Rate converter

Abstract

Embodiments of the invention may be used to implement a rate converter that includes: 6 channels in forward (audio) path, each channel having a 24-bit signal path per channel, an End-to-end SNR of 110 dB, all within the 20 Hz to 20 KHz bandwidth. Embodiment may also be used to implement a rate converter having: 2 channels in a reverse path, such as for voice signals, 16-bit signal path per channel, an End-to-end SNR of 93 dB, all within 20 Hz to 20 KHz bandwidth. The rate converter may include sample rates such as 8, 11.025, 12, 16, 22.05, 24, 32 44.1, 48, and 96 KHz. Further, rate converters according to embodiments may include a gated clock in low-power mode to conserve power.

Description

FIELD OF THE INVENTION

This disclosure is directed to signal processing, and, more specifically, to a system for converting data samples at a first rate to data samples at a second rate.

BACKGROUND

In general, conventional rate converters include two major computation blocks, as illustrated in FIG. 1. A first block is a conversion rate tracking loop that determines the ratio between an input sample rate and output sample rate. Once the ratio is determined, the conversion rate tracking loop generates the corresponding input sample index for each output sample.

The second computation block is the sample interpolator. The function of this block is to interpolate the input data sample and to generate the output data sample with the real value index. One problem with conventional rate converters is that they create aliasing from the resampling, and it is relatively difficult to produce an output signal having a high Signal-to-Noise Ratio (SNR) and having a relatively flat frequency response within standard frequency ranges for audio signals.

Embodiments of the invention address this and other limitations of the prior art.

SUMMARY OF THE INVENTION

Embodiments of the invention may be used to implement a rate converter that includes: 6 channels in forward (audio) path, each channel having a 24-bit signal path per channel, an End-to-end SNR of 110 dB, all within the 20 Hz to 20 KHz bandwidth. Embodiment may also be used to implement a rate converter having: 2 channels in a reverse path, such as for voice signals, 16-bit signal path per channel, an End-to-end SNR of 93 dB, all within 20 Hz to 20 KHz bandwidth.

The rate converter may include sample rates such as 8, 11.025, 12, 16, 22.05, 24, 32.44.1, 48, and 96 KHz.

Further, rate converters according to embodiments may include a gated clock in low-power mode to conserve power.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a block diagram of a conventional rate converter.

FIG. 2 is a block diagram for a conversion rate tracking loop of a first type of rate converter.

FIG. 3 is a block diagram for a conversion rate tracking loop of a rate converter according to embodiments of the invention.

FIG. 4 is a graph showing passband behavior for anti-alias filters according to embodiments of the invention.

FIG. 5 is a graph illustrating total wrapped aliases for all anti-alias filters according to embodiments of the invention.

FIGS. 6A, 6B, 6C, 6D, 6E, and 6F illustrate frequency responses for 1×, 1.5×, 2×, 3×, 4×, and 6× filters according to embodiments of the invention.

FIG. 7 is a block diagram of an interpolator portion of a rate controller according to embodiments of the invention.

FIGS. 8A-8B, 9A-9B, 10A-10B, 11A-11B, 12A-12B, 13A-13B, 14A-14B, are graphs illustrating performance differences between an old rate convertor and an example rate convertor according to embodiments of the invention.

DETAILED DESCRIPTION

For comparison, FIG. 2 is a block diagram for a conversion rate tracking loop of a first type of rate converter, while FIG. 3 is a block diagram for a conversion rate tracking loop of a rate converter according to embodiments of the invention.

In general, with reference to FIG. 3, inputs for a rate converter block include an input audio sample, an input audio channel id, a signal that the input sample is ready, and a signal that indicates that the audio destination is ready to take the next sample. In particular, the input audio samples are packed into one word for all the audio channels, which is time multiplexed for all channels. The input audio channel id tells which channel the sample belongs to. The input sample ready signal signifies that the next audio sample is ready, while the output sample acknowledgement signifies that the audio destination is ready to take the next sample. Other standard interfaces signals like clk, reset, enable, etc. may be present, as well as one or more control signals from a Direct Memory Address. Finally, there may be one or more register bank access signals.

There are also several output signals. An output audio sample includes a single word with all of the audio channels output from the converter. It is time multiplexed for all channels. The output audio channel id tells which channel the sample belongs to. An output sample ready signal signifies that the next audio sample is ready. An input sample ack signifies that the converter is ready to accept the next signal from the audio source. Also, a register bank access signal may be output.

In general, with reference to FIG. 3, the rate tracking loop tries to estimate the ratio between an input sample rate and output sample rate. In addition, it generates the input sample index that corresponds to every output sample.

Both input and output sample rates may be jittery. The estimated sample rate ratio should be stable enough to yield a high SNR, and also should maintain a stable average value for the sample index. Ideally, for maximum performance input buffers should not overflow or underflow, and audio latency variation should be minimized.

In operation, initially, an estimate of drate (decimation rate) is obtained by either a user setting, or by measuring input and out sample rates. The estimated value is stored in drate reg, which is a register that stores the decimation rate.

The tracking loop circuit includes one or more different “gears.” In one embodiment there are four gears. The difference between the gears is the value for the gain elements G1, G2 and G3. The gain values for each of the gear levels are programmed in registers and can be changed by firmware. The gear control decides when to move the gear up or down. The change is controlled by two factors. In some embodiments, the converter stays in each gear for a minimal amount of time before switching to another gear. Also in some embodiments, this minimal time doubles for each higher gear. In one embodiment, the rate converter of FIG. 3 moves to a higher gear when the absolute value of error and the change slope of error are both under certain thresholds. Likewise, if the absolute value of error or the change slope is above the thresholds, the tracking loop would move down the gear.

Table 1 includes default values for tracking loop parameter registers according to embodiments of the invention

	TABLE 1
	Gear
	0	1	2	3
Log2 (G1)	−3	−5	−7	−9
Log2 (G2)	−7	−9	−11	−13
Log2 (G3)	−6	−8	−10	−12
Log2 (gear up abs)	9.5	7.5	5.5	N/A
Log2 (gear up slope)	8	6	4	N/A
Log2 (gear down abs)	N/A	10.5	8.5	6.5
Log2 (gear down slope)	N/A	9	7	5

The tracking loop of FIG. 3 converges with the default values. In some embodiment it converges within 1 second before the SNR is above 100 dB, and reaches final lock with the next few seconds. The final tracking yields a latency variation of less than 100 ns. The overall SNR impact is also small. More than 120 dB SNR can be reached.

In a particular implementation of embodiments of the invention, the index error is sampled at 24 MHz to get time accuracy that is way below the sample period. The average by 2¹⁴ enables all of 2nd order phase lock loop and the IIR in error smoothing to operate at 1.5 KHz clock, which allows a very low current draw. The sample index buffer has 28 bits. With one input sample counts as 2²³, that gives enough room for 16 samples. Since the input buffer is only 16 samples more than the filter requirement. It is enough to cover all possible cases. The drate register contains 41 bits. That is 13 more LSB than the sample index buffer. The 13 LSBs may be a conservative number, but it costs very little in area and current. The error IIR buffer contains 45 bits. It adds 13 LSBs to the input from index error average. Again, 13 LSBs may be a conservative number, but it costs very little in area and current since no multipliers are required. The operation frequency is 1.5 KHz. Dithering is done at reducing drate from 41 bit to 28 bit before adds to sample buffer.

Compared to the rate converter illustrated in FIG. 2, the rate converter of FIG. 3 is much more efficient. The rate converter of FIG. 3 includes a simple 1st order IIR filter, while the converter of FIG. 2 has four more complex low pass filters. The simple 1st order IIR filter is certainly much simpler and costs a lot less area and current. Each filter has different gain and different spectral/impulse response. All four filters are running all the time. The gear shift decides which filter result to pick. The average is over 2¹⁰ samples instead of 2¹⁴.

In one embodiment, the gear selection of the new rate converter depends only on time. For each gear level, it stays for a fixed amount time. The stay time for gear level n+1 is twice as the time for level n. The time to stay in each gear depends on the phase lock quality. Different initial conditions and sample conversion ratios decides how long to stay. A threshold based decision is a good implementation method.

There is no 2nd order tracking loop. Instead, the drate output is dumped into the register four times during the operation of the rate converter.

The first time is at beginning of rate conversion and it is set to the initially estimated drate.

Each time gear changes up, the current drate is dumped into a register.

The sample index buffer is not adjusted for every output sample, but instead it is adjusted when the accumulative increase is above one input sample.

Introducing the 2nd order tracking loop is the biggest change for performance compared to the converter of FIG. 2. The converter of FIG. 2 always converges, but when it converges, the sample index is not 0. That causes some variance in audio latency. If the data is dumped too often, it causes divergence. If the amount of change is reduced each time, then it becomes a 2nd order tracking loop. If it is updated every time the IIR gets a new value, it behaves as the 2nd order tracking loop of FIG. 3.

The older design of FIG. 2 updates the sample index buffer every time the accumulation across the input sample, which is unnecessary complex. The design of FIG. 3 does not update according to the same schedule.

While the above description has been focused on rate tracking, embodiments of the invention additionally include a new design for the other major block of a rate converter, the sample interpolator.

Theoretical Background

In theory, the approach is to apply a continuous time filter with certain frequency response to a discretely sampled input and then discretely resample it at another sample rate.

The frequency response of such a filter should reject the alias generated by the discrete time input samples as much as possible while maintaining the audio passband as much as possible.

For implementation simplicity, embodiments of the invention use a filter length of 80 times input sample period. For example, let's suppose the filter value is f(t) for −40T_s<t≤40T_s. Let the input samples be s_n, the output sample s_m′=Σ_k=−40⁴⁰s_└mr┘+k·f((mr−└mr┘+k)·T_s) where r is the output sample to input sample ratio.

It is conventionally difficult to get the values of filter f. The filter can be pre-computed and stored, but the memory requirement is very big. To achieve 120 dB alias rejection would require more than a million entries stored, which is an impractically large number. Instead, the whole 80 sample period may be broken into many pieces and given a polynomial for each piece to approximate the continuous time filter. One embodiment includes a 3rd order polynomial over 640 time pieces. That selection provides an alias less than 120 dB lower than signal for the entire audio pass band.

The way to generate such polynomials is first generate a vastly oversampled filter with the desired frequency response and length. Then, within each of the 640 time pieces, the points of the filter are fit to a 3rd order polynomial with minimum square error. The performance of such polynomials can be verified by computing 30 points in each time piece and look at the overall frequency response.

FIG. 4 is a graph showing passband behavior for anti-alias filters according to embodiments of the invention.

From the graph, it can be seen that the passband ripple for 6× decimation is 0.1 dB, for 4× decimation is 0.01 dB and even lower for other filters.

FIG. 5 is a graph illustrating total wrapped aliases for all anti-alias filters according to embodiments of the invention. This figure shows that the total alias for 6× decimation is between −97 dB to −100 dB. The total alias for 4× decimation is about 120 dB. The total alias for 1× filter is around −130 dB up to 10 KHz; then it goes up to −120 dB at 16 KHz and to −112 dB at 20 KHz.

For all other filters, the alias rejection varies between −130 dB to −125 dB. There are occasional strays going up to −122 dB.

FIGS. 6A-6E illustrate individual frequency responses for 1×, 1.5×, 2×, 3×, 4×, and 6× filters according to embodiments of the invention. The humps on the 1× filter and 1.5× filter is a result of the polynomial approximation. For the other filters, since the cutoff frequency is much lower, the polynomial approximation error is under the noise floor.

FIG. 7 is a block diagram of an interpolator portion of a rate controller according to embodiments of the invention. Although not illustrated, the multiplexers (muxes) are controlled by state machine.

In operation, one operation cycle is performed on each output sample. In one embodiment, each operation cycle goes through 80 samples corresponding to the filter size. For each of the 80 samples, first is coefficient generation, and then the multiplier and adding over one input sample for each audio channel. When the whole computation is over, the output sample is stored in the 2 sample FIFO buffer.

In the following example, let N be the number of audio channels.

Each operation cycle takes (N+3)*80 cycles.

Let k be the sample index between 0 and 79.

The coefficient ROM is read in the following cycles:

• • For cycle k*(N+3)+1, the coef ROM reads out the 3rd order coefficient and stores it to coef buffer;
  • For cycle k*(N+3)+2, the coef ROM read out the 2nd order coefficient and passes it to the adder;
  • For cycle k*(N+3)+3, the coef ROM read out the 1st order coefficient and passes it to the adder;
  • For cycle k*(N+3)+4, the coef ROM read out the 0th order coefficient and passes it to the adder.

The coefficient buffer is updated during cycles k*(N+3)+1 to k*(N+3)+4. It takes input from coefficient ROM in cycle k*(N+3)+1 and takes input from multiplier accumulator in other cycles.

The input RAM is read during cycles k*(N+3)+5 to k*(N+3)+N+4. Note that the last cycle of input RAM access overlaps with first cycle of coefficient ROM access. But this does not cause any problem.

The output accumulator is reset to 0 at cycle 0 and is updated during cycles k*(N+3)+5 to k*(N+3)+N+4.

The multiplier selector takes LSB of sample index during cycles k*(N+3)+2 to k*(N+3)+4 and takes input RAM during cycles k*(N+3)+5 to k*(N+3)+N+4.

The adder selector takes coefficient buffer during cycles k*(N+3)+2 to k*(N+3)+4 and takes output accumulator during cycles k*(N+3)+5 to k*(N+3)+N+4.

Let x be the LSB of sample index. The operation being done are:

Cycle k*(N+3)+1 gets c′=c₃

Cycle k*(N+3)+2 gets c′=cx+c₂=c₃x+c₂

Cycle k*(N+3)+3 gets c′=cx+c₁=c₃x²+c₂x+c₁

Cycle k*(N+3)+4 gets c′=cx+c₀=c₃x³+c₂x²+c₁x+c₀. That is the coefficient applied over all audio channels.

Cycle k*(N+3)+j, 0≤j<N, gets b_j′=b_j+c·a_k. Here b is the output accumulator for channel j and a_j,k is the input sample k for channel j.

At the end of (N+3)*80 cycles, the output accumulators are dumped into the output buffer and ready for output over output strobe.

Implementation Details

In an example implementation, for each filter, there are 640 time intervals. Each contains a 3rd order polynomial. Therefore, 2560 words are needed for each filter coefficient ROM. However, symmetry reduces the coefficient ROM. Since f (−t)=f (t), we only need to store half of the filter polynomials. That is, 1280 words per filter coefficient ROM.

There is one input buffer and one output buffer for each audio channel. Each input or output buffer contains two 24 bit words. It serves as a FIFO to temporally store the input or output sample before written by input RAM or output to the next block.

The input RAM contains 96 words of 24 bit width for each audio channel. In the 96 words, 80 are used for the anti-aliasing filter. The remaining 16 are used for possible sample jitter and input/output rate mismatch before the phaselock loop completely locks.

The address gen generates the access address for the input sample RAM and the coefficient ROM. Suppose the input buffer start address is i, the sample index is a·2²³+b·2²⁰+c when an operation cycle starts. Then the kth sample index generated is (a+i+8+k) mod 96. The coefficient ROM address is 32·(a+k)+4·b+j, here j means the coefficient order of the polynomial.

The multiplier is a 24 bit by 24 bit signed multiplier. The adder is a 28 bit adder. The output accumulator is 28 bits too. Rounding is performed before storing to the output sample buffer.

Compared to the operation of the interpolator that operates in conjunction with the rate tracker of FIG. 2, the major differences with the interpolator are:

The conventional multiplier is a pipelined multiplier that takes 12 cycles. The multiplier in the interpolator of FIG. 7 is a single cycle booth multiplier come from synthesizer.

In the old interpolator x² and x³ must be computed because of the 12 cycle multiplier latency. In the old interpolator, the new coefficient is computes as c₃x³+c₂x²+c₁x+c₀ while the new interpolator compute it as ((c₃x+c₂)x+c₁)x+c₀ which does not need to compute x² and x³.

Each operation cycle of the old interpolator requires 92N+288 cycles and 80N+240 cycles in the interpolator of FIG. 7. That means a higher clock rate is required in the old interpolator.

The old interpolator only has two filters, one for full bandwidth and the other for half bandwidth. The new design has 6 filters, supporting 1×. 1.5×, 2×, 3×, 4×, and 6× decimation. That practically allows any rate to any rate conversion. Also, the old interpolator has 2560 words for each filter and the new interpolator only has 1280 words for each filter.

The old rate converter computes all 80 coefficients before applying them. Therefore, it needs 80 word coefficient RAM. The new converter computes one coefficient and apples it to all of the channels before going to the next one. Therefore, it only needs one word to store the coefficient.

The impacts of the change include: the single cycle multiplier has about half area and a third current compare to the pipelined multiplier; not requiring x² and x³ saves area and current too; and a lower clock rate means less current.

More filters take more area which is a tradeoff, but it makes the design more flexible and is able to handle all rate to all rate conversion. Also, the total words is 7680 words for the new design compared to 5120 words for the old one.

Not requiring the coefficient RAM makes a big difference in area and current.

FIGS. 8A and 8B are graphs that illustrate amplitude response and SNR for similar rate conversions comparing an existing rate controller (labeled in the graphs as “old filter”) to a rate controller according to embodiments of the invention (labeled in the graphs as “new filter”).

FIGS. 9A and 9B are graphs that illustrate amplitude response and SNR for 1.5× decimation comparing an existing rate controller (labeled in the graphs as “old filter”) to a rate controller according to embodiments of the invention (labeled in the graphs as “new filter”).

FIGS. 10A and 10B are graphs that illustrate amplitude response and SNR for 2× decimation comparing an existing rate controller (labeled in the graphs as “old filter”) to a rate controller according to embodiments of the invention (labeled in the graphs as “new filter”).

FIGS. 11A and 11B are graphs that illustrate amplitude response and SNR for 3× decimation comparing an existing rate controller (labeled in the graphs as “old filter”) to a rate controller according to embodiments of the invention (labeled in the graphs as “new filter”).

FIGS. 12A and 12B are graphs that illustrate amplitude response and SNR for 4× decimation comparing an existing rate controller (labeled in the graphs as “old filter”) to a rate controller according to embodiments of the invention (labeled in the graphs as “new filter”).

FIGS. 13A and 13B are graphs that illustrate amplitude response and SNR for 6× decimation comparing an existing rate controller (labeled in the graphs as “old filter”) to a rate controller according to embodiments of the invention (labeled in the graphs as “new filter”).

FIGS. 14A and 14B are graphs that illustrate amplitude response and SNR for interpolation rate conversion comparing an existing rate controller (labeled in the graphs as “old filter”) to a rate controller according to embodiments of the invention (labeled in the graphs as “new filter”).

From inspection of FIGS. 8A-14B, it is seen that, for most of the cases, a rate coder according to embodiments of the invention ha have 1-2 dB higher SNR compare to the old rate coder. The exceptions are, for 6× decimation, the new rate coder has 95 dB˜100 dB SNR while the old rate coder has more than 10 dB higher SNR. This is because the new coder provides very tight alias rejection compared to the old decoder. Similarly, for 4× decimation, the new coder has about 4 dB lower SNR, which is also due to increased computation for tight alias rejection.

When the input rate is very close to integer multiple of output rate, the alias falls right on the signal. In these cases, the old and new coders have similar SNRs. That is because the alias falls almost exactly on the signal and cannot be distinguished.

Embodiments of the invention provide: up to 8 channels of audio with a sample rate <=48 KHz with a 48 MHz clock rate, and coverage of all rate to all rate conversion if the decimation rate is not more than 6×.

Embodiments of the invention may be incorporated into integrated circuits such as sound processing circuits, or other audio circuitry. In turn, the integrated circuits may be used in audio devices such as headphones, sound bars, audio docks, amplifiers, speakers, etc.

Also, although embodiments of the invention have been described using functional blocks, the block may be implemented in any physical embodiment, as is known in the art. For example blocks may be implemented in application specific integrated circuits (ASICs), FPGAs or other programmable firmware, software running on a specialized processor, software running on a general purpose processor, or any combination of the above.

Having described and illustrated the principles of the invention with reference to illustrated embodiments, it will be recognized that the illustrated embodiments may be modified in arrangement and detail without departing from such principles, and may be combined in any desired manner. And although the foregoing discussion has focused on particular embodiments, other configurations are contemplated.

In particular, even though expressions such as “according to an embodiment of the invention” or the like are used herein, these phrases are meant to generally reference embodiment possibilities, and are not intended to limit the invention to particular embodiment configurations. As used herein, these terms may reference the same or different embodiments that are combinable into other embodiments.

Consequently, in view of the wide variety of permutations to the embodiments described herein, this detailed description and accompanying material is intended to be illustrative only, and should not be taken as limiting the scope of the invention.

Claims

1. An audio processing circuit, comprising: a multiplier accumulator having a multiplier and an adder, the multiplier accumulator configured to output an output sample;a first multiplexer configured to receive a sample index and a sample input and output one of the sample index and the sample output to the multiplier of the multiplier accumulator;a second multiplexer configured to receive a coefficient and the output sample and output one of the coefficient and the output sample to the multiplier;an output accumulator configured to receive and store the output sample; anda third multiplexer configured to receive the output sample from the output accumulator and the coefficient and output one of the coefficient and output sampler to the adder.

2. The audio processing circuit of claim 1 further comprising a coefficient buffer configured to store one or more coefficients.

3. The audio processing circuit of claim 1 further comprising a sample index accumulator for determining the sample index for each output audio sample at an output sample rate different from an input sample rate.

4. The audio processing circuit of claim 3 further comprising a gain value controller configured to adjust gain values during conversion between the input sample rate and an output sample rate.

5. The audio processing circuit of claim 4 further comprising a filter selector structured to select a filter from a plurality of filters for application to audio data based on output of the gain value controller.

6. The audio processing circuit of claim 5 in which each of the plurality of filters is configured to provide a different impulse response to audio data during rate conversion.

7. The audio processing circuit of claim 6 in which the plurality of filters are infinite impulse response (IIR) filters.

8. The audio processing circuit of claim 5 in which the filter selector selects a new filter when an absolute value of an index error and a change of slope of the index error are both under or both over a threshold.

9. The audio processing circuit of claim 5 in which the filter selector selects the filter for a fixed amount of time.

10. The audio processing circuit of claim 9 in which the plurality of filters are ordered, and in which the fixed amount of time doubles for each higher order filter.

11. A method for converting an input sample having an input sample rate to an output sample having an output sample rate, comprising: selecting one of a sample index or a sample input to output to a multiplier;selecting one of a coefficient or an output of an adder to output to the multiplier;multiplying at a multiplier the selected one of the input sample or the sample index with the selected one of the coefficient or the output of an adder to generate a multiplier output;selecting one of the coefficient or an output from an output accumulator to output to the adder;adding the selected one of the output from the output accumulator or the coefficient with the output of the multiplier to generate an output sample; andstoring the output sample in the output accumulator.

12. The method of claim 11 further comprising determining the sample index for each output audio sample at an output sample rate different from an input sample rate.

13. The method of claim 12 further comprising adjusting by a gain value controller gain values during conversion between the input sample rate and an output sample rate.

14. The method of claim 13 further comprising selecting a filter from a plurality of filters for application to audio data based on output of the gain value controller.

15. The method of claim 14 further comprising selecting a new filter when an absolute value of an index error and a change of slope of the index error are both under or both over a threshold.

16. The method of claim 14 further comprising selecting the filter for a fixed amount of time.

17. A method for converting an input sample having an input sample rate to an output sample having an output sample rate, comprising: determining an index error;adjusting gain values based on the index error;generating a sample index based on the gain values;multiplying at a multiplier one of the input sample or the sample index with one of a coefficient or an output of an adder to generate a multiplier output;selecting one of the coefficient or an output from an output accumulator to output to the adder;adding the selected one of the output from the output accumulator or the coefficient with the output of the multiplier to generate an output sample; andstoring the output sample in the output accumulator.

18. The method of claim 17 further comprising selecting a filter from a plurality of filters for application to audio data based on the gain values.

19. The method of claim 18 further comprising selecting a new filter when an absolute value of the index error and a change of slope of the index error are both under or both over a threshold.

20. The method of claim 18 further comprising selecting the filter for a fixed amount of time.