Misc changes to address Robert Sparks' comments
See http://www.ietf.org/mail-archive/web/codec/current/msg02833.html Still more changes to come
This commit is contained in:
Jean-Marc Valin
parent
8365b5d00d
commit
72273000ec
9 changed files with 158 additions and 92 deletions
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@ -36,8 +36,7 @@ LIBPREFIX = lib
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LIBSUFFIX = .a
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OBJSUFFIX = .o
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CC = $(TOOLCHAIN_PREFIX)gcc$(TOOLCHAIN_SUFFIX)
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CXX = $(TOOLCHAIN_PREFIX)g++$(TOOLCHAIN_SUFFIX)
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CC = $(TOOLCHAIN_PREFIX)cc$(TOOLCHAIN_SUFFIX)
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AR = $(TOOLCHAIN_PREFIX)ar
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RANLIB = $(TOOLCHAIN_PREFIX)ranlib
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CP = $(TOOLCHAIN_PREFIX)cp
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@ -79,7 +78,6 @@ LDFLAGS += $(call ldflags-from-ldlibdirs,$(LDLIBDIRS))
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LDLIBS += $(call ldlibs-from-libs,$(LIBS))
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COMPILE.c.cmdline = $(CC) -c $(CFLAGS) -o $@ $<
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COMPILE.cpp.cmdline = $(CXX) -c $(CFLAGS) -o $@ $<
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LINK.o = $(CC) $(LDPREFLAGS) $(LDFLAGS)
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LINK.o.cmdline = $(LINK.o) $^ $(LDLIBS) -o $@$(EXESUFFIX)
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@ -6,10 +6,11 @@ If this does not work, or if you want to change the default configuration (e.g.,
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to compile for a fixed-point architecture), simply edit the options in the
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Makefile.
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To build from the git repository instead of using this draft, follow these
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To build from the git repository instead of using this RFC, follow these
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steps:
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1) Clone the repository:
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1) Clone the repository (latest implementation of this standard at the time
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of publication)
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% git clone git://git.opus-codec.org/opus.git
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% cd opus
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@ -99,8 +99,7 @@ void compute_band_energies(const CELTMode *m, const celt_sig *X, celt_ener *band
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sum = MAC16_16(sum, EXTRACT16(VSHR32(X[j+c*N],shift)),
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EXTRACT16(VSHR32(X[j+c*N],shift)));
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} while (++j<M*eBands[i+1]);
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/* We're adding one here to make damn sure we never end up with a pitch vector that's
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larger than unity norm */
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/* We're adding one here to ensure the normalized band isn't larger than unity norm */
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bandE[i+c*m->nbEBands] = EPSILON+VSHR32(EXTEND32(celt_sqrt(sum)),-shift);
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} else {
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bandE[i+c*m->nbEBands] = EPSILON;
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@ -37,19 +37,6 @@
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extern "C" {
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#endif
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/*
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ATTENTION!
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If you would like a :
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-- a utility that will handle the caching of fft objects
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-- real-only (no imaginary time component ) FFT
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-- a multi-dimensional FFT
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-- a command-line utility to perform ffts
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-- a command-line utility to perform fast-convolution filtering
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Then see kfc.h kiss_fftr.h kiss_fftnd.h fftutil.c kiss_fastfir.c
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in the tools/ directory.
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*/
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#ifdef USE_SIMD
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# include <xmmintrin.h>
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# define kiss_fft_scalar __m128
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22
celt/vq.c
22
celt/vq.c
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@ -70,14 +70,7 @@ static void exp_rotation(celt_norm *X, int len, int dir, int stride, int K, int
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opus_val16 gain, theta;
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int stride2=0;
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int factor;
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/*int i;
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if (len>=30)
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{
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for (i=0;i<len;i++)
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X[i] = 0;
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X[14] = 1;
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K=5;
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}*/
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if (2*K>=len || spread==SPREAD_NONE)
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return;
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factor = SPREAD_FACTOR[spread-1];
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@ -91,9 +84,8 @@ static void exp_rotation(celt_norm *X, int len, int dir, int stride, int K, int
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if (len>=8*stride)
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{
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stride2 = 1;
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/* This is just a simple way of computing sqrt(len/stride) with rounding.
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It's basically incrementing long as (stride2+0.5)^2 < len/stride.
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I _think_ it is bit-exact */
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/* This is just a simple (equivalent) way of computing sqrt(len/stride) with rounding.
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It's basically incrementing long as (stride2+0.5)^2 < len/stride. */
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while ((stride2*stride2+stride2)*stride + (stride>>2) < len)
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stride2++;
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}
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@ -113,13 +105,6 @@ static void exp_rotation(celt_norm *X, int len, int dir, int stride, int K, int
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exp_rotation1(X+i*len, len, stride2, s, -c);
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}
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}
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/*if (len>=30)
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{
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for (i=0;i<len;i++)
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printf ("%f ", X[i]);
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printf ("\n");
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exit(0);
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}*/
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}
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/** Takes the pitch vector and the decoded residual vector, computes the gain
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@ -233,7 +218,6 @@ unsigned alg_quant(celt_norm *X, int N, int K, int spread, int B, ec_enc *enc
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while (++j<N);
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sum = QCONST16(1.f,14);
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}
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/* Do we have sufficient accuracy here? */
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rcp = EXTRACT16(MULT16_32_Q16(K-1, celt_rcp(sum)));
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j=0; do {
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#ifdef FIXED_POINT
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@ -80,8 +80,8 @@ Opus uses both linear prediction (LP) and the Modified Discrete Cosine
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The Opus codec is a real-time interactive audio codec designed to meet the requirements
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described in <xref target="requirements"></xref>.
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It is composed of a linear
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prediction (LP)-based layer and a Modified Discrete Cosine Transform
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(MDCT)-based layer.
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prediction (LP)-based <xref target="LPC"/> layer and a Modified Discrete Cosine Transform
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(MDCT)-based <xref target="MDCT"/> layer.
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The main idea behind using two layers is that in speech, linear prediction
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techniques (such as CELP) code low frequencies more efficiently than transform
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(e.g., MDCT) domain techniques, while the situation is reversed for music and
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@ -273,8 +273,7 @@ Therefore, if an application wishes to process a signal sampled at 32 kHz,
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</t>
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<t>
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The LP layer is based on the
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<eref target='http://developer.skype.com/silk'>SILK</eref> codec
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The LP layer is based on the SILK codec
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<xref target="SILK"></xref>.
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It supports NB, MB, or WB audio and frame sizes from 10 ms to 60 ms,
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and requires an additional 5 ms look-ahead for noise shaping estimation.
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@ -290,9 +289,7 @@ This document does not serve to define that format, but those interested in the
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</t>
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<t>
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The MDCT layer is based on the
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<eref target='http://www.celt-codec.org/'>CELT</eref> codec
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<xref target="CELT"></xref>.
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The MDCT layer is based on the CELT codec <xref target="CELT"></xref>.
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It supports NB, WB, SWB, or FB audio and frame sizes from 2.5 ms to
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20 ms, and requires an additional 2.5 ms look-ahead due to the
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overlapping MDCT windows.
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@ -436,7 +433,7 @@ encoder, the complexity is selected using an integer from 0 to 10, where
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0 is the lowest complexity and 10 is the highest. Examples of
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computations for which such trade-offs may occur are:
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<list style="symbols">
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<t>The order of the pitch analysis whitening filter,</t>
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<t>The order of the pitch analysis whitening filter <xref target="Whitening"/>,</t>
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<t>The order of the short-term noise shaping filter,</t>
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<t>The number of states in delayed decision quantization of the
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residual signal, and</t>
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@ -474,9 +471,8 @@ the default. However, in some (rare) applications, constant bitrate (CBR)
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is required. There are two main reasons to operate in CBR mode:
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<list style="symbols">
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<t>When the transport only supports a fixed size for each compressed frame</t>
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<t>When security is important <spanx style="emph">and</spanx> the input audio
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not a normal conversation but is highly constrained (e.g. yes/no, recorded prompts)
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<xref target="SRTP-VBR"></xref> </t>
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<t>When encryption is used for an audio stream that is either highly constrained
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(e.g. yes/no, recorded prompts) or highly sensitive <xref target="SRTP-VBR"></xref> </t>
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</list>
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When low-latency transmission is required over a relatively slow connection, then
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@ -734,9 +730,9 @@ This makes, for example, a 2-byte code 2 packet with a second byte in the range
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</figure>
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</section>
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<section title="Code 3: An Arbitrary Number of Frames in the Packet">
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<section title="Code 3: A Signaled Number of Frames in the Packet">
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<t>
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Code 3 packets may encode an arbitrary number of frames, as well as additional
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Code 3 packets signal the number of frames, as well as additional
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padding, called "Opus padding" to indicate that this padding is added at the
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Opus layer, rather than at the transport layer.
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Code 3 packets MUST have at least 2 bytes.
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@ -1271,10 +1267,10 @@ The raw bits used by the CELT layer are packed at the end of the packet, with
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The reference implementation reads them using ec_dec_bits() (entdec.c).
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Because the range decoder must read several bytes ahead in the stream, as
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described in <xref target="range-decoder-renorm"/>, the input consumed by the
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raw bits MAY overlap with the input consumed by the range coder, and a decoder
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raw bits may overlap with the input consumed by the range coder, and a decoder
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MUST allow this.
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The format should render it impossible to attempt to read more raw bits than
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there are actual bits in the frame, though a decoder MAY wish to check for
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there are actual bits in the frame, though a decoder may wish to check for
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this and report an error.
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</t>
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</section>
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@ -1388,9 +1384,9 @@ Reading raw bits increases nbits_total by the number of raw bits read.
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<section anchor="ec_tell" title="ec_tell()">
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<t>
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The whole number of bits buffered in rng may be estimated via l = ilog(rng).
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The whole number of bits buffered in rng may be estimated via lg = ilog(rng).
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ec_tell() then becomes a simple matter of removing these bits from the total.
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It returns (nbits_total - l).
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It returns (nbits_total - lg).
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</t>
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<t>
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In a newly initialized decoder, before any symbols have been read, this reports
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@ -1403,7 +1399,7 @@ This is the bit reserved for termination of the encoder.
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<t>
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ec_tell_frac() estimates the number of bits buffered in rng to fractional
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precision.
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Since rng must be greater than 2**23 after renormalization, l must be at least
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Since rng must be greater than 2**23 after renormalization, lg must be at least
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24.
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Let
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<figure align="center">
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@ -1414,7 +1410,7 @@ r_Q15 = rng >> (l-16) ,
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</figure>
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so that 32768 <= r_Q15 < 65536, an unsigned Q15 value representing the
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fractional part of rng.
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Then the following procedure can be used to add one bit of precision to l.
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Then the following procedure can be used to add one bit of precision to lg.
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First, update
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<figure align="center">
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<artwork align="center">
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@ -1422,11 +1418,11 @@ First, update
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r_Q15 = (r_Q15*r_Q15) >> 15 .
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]]></artwork>
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</figure>
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Then add the 16th bit of r_Q15 to l via
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Then add the 16th bit of r_Q15 to lg via
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<figure align="center">
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<artwork align="center">
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<![CDATA[
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l = 2*l + (r_Q15 >> 16) .
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lg = 2*lg + (r_Q15 >> 16) .
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]]></artwork>
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</figure>
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Finally, if this bit was a 1, reduce r_Q15 by a factor of two via
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@ -1439,8 +1435,8 @@ r_Q15 = r_Q15 >> 1 ,
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so that it once again lies in the range 32768 <= r_Q15 < 65536.
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</t>
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<t>
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This procedure is repeated three times to extend l to 1/8th bit precision.
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ec_tell_frac() then returns (nbits_total*8 - l).
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This procedure is repeated three times to extend lg to 1/8th bit precision.
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ec_tell_frac() then returns (nbits_total*8 - lg).
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</t>
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</section>
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@ -5301,7 +5297,7 @@ resolution is shown in the tables below.
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<t>
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A negative TF adjustment means that the temporal resolution is increased,
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while a positive TF adjustment means that the frequency resolution is increased.
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Changes in TF resolution are implemented using the Hadamard transform. To increase
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Changes in TF resolution are implemented using the Hadamard transform <xref target="Hadamard"/>. To increase
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the time resolution by N, N "levels" of the Hadamard transform are applied to the
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decoded vector for each interleaved MDCT vector. To increase the frequency resolution
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(assumes a transient frame), then N levels of the Hadamard transform are applied
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@ -5459,9 +5455,9 @@ artifact than if the frame were dropped after decoding.
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<t>
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A decoder MAY employ a more sophisticated drift compensation method. For
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example, the
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<eref target='http://code.google.com/p/webrtc/source/browse/trunk/src/modules/audio_coding/NetEQ/main/source/?r=583'>NetEQ component</eref>
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<xref target='Google-NetEQ'>NetEQ component</xref>
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of the
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<eref target='http://code.google.com/p/webrtc/'>WebRTC.org codebase</eref>
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<xref target='Google-WebRTC'>Google WebRTC codebase</xref>
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compensates for drift by adding or removing
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one period when the signal is highly periodic. The reference implementation of
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Opus allows a caller to learn whether the current frame's signal is highly
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@ -6822,7 +6818,7 @@ of the scalar quantizer, and as a result the quantization error of
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each value depends on the quantization decision of the previous value.
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This dependency is exploited by the delayed decision mechanism to
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search for a quantization sequency with best R/D performance
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with a Viterbi-like algorithm .
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with a Viterbi-like algorithm <xref target="Viterbi"/>.
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The quantizer processes the residual LSF vector in reverse order
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(i.e., it starts with the highest residual LSF value).
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This is done because the prediction works slightly
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@ -7274,14 +7270,15 @@ are built and <vector path> is the directory containing the test vectors.
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<section title="Opus Custom">
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<t>
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To complement the Opus specification, the "Opus Custom" codec is defined to
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Opus Custom is an OPTIONAL part of the specification that is defined to
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handle special sample rates and frame rates that are not supported by the
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main Opus specification. Use of Opus Custom is discouraged for all but very
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special applications for which a frame size different from 2.5, 5, 10, or 20 ms is
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needed (for either complexity or latency reasons). Such applications will not
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be compatible with the "main" Opus codec. In Opus Custom operation,
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only the CELT layer is available, which is available using the celt_* function
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calls in celt.h.
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needed (for either complexity or latency reasons). Because Opus Custom is
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optional, applications using that part of the specification may not be compatible
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with other applications implementing Opus. In Opus Custom operation,
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only the CELT layer is available, using the opus_custom_* function
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calls in opus_custom.h.
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</t>
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</section>
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@ -7338,7 +7335,7 @@ Sending the decoder packets generated by a version of the reference encoder
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</t>
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</list>
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In all of the conditions above, both the encoder and the decoder were run
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inside the <eref target="http://valgrind.org/">Valgrind</eref> memory
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inside the <xref target="Valgrind">Valgrind</xref> memory
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debugger, which tracks reads and writes to invalid memory regions as well as
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the use of uninitialized memory.
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There were no errors reported on any of the tested conditions.
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@ -7407,7 +7404,7 @@ name of work, or endorsement information.</t>
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<format type='TXT' target='http://tools.ietf.org/rfc/rfc6366.txt' />
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</reference>
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<reference anchor='SILK'>
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<reference anchor='SILK' target='http://developer.skype.com/silk'>
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<front>
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<title>SILK Speech Codec</title>
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<author initials='K.' surname='Vos' fullname='K. Vos'>
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@ -7442,7 +7439,7 @@ Robust and Efficient Quantization of Speech LSP Parameters Using Structured Vect
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<seriesInfo name="ICASSP-1991, Proc. IEEE Int. Conf. Acoust., Speech, Signal Processing, pp. 641-644, October" value="1991"/>
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</reference>
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<reference anchor='CELT'>
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<reference anchor='CELT' target='http://celt-codec.org/'>
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<front>
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<title>Constrained-Energy Lapped Transform (CELT) Codec</title>
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<author initials='J-M.' surname='Valin' fullname='J-M. Valin'>
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@ -7472,8 +7469,8 @@ Robust and Efficient Quantization of Speech LSP Parameters Using Structured Vect
|
|||
<abstract>
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<t></t>
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||||
</abstract></front>
|
||||
<seriesInfo name='Internet-Draft' value='draft-ietf-avtcore-srtp-vbr-audio-03' />
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||||
<format type='TXT' target='http://tools.ietf.org/html/draft-ietf-avtcore-srtp-vbr-audio-03' />
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<seriesInfo name='RFC' value='6562' />
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<format type='TXT' target='http://tools.ietf.org/html/rfc6562' />
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</reference>
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||||
|
||||
<reference anchor='DOS'>
|
||||
|
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@ -7536,6 +7533,98 @@ Robust and Efficient Quantization of Speech LSP Parameters Using Structured Vect
|
|||
<seriesInfo name="IEEE Trans. on Information Theory, Vol. 32" value="pp. 568-583" />
|
||||
</reference>
|
||||
|
||||
<reference anchor="Valgrind" target="http://valgrind.org/">
|
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<front>
|
||||
<title>Valgrind website</title>
|
||||
<author></author>
|
||||
</front>
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</reference>
|
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|
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<reference anchor="Google-NetEQ" target="http://code.google.com/p/webrtc/source/browse/trunk/src/modules/audio_coding/NetEQ/main/source/?r=583">
|
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<front>
|
||||
<title>Google NetEQ code</title>
|
||||
<author></author>
|
||||
</front>
|
||||
</reference>
|
||||
|
||||
<reference anchor="Google-WebRTC" target="http://code.google.com/p/webrtc/">
|
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<front>
|
||||
<title>Google WebRTC code</title>
|
||||
<author></author>
|
||||
</front>
|
||||
</reference>
|
||||
|
||||
|
||||
<reference anchor="Opus-git" target="git://git.xiph.org/opus.git">
|
||||
<front>
|
||||
<title>Opus Git Repository</title>
|
||||
<author></author>
|
||||
</front>
|
||||
</reference>
|
||||
|
||||
<reference anchor="Opus-website" target="http://opus-codec.org/">
|
||||
<front>
|
||||
<title>Opus website</title>
|
||||
<author></author>
|
||||
</front>
|
||||
</reference>
|
||||
|
||||
<reference anchor="Vectors-website" target="http://opus-codec.org/testvectors/">
|
||||
<front>
|
||||
<title>Opus Testvectors (webside)</title>
|
||||
<author></author>
|
||||
</front>
|
||||
</reference>
|
||||
|
||||
<reference anchor="Vectors-proc" target="http://www.ietf.org/proceedings/83/slides/slides-83-codec-0.gz">
|
||||
<front>
|
||||
<title>Opus Testvectors (proceedings)</title>
|
||||
<author></author>
|
||||
</front>
|
||||
</reference>
|
||||
|
||||
<reference anchor="Hadamard" target="http://en.wikipedia.org/wiki/Hadamard_transform">
|
||||
<front>
|
||||
<title>Hadamard Transform</title>
|
||||
<author><organization>Wikipedia</organization></author>
|
||||
</front>
|
||||
</reference>
|
||||
|
||||
<reference anchor="Viterbi" target="http://en.wikipedia.org/wiki/Viterbi_algorithm">
|
||||
<front>
|
||||
<title>Viterbi Algorithm</title>
|
||||
<author><organization>Wikipedia</organization></author>
|
||||
</front>
|
||||
</reference>
|
||||
|
||||
<reference anchor="Whitening" target="http://en.wikipedia.org/wiki/White_noise">
|
||||
<front>
|
||||
<title>White Noise</title>
|
||||
<author><organization>Wikipedia</organization></author>
|
||||
</front>
|
||||
</reference>
|
||||
|
||||
<reference anchor="LPC" target="http://en.wikipedia.org/wiki/Linear_prediction">
|
||||
<front>
|
||||
<title>Linear Prediction</title>
|
||||
<author><organization>Wikipedia</organization></author>
|
||||
</front>
|
||||
</reference>
|
||||
|
||||
<reference anchor="MDCT" target="http://en.wikipedia.org/wiki/Modified_discrete_cosine_transform">
|
||||
<front>
|
||||
<title>Modified Discrete Cosine Transform</title>
|
||||
<author><organization>Wikipedia</organization></author>
|
||||
</front>
|
||||
</reference>
|
||||
|
||||
<reference anchor="FFT" target="http://en.wikipedia.org/wiki/Fast_Fourier_transform">
|
||||
<front>
|
||||
<title>Fast Fourier Transform</title>
|
||||
<author><organization>Wikipedia</organization></author>
|
||||
</front>
|
||||
</reference>
|
||||
|
||||
</references>
|
||||
|
||||
<section anchor="ref-implementation" title="Reference Implementation">
|
||||
|
|
@ -7551,7 +7640,7 @@ available in the README file.
|
|||
<t>The implementation can be compiled with either a C89 or a C99
|
||||
compiler. It is reasonably optimized for most platforms such that
|
||||
only architecture-specific optimizations are likely to be useful.
|
||||
The FFT used is a slightly modified version of the KISS-FFT library,
|
||||
The FFT <xref target="FFT"/> used is a slightly modified version of the KISS-FFT library,
|
||||
but it is easy to substitute any other FFT library.
|
||||
</t>
|
||||
|
||||
|
|
@ -7586,7 +7675,7 @@ following command line:
|
|||
|
||||
<list style="symbols">
|
||||
<t><![CDATA[
|
||||
cat draft-ietf-codec-opus.txt | grep '^\ \ \ ###' | sed -e 's/\s\s\s###//' | base64 -d > opus_source.tar.gz
|
||||
cat draft-ietf-codec-opus.txt | grep '^\ \ \ ###' | sed -e 's/...###//' | base64 -d > opus_source.tar.gz
|
||||
]]></t>
|
||||
<t>
|
||||
tar xzvf opus_source.tar.gz
|
||||
|
|
@ -7594,11 +7683,19 @@ tar xzvf opus_source.tar.gz
|
|||
<t>cd opus_source</t>
|
||||
<t>make</t>
|
||||
</list>
|
||||
On systems where the provided Makefile does not work, the following command line may be used to compile
|
||||
the source code:
|
||||
<list style="symbols">
|
||||
<t><![CDATA[
|
||||
cc -O2 -g -o opus_demo src/opus_demo.c `cat *.mk | grep -v fixed | sed -e 's/.*=//' -e 's/\\\\//'` -DOPUS_BUILD -Iinclude -Icelt -Isilk -Isilk/float -Drestrict= -lm
|
||||
]]></t></list>
|
||||
</t>
|
||||
|
||||
<t>
|
||||
On systems where the base64 utility is not present, the following commands can be used instead:
|
||||
<list style="symbols">
|
||||
<t><![CDATA[
|
||||
cat draft-ietf-codec-opus.txt | grep '^\ \ \ ###' | sed -e 's/\s\s\s###//' > opus.b64
|
||||
cat draft-ietf-codec-opus.txt | grep '^\ \ \ ###' | sed -e 's/...###//' > opus.b64
|
||||
]]></t>
|
||||
<t>openssl base64 -d -in opus.b64 > opus_source.tar.gz</t>
|
||||
</list>
|
||||
|
|
@ -7606,12 +7703,13 @@ cat draft-ietf-codec-opus.txt | grep '^\ \ \ ###' | sed -e 's/\s\s\s###//' > opu
|
|||
</t>
|
||||
</section>
|
||||
|
||||
<section title="Development Versions">
|
||||
<section title="Up-to-date Implementation">
|
||||
<t>
|
||||
The current development version of the source code is available in a
|
||||
<eref target='git://git.opus-codec.org/opus.git'>Git repository</eref>.
|
||||
Development snapshots are provided at
|
||||
<eref target='http://opus-codec.org/'/>.
|
||||
As of the time of publication of this memo, up-to-date source code implementing
|
||||
this standard is available in a
|
||||
<xref target='Opus-git'>Git repository</xref>.
|
||||
Releases and other resources are available at
|
||||
<xref target='Opus-website'/>.
|
||||
</t>
|
||||
</section>
|
||||
|
||||
|
|
@ -7624,9 +7722,8 @@ Development snapshots are provided at
|
|||
<section anchor="test-vectors" title="Test Vectors">
|
||||
<t>
|
||||
Because of size constraints, the Opus test vectors are not distributed in this
|
||||
draft. They are available from the Opus codec website at
|
||||
<eref target="http://opus-codec.org/testvectors/"/> and will also be made available
|
||||
in IETF meeting proceedings. These test vectors were created specifically to exercise
|
||||
draft. They are available in the proceedings of the 83th IETF meeting (Paris) <xref target="Vectors-proc"/> and from the Opus codec website at
|
||||
<xref target="Vectors-website"/>. These test vectors were created specifically to exercise
|
||||
all aspects of the decoder and therefore the audio quality of the decoded output is
|
||||
significantly lower than what Opus can achieve in normal operation.
|
||||
</t>
|
||||
|
|
|
|||
|
|
@ -120,7 +120,7 @@ extern "C" {
|
|||
#define OPUS_AUTO -1000 /**<Auto/default setting @hideinitializer*/
|
||||
#define OPUS_BITRATE_MAX -1 /**<Maximum bitrate @hideinitializer*/
|
||||
|
||||
/** Best for "standard" VoIP/videoconference applications where listening quality and intelligibility matter most
|
||||
/** Best for most VoIP/videoconference applications where listening quality and intelligibility matter most
|
||||
* @hideinitializer */
|
||||
#define OPUS_APPLICATION_VOIP 2048
|
||||
/** Best for broadcast/high-fidelity application where the decoded audio should be as close as possible to the input
|
||||
|
|
|
|||
|
|
@ -292,7 +292,7 @@ static int opus_decode_frame(OpusDecoder *st, const unsigned char *data,
|
|||
if (st->prev_mode==MODE_CELT_ONLY)
|
||||
silk_InitDecoder( silk_dec );
|
||||
|
||||
/* The SILK PLC cannot support produce frames of less than 10 ms */
|
||||
/* The SILK PLC cannot produce frames of less than 10 ms */
|
||||
st->DecControl.payloadSize_ms = IMAX(10, 1000 * audiosize / st->Fs);
|
||||
|
||||
if (data != NULL)
|
||||
|
|
@ -574,7 +574,7 @@ static int opus_packet_parse_impl(const unsigned char *data, int len,
|
|||
last_size = len-size[0];
|
||||
break;
|
||||
/* Multiple CBR/VBR frames (from 0 to 120 ms) */
|
||||
case 3:
|
||||
default: /*case 3:*/
|
||||
if (len<1)
|
||||
return OPUS_INVALID_PACKET;
|
||||
/* Number of frames encoded in bits 0 to 5 */
|
||||
|
|
|
|||
|
|
@ -36,7 +36,7 @@ echo Testing mono
|
|||
echo "=============="
|
||||
echo
|
||||
|
||||
for file in `seq -w 1 11`
|
||||
for file in 01 02 03 04 05 06 07 08 09 10 11 12
|
||||
do
|
||||
if [ -e $VECTOR_PATH/testvector$file.bit ]; then
|
||||
echo Testing testvector$file
|
||||
|
|
@ -66,7 +66,7 @@ echo Testing stereo
|
|||
echo "=============="
|
||||
echo
|
||||
|
||||
for file in `seq -w 1 11`
|
||||
for file in 01 02 03 04 05 06 07 08 09 10 11 12
|
||||
do
|
||||
if [ -e $VECTOR_PATH/testvector$file.bit ]; then
|
||||
echo Testing testvector$file
|
||||
|
|
|
|||
Loading…
Add table
Add a link
Reference in a new issue