changeset 1318:f830a10bfbd6 3.0-integration

Start pruning other resampler logic than bqresample

--- a/base/Resampler.h	Mon Dec 05 16:39:03 2016 +0000
+++ /dev/null	Thu Jan 01 00:00:00 1970 +0000
@@ -1,51 +0,0 @@
-/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
-
-/*
-    Sonic Visualiser
-    An audio file viewer and annotation editor.
-    Centre for Digital Music, Queen Mary, University of London.
-    
-    This program is free software; you can redistribute it and/or
-    modify it under the terms of the GNU General Public License as
-    published by the Free Software Foundation; either version 2 of the
-    License, or (at your option) any later version.  See the file
-    COPYING included with this distribution for more information.
-*/
-
-/*
-   This is a modified version of a source file from the 
-   Rubber Band audio timestretcher library.
-   This file copyright 2007 Chris Cannam.
-*/
-
-#ifndef _RESAMPLER_H_
-#define _RESAMPLER_H_
-
-#include "BaseTypes.h"
-
-#include <sys/types.h>
-
-class Resampler
-{
-public:
-    enum Quality { Best, FastestTolerable, Fastest };
-
-    Resampler(Quality quality, int channels, sv_frame_t chunkSize = 0);
-    ~Resampler();
-
-    sv_frame_t resample(float **in, float **out,
-                        sv_frame_t incount, double ratio,
-                        bool final = false);
-
-    sv_frame_t resampleInterleaved(float *in, float *out,
-                                   sv_frame_t incount, double ratio,
-                                   bool final = false);
-
-    void reset();
-
-protected:
-    class D;
-    D *m_d;
-};
-
-#endif

--- a/base/ResamplerSV.cpp	Mon Dec 05 16:39:03 2016 +0000
+++ /dev/null	Thu Jan 01 00:00:00 1970 +0000
@@ -1,196 +0,0 @@
-/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */
-/*
-    Sonic Visualiser
-    An audio file viewer and annotation editor.
-    Centre for Digital Music, Queen Mary, University of London.
-    
-    This program is free software; you can redistribute it and/or
-    modify it under the terms of the GNU General Public License as
-    published by the Free Software Foundation; either version 2 of the
-    License, or (at your option) any later version.  See the file
-    COPYING included with this distribution for more information.
-*/
-
-/*
-   This is a modified version of a source file from the 
-   Rubber Band audio timestretcher library.
-   This file copyright 2007 Chris Cannam.
-*/
-
-#include "Resampler.h"
-
-#include <cstdlib>
-#include <cmath>
-
-#include <iostream>
-
-#include <samplerate.h>
-
-#include "Debug.h"
-
-class Resampler::D
-{
-public:
-    D(Quality quality, int channels, sv_frame_t chunkSize);
-    ~D();
-
-    sv_frame_t resample(float **in, float **out,
-                 sv_frame_t incount, double ratio,
-                 bool final);
-
-    sv_frame_t resampleInterleaved(float *in, float *out,
-                            sv_frame_t incount, double ratio,
-                            bool final);
-
-    void reset();
-
-protected:
-    SRC_STATE *m_src;
-    float *m_iin;
-    float *m_iout;
-    int m_channels;
-    sv_frame_t m_iinsize;
-    sv_frame_t m_ioutsize;
-};
-
-Resampler::D::D(Quality quality, int channels, sv_frame_t chunkSize) :
-    m_src(0),
-    m_iin(0),
-    m_iout(0),
-    m_channels(channels),
-    m_iinsize(0),
-    m_ioutsize(0)
-{
-    int err = 0;
-    m_src = src_new(quality == Best ? SRC_SINC_BEST_QUALITY :
-                    quality == Fastest ? SRC_LINEAR :
-                    SRC_SINC_FASTEST,
-                    channels, &err);
-
-    //!!! check err, throw
-
-    if (chunkSize > 0 && m_channels > 1) {
-        //!!! alignment?
-        m_iinsize = chunkSize * m_channels;
-        m_ioutsize = chunkSize * m_channels * 2;
-        m_iin = (float *)malloc(m_iinsize * sizeof(float));
-        m_iout = (float *)malloc(m_ioutsize * sizeof(float));
-    }
-}
-
-Resampler::D::~D()
-{
-    src_delete(m_src);
-    if (m_iinsize > 0) {
-        free(m_iin);
-    }
-    if (m_ioutsize > 0) {
-        free(m_iout);
-    }
-}
-
-sv_frame_t
-Resampler::D::resample(float **in, float **out,
-                       sv_frame_t incount, double ratio,
-                       bool final)
-{
-    if (m_channels == 1) {
-        return resampleInterleaved(*in, *out, incount, ratio, final);
-    }
-
-    sv_frame_t outcount = lrint(ceil(double(incount) * ratio));
-
-    if (incount * m_channels > m_iinsize) {
-        m_iinsize = incount * m_channels;
-        m_iin = (float *)realloc(m_iin, m_iinsize * sizeof(float));
-    }
-    if (outcount * m_channels > m_ioutsize) {
-        m_ioutsize = outcount * m_channels;
-        m_iout = (float *)realloc(m_iout, m_ioutsize * sizeof(float));
-    }
-    for (sv_frame_t i = 0; i < incount; ++i) {
-        for (int c = 0; c < m_channels; ++c) {
-            m_iin[i * m_channels + c] = in[c][i];
-        }
-    }
-    
-    sv_frame_t gen = resampleInterleaved(m_iin, m_iout, incount, ratio, final);
-
-    for (sv_frame_t i = 0; i < gen; ++i) {
-        for (int c = 0; c < m_channels; ++c) {
-            out[c][i] = m_iout[i * m_channels + c];
-        }
-    }
-
-    return gen;
-}
-
-sv_frame_t
-Resampler::D::resampleInterleaved(float *in, float *out,
-                                  sv_frame_t incount, double ratio,
-                                  bool final)
-{
-    SRC_DATA data;
-
-    sv_frame_t outcount = lrint(ceil(double(incount) * ratio));
-
-    data.data_in = in;
-    data.data_out = out;
-    data.input_frames = incount;
-    data.output_frames = outcount;
-    data.src_ratio = ratio;
-    data.end_of_input = (final ? 1 : 0);
-
-    int err = src_process(m_src, &data);
-
-    if (err) {
-        cerr << "Resampler: ERROR: src_process returned error: " <<
-            src_strerror(err) << endl;
-        return 0;
-    }
-
-    if (data.input_frames_used != incount) {
-        cerr << "Resampler: NOTE: input_frames_used == " << data.input_frames_used << " (while incount = " << incount << ")" << endl;
-    }
-
-    return data.output_frames_gen;
-}
-
-void
-Resampler::D::reset()
-{
-    src_reset(m_src);
-}
-
-Resampler::Resampler(Quality quality, int channels, sv_frame_t chunkSize)
-{
-    m_d = new D(quality, channels, chunkSize);
-}
-
-Resampler::~Resampler()
-{
-    delete m_d;
-}
-
-sv_frame_t 
-Resampler::resample(float **in, float **out,
-                    sv_frame_t incount, double ratio,
-                    bool final)
-{
-    return m_d->resample(in, out, incount, ratio, final);
-}
-
-sv_frame_t 
-Resampler::resampleInterleaved(float *in, float *out,
-                               sv_frame_t incount, double ratio,
-                               bool final)
-{
-    return m_d->resampleInterleaved(in, out, incount, ratio, final);
-}
-
-void
-Resampler::reset()
-{
-    m_d->reset();
-}
-

--- a/data/fileio/CodedAudioFileReader.cpp	Mon Dec 05 16:39:03 2016 +0000
+++ b/data/fileio/CodedAudioFileReader.cpp	Mon Dec 05 16:54:19 2016 +0000
@@ -20,9 +20,10 @@
 #include "base/Exceptions.h"
 #include "base/Profiler.h"
 #include "base/Serialiser.h"
-#include "base/Resampler.h"
 #include "base/StorageAdviser.h"
 
+#include <bqresample/Resampler.h>
+
 #include <stdint.h>
 #include <iostream>
 #include <QDir>
@@ -142,9 +143,10 @@
     }
     if (m_fileRate != m_sampleRate) {
         SVDEBUG << "CodedAudioFileReader: resampling " << m_fileRate << " -> " <<  m_sampleRate << endl;
-        m_resampler = new Resampler(Resampler::FastestTolerable,
-                                    m_channelCount,
-                                    m_cacheWriteBufferSize);
+        m_resampler = new breakfastquay::Resampler
+            (breakfastquay::Resampler::FastestTolerable,
+             m_channelCount,
+             m_cacheWriteBufferSize);
         double ratio = m_sampleRate / m_fileRate;
         m_resampleBuffer = new float
             [lrint(ceil(double(m_cacheWriteBufferSize) * m_channelCount * ratio + 1))];

--- a/data/fileio/CodedAudioFileReader.h	Mon Dec 05 16:39:03 2016 +0000
+++ b/data/fileio/CodedAudioFileReader.h	Mon Dec 05 16:54:19 2016 +0000
@@ -24,7 +24,10 @@
 
 class WavFileReader;
 class Serialiser;
-class Resampler;
+
+namespace breakfastquay {
+    class Resampler;
+}
 
 class CodedAudioFileReader : public AudioFileReader
 {
@@ -105,7 +108,7 @@
     sv_frame_t m_cacheWriteBufferIndex; // samples
     sv_frame_t m_cacheWriteBufferSize; // frames
 
-    Resampler *m_resampler;
+    breakfastquay::Resampler *m_resampler;
     float *m_resampleBuffer;
     sv_frame_t m_fileFrameCount;
 

--- a/files.pri	Mon Dec 05 16:39:03 2016 +0000
+++ b/files.pri	Mon Dec 05 16:54:19 2016 +0000
@@ -23,7 +23,6 @@
            base/RangeMapper.h \
            base/RealTime.h \
            base/RecentFiles.h \
-           base/Resampler.h \
            base/ResourceFinder.h \
            base/RingBuffer.h \
            base/Scavenger.h \
@@ -159,7 +158,6 @@
            base/RangeMapper.cpp \
            base/RealTimeSV.cpp \
            base/RecentFiles.cpp \
-           base/ResamplerSV.cpp \
            base/ResourceFinder.cpp \
            base/Selection.cpp \
            base/Serialiser.cpp \