1. Only for fixed point on x86 platform (32bit and 64bit, uses SIMD
intrinsics up to SSE4.2)
2. Use "configure --enable-fixed-point --enable-intrinsics" to enable
optimization, default is disabled.
3. Official test cases are verified and passed.
Signed-off-by: Timothy B. Terriberry <tterribe@xiph.org>
By not reinitializing the stereo state during a switch, the old stereo
memory would create a false impulse (i.e. single sample) in the decoded audio.
This change affects the normative part of the decoder. Fortunately,
the modified decoder is still compliant with the specification because
it still easily passes the testvectors. For example, for the float decoder
at 48 kHz, the opus_compare (arbitrary) "quality score" changes from
from 99.9333% to 99.925%.
This allows the decoder to be compiled with
NONTHREADSAFE_PSEUDOSTACK to move the memory for large buffers off
the stack for devices where it is very limited.
This patch only attempts to do this for the decoder.
The encoder still requires more than 10 kB of stack.
Signed-off-by: Gregory Maxwell <greg@xiph.org>
silk_get_TOC is unused, unreachable, and not useful without some
packet pre-processing, but until we remove it it should be correct.
This was also reported by Mozilla.
- Safer gain clamping for PLC
- Makes opus_decoder_get_nb_samples() report an error on invalid ToC
- Giving a free license to the text ofthe draft (not just the code)
decoder:
- fixed incorrect scaling of filter states for the smallest quantization
step sizes
- NLSF2A now limits the prediction gain of LPC filters
encoder:
- increased damping of LTP coefficients in LTP analysis
- increased white noise fraction in noise shaping LPC analysis
- introduced maximum total prediction gain. Used by Burg's method to
exit early if prediction gain is exceeded. This improves packet
loss robustness and numerical robustness in Burg's method
- Prefiltered signal is now in int32 Q10 domain, from int16 Q0
- Increased max number of iterations in CBR gain control loop from 5 to 6
- Removed useless code from LTP scaling control
- Optimization: smarter LPC loop unrolling
- Switched default win32 compile mode to be floating-point
resampler:
- made resampler have constant delay of 0.75 ms; removed delay
compensation from silk code.
- removed obsolete table entries (~850 Bytes)
- increased downsampling filter order from 16 to 18/24/36 (depending on
frequency ratio)
- reoptimized filter coefficients
Neither the encoder nor decoder were incrementing the side-channel
index for a mid-only frame.
Unfortunately, they used that index to index the VAD flags and LBRR
flags.
This made the VAD flags for the side channel particularly useless,
as you couldn't tell which frame a flag belonged to without
decoding most of the packet.
It also desynched the LBRR information, as look at the wrong LBRR
flags to decide when it had to code a mid-only flag.
If some frames were skipped in the side channel, then the last few
VAD flags and LBRR flags would be garbage, but still get encoded.
This patch fixes this by continuing to increment nFramesDecoded or
nFramesEncoded, even when skipping a frame in the side channel.
This makes the side-channel VAD and LBRR flags correspond to the
correct time periods for frames greater than 20 ms.
It also fixes a bug where if DTX was not used on the packet where
the side channel got turned off, it would never get used again
until the encoder attempted to encode something in the side
channel again.
b24e5746 introduced changes to LastGainIndex which broke
conditional coding for side frames after a mid-only frame (i.e.,
in a 60 ms frame where the side is coded, not coded, then coded
again).
These rules were a mess in general, however, because the side
channel state kept a different nFramesDecoded count from the mid
channel state, and had no way to tell if the prior side frame was
coded.
This patch attempts to rationalize them by moving the conditional
coding decision up to the top level, where all this information is
available.
The first coded side frame after an uncoded side frame now always
uses independent coding.
If such a frame is also not the first side frame in an Opus frame,
then it doesn't include an LTP scaling parameter (because the LTP
state is well-defined).
- There was a bug where the decoder resampler was not properly initialized
when fs_kHz == API_fs_kHz. In that case the resampler would continue to
upsample, and the output was corrupt.
- The delay value in the decoder was taken from the state before it was
potentially updated. This caused the decoder to apply the new dalay value one
frame late
- The encoder and decoder states are now updated more consistently, when
the sampling rate changes (pesq liked these changes)
- Properly resetting the side channel encoder and decoder for the first
frame with side coding active again
- Faster updating the "ratio" value in the LR_to_MS() code for large
prediction values means that for certain extreme/artificial input
signals the output looks better
- compile warning in opus_decoder.c
- decoder state reduced by ~3 kB (by merging buffers, as Tim suggested)
- some minor decoder optimizations (only the PLC is non-bit exact, so should be ok)
Adds SILK delay compensation that depends on encode and decode sampling
rate, as well as SILK internal coding rate. This ensures that the SILK
part of Opus is always in sync with the CELT part no matter what the
sampling rates are. It also increases the resampling delay to 1.15 ms
(was previously 0.48 ms).