From d1b2e076776ca94d28520dfe1c72e375745f62ba Mon Sep 17 00:00:00 2001
From: Edouard DUPIN <yui.heero@gmail.com>
Date: Tue, 21 Apr 2015 21:57:32 +0200
Subject: [PATCH] [DEV] create new API i,terface for speex ==> more stable, ABI
 stable

---
 audio/algo/speex/Resampler.cpp | 194 +++++++++++++++++++--------------
 audio/algo/speex/Resampler.h   |  18 ++-
 lutin_audio_algo_speex.py      |   2 +-
 test/main.cpp                  |  12 +-
 4 files changed, 126 insertions(+), 100 deletions(-)

diff --git a/audio/algo/speex/Resampler.cpp b/audio/algo/speex/Resampler.cpp
index bad4830..e3af626 100644
--- a/audio/algo/speex/Resampler.cpp
+++ b/audio/algo/speex/Resampler.cpp
@@ -8,44 +8,125 @@
 #include <audio/algo/speex/Resampler.h>
 #include <audio/algo/speex/debug.h>
 #include <cmath>
+#ifdef HAVE_SPEEX_DSP
+	#include <speex/speex_resampler.h>
+#endif
+
+#undef __class__
+#define __class__ "algo::speex::ResamplerPrivate"
+namespace audio {
+	namespace algo {
+		namespace speex {
+			class ResamplerPrivate {
+				private:
+					#ifdef HAVE_SPEEX_DSP
+						SpeexResamplerState* m_speexResampler;
+					#endif
+					enum audio::format m_format;
+				public:
+					ResamplerPrivate(int8_t _nbChannel, float _inputSampleRate, float _outputSampleRate, int8_t _quality, enum audio::format _format) :
+					  #ifdef HAVE_SPEEX_DSP
+					    m_speexResampler(nullptr),
+					  #endif
+					  m_format(_format) {
+						#ifdef HAVE_SPEEX_DSP
+							if (m_speexResampler != nullptr) {
+								speex_resampler_destroy(m_speexResampler);
+								m_speexResampler = nullptr;
+							}
+							AA_SPEEX_DEBUG("Create resampler for : " << _inputSampleRate << " to " << _outputSampleRate);
+							int err = 0;
+							m_speexResampler = speex_resampler_init(_nbChannel,
+							                                        _inputSampleRate,
+							                                        _outputSampleRate,
+							                                        _quality, &err);
+						#else
+							AA_SPEEX_WARNING("SPEEX DSP lib not accessible ==> can not resample");
+						#endif
+					}
+				
+				~ResamplerPrivate() {
+					#ifdef HAVE_SPEEX_DSP
+						if (m_speexResampler != nullptr) {
+							speex_resampler_destroy(m_speexResampler);
+							m_speexResampler = nullptr;
+						}
+					#endif
+				}
+				
+				void process(void* _output, size_t& _nbChunkOut, const void* _input, size_t _nbChunk) {
+					#ifdef HAVE_SPEEX_DSP
+						switch (m_format) {
+							case audio::format_int16:
+								{
+									uint32_t nbChunkInput = _nbChunk;
+									uint32_t nbChunkOutput = _nbChunkOut;
+									int ret = speex_resampler_process_interleaved_int(m_speexResampler,
+									                                                  reinterpret_cast<const int16_t*>(_input),
+									                                                  &nbChunkInput,
+									                                                  reinterpret_cast<int16_t*>(_output),
+									                                                  &nbChunkOutput);
+									// Check all input and output ...
+									if (nbChunkInput != _nbChunk) {
+										AA_SPEEX_ERROR("inputSize (not all read ...) proceed=" << nbChunkInput << " requested=" << _nbChunk);
+										// TODO : manage this case ...
+									}
+									if (nbChunkOutput == _nbChunkOut) {
+										AA_SPEEX_ERROR("Might have not enought data in output... output size=" << _nbChunkOut);
+										// TODO : manage this case ...
+									}
+									_nbChunkOut = nbChunkOutput;
+								}
+								break;
+							case audio::format_float:
+								{
+									uint32_t nbChunkInput = _nbChunk;
+									uint32_t nbChunkOutput = _nbChunkOut;
+									int ret = speex_resampler_process_interleaved_float(m_speexResampler,
+									                                                    reinterpret_cast<const float*>(_input),
+									                                                    &nbChunkInput,
+									                                                    reinterpret_cast<float*>(_output),
+									                                                    &nbChunkOutput);
+									// Check all input and output ...
+									if (nbChunkInput != _nbChunk) {
+										AA_SPEEX_ERROR("inputSize (not all read ...) proceed=" << nbChunkInput << " requested=" << _nbChunk);
+										// TODO : manage this case ...
+									}
+									if (nbChunkOutput == _nbChunkOut) {
+										AA_SPEEX_ERROR("Might have not enought data in output... output size=" << _nbChunkOut);
+										// TODO : manage this case ...
+									}
+									_nbChunkOut = nbChunkOutput;
+								}
+								break;
+							default:
+								AA_SPEEX_ERROR("Can not Limit with unsupported format : " << m_format);
+								break;
+						}
+					#else
+						AA_SPEEX_ERROR("Not build with speex DSP ... ");
+						_nbChunkOut = _nbChunk/10;
+					#endif
+				}
+			};
+		}
+	}
+}
 
 #undef __class__
 #define __class__ "algo::speex::Resampler"
 
-audio::algo::speex::Resampler::Resampler() :
-  #ifdef HAVE_SPEEX_DSP
-    m_speexResampler(nullptr),
-  #endif
-  m_isConfigured(false) {
+audio::algo::speex::Resampler::Resampler() {
 	
 }
 
 audio::algo::speex::Resampler::~Resampler() {
-	#ifdef HAVE_SPEEX_DSP
-		if (m_speexResampler != nullptr) {
-			speex_resampler_destroy(m_speexResampler);
-			m_speexResampler = nullptr;
-		}
-	#endif
+	
 }
 
-void audio::algo::speex::Resampler::init(int8_t _nbChannel, float _inputSampleRate, float _outputSampleRate, int8_t _quality) {
-	#ifdef HAVE_SPEEX_DSP
-		if (m_speexResampler != nullptr) {
-			speex_resampler_destroy(m_speexResampler);
-			m_speexResampler = nullptr;
-		}
-		AA_SPEEX_DEBUG("Create resampler for : " << _inputSampleRate << " to " << _outputSampleRate);
-		int err = 0;
-		m_speexResampler = speex_resampler_init(_nbChannel,
-		                                        _inputSampleRate,
-		                                        _outputSampleRate,
-		                                        _quality, &err);
-		m_isConfigured = true;
-	#else
-		AA_SPEEX_WARNING("SPEEX DSP lib not accessible ==> can not resample");
-		m_isConfigured = false;
-	#endif
+void audio::algo::speex::Resampler::init(int8_t _nbChannel, float _inputSampleRate, float _outputSampleRate, int8_t _quality, enum audio::format _format) {
+	m_private.reset();
+	m_private = std11::make_shared<audio::algo::speex::ResamplerPrivate>(_nbChannel, _inputSampleRate, _outputSampleRate, _quality, _format);
 }
 
 std::vector<enum audio::format> audio::algo::speex::Resampler::getSupportedFormat() {
@@ -60,61 +141,10 @@ std::vector<enum audio::format> audio::algo::speex::Resampler::getNativeSupporte
 	return out;
 }
 
-void audio::algo::speex::Resampler::process(void* _output, size_t& _nbChunkOut, const void* _input, size_t _nbChunk, enum audio::format _format) {
-	if (m_isConfigured == false) {
+void audio::algo::speex::Resampler::process(void* _output, size_t& _nbChunkOut, const void* _input, size_t _nbChunk) {
+	if (m_private == nullptr) {
 		AA_SPEEX_ERROR("Algo is not initialized...");
 	}
-	#ifdef HAVE_SPEEX_DSP
-		switch (_format) {
-			case audio::format_int16:
-				{
-					uint32_t nbChunkInput = _nbChunk;
-					uint32_t nbChunkOutput = _nbChunkOut;
-					int ret = speex_resampler_process_interleaved_int(m_speexResampler,
-					                                                  reinterpret_cast<const int16_t*>(_input),
-					                                                  &nbChunkInput,
-					                                                  reinterpret_cast<int16_t*>(_output),
-					                                                  &nbChunkOutput);
-					// Check all input and output ...
-					if (nbChunkInput != _nbChunk) {
-						AA_SPEEX_ERROR("inputSize (not all read ...) proceed=" << nbChunkInput << " requested=" << _nbChunk);
-						// TODO : manage this case ...
-					}
-					if (nbChunkOutput == _nbChunkOut) {
-						AA_SPEEX_ERROR("Might have not enought data in output... output size=" << _nbChunkOut);
-						// TODO : manage this case ...
-					}
-					_nbChunkOut = nbChunkOutput;
-				}
-				break;
-			case audio::format_float:
-				{
-					uint32_t nbChunkInput = _nbChunk;
-					uint32_t nbChunkOutput = _nbChunkOut;
-					int ret = speex_resampler_process_interleaved_float(m_speexResampler,
-					                                                    reinterpret_cast<const float*>(_input),
-					                                                    &nbChunkInput,
-					                                                    reinterpret_cast<float*>(_output),
-					                                                    &nbChunkOutput);
-					// Check all input and output ...
-					if (nbChunkInput != _nbChunk) {
-						AA_SPEEX_ERROR("inputSize (not all read ...) proceed=" << nbChunkInput << " requested=" << _nbChunk);
-						// TODO : manage this case ...
-					}
-					if (nbChunkOutput == _nbChunkOut) {
-						AA_SPEEX_ERROR("Might have not enought data in output... output size=" << _nbChunkOut);
-						// TODO : manage this case ...
-					}
-					_nbChunkOut = nbChunkOutput;
-				}
-				break;
-			default:
-				AA_SPEEX_ERROR("Can not Limit with unsupported format : " << _format);
-				break;
-		}
-	#else
-		AA_SPEEX_ERROR("Not build with speex DSP ... ");
-		_nbChunkOut = _nbChunk/10;
-	#endif
+	m_private->process(_output, _nbChunkOut, _input, _nbChunk);
 }
 
diff --git a/audio/algo/speex/Resampler.h b/audio/algo/speex/Resampler.h
index 3020198..160e4b1 100644
--- a/audio/algo/speex/Resampler.h
+++ b/audio/algo/speex/Resampler.h
@@ -11,20 +11,14 @@
 #include <etk/types.h>
 #include <audio/format.h>
 #include <etk/chrono.h>
+#include <etk/memory.h>
 #include <vector>
-#ifdef HAVE_SPEEX_DSP
-	#include <speex/speex_resampler.h>
-#endif
 
 namespace audio {
 	namespace algo {
 		namespace speex {
+			class ResamplerPrivate;
 			class Resampler {
-				protected:
-					#ifdef HAVE_SPEEX_DSP
-						SpeexResamplerState* m_speexResampler;
-					#endif
-					bool m_isConfigured;
 				public:
 					Resampler();
 					virtual ~Resampler();
@@ -35,8 +29,9 @@ namespace audio {
 					 * @param[in] _inputSampleRate Input sample rate.
 					 * @param[in] _outputSampleRate Output sample rate.
 					 * @param[in] _quality Resampler quality [1..10].
+					 * @param[in] _format Input/output data format.
 					 */
-					virtual void init(int8_t _nbChannel, float _inputSampleRate, float _outputSampleRate, int8_t _quality);
+					virtual void init(int8_t _nbChannel, float _inputSampleRate, float _outputSampleRate, int8_t _quality=4, enum audio::format _format = audio::format_float);
 					/**
 					 * @brief Get list of format suported in input.
 					 * @return list of supported format
@@ -54,9 +49,10 @@ namespace audio {
 					 * @param[in] _input Input data.
 					 * @param[in] _nbChunk Number of chunk in the input buffer.
 					 * @param[in] _nbChannel Number of channel in the stream.
-					 * @param[in] _format Input data format.
 					 */
-					virtual void process(void* _output, size_t& _nbChunkOut, const void* _input, size_t _nbChunk, enum audio::format _format = audio::format_float);
+					virtual void process(void* _output, size_t& _nbChunkOut, const void* _input, size_t _nbChunk);
+				protected:
+					std11::shared_ptr<ResamplerPrivate> m_private; // private data.
 			};
 		}
 	}
diff --git a/lutin_audio_algo_speex.py b/lutin_audio_algo_speex.py
index 681088c..82f5143 100644
--- a/lutin_audio_algo_speex.py
+++ b/lutin_audio_algo_speex.py
@@ -15,7 +15,7 @@ def create(target):
 		])
 	myModule.add_module_depend(['etk', 'audio'])
 	
-	myModule.add_optionnal_module_depend('speexdsp', "HAVE_SPEEX_DSP", True)
+	myModule.add_optionnal_module_depend('speexdsp', "HAVE_SPEEX_DSP")
 	
 	myModule.add_export_path(tools.get_current_path(__file__))
 	# return module
diff --git a/test/main.cpp b/test/main.cpp
index 9351136..4ed11f5 100644
--- a/test/main.cpp
+++ b/test/main.cpp
@@ -79,11 +79,11 @@ float performanceResamplerStepFloat(float _sampleRateIn, float _sampleRateOut, i
 	APPL_INFO("Start Resampler performance ... " << _sampleRateIn << " -> " << _sampleRateOut << " float");
 	Performance perfo;
 	audio::algo::speex::Resampler algo;
-	algo.init(1, _sampleRateIn, _sampleRateOut, _quality);
+	algo.init(1, _sampleRateIn, _sampleRateOut, _quality, audio::format_float);
 	for (int32_t iii=0; iii<1024; ++iii) {
 		perfo.tic();
 		size_t sizeOut = output.size();
-		algo.process(&output[0], sizeOut, &input[0], input.size(), audio::format_float);
+		algo.process(&output[0], sizeOut, &input[0], input.size());
 		perfo.toc();
 		usleep(1000);
 	}
@@ -119,11 +119,11 @@ float performanceResamplerStepI16(float _sampleRateIn, float _sampleRateOut, int
 	APPL_INFO("Start Resampler performance ... " << _sampleRateIn << " -> " << _sampleRateOut << " int16_t");
 	Performance perfo;
 	audio::algo::speex::Resampler algo;
-	algo.init(1, _sampleRateIn, _sampleRateOut, _quality);
+	algo.init(1, _sampleRateIn, _sampleRateOut, _quality, audio::format_int16);
 	for (int32_t iii=0; iii<1024; ++iii) {
 		perfo.tic();
 		size_t sizeOut = output.size();
-		algo.process(&output[0], sizeOut, &input[0], input.size(), audio::format_int16);
+		algo.process(&output[0], sizeOut, &input[0], input.size());
 		perfo.toc();
 		usleep(1000);
 	}
@@ -247,7 +247,7 @@ int main(int _argc, const char** _argv) {
 		
 		Performance perfo;
 		audio::algo::speex::Resampler algo;
-		algo.init(nbChan, sampleRateIn, sampleRateOut, quality);
+		algo.init(nbChan, sampleRateIn, sampleRateOut, quality, audio::format_int16);
 		int32_t lastPourcent = -1;
 		size_t outputPosition = 0;
 		for (int32_t iii=0; iii<inputData.size()/blockSize; ++iii) {
@@ -259,7 +259,7 @@ int main(int _argc, const char** _argv) {
 			}
 			size_t availlableSize = (output.size() - outputPosition) / nbChan;
 			perfo.tic();
-			algo.process(&output[outputPosition], availlableSize, &inputData[iii*blockSize], blockSize, audio::format_int16);
+			algo.process(&output[outputPosition], availlableSize, &inputData[iii*blockSize], blockSize);
 			if (perf == true) {
 				perfo.toc();
 				usleep(1000);