diff --git a/daemon/src/audio/alsa/alsalayer.cpp b/daemon/src/audio/alsa/alsalayer.cpp
index 473e6b69e6db186b51c4b43ba7636a3c834c6672..d91fc5eec3d9bfdefe5fcb2aee712d637c47133f 100644
--- a/daemon/src/audio/alsa/alsalayer.cpp
+++ b/daemon/src/audio/alsa/alsalayer.cpp
@@ -706,7 +706,7 @@ void AlsaLayer::capture()
AudioBuffer in(toGetSamples, 1, sampleRate_);
// TODO: handle ALSA multichannel capture
- const int toGetBytes = in.samples() * sizeof(SFLAudioSample);
+ const int toGetBytes = in.frames() * sizeof(SFLAudioSample);
SFLAudioSample * const in_ptr = in.getChannel(0)->data();
if (read(in_ptr, toGetBytes) != toGetBytes) {
diff --git a/daemon/src/audio/audiobuffer.cpp b/daemon/src/audio/audiobuffer.cpp
index 32b656b591b0cf054f4de328b74d2e390672fdc4..d934758e4fb7af2aef520c68407448cd1b510421 100644
--- a/daemon/src/audio/audiobuffer.cpp
+++ b/daemon/src/audio/audiobuffer.cpp
@@ -48,7 +48,7 @@ AudioBuffer::AudioBuffer(const SFLAudioSample* in, size_t sample_num, unsigned c
AudioBuffer::AudioBuffer(const AudioBuffer& other, bool copy_content /* = false */)
: sampleRate_(other.sampleRate_),
samples_(copy_content ? other.samples_ :
- std::vector<std::vector<SFLAudioSample> >(other.samples_.size(), std::vector<SFLAudioSample>(other.samples())))
+ std::vector<std::vector<SFLAudioSample> >(other.samples_.size(), std::vector<SFLAudioSample>(other.frames())))
{}
int AudioBuffer::getSampleRate() const
@@ -68,7 +68,7 @@ void AudioBuffer::setChannelNum(unsigned n, bool copy_content /* = false */)
if (n == samples_.size())
return;
- const size_t start_size = samples();
+ const size_t start_size = frames();
if (copy_content and not samples_.empty())
samples_.resize(n, samples_[0]);
@@ -78,7 +78,7 @@ void AudioBuffer::setChannelNum(unsigned n, bool copy_content /* = false */)
void AudioBuffer::resize(size_t sample_num)
{
- if (samples() == sample_num)
+ if (frames() == sample_num)
return;
for (unsigned i = 0; i < samples_.size(); i++)
@@ -114,20 +114,20 @@ void AudioBuffer::applyGain(double gain)
size_t AudioBuffer::interleave(SFLAudioSample* out) const
{
- for (unsigned i = 0; i < samples(); i++)
+ for (unsigned i = 0; i < frames(); i++)
for (unsigned j = 0; j < samples_.size(); j++)
*out++ = samples_[j][i];
- return samples() * samples_.size();
+ return frames() * samples_.size();
}
size_t AudioBuffer::interleaveFloat(float* out) const
{
- for (unsigned i = 0; i < samples(); i++)
+ for (unsigned i = 0; i < frames(); i++)
for (unsigned j = 0; j < samples_.size(); j++)
*out++ = (float) samples_[j][i] * .000030517578125f;
- return samples() * samples_.size();
+ return frames() * samples_.size();
}
void AudioBuffer::deinterleave(const SFLAudioSample* in, size_t sample_num, unsigned channel_num)
@@ -139,7 +139,7 @@ void AudioBuffer::deinterleave(const SFLAudioSample* in, size_t sample_num, unsi
setChannelNum(channel_num);
resize(sample_num);
- for (unsigned i = 0; i < samples(); i++)
+ for (unsigned i = 0; i < frames(); i++)
for (unsigned j = 0; j < samples_.size(); j++)
samples_[j][i] = *in++;
}
@@ -147,7 +147,7 @@ void AudioBuffer::deinterleave(const SFLAudioSample* in, size_t sample_num, unsi
size_t AudioBuffer::mix(const AudioBuffer& other, bool up /* = true */)
{
const bool upmix = up && (other.samples_.size() < samples_.size());
- const size_t samp_num = std::min(samples(), other.samples());
+ const size_t samp_num = std::min(frames(), other.frames());
const unsigned chan_num = upmix ? samples_.size() : std::min(samples_.size(), other.samples_.size());
for (unsigned i = 0; i < chan_num; i++) {
@@ -163,16 +163,16 @@ size_t AudioBuffer::mix(const AudioBuffer& other, bool up /* = true */)
size_t AudioBuffer::copy(AudioBuffer& in, int sample_num /* = -1 */, size_t pos_in /* = 0 */, size_t pos_out /* = 0 */, bool up /* = true */)
{
if (sample_num == -1)
- sample_num = in.samples();
+ sample_num = in.frames();
- int to_copy = std::min((int)in.samples() - (int)pos_in, sample_num);
+ int to_copy = std::min((int)in.frames() - (int)pos_in, sample_num);
if (to_copy <= 0) return 0;
const bool upmix = up && (in.samples_.size() < samples_.size());
const size_t chan_num = upmix ? samples_.size() : std::min(in.samples_.size(), samples_.size());
- if ((pos_out + to_copy) > samples())
+ if ((pos_out + to_copy) > frames())
resize(pos_out + to_copy);
sampleRate_ = in.sampleRate_;
@@ -190,7 +190,7 @@ size_t AudioBuffer::copy(SFLAudioSample* in, size_t sample_num, size_t pos_out /
{
if (in == NULL) return 0;
- if ((pos_out + sample_num) > samples())
+ if ((pos_out + sample_num) > frames())
resize(pos_out + sample_num);
const size_t chan_num = samples_.size();
@@ -205,7 +205,7 @@ size_t AudioBuffer::copy(SFLAudioSample* in, size_t sample_num, size_t pos_out /
std::ostream& operator<<(std::ostream& os, const AudioBuffer& buf)
{
- for (unsigned i = 0; i < buf.samples(); i++) {
+ for (unsigned i = 0; i < buf.frames(); i++) {
for (unsigned j = 0; j < buf.samples_.size(); j++)
os << buf.samples_[j][i];
}
diff --git a/daemon/src/audio/audiobuffer.h b/daemon/src/audio/audiobuffer.h
index 878c8f284aa592a3b79edaf3b61457e12e55faa5..1c176b235e2d6312d92a13278f5d3c40614ef76d 100644
--- a/daemon/src/audio/audiobuffer.h
+++ b/daemon/src/audio/audiobuffer.h
@@ -61,7 +61,7 @@ class AudioBuffer {
}
inline size_t size() {
- return samples() * channels() * sizeof(SFLAudioSample);
+ return frames() * channels() * sizeof(SFLAudioSample);
}
/**
@@ -91,9 +91,9 @@ class AudioBuffer {
void setChannelNum(unsigned n, bool copy_first = false);
/**
- * Returns the number of (multichannel) samples in this buffer.
+ * Returns the number of (multichannel) frames in this buffer.
*/
- inline size_t samples() const {
+ inline size_t frames() const {
if (not samples_.empty())
return samples_[0].size();
else
@@ -101,10 +101,10 @@ class AudioBuffer {
}
/**
- * Return the total number of single samples in the buffer (same as samples()*channels()).
+ * Return the total number of single samples in the buffer
*/
inline size_t capacity() const {
- return samples() * channels();
+ return frames() * channels();
}
/**
@@ -145,7 +145,7 @@ class AudioBuffer {
/**
* Write interleaved multichannel data to the out buffer, while samples are converted to float.
- * The buffer must be at least of size getChannelNum()*samples()*sizeof(float).
+ * The buffer must be at least of size getChannelNum()*frames()*sizeof(float).
*
* @returns Number of samples writen.
*/
diff --git a/daemon/src/audio/audioloop.cpp b/daemon/src/audio/audioloop.cpp
index 017725ad427c3620a6f1803e8cdd77031e7fe1e0..ff73b2ec6bef5f8298c274ca4566a5cc341eb55c 100644
--- a/daemon/src/audio/audioloop.cpp
+++ b/daemon/src/audio/audioloop.cpp
@@ -58,7 +58,7 @@ AudioLoop::~AudioLoop()
void
AudioLoop::seek(double relative_position)
{
- pos_ = static_cast<double>(buffer_->samples() * relative_position * 0.01);
+ pos_ = static_cast<double>(buffer_->frames() * relative_position * 0.01);
}
void
@@ -69,9 +69,9 @@ AudioLoop::getNext(AudioBuffer& output, double gain)
return;
}
- const size_t buf_samples = buffer_->samples();
+ const size_t buf_samples = buffer_->frames();
size_t pos = pos_;
- size_t total_samples = output.samples();
+ size_t total_samples = output.frames();
size_t output_pos = 0;
if (buf_samples == 0) {
diff --git a/daemon/src/audio/audioloop.h b/daemon/src/audio/audioloop.h
index 9d58060dc3d8c99429686607ca1c545f4849e051..72073b8067ed890e26269db7cb17d52655f2168e 100644
--- a/daemon/src/audio/audioloop.h
+++ b/daemon/src/audio/audioloop.h
@@ -72,7 +72,7 @@ class AudioLoop {
* @return unsigned int The size
*/
size_t getSize() {
- return buffer_->samples();
+ return buffer_->frames();
}
protected:
diff --git a/daemon/src/audio/audiorecord.cpp b/daemon/src/audio/audiorecord.cpp
index d2d22b7642bc9444b3e7258d09812e03a265cc70..8ceaeb9ea81a5faad5fa6595102fd61750aa07af 100644
--- a/daemon/src/audio/audiorecord.cpp
+++ b/daemon/src/audio/audiorecord.cpp
@@ -231,7 +231,7 @@ void AudioRecord::recData(AudioBuffer& buffer)
return;
}
- const int nSamples = buffer.samples();
+ const int nSamples = buffer.frames();
// FIXME: mono only
if (fileHandle_->write(buffer.getChannel(0)->data(), nSamples) != nSamples) {
diff --git a/daemon/src/audio/audiortp/audio_rtp_record_handler.cpp b/daemon/src/audio/audiortp/audio_rtp_record_handler.cpp
index 563b3a063ce20da16afdc0b0ed5c5a80b9094257..33e8bd4d0d12039a3ec7101e768c3b800860ef8f 100644
--- a/daemon/src/audio/audiortp/audio_rtp_record_handler.cpp
+++ b/daemon/src/audio/audiortp/audio_rtp_record_handler.cpp
@@ -375,7 +375,7 @@ void AudioRtpRecordHandler::processDataDecode(unsigned char *spkrData, size_t si
}
}
- size = std::min(size, audioRtpRecord_.decData_.samples());
+ size = std::min(size, audioRtpRecord_.decData_.frames());
{
ScopedLock lock(audioRtpRecord_.audioCodecMutex_);
diff --git a/daemon/src/audio/dcblocker.cpp b/daemon/src/audio/dcblocker.cpp
index bb238e532f0a0cd8807dce71f3d143b5954eab66..73bef43c9c9b6eecb46d9c10acb1328f8e7a81f2 100644
--- a/daemon/src/audio/dcblocker.cpp
+++ b/daemon/src/audio/dcblocker.cpp
@@ -62,7 +62,7 @@ void DcBlocker::process(SFLAudioSample *out, SFLAudioSample *in, int samples)
void DcBlocker::process(AudioBuffer& buf)
{
const size_t chans = buf.channels();
- const size_t samples = buf.samples();
+ const size_t samples = buf.frames();
if(chans > states.size())
states.resize(buf.channels(), (struct StreamState){0, 0, 0, 0});
diff --git a/daemon/src/audio/gaincontrol.cpp b/daemon/src/audio/gaincontrol.cpp
index db9cb90f34afa01c10c0bf27e3355a00e4f66b21..a754c0cd6abea9b15ea0eb6acdcecda9ef821184 100644
--- a/daemon/src/audio/gaincontrol.cpp
+++ b/daemon/src/audio/gaincontrol.cpp
@@ -58,7 +58,7 @@ GainControl::GainControl(double sr, double target) : averager_(sr, SFL_GAIN_ATTA
void GainControl::process(AudioBuffer& buf)
{
- process(buf.getChannel(0)->data(), buf.samples());
+ process(buf.getChannel(0)->data(), buf.frames());
}
void GainControl::process(SFLAudioSample *buf, int samples)
diff --git a/daemon/src/audio/pulseaudio/pulselayer.cpp b/daemon/src/audio/pulseaudio/pulselayer.cpp
index e05f9a13c2cb6b1e8a4b9d8d0c797923c2ab7c69..d54c2ef513c3f74eab64802b754b5d40bf054f21 100644
--- a/daemon/src/audio/pulseaudio/pulselayer.cpp
+++ b/daemon/src/audio/pulseaudio/pulselayer.cpp
@@ -544,7 +544,7 @@ void PulseLayer::readFromMic()
Manager::instance().getMainBuffer().putData(*out, MainBuffer::DEFAULT_ID);
#ifdef RECTODISK
- outfileResampled.write((const char *)out->getChannel(0), out->samples() * sizeof(SFLAudioSample));
+ outfileResampled.write((const char *)out->getChannel(0), out->frames() * sizeof(SFLAudioSample));
#endif
if (pa_stream_drop(record_->pulseStream()) < 0)
diff --git a/daemon/src/audio/ringbuffer.cpp b/daemon/src/audio/ringbuffer.cpp
index ba8b74b55e53cd1225f38f92b31d60a1ac3bf5e6..9ba79ef50c94f78e595a33a1e9d0616a95151900 100644
--- a/daemon/src/audio/ringbuffer.cpp
+++ b/daemon/src/audio/ringbuffer.cpp
@@ -73,21 +73,21 @@ RingBuffer::flushAll()
size_t
RingBuffer::putLength() const
{
- const size_t buffer_size = buffer_.samples();
+ const size_t buffer_size = buffer_.frames();
const size_t startPos = (not readpointers_.empty()) ? getSmallestReadPointer() : 0;
return (endPos_ + buffer_size - startPos) % buffer_size;
}
size_t RingBuffer::getLength(const std::string &call_id) const
{
- const size_t buffer_size = buffer_.samples();
+ const size_t buffer_size = buffer_.frames();
return (endPos_ + buffer_size - getReadPointer(call_id)) % buffer_size;
}
void
RingBuffer::debug()
{
- DEBUG("Start=%d; End=%d; BufferSize=%d", getSmallestReadPointer(), endPos_, buffer_.samples());
+ DEBUG("Start=%d; End=%d; BufferSize=%d", getSmallestReadPointer(), endPos_, buffer_.frames());
}
size_t RingBuffer::getReadPointer(const std::string &call_id) const
@@ -105,7 +105,7 @@ RingBuffer::getSmallestReadPointer() const
if (hasNoReadPointers())
return 0;
- size_t smallest = buffer_.samples();
+ size_t smallest = buffer_.frames();
ReadPointer::const_iterator iter;
@@ -166,8 +166,8 @@ bool RingBuffer::hasNoReadPointers() const
void RingBuffer::put(AudioBuffer& buf)
{
const size_t len = putLength();
- const size_t sample_num = buf.samples();
- const size_t buffer_size = buffer_.samples();
+ const size_t sample_num = buf.frames();
+ const size_t buffer_size = buffer_.frames();
size_t toCopy = sample_num;
// Add more channels if the input buffer holds more channels than the ring.
@@ -216,8 +216,8 @@ size_t RingBuffer::get(AudioBuffer& buf, const std::string &call_id)
return 0;
const size_t len = getLength(call_id);
- const size_t sample_num = buf.samples();
- const size_t buffer_size = buffer_.samples();
+ const size_t sample_num = buf.frames();
+ const size_t buffer_size = buffer_.frames();
size_t toCopy = std::min(sample_num, len);
const size_t copied = toCopy;
@@ -250,7 +250,7 @@ RingBuffer::discard(size_t toDiscard, const std::string &call_id)
if (toDiscard > len)
toDiscard = len;
- size_t buffer_size = buffer_.samples();
+ size_t buffer_size = buffer_.frames();
size_t startPos = (getReadPointer(call_id) + toDiscard) % buffer_size;
storeReadPointer(startPos, call_id);
diff --git a/daemon/src/audio/samplerateconverter.cpp b/daemon/src/audio/samplerateconverter.cpp
index acbf4d1c3aa37fda42aabc3823deb82d4f12c060..b3c57755977d3b65cfe8e7b3b065a5c681803046 100644
--- a/daemon/src/audio/samplerateconverter.cpp
+++ b/daemon/src/audio/samplerateconverter.cpp
@@ -69,7 +69,7 @@ void SamplerateConverter::resample(const AudioBuffer &dataIn, AudioBuffer &dataO
if (sampleFactor == 1.0)
return;
- const size_t nbFrames = dataIn.samples();
+ const size_t nbFrames = dataIn.frames();
const size_t nbChans = dataIn.channels();
if (nbChans != channels_) {
diff --git a/daemon/src/audio/sound/audiofile.cpp b/daemon/src/audio/sound/audiofile.cpp
index bc655ecb792797f03418eac19e7fbeeca69637f0..adca75d394d4f4b569771a2ad01491a44cdd93b5 100644
--- a/daemon/src/audio/sound/audiofile.cpp
+++ b/daemon/src/audio/sound/audiofile.cpp
@@ -58,7 +58,7 @@ AudioFile::onBufferFinish()
if ((updatePlaybackScale_ % 5) == 0) {
CallManager *cm = Manager::instance().getClient()->getCallManager();
- cm->updatePlaybackScale(filepath_, pos_ / divisor, buffer_->samples() / divisor);
+ cm->updatePlaybackScale(filepath_, pos_ / divisor, buffer_->frames() / divisor);
}
updatePlaybackScale_++;
diff --git a/daemon/test/audiobuffertest.cpp b/daemon/test/audiobuffertest.cpp
index 7a1aa9052da8ebb5c783042372545dd2f9000871..4ccc48d6f84c82a237b3dd71e4788834c404f5e0 100644
--- a/daemon/test/audiobuffertest.cpp
+++ b/daemon/test/audiobuffertest.cpp
@@ -43,24 +43,24 @@ void AudioBufferTest::testAudioBufferConstructors()
SFLAudioSample test_samples2[] = {10, 11, 12, 13, 14, 15, 16, 17};
AudioBuffer empty_buf(0);
- CPPUNIT_ASSERT(empty_buf.samples() == 0);
+ CPPUNIT_ASSERT(empty_buf.frames() == 0);
CPPUNIT_ASSERT(empty_buf.channels() == 1);
CPPUNIT_ASSERT(empty_buf.getChannel(0)->size() == 0);
AudioBuffer test_buf1(8, 2);
- CPPUNIT_ASSERT(test_buf1.samples() == 8);
+ CPPUNIT_ASSERT(test_buf1.frames() == 8);
CPPUNIT_ASSERT(test_buf1.channels() == 2);
CPPUNIT_ASSERT(test_buf1.getChannel(0)->size() == 8);
CPPUNIT_ASSERT(test_buf1.getChannel(1)->size() == 8);
CPPUNIT_ASSERT(test_buf1.getChannel(2) == NULL);
AudioBuffer test_buf2(test_samples1, 0, 0);
- CPPUNIT_ASSERT(test_buf2.samples() == 0);
+ CPPUNIT_ASSERT(test_buf2.frames() == 0);
CPPUNIT_ASSERT(test_buf2.channels() == 1);
CPPUNIT_ASSERT(test_buf2.getChannel(0)->size() == 0);
AudioBuffer test_buf3(test_samples2, 4, 2);
- CPPUNIT_ASSERT(test_buf3.samples() == 4);
+ CPPUNIT_ASSERT(test_buf3.frames() == 4);
CPPUNIT_ASSERT(test_buf3.channels() == 2);
CPPUNIT_ASSERT(test_buf3.getChannel(0)->size() == 4);
}
@@ -98,7 +98,7 @@ void AudioBufferTest::testAudioBufferMix()
test_buf1.mix(test_buf2);
CPPUNIT_ASSERT(test_buf1.channels() == 2);
- CPPUNIT_ASSERT(test_buf1.samples() == 4);
+ CPPUNIT_ASSERT(test_buf1.frames() == 4);
CPPUNIT_ASSERT((*test_buf1.getChannel(0))[0] == test_samples1[0]+test_samples2[0]);
CPPUNIT_ASSERT((*test_buf1.getChannel(1))[0] == test_samples1[0]+test_samples2[1]);
}
diff --git a/daemon/test/resamplertest.cpp b/daemon/test/resamplertest.cpp
index 4f7e480cf234021dbe0da1d8ea326c81ef4cc2fc..5fc1ab16555e7664e8827d45aedef99c7461eb0c 100644
--- a/daemon/test/resamplertest.cpp
+++ b/daemon/test/resamplertest.cpp
@@ -215,7 +215,7 @@ void ResamplerTest::performUpsampling(SamplerateConverter &converter)
AudioBuffer tmpInputBuffer(TMP_LOWSMPLR_BUFFER_LENGTH, 1, 8000);
AudioBuffer tmpOutputBuffer(TMP_HIGHSMPLR_BUFFER_LENGTH, 1, 16000);
- for (size_t i = 0, j = 0; i < (inputBuffer.samples() / 2); i += tmpInputBuffer.samples(), j += tmpOutputBuffer.samples()) {
+ for (size_t i = 0, j = 0; i < (inputBuffer.frames() / 2); i += tmpInputBuffer.frames(), j += tmpOutputBuffer.frames()) {
tmpInputBuffer.copy(inputBuffer, i);
converter.resample(tmpInputBuffer, tmpOutputBuffer);
outputBuffer.copy(tmpOutputBuffer, -1, 0, j);
@@ -227,7 +227,7 @@ void ResamplerTest::performDownsampling(SamplerateConverter &converter)
AudioBuffer tmpInputBuffer(TMP_HIGHSMPLR_BUFFER_LENGTH, 1, 16000);
AudioBuffer tmpOutputBuffer(TMP_LOWSMPLR_BUFFER_LENGTH, 1, 8000);
- for (size_t i = 0, j = 0; i < inputBuffer.samples(); i += tmpInputBuffer.samples(), j += tmpOutputBuffer.samples()) {
+ for (size_t i = 0, j = 0; i < inputBuffer.frames(); i += tmpInputBuffer.frames(), j += tmpOutputBuffer.frames()) {
tmpInputBuffer.copy(inputBuffer, i);
converter.resample(tmpInputBuffer, tmpOutputBuffer);
outputBuffer.copy(tmpOutputBuffer, -1, 0, j);