Optimization of AMR NB and WB decoders for MIPS

AMR NB and WB decoders are optimized for MIPS architecture.
Appropriate Makefiles are changed accordingly.

Cnfigure script is changed in order to support optimizations.
 Optimizations are enabled by default when compiling is done for
  mips architecture.
 Appropriate cflags are automatically set.
 Support for several mips CPUs is added in configure script.

New ffmpeg options are added for disabling optimizations.

The FFMPEG option --disable-mipsfpu disables MIPS floating point
 optimizations.
The FFMPEG option --disable-mips32r2 disables MIPS32R2
 optimizations.
The FFMPEG option --disable-mipsdspr1 disables MIPS DSP ASE R1
 optimizations.
The FFMPEG option --disable-mipsdspr2 disables MIPS DSP ASE R2
 optimizations.

Signed-off-by: Nedeljko Babic <nbabic@mips.com>
Reviewed-by: Vitor Sessak <vitor1001@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

1 files changed, 35 insertions, 19 deletions

diff --git a/libavcodec/amrnbdec.c b/libavcodec/amrnbdec.c
index 6b658c0a1a..46e4856beb 100644
--- a/libavcodec/amrnbdec.c
+++ b/libavcodec/amrnbdec.c
@@ -136,6 +136,11 @@ typedef struct AMRContext {
 
     float samples_in[LP_FILTER_ORDER + AMR_SUBFRAME_SIZE]; ///< floating point samples
 
+    ACELPFContext                     acelpf_ctx; ///< context for filters for ACELP-based codecs
+    ACELPVContext                     acelpv_ctx; ///< context for vector operations for ACELP-based codecs
+    CELPFContext                       celpf_ctx; ///< context for filters for CELP-based codecs
+    CELPMContext                       celpm_ctx; ///< context for fixed point math operations
+
 } AMRContext;
 
 /** Double version of ff_weighted_vector_sumf() */
@@ -171,6 +176,11 @@ static av_cold int amrnb_decode_init(AVCodecContext *avctx)
     avcodec_get_frame_defaults(&p->avframe);
     avctx->coded_frame = &p->avframe;
 
+    ff_acelp_filter_init(&p->acelpf_ctx);
+    ff_acelp_vectors_init(&p->acelpv_ctx);
+    ff_celp_filter_init(&p->celpf_ctx);
+    ff_celp_math_init(&p->celpm_ctx);
+
     return 0;
 }
 
@@ -214,15 +224,16 @@ static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf,
  * Interpolate the LSF vector (used for fixed gain smoothing).
  * The interpolation is done over all four subframes even in MODE_12k2.
  *
+ * @param[in]     ctx       The Context
  * @param[in,out] lsf_q     LSFs in [0,1] for each subframe
  * @param[in]     lsf_new   New LSFs in [0,1] for subframe 4
  */
-static void interpolate_lsf(float lsf_q[4][LP_FILTER_ORDER], float *lsf_new)
+static void interpolate_lsf(ACELPVContext *ctx, float lsf_q[4][LP_FILTER_ORDER], float *lsf_new)
 {
     int i;
 
     for (i = 0; i < 4; i++)
-        ff_weighted_vector_sumf(lsf_q[i], lsf_q[3], lsf_new,
+        ctx->weighted_vector_sumf(lsf_q[i], lsf_q[3], lsf_new,
                                 0.25 * (3 - i), 0.25 * (i + 1),
                                 LP_FILTER_ORDER);
 }
@@ -266,7 +277,7 @@ static void lsf2lsp_for_mode12k2(AMRContext *p, double lsp[LP_FILTER_ORDER],
     ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
 
     if (update)
-        interpolate_lsf(p->lsf_q, lsf_q);
+        interpolate_lsf(&p->acelpv_ctx, p->lsf_q, lsf_q);
 
     ff_acelp_lsf2lspd(lsp, lsf_q, LP_FILTER_ORDER);
 }
@@ -329,7 +340,7 @@ static void lsf2lsp_3(AMRContext *p)
     ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
 
     // store data for computing the next frame's LSFs
-    interpolate_lsf(p->lsf_q, lsf_q);
+    interpolate_lsf(&p->acelpv_ctx, p->lsf_q, lsf_q);
     memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
 
     ff_acelp_lsf2lspd(p->lsp[3], lsf_q, LP_FILTER_ORDER);
@@ -395,7 +406,8 @@ static void decode_pitch_vector(AMRContext *p,
 
     /* Calculate the pitch vector by interpolating the past excitation at the
        pitch lag using a b60 hamming windowed sinc function.   */
-    ff_acelp_interpolatef(p->excitation, p->excitation + 1 - pitch_lag_int,
+    p->acelpf_ctx.acelp_interpolatef(p->excitation,
+                          p->excitation + 1 - pitch_lag_int,
                           ff_b60_sinc, 6,
                           pitch_lag_frac + 6 - 6*(pitch_lag_frac > 0),
                           10, AMR_SUBFRAME_SIZE);
@@ -780,12 +792,12 @@ static int synthesis(AMRContext *p, float *lpc,
         for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
             p->pitch_vector[i] *= 0.25;
 
-    ff_weighted_vector_sumf(excitation, p->pitch_vector, fixed_vector,
+    p->acelpv_ctx.weighted_vector_sumf(excitation, p->pitch_vector, fixed_vector,
                             p->pitch_gain[4], fixed_gain, AMR_SUBFRAME_SIZE);
 
     // emphasize pitch vector contribution
     if (p->pitch_gain[4] > 0.5 && !overflow) {
-        float energy = ff_dot_productf(excitation, excitation,
+        float energy = p->celpm_ctx.dot_productf(excitation, excitation,
                                        AMR_SUBFRAME_SIZE);
         float pitch_factor =
             p->pitch_gain[4] *
@@ -800,7 +812,8 @@ static int synthesis(AMRContext *p, float *lpc,
                                                 AMR_SUBFRAME_SIZE);
     }
 
-    ff_celp_lp_synthesis_filterf(samples, lpc, excitation, AMR_SUBFRAME_SIZE,
+    p->celpf_ctx.celp_lp_synthesis_filterf(samples, lpc, excitation,
+                                 AMR_SUBFRAME_SIZE,
                                  LP_FILTER_ORDER);
 
     // detect overflow
@@ -846,10 +859,11 @@ static void update_state(AMRContext *p)
 /**
  * Get the tilt factor of a formant filter from its transfer function
  *
+ * @param p     The Context
  * @param lpc_n LP_FILTER_ORDER coefficients of the numerator
  * @param lpc_d LP_FILTER_ORDER coefficients of the denominator
  */
-static float tilt_factor(float *lpc_n, float *lpc_d)
+static float tilt_factor(AMRContext *p, float *lpc_n, float *lpc_d)
 {
     float rh0, rh1; // autocorrelation at lag 0 and 1
 
@@ -859,11 +873,12 @@ static float tilt_factor(float *lpc_n, float *lpc_d)
 
     hf[0] = 1.0;
     memcpy(hf + 1, lpc_n, sizeof(float) * LP_FILTER_ORDER);
-    ff_celp_lp_synthesis_filterf(hf, lpc_d, hf, AMR_TILT_RESPONSE,
+    p->celpf_ctx.celp_lp_synthesis_filterf(hf, lpc_d, hf,
+                                 AMR_TILT_RESPONSE,
                                  LP_FILTER_ORDER);
 
-    rh0 = ff_dot_productf(hf, hf,     AMR_TILT_RESPONSE);
-    rh1 = ff_dot_productf(hf, hf + 1, AMR_TILT_RESPONSE - 1);
+    rh0 = p->celpm_ctx.dot_productf(hf, hf,     AMR_TILT_RESPONSE);
+    rh1 = p->celpm_ctx.dot_productf(hf, hf + 1, AMR_TILT_RESPONSE - 1);
 
     // The spec only specifies this check for 12.2 and 10.2 kbit/s
     // modes. But in the ref source the tilt is always non-negative.
@@ -883,7 +898,7 @@ static void postfilter(AMRContext *p, float *lpc, float *buf_out)
     int i;
     float *samples          = p->samples_in + LP_FILTER_ORDER; // Start of input
 
-    float speech_gain       = ff_dot_productf(samples, samples,
+    float speech_gain       = p->celpm_ctx.dot_productf(samples, samples,
                                               AMR_SUBFRAME_SIZE);
 
     float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER];  // Output of pole filter
@@ -904,16 +919,16 @@ static void postfilter(AMRContext *p, float *lpc, float *buf_out)
     }
 
     memcpy(pole_out, p->postfilter_mem, sizeof(float) * LP_FILTER_ORDER);
-    ff_celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples,
+    p->celpf_ctx.celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples,
                                  AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
     memcpy(p->postfilter_mem, pole_out + AMR_SUBFRAME_SIZE,
            sizeof(float) * LP_FILTER_ORDER);
 
-    ff_celp_lp_zero_synthesis_filterf(buf_out, lpc_n,
+    p->celpf_ctx.celp_lp_zero_synthesis_filterf(buf_out, lpc_n,
                                       pole_out + LP_FILTER_ORDER,
                                       AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
 
-    ff_tilt_compensation(&p->tilt_mem, tilt_factor(lpc_n, lpc_d), buf_out,
+    ff_tilt_compensation(&p->tilt_mem, tilt_factor(p, lpc_n, lpc_d), buf_out,
                          AMR_SUBFRAME_SIZE);
 
     ff_adaptive_gain_control(buf_out, buf_out, speech_gain, AMR_SUBFRAME_SIZE,
@@ -990,7 +1005,7 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data,
 
         p->fixed_gain[4] =
             ff_amr_set_fixed_gain(fixed_gain_factor,
-                       ff_dot_productf(p->fixed_vector, p->fixed_vector,
+                       p->celpm_ctx.dot_productf(p->fixed_vector, p->fixed_vector,
                                        AMR_SUBFRAME_SIZE)/AMR_SUBFRAME_SIZE,
                        p->prediction_error,
                        energy_mean[p->cur_frame_mode], energy_pred_fac);
@@ -1034,7 +1049,8 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data,
         update_state(p);
     }
 
-    ff_acelp_apply_order_2_transfer_function(buf_out, buf_out, highpass_zeros,
+    p->acelpf_ctx.acelp_apply_order_2_transfer_function(buf_out,
+                                             buf_out, highpass_zeros,
                                              highpass_poles,
                                              highpass_gain * AMR_SAMPLE_SCALE,
                                              p->high_pass_mem, AMR_BLOCK_SIZE);
@@ -1045,7 +1061,7 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data,
      * for fixed_gain_smooth.
      * The specification has an incorrect formula: the reference decoder uses
      * qbar(n-1) rather than qbar(n) in section 6.1(4) equation 71. */
-    ff_weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3],
+    p->acelpv_ctx.weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3],
                             0.84, 0.16, LP_FILTER_ORDER);
 
     *got_frame_ptr   = 1;