[DEV] create new API i,terface for speex ==> more stable, ABI stable

audio/algo/speex/Resampler.cpp

@@ -8,28 +8,27 @@
 #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::Resampler"
+#define __class__ "algo::speex::ResamplerPrivate"
-
+namespace audio {
-audio::algo::speex::Resampler::Resampler() :
+	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_isConfigured(false) {
+					  m_format(_format) {
-	
-}
-
-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);
@@ -41,31 +40,23 @@ void audio::algo::speex::Resampler::init(int8_t _nbChannel, float _inputSampleRa
 							                                        _inputSampleRate,
 							                                        _outputSampleRate,
 							                                        _quality, &err);
-		m_isConfigured = true;
 						#else
 							AA_SPEEX_WARNING("SPEEX DSP lib not accessible ==> can not resample");
-		m_isConfigured = false;
 						#endif
-}
-
-std::vector<enum audio::format> audio::algo::speex::Resampler::getSupportedFormat() {
-	std::vector<enum audio::format> out = getNativeSupportedFormat();
-	return out;
-}
-
-std::vector<enum audio::format> audio::algo::speex::Resampler::getNativeSupportedFormat() {
-	std::vector<enum audio::format> out;
-	out.push_back(audio::format_float);
-	out.push_back(audio::format_int16);
-	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) {
-		AA_SPEEX_ERROR("Algo is not initialized...");
 					}
+				
+				~ResamplerPrivate() {
 					#ifdef HAVE_SPEEX_DSP
-		switch (_format) {
+						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;
@@ -109,12 +100,51 @@ void audio::algo::speex::Resampler::process(void* _output, size_t& _nbChunkOut,
 								}
 								break;
 							default:
-				AA_SPEEX_ERROR("Can not Limit with unsupported format : " << _format);
+								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() {
+	
+}
+
+audio::algo::speex::Resampler::~Resampler() {
+	
+}
+
+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() {
+	std::vector<enum audio::format> out = getNativeSupportedFormat();
+	return out;
+}
+
+std::vector<enum audio::format> audio::algo::speex::Resampler::getNativeSupportedFormat() {
+	std::vector<enum audio::format> out;
+	out.push_back(audio::format_float);
+	out.push_back(audio::format_int16);
+	return out;
+}
+
+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...");
+	}
+	m_private->process(_output, _nbChunkOut, _input, _nbChunk);
 }
 

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.
 			};
 		}
 	}

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

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);