Add sample endianness info to all audio providers (except one, intentionally; see the stream provider changeset), and make the converting audio provider convert to native endian when required. Updates #725 but needs some testing. Might break compilation in some places, but shouldn't. ("Works for me.")

Originally committed to SVN as r3016.

aegisub/src/audio_provider_avs.h

@@ -61,6 +61,9 @@ public:
 
 	wxString GetFilename();
 
+	// Only exists for x86 Windows, always delivers machine (little) endian
+	bool AreSamplesNativeEndian() { return true; }
+
 	void GetAudio(void *buf, int64_t start, int64_t count);
 	void GetWaveForm(int *min,int *peak,int64_t start,int w,int h,int samples,float scale);
 };

aegisub/src/audio_provider_convert.cpp

@@ -40,6 +40,7 @@
 
 #include "audio_provider_convert.h"
 #include "audio_provider_downmix.h"
+#include "aegisub_endian.h"
 
 
 ///////////////
@@ -70,7 +71,7 @@ ConvertAudioProvider::~ConvertAudioProvider() {
 // Convert to 16-bit
 void ConvertAudioProvider::Make16Bit(const char *src, short *dst, int64_t count) {
 	for (int64_t i=0;i<count;i++) {
-		dst[i] = (src[i]-128)*255;
+		dst[i] = (short(src[i])-128)*255;
 	}
 }
 
@@ -78,7 +79,9 @@ void ConvertAudioProvider::Make16Bit(const char *src, short *dst, int64_t count)
 //////////////////////
 // Change sample rate
 // This requres 16-bit input
-void ConvertAudioProvider::ChangeSampleRate(const short *src, short *dst, int64_t count) {
+// The SampleConverter is a class overloading operator() with a function from short to short
+template<class SampleConverter>
+void ConvertAudioProvider::ChangeSampleRate(const short *src, short *dst, int64_t count, const SampleConverter &converter) {
 	// Upsample by 2
 	if (sampleMult == 2) {
 		int64_t size = count/2;
@@ -86,7 +89,7 @@ void ConvertAudioProvider::ChangeSampleRate(const short *src, short *dst, int64_
 		short next = 0;
 		for (int64_t i=0;i<size;i++) {
 			cur = next;
-			next = *src++;
+			next = converter(*src++);
 			*(dst++) = cur;
 			*(dst++) = (cur+next)/2;
 		}
@@ -100,7 +103,7 @@ void ConvertAudioProvider::ChangeSampleRate(const short *src, short *dst, int64_
 		short next = 0;
 		for (int64_t i=0;i<size;i++) {
 			cur = next;
-			next = *(src++);
+			next = converter(*src++);
 			*(dst++) = cur;
 			*(dst++) = (cur*3+next)/4;
 			*(dst++) = (cur+next)/2;
@@ -109,11 +112,30 @@ void ConvertAudioProvider::ChangeSampleRate(const short *src, short *dst, int64_
 		for (int i=0;i<count%4;i++) *(dst++) = next;
 	}
 
-	// Nothing to do (shouldn't really get here, but...)
-	else if (sampleMult == 1) memcpy((void*)src,dst,count);
+	// Nothing much to do, just ensure correct endedness
+	else if (sampleMult == 1) {
+		while (count-- > 0) {
+			*dst++ = converter(*src++);
+		}
+	}
 }
 
 
+// Do-nothing sample converter for ChangeSampleRate
+struct NullSampleConverter {
+	inline short operator()(const short val) const {
+		return val;
+	}
+};
+
+// Endian-swapping sample converter for ChangeSampleRate
+struct EndianSwapSampleConverter {
+	inline short operator()(const short val) const {
+		return (short)Endian::Reverse((uint16_t)val);
+	};
+};
+
+
 /////////////
 // Get audio
 void ConvertAudioProvider::GetAudio(void *destination, int64_t start, int64_t count) {
@@ -154,8 +176,11 @@ void ConvertAudioProvider::GetAudio(void *destination, int64_t start, int64_t co
 		else if (srcBps == 2) last = buffer1;
 
 		// Convert sample rate
-		if (sampleMult != 1) {
-			ChangeSampleRate(last,(short*)destination,count * channels);
+		if (sampleMult != 1 && source->AreSamplesNativeEndian()) {
+			ChangeSampleRate(last,(short*)destination,count * channels, NullSampleConverter());
+		}
+		else if (!source->AreSamplesNativeEndian()) {
+			ChangeSampleRate(last,(short*)destination,count * channels, EndianSwapSampleConverter());
 		}
 
 		delete [] buffer1;
@@ -166,11 +191,22 @@ void ConvertAudioProvider::GetAudio(void *destination, int64_t start, int64_t co
 // See if we need to downmix the number of channels
 AudioProvider *CreateConvertAudioProvider(AudioProvider *source_provider) {
 	AudioProvider *provider = source_provider;
-	if (provider->GetBytesPerSample() != 2 || provider->GetSampleRate() < 32000)
-		provider = new ConvertAudioProvider(provider);
 
+	// Aegisub requires 16 bit samples,
+	// some audio players break with low samplerates,
+	// everything breaks with wrong-ended samples.
+	if (provider->GetBytesPerSample() != 2 ||
+		provider->GetSampleRate() < 32000 ||
+		!provider->AreSamplesNativeEndian())
+	{
+		provider = new ConvertAudioProvider(provider);
+	}
+
+	// We also require mono audio for historical reasons
 	if (provider->GetChannels() != 1)
+	{
 		provider = new DownmixingAudioProvider(provider);
+	}
 
 	return provider;
 }

aegisub/src/audio_provider_convert.h

@@ -50,12 +50,17 @@ private:
 
 	AudioProvider *source;
 	void Make16Bit(const char *src, short *dst, int64_t count);
-	void ChangeSampleRate(const short *src, short *dst, int64_t count);
+	template<class SampleConverter>
+	void ChangeSampleRate(const short *src, short *dst, int64_t count, const SampleConverter &converter);
 
 public:
 	ConvertAudioProvider(AudioProvider *source);
 	~ConvertAudioProvider();
 
+	// By its nature, the ConvertAudioProvider always delivers machine endian:
+	// That's one of the points of it!
+	bool AreSamplesNativeEndian() { return true; }
+
 	void GetAudio(void *buf, int64_t start, int64_t count);
 	wxString GetFilename() { return source->GetFilename(); }
 };

aegisub/src/audio_provider_downmix.cpp

@@ -55,6 +55,8 @@ DownmixingAudioProvider::DownmixingAudioProvider(AudioProvider *source) {
 
 	if (!(bytes_per_sample == 1 || bytes_per_sample == 2))
 		throw _T("Downmixing Audio Provider: Can only downmix 8 and 16 bit audio");
+	if (!source->AreSamplesNativeEndian())
+		throw _T("Downmixing Audio Provider: Source must have machine endian samples");
 }
 
 /////////////////

aegisub/src/audio_provider_downmix.h

@@ -44,6 +44,9 @@ public:
 	DownmixingAudioProvider(AudioProvider *source);
 	~DownmixingAudioProvider();
 
+	// Downmixing requires samples to be native endian beforehand
+	bool AreSamplesNativeEndian() { return true; }
+
 	void GetAudio(void *buf, int64_t start, int64_t count);
 
 };

aegisub/src/audio_provider_dummy.h

@@ -52,5 +52,7 @@ public:
 	DummyAudioProvider(unsigned long dur_ms, bool _noise);
 	~DummyAudioProvider();
 
+	bool AreSamplesNativeEndian() { return true; }
+
 	void GetAudio(void *buf, int64_t start, int64_t count);
 };

aegisub/src/audio_provider_ffmpegsource.h

@@ -60,6 +60,9 @@ public:
 	FFmpegSourceAudioProvider(Aegisub::String filename);
 	virtual ~FFmpegSourceAudioProvider();
 
+	// FFMS always delivers samples in machine endian
+	bool AreSamplesNativeEndian() { return true; }
+
 	virtual void GetAudio(void *buf, int64_t start, int64_t count);
 
 };

aegisub/src/audio_provider_hd.cpp

@@ -59,6 +59,7 @@ HDAudioProvider::HDAudioProvider(AudioProvider *source) {
 	channels = source->GetChannels();
 	sample_rate = source->GetSampleRate();
 	filename = source->GetFilename();
+	samples_native_endian = source->AreSamplesNativeEndian();
 
 	// Check free space
 	wxLongLong freespace;

aegisub/src/audio_provider_hd.h

@@ -50,6 +50,7 @@ private:
 	wxMutex diskmutex;
 	wxFile file_cache;
 	wxString diskCacheFilename;
+	bool samples_native_endian;
 
 	static wxString DiskCachePath();
 	static wxString DiskCacheName();
@@ -58,5 +59,7 @@ public:
 	HDAudioProvider(AudioProvider *source);
 	~HDAudioProvider();
 
+	bool AreSamplesNativeEndian() { return samples_native_endian; }
+
 	void GetAudio(void *buf, int64_t start, int64_t count);
 };

aegisub/src/audio_provider_lavc.h

@@ -89,6 +89,10 @@ private:
 public:
 	LAVCAudioProvider(Aegisub::String _filename);
 	virtual ~LAVCAudioProvider();
+
+	// Supposedly lavc always returns machine endian samples
+	bool AreSamplesNativeEndian() { return true; }
+
 	virtual void GetAudio(void *buf, int64_t start, int64_t count);
 };
 

aegisub/src/audio_provider_pcm.cpp

@@ -365,6 +365,16 @@ public:
 			filepos += (Endian::LittleToMachine(ch.size) + 1) & ~1;
 		}
 	}
+
+
+	bool AreSamplesNativeEndian()
+	{
+		// 8 bit samples don't consider endianness
+		if (bytes_per_sample < 2) return true;
+		// Otherwise test whether we're little endian
+		uint32_t testvalue = 0x008800ff;
+		return testvalue == Endian::LittleToMachine(testvalue);
+	}
 };
 
 
@@ -508,6 +518,16 @@ public:
 			filepos += (chunk_size + 7) & ~7;
 		}
 	}
+
+
+	bool AreSamplesNativeEndian()
+	{
+		// 8 bit samples don't consider endianness
+		if (bytes_per_sample < 2) return true;
+		// Otherwise test whether we're little endian
+		uint32_t testvalue = 0x008800ff;
+		return testvalue == Endian::LittleToMachine(testvalue);
+	}
 };
 
 

aegisub/src/audio_provider_ram.cpp

@@ -57,6 +57,7 @@ RAMAudioProvider::RAMAudioProvider(AudioProvider *source) {
 	// Init
 	blockcache = NULL;
 	blockcount = 0;
+	samples_native_endian = source->AreSamplesNativeEndian();
 
 	// Allocate cache
 	int64_t ssize = source->GetNumSamples() * source->GetBytesPerSample();

aegisub/src/audio_provider_ram.h

@@ -48,6 +48,7 @@ class RAMAudioProvider : public AudioProvider {
 private:
 	char** blockcache;
 	int blockcount;
+	bool samples_native_endian;
 
 	void Clear();
 
@@ -55,5 +56,7 @@ public:
 	RAMAudioProvider(AudioProvider *source);
 	~RAMAudioProvider();
 
+	bool AreSamplesNativeEndian() { return samples_native_endian; }
+
 	void GetAudio(void *buf, int64_t start, int64_t count);
 };

aegisub/src/audio_provider_stream.h

@@ -68,6 +68,12 @@ public:
 	StreamAudioProvider();
 	~StreamAudioProvider();
 
+	// The AreSamplesNativeEndian() method is intentionally missing.
+	// This class was used with the experimental scrubbing code but never
+	// used outside that.
+	// The method is left out to make it break compilation in case it does
+	// get used again, so it can get a review and stuff.
+
 	void GetAudio(void *buf, int64_t start, int64_t count);
 	void Append(void *buf, int64_t count);
 	void SetParams(int channels,int rate,int bps);

aegisub/src/include/aegisub/audio_provider.h

@@ -74,6 +74,7 @@ public:
 	int GetSampleRate();
 	int GetBytesPerSample();
 	int GetChannels();
+	virtual bool AreSamplesNativeEndian() = 0;
 
 	void GetWaveForm(int *min,int *peak,int64_t start,int w,int h,int samples,float scale);
 };