16 +2,14 @@
namespace SpeexWebRTCTest {
AudioEffect::AudioEffect(unsigned int frameSize,
const QAudioFormat& mainFormat,
const QAudioFormat& auxFormat)
: frameSize_(frameSize), mainFormat_(mainFormat), auxFormat_(auxFormat)
AudioEffect::AudioEffect(const QAudioFormat& mainFormat, const QAudioFormat& auxFormat)
: mainFormat_(mainFormat), auxFormat_(auxFormat)
{
}
unsigned int AudioEffect::getFrameSize() const
{
return frameSize_;
return mainFormat_.sampleRate() * requiredFrameSizeMs() / 1000;
}
const QAudioFormat& AudioEffect::getMainFormat() const

5
src/AudioEffect.h
View file

@ -12,7 +12,7 @@ class AudioEffect : public QObject
{
Q_OBJECT
public:
AudioEffect(unsigned int frameSize, const QAudioFormat& mainFormat, const QAudioFormat& auxFormat);
AudioEffect(const QAudioFormat& mainFormat, const QAudioFormat& auxFormat);
virtual void processFrame(QAudioBuffer& mainBuffer, const QAudioBuffer& auxBuffer) = 0;
@ -30,11 +30,12 @@ protected:
voiceActive_ = active;
}
virtual unsigned int requiredFrameSizeMs() const = 0;
signals:
void voiceActivityChanged(bool voice);
private:
const unsigned int frameSize_;
const QAudioFormat mainFormat_;
const QAudioFormat auxFormat_;

12
src/AudioProcessor.cpp
View file

@ -196,15 +196,11 @@ Backend AudioProcessor::getCurrentBackend() const
void AudioProcessor::switchBackend(Backend backend)
{
if (backend == Backend::Speex)
{
bufferSize_ = format_.sampleRate() * 0.025; // 25ms
dsp_.reset(new SpeexDSP(bufferSize_, format_, monitorFormat_));
}
dsp_.reset(new SpeexDSP(format_, monitorFormat_));
else
{
bufferSize_ = format_.sampleRate() * 0.01; // 10ms
dsp_.reset(new WebRTCDSP(bufferSize_, format_, monitorFormat_));
}
dsp_.reset(new WebRTCDSP(format_, monitorFormat_));
bufferSize_ = dsp_->getFrameSize();
connect(dsp_.get(), &AudioEffect::voiceActivityChanged, this,
&AudioProcessor::voiceActivityChanged);

9
src/SpeexDSP.cpp
View file

@ -13,8 +13,8 @@ int on = 1;
int off = 0;
} // namespace
SpeexDSP::SpeexDSP(unsigned int frameSize, const QAudioFormat& mainFormat, const QAudioFormat& auxFormat)
: AudioEffect(frameSize, mainFormat, auxFormat)
SpeexDSP::SpeexDSP(const QAudioFormat& mainFormat, const QAudioFormat& auxFormat)
: AudioEffect(mainFormat, auxFormat)
{
preprocess_ = speex_preprocess_state_init(getFrameSize(), getMainFormat().sampleRate());
echo_ = speex_echo_state_init_mc(getFrameSize(), getFrameSize() * 10,
@ -78,4 +78,9 @@ void SpeexDSP::setParameter(const QString& param, QVariant value)
throw std::invalid_argument("Invalid param");
}
unsigned int SpeexDSP::requiredFrameSizeMs() const
{
return 25;
}
} // namespace SpeexWebRTCTest

6
src/SpeexDSP.h
View file

@ -10,17 +10,19 @@ typedef struct SpeexEchoState_ SpeexEchoState;
namespace SpeexWebRTCTest {
class SpeexDSP : public AudioEffect
class SpeexDSP final : public AudioEffect
{
Q_OBJECT
public:
SpeexDSP(unsigned int frameSize, const QAudioFormat& mainFormat, const QAudioFormat& auxFormat);
SpeexDSP(const QAudioFormat& mainFormat, const QAudioFormat& auxFormat);
~SpeexDSP() override;
void processFrame(QAudioBuffer& mainBuffer, const QAudioBuffer& auxBuffer) override;
void setParameter(const QString& param, QVariant value) override;
private:
unsigned int requiredFrameSizeMs() const override;
SpeexPreprocessState* preprocess_ = nullptr;
SpeexEchoState* echo_ = nullptr;

11
src/WebRTCDSP.cpp
View file

@ -37,10 +37,8 @@ void convert(const webrtc::AudioFrame& from, QAudioBuffer& to)
} // namespace
WebRTCDSP::WebRTCDSP(unsigned int frameSize,
const QAudioFormat& mainFormat,
const QAudioFormat& auxFormat)
: AudioEffect(frameSize, mainFormat, auxFormat)
WebRTCDSP::WebRTCDSP(const QAudioFormat& mainFormat, const QAudioFormat& auxFormat)
: AudioEffect(mainFormat, auxFormat)
{
apm_ = webrtc::AudioProcessing::Create();
if (!apm_)
@ -181,4 +179,9 @@ void WebRTCDSP::setParameter(const QString& param, QVariant value)
throw std::invalid_argument("Invalid param");
}
unsigned int WebRTCDSP::requiredFrameSizeMs() const
{
return 10;
}
} // namespace SpeexWebRTCTest

6
src/WebRTCDSP.h
View file

@ -9,17 +9,19 @@ class AudioProcessing;
namespace SpeexWebRTCTest {
class WebRTCDSP : public AudioEffect
class WebRTCDSP final : public AudioEffect
{
Q_OBJECT
public:
WebRTCDSP(unsigned int frameSize, const QAudioFormat& mainFormat, const QAudioFormat& auxFormat);
WebRTCDSP(const QAudioFormat& mainFormat, const QAudioFormat& auxFormat);
~WebRTCDSP() override;
void processFrame(QAudioBuffer& mainBuffer, const QAudioBuffer& auxBuffer) override;
void setParameter(const QString& param, QVariant value) override;
private:
unsigned int requiredFrameSizeMs() const override;
webrtc::AudioProcessing* apm_;
};