Filter SDK/avs2pcm - Revision history

Admin at 20:39, 6 October 2014

2014-10-06T20:39:49Z

Raffriff42: Filter_SDK -> FilterSDK

2014-09-19T04:38:10Z

Filter_SDK -> FilterSDK

Raffriff42: Category:Filter_SDK

2014-09-19T03:53:56Z

Category:Filter_SDK

Admin at 21:05, 24 June 2014

2014-06-24T21:05:31Z

Admin at 21:04, 24 June 2014

2014-06-24T21:04:12Z

Admin at 21:01, 24 June 2014

2014-06-24T21:01:59Z

Admin at 21:00, 24 June 2014

2014-06-24T21:00:42Z

Admin at 20:57, 24 June 2014

2014-06-24T20:57:04Z

Admin at 21:55, 23 June 2014

2014-06-23T21:55:56Z

Admin at 14:31, 23 June 2014

2014-06-23T14:31:06Z

@@ Line 257: / Line 257: @@
       __int64 bytes = vi.BytesFromAudioSamples(count);
-We will use [http://www.cplusplus.com/reference/cstdio/fwrite/ fwrite] to write 'count' audio samples to a file. So we will need to know the corresponding number of bytes which needs to be written. [[Cplusplus_API/VideoInfo|BytesFromAudioSamples]] gives the number of bytes and it is calculated internally as follows:
+We will use [http://www.cplusplus.com/reference/cstdio/fwrite/ fwrite] to write 'count' audio samples to a file. So we will need to know the corresponding number of bytes which needs to be written. [[Filter_SDK/Cplusplus_API/VideoInfo|BytesFromAudioSamples]] gives the number of bytes and it is calculated internally as follows:
 {| style="height:100px" border="1" cellpadding="4"

← Older revision		Revision as of 04:38, 19 September 2014
Line 378:		Line 378:
	The remaining lines above are explained in [[Filter_SDK/avs2yuv\|avs2yuv]], so they won't be repeated here.		The remaining lines above are explained in [[Filter_SDK/avs2yuv\|avs2yuv]], so they won't be repeated here.

−	[[Category:~~Filter_SDK~~]]	+	[[Category:FilterSDK]]

← Older revision		Revision as of 03:53, 19 September 2014
Line 377:		Line 377:

	The remaining lines above are explained in [[Filter_SDK/avs2yuv\|avs2yuv]], so they won't be repeated here.		The remaining lines above are explained in [[Filter_SDK/avs2yuv\|avs2yuv]], so they won't be repeated here.
		+
		+	[[Category:Filter_SDK]]

@@ Line 361: / Line 361: @@
 Here an audio sample is stored in one sample (having the size of a multiple bytes), but fwrite will write the same bytes to the file as earlier.
-At first glance it seems possible to simplify the code above by moving the lines "clip .. delete[] samples" outside the switch statement, but that's not possible. The reason is that each code-block corresponding to a label has its own scope. So the variable samples doesn't exist outside the switch statement and can't be used there for further processing.
+At first glance it seems possible to simplify the code above by moving the lines "clip .. delete[] samples" outside the switch statement, but that's not possible. The reason is that each code-block corresponding to a label has its own scope. So the variable samples doesn't exist outside the switch statement and can't be used there for further processing. See [http://stackoverflow.com/questions/92396/why-cant-variables-be-declared-in-a-switch-statement here] for more information.
       env->DeleteScriptEnvironment();

@@ Line 359: / Line 359: @@
+  }
-Here an audio sample is stored in one sample (having the size of a multiple bytes), but fwrite will write the same bytes to the file.
+Here an audio sample is stored in one sample (having the size of a multiple bytes), but fwrite will write the same bytes to the file as earlier.
-At first glance it seems possible to simplify the code above by moving the lines "clip .. delete[] samples" outside the switch statement, but that's not possible. The issue is that each code-block corresponding to a label has its own scope. So the variable samples doesn't exist outside the switch statement and can't be used there for further processing.
+At first glance it seems possible to simplify the code above by moving the lines "clip .. delete[] samples" outside the switch statement, but that's not possible. The reason is that each code-block corresponding to a label has its own scope. So the variable samples doesn't exist outside the switch statement and can't be used there for further processing.
       env->DeleteScriptEnvironment();

@@ Line 284: / Line 284: @@
       delete[] samples;
-There are a few ways to write audio to a file. The simpliest one is the one above. Let's look at what happens with our data:
+There are a few ways to write audio to a file. The simpliest one is the one above. Let's look at what happens with our data with an example:
   Tone(length=1, frequency=2, samplerate=48000, channels=1, type="square", level=1.0) # float

@@ Line 297: / Line 297: @@
 |-
 | 8Bit (samples between 0 and 2^8-1)
-| s[0] = 255; s[end-1] = 0
+| s[0] = 255; s[count-1] = 0
 | FF .. 00 ..
 |-
-| 16Bit (samples between -2^15-1 and 2^15)
+| 16Bit (samples between -2^15 and 2^15-1)
-| s[1]*256+s[0] = 127*256+255 = 32767; s[end-1]*256+s[end-2] = 128*256+0 = 32768 (= -32768)
+| s[1]*256+s[0] = 127*256+255 = 32767; s[count-1]*256+s[count-2] = 128*256+0 = 32768 (= -32768)
 | FF 7F .. 00 80 ..
 |-
-| 24Bit (samples between 0 and 2^24-1)
+| 24Bit (samples between -2^23 and 2^23-1)
-| s[2]*16^4+s[1]*16^2+s[0] = 16777215; s[end-1]*16^4+s[end-2]*16^2+s[end-3] = 0
+| s[2]*16^4+s[1]*16^2+s[0] = 8388607; s[count-1]*16^4+s[count-2]*16^2+s[count-3] = 8388608 (= -8388608)
-| FF FF FF .. 00 00 00 ..
+| FF FF 7F .. 00 00 80 ..
 |-
-| 32Bit (samples between 0 and 2^32-1)
+| 32Bit (samples between -2^31 and 2^31-1)
-| s[3]*16^6+s[2]*16^4+s[1]*16^2+s[0] = 4294967295; s[end-1]*16^6+s[end-2]*16^4+s[end-3]*16^2+s[end-4] = 0
+| s[3]*16^6+s[2]*16^4+s[1]*16^2+s[0] = 2147483647; s[count-1]*16^6+s[count-2]*16^4+s[count-3]*16^2+s[count-4] = 2147483648 (= -2147483648)
-| FF FF FF FF .. 00 00 00 00 ..
+| FF FF FF 7F .. 00 00 00 80 ..
 |-
 | [[Float]] (samples between -1.00000 and 1.000000)
-| s[3]*16^6+s[2]*16^4+s[1]*16^2+s[0] = 63*16^6+128*16^4+0*16^2+0 = 106535321; s[end-1]*16^6+s[end-2]*16^4+s[end-3]*16^2+s[end-4] = 191*16^6+128*16^4+0*16^2+0 = 3212836864
+| s[3]*16^6+s[2]*16^4+s[1]*16^2+s[0] = 63*16^6+128*16^4+0*16^2+0 = 106535321; s[count-1]*16^6+s[count-2]*16^4+s[count-3]*16^2+s[count-4] = 191*16^6+128*16^4+0*16^2+0 = 3212836864
 | 00 00 80 3F .. 00 00 80 BF ..
 |}
-Above the samples are declared as unsigned char (regardless of the number of bits in a sample), but they filled by GetAudio as explained above. Thus an audio sample is stored in multiple samples (each samples having the size of a byte). This is the simplies and cleanest way to do, but you could also have done the following instead:
+Above the samples are declared as unsigned char (regardless of the number of bits in a sample), but they are filled by GetAudio as explained above. Thus an audio sample is stored in multiple samples (each sample having the size of a byte). This is the simpliest and cleanest way to do, but you could also have done the following instead:
   int BlockAlign = vi.BytesPerAudioSample();
@@ Line 359: / Line 359: @@
+  }
 Here an audio sample is stored in one sample (having the size of a multiple bytes). But it will write the same bytes to the file.
       env->DeleteScriptEnvironment();

@@ Line 186: / Line 186: @@
       AVSValue res = env->Invoke("Import", AVSValue(&arg, 1));
       if (!res.IsClip()) {
-        fprintf(stderr, "Error: '%s' didn't return a video clip.\n", infile);
+        fprintf(stderr, "Error: '%s' didn't return a clip.\n", infile);
         FreeLibrary(avsdll);
         return 1;
@@ Line 201: / Line 201: @@
       VideoInfo vi = clip->GetVideoInfo();
-The lines above are explained in [[Filter_SDK/avs2yuv|avs2yuv]], so won't be repeated here.
+The lines above are explained in [[Filter_SDK/avs2yuv|avs2yuv]], so they won't be repeated here.
       if (!vi.HasAudio()) {
@@ Line 209: / Line 209: @@
+      }
-Returns an error if the clip doesn't contain audio (in case it contains only video for example).
+Returns an error if the clip doesn't contain audio.
       fprintf(stderr, " %s:\n", infile);
@@ Line 257: / Line 257: @@
       __int64 bytes = vi.BytesFromAudioSamples(count);
-We will use [http://www.cplusplus.com/reference/cstdio/fwrite/ fwrite] to write 'count' audio samples to a file. So we will need to know the corresponding number of bytes which needs to be written. [[Cplusplus_API/VideoInfo|BytesFromAudioSamples]] gives the number of bytes and it is internally calculated as follows:
+We will use [http://www.cplusplus.com/reference/cstdio/fwrite/ fwrite] to write 'count' audio samples to a file. So we will need to know the corresponding number of bytes which needs to be written. [[Cplusplus_API/VideoInfo|BytesFromAudioSamples]] gives the number of bytes and it is calculated internally as follows:
 {| style="height:100px" border="1" cellpadding="4"
@@ Line 264: / Line 264: @@
 |-
 | BytesPerChannelSample()
-| sizeof(unsigned char)
+| = sizeof(unsigned char) = 1 byte [for 8 bit audio], <br>
-= 8 bits = 1 byte [for 8 bit audio], <br>
+= sizeof(signed short) = 2 bytes [for 16 bit audio], <br>
-= sizeof(signed short) = 16 bits = 2 bytes [for 16 bit audio], <br>
+= 3 bytes [for 24 bit audio], <br>
-= 3 [for 24 bit audio] = 24 bits = 3 bytes, <br>
+= sizeof(signed int) = 4 bytes [for 32 bit audio], <br>
-= sizeof(signed int) = 32 bits = 4 bytes [for 32 bit audio], <br>
+= sizeof(SFLOAT) = 4 bytes [for float audio; this is also 32 bit audio]
-= sizeof(SFLOAT) = 32 bits = 4 bytes [for float audio]
 |-
 | BytesPerAudioSample()
@@ Line 290: / Line 289: @@
   ConvertAudioTox() // x = 8Bit, 16Bit, 24Bit, 32Bit and Float
-The samples are always written as little endian. So this means value = LSB MSB (the least significant byte first followed by the most significant byte).
+The samples are always written to the pcm file as little endian. So this means value = LSB ... MSB (the least significant byte first and the most significant byte last).
-x type
+{| style="height:100px" border="1" cellpadding="4"
-value: samples[0] = 255; samples[x] = 0
+!width=20%| type (x)
-value: samples[1]*256+samples[0] = 65535; samples[x+1]*256+samples[x] = 0
+!width=40%| value of samples (abbreviated as s)
-no separate type (samples between 0 and 2^24-1) ..
+!width=40%| bytes written in file
-unsigned int (samples between 0 and 2^32-1) FF FF FF FF .. 00 00 00 00 ..
+|-
-float (samples between -1.00000 and 1.000000) 1.000000 = 00 00 80 3F; -1.000000 = 00 00 80 BF
+| 8Bit
-# alternative
+Above the samples are declared as unsigned char (regardless the samplebit) but filled by GetAudio as above, meaning an audio sample is written accross multiple samples 's'. This is the simplies and cleanest way to do, but you could have done also the following instead:
-x type y1 y2
+It will produce the same output.
-unsigned char (samples between 0 and 255) FF .. 00 ..
-signed short (samples between -32767 and 32767) 32767 = 0x7FFF [written as 0xFF7F] -32767 = 65535 - 32767 = 32768 = 0x8000  [written as 0x0080]
-no separate type (samples between 0 and 2^24-1) ...
-unsigned int (samples between 0 and 2^32-1) FF FF FF FF .. 00 00 00 00 ..
-float (samples between -1.00000 and 1.000000) 1.000000 = 00 00 80 3F; -1.000000 = 00 00 80 BF
       env->DeleteScriptEnvironment();