* #30063: audiobuffer: rename samples() to frames()

An audiobuffer of a given duration will return the same number of frames
regardless of how many channels it has.

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) {

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];
     }

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.
          */

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) {

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:

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) {

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_);

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});
 

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)

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)

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);

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_) {

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_++;

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]);
 }

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);