Ima adpcm
Author: k | 2025-04-24
ADPCM IMA Dialogic OKI: ADPCM IMA Duck DK3: ADPCM IMA Duck DK4: ADPCM IMA Electronic Arts EACS: ADPCM IMA Electronic Arts SEAD: ADPCM IMA Eurocom DAT4: ADPCM IMA QuickTime: X: X: ADPCM IMA Simon Schuster Interactive: X: X: ADPCM IMA Ubisoft APM: X: X: ADPCM IMA Loki SDL MJPEG: X: ADPCM IMA WAV: X: X: ADPCM IMA
IMA-ADPCM/ima_adpcm.c at master aikiriao/IMA-ADPCM
State, I keep my state, since it's more accurate.Comparing Microsoft's and Apple's IMA ADPCM CodecsFrom a coding point of view, Apple's implementation of IMA ADPCM is simpler, mostly because you don't need to worry about two different packet formats. It also provides slightly better quality; Microsoft throws away eight bits of decompressor state at the beginning of each mono packet, which results in a small jump in sample value at the start of each packet.Microsoft's approach is more compact. Assuming 2048-byte stereo packets, Microsoft's approach incurs only eight bytes of overhead for each 2041 bytes of compressed data (less than 0.4 percent overhead), while Apple incurs two bytes for each 32 bytes of compressed data (over six percent overhead). It also provides lower latency; if stereo data is being compressed and decompressed simultaneously, Microsoft only incurs an eight-sample delay, while Apple must wait for a complete 64-sample packet before it can have stereo data available. However, this is not a very serious issue even at low sampling rates; at 11,025 samples per second, 64 samples is less than six milliseconds.ConclusionAlthough originally defined to be a platform-independent standard for audio-file compression, the IMA ADPCM format doesn't quite meet that requirement. The low-level compression is not a concern; it's independent of byte order and is defined using only low-level bit manipulations, which give it good performance even on processors that don't have fast integer division.However, the differences between the Microsoft and Apple versions make it impossible to mix their codecs. You can't extract IMA ADPCM compressed data from an AIFF-C file and feed it to the Microsoft codec, nor can you feed IMA ADPCM WAVE data to the Apple codec. Fortunately, IMA ADPCM codecs are compact; the code I've provided here should allow you to easily support both common variants, and you should also be able to adapt this code to support any other IMA ADPCM variants you may encounter.AcknowledgmentsThe code presented here is partially based on an implementation of Intel's DVI ADPCM algorithm developed by Jack Jansen at the Stichting Mathematisch Centrum, Amsterdam, The Netherlands.Listing Oneshort ImaAdpcmDecode(unsigned char deltaCode, ImaState &state) ADPCM IMA Dialogic OKI: ADPCM IMA Duck DK3: ADPCM IMA Duck DK4: ADPCM IMA Electronic Arts EACS: ADPCM IMA Electronic Arts SEAD: ADPCM IMA Eurocom DAT4: ADPCM IMA QuickTime: X: X: ADPCM IMA Simon Schuster Interactive: X: X: ADPCM IMA Ubisoft APM: X: X: ADPCM IMA Loki SDL MJPEG: X: ADPCM IMA WAV: X: X: ADPCM IMA IMA ADPCM Audio Compression AcceleratorRead the documentation for projectWhat is this projectThis HDL block accepts a pulse density modulated (PDM) microphone signal and produces an encoded output at a lower sampling frequency while maintaining audio intelligibility.Expected Inputs: clk (clk) slow_clk (ui_in[1]) for the ADPCM block at 1/8 frequency of clk Pulse Density Modulated input pdm_in (ui_in[2]) at clocked with clk block_enable (ui_in[3]) (active high): single bit enable for the entire blockOutputs: encPcm (uo_out[4:1]): the final 4 bit ADPCM encoded output outValid (uo_out[0]): Output Valid flag for the ADPCM block, goes high for one cycle of slow_clk each time a new valid adpcm value is outputVerilog FilesCICDecimatorVerilogBlock.vThe CIC decimation filter to parallelize serial pdm microphone data and decimate by a factor of 64CIC_ADPCM_Wrapper.vThe wrapper block with control logic for block_enable and connections between the CIC and encoderima_adpcm_enc.vThe IMA ADPCM encoder which encodes the 16 bit output of the CIC filter to 4 bit adpcm samplestt_um_ADPCM_COMPRESSOR.vA Top level wrapper to match the tinytapeout signal namesResourcesCIC Filter ExplanationCIC Filter Continued[IMA ADPCM Standard] (Comments
State, I keep my state, since it's more accurate.Comparing Microsoft's and Apple's IMA ADPCM CodecsFrom a coding point of view, Apple's implementation of IMA ADPCM is simpler, mostly because you don't need to worry about two different packet formats. It also provides slightly better quality; Microsoft throws away eight bits of decompressor state at the beginning of each mono packet, which results in a small jump in sample value at the start of each packet.Microsoft's approach is more compact. Assuming 2048-byte stereo packets, Microsoft's approach incurs only eight bytes of overhead for each 2041 bytes of compressed data (less than 0.4 percent overhead), while Apple incurs two bytes for each 32 bytes of compressed data (over six percent overhead). It also provides lower latency; if stereo data is being compressed and decompressed simultaneously, Microsoft only incurs an eight-sample delay, while Apple must wait for a complete 64-sample packet before it can have stereo data available. However, this is not a very serious issue even at low sampling rates; at 11,025 samples per second, 64 samples is less than six milliseconds.ConclusionAlthough originally defined to be a platform-independent standard for audio-file compression, the IMA ADPCM format doesn't quite meet that requirement. The low-level compression is not a concern; it's independent of byte order and is defined using only low-level bit manipulations, which give it good performance even on processors that don't have fast integer division.However, the differences between the Microsoft and Apple versions make it impossible to mix their codecs. You can't extract IMA ADPCM compressed data from an AIFF-C file and feed it to the Microsoft codec, nor can you feed IMA ADPCM WAVE data to the Apple codec. Fortunately, IMA ADPCM codecs are compact; the code I've provided here should allow you to easily support both common variants, and you should also be able to adapt this code to support any other IMA ADPCM variants you may encounter.AcknowledgmentsThe code presented here is partially based on an implementation of Intel's DVI ADPCM algorithm developed by Jack Jansen at the Stichting Mathematisch Centrum, Amsterdam, The Netherlands.Listing Oneshort ImaAdpcmDecode(unsigned char deltaCode, ImaState &state)
2025-04-06IMA ADPCM Audio Compression AcceleratorRead the documentation for projectWhat is this projectThis HDL block accepts a pulse density modulated (PDM) microphone signal and produces an encoded output at a lower sampling frequency while maintaining audio intelligibility.Expected Inputs: clk (clk) slow_clk (ui_in[1]) for the ADPCM block at 1/8 frequency of clk Pulse Density Modulated input pdm_in (ui_in[2]) at clocked with clk block_enable (ui_in[3]) (active high): single bit enable for the entire blockOutputs: encPcm (uo_out[4:1]): the final 4 bit ADPCM encoded output outValid (uo_out[0]): Output Valid flag for the ADPCM block, goes high for one cycle of slow_clk each time a new valid adpcm value is outputVerilog FilesCICDecimatorVerilogBlock.vThe CIC decimation filter to parallelize serial pdm microphone data and decimate by a factor of 64CIC_ADPCM_Wrapper.vThe wrapper block with control logic for block_enable and connections between the CIC and encoderima_adpcm_enc.vThe IMA ADPCM encoder which encodes the 16 bit output of the CIC filter to 4 bit adpcm samplestt_um_ADPCM_COMPRESSOR.vA Top level wrapper to match the tinytapeout signal namesResourcesCIC Filter ExplanationCIC Filter Continued[IMA ADPCM Standard] (
2025-03-31In this month's column, Tim goes inside the IMA ADPCM audio-compression format and examines how it is implemented by Microsoft and Apple. Dr. Dobb's Journal November 1997: Algorithm AlleyInside IMA ADPCMBy Tim KientzleTim is senior technical editor for DDJ and author of the forthcoming Programmer's Guide to Sound (Addison-Wesley, 1998). He can be reached at [email protected].Audio compression is a tricky business. Although the 10:1 or better compression offered by MPEG audio and Dolby AC-2 is impressive, the CPU requirements are, unfortunately, equally impressive. This is fine if you can afford to devote a high-speed CPU entirely to audio decompression, but that's not always an option. Frequently, you'll want to juggle animation or other CPU-intensive actions at the same time.Intel's DVI video compression engine includes a method for audio compression that offers moderate compression with very low CPU requirements. This engine was later adopted by the Interactive Multimedia Association (IMA), and the audio algorithm has become widely known as "IMA ADPCM." In this article, I'll describe how IMA ADPCM works, and discuss two specific implementations of it, one by Microsoft and the other by Apple.IMA ADPCMIMA ADPCM stores four bits of information for each sample. The compressor takes the difference between two samples, divides it by the current step size, and uses that as the next four-bit compressed output. Conversely, the decompressor takes this four-bit value, multiplies it by the current step size, and adds the result to the previous sample to create the next sample.To keep the amount of state information low, the step size is not stored directly. Rather, there is a table of 88 possible step sizes (Example 1), and the decompressor maintains an index into this table. These step sizes roughly follow an exponential curve. Thanks to this table, the total amount of state information is only 23 bits: seven bits for the index into the step size table and 16 bits for the current decompressed value. In embedded applications, it may be possible to keep the entire state in registers, requiring no RAM at all.The important part of IMA ADPCM is how it adjusts the
2025-04-13The Sound Normalizer improves and regains a quality of Mp3 and Wav (PCM 8, 16, 24, 32 bits, DSP, GSM, IMA ADPCM, MS ADPCM, AC3, MP3, MP2, OGG, A-LAW, u-LAW) files. It is reached by the test and normalization of the volume level of Mp3 and Wav files.The Sound Normalizer contains Batch processor and it allows to fulfill the batch test, batch normalization and batch converting Mp3 and Wav (PCM 8, 16, 24, 32 bits, DSP, GSM, IMA ADPCM, MS ADPCM, AC3, MP3, MP2, OGG, A-LAW, u-LAW) files.The Sound Normalizer allows to fulfill the test and normalization separately for each channel.Key Features: * Advanced Wav Normalizer and MP3 Normalizer; * Batch processor for Mp3 and Wav (PCM 8, 16, 24, 32 bit) files; * Batch normalizing for Mp3 and Wav (PCM 8, 16, 24, 32 bit) files; * Batch converting for Mp3 and Wav (PCM 8, 16, 24, 32 bit) files; * Batch test for Mp3 and Wav (PCM 8, 16, 24, 32 bit) files; * Converting Mp3 to Wav files using LAME MP3 Encoder 3.97; * Converting Wav to Mp3 files using Lame MP3 Encoder 3.97; * Support various encoder quality/speed modes; * Extensive encoder options; * Resampling for Mp3 files; * Support pre-listen audio file; * Support ID3 v1 and v2 tag (Artist/Title/Genre and so on); * Simple, easy-to-use and highly customizable Windows XP interface; * Support Themes.Fixed some bugs.
2025-03-26