x86: Refactor PSWAPD fallback implementations and port to cpuflags

[ffmpeg] / libavcodec / amrnbdec.c

diff --git a/libavcodec/amrnbdec.c b/libavcodec/amrnbdec.c
index a7d0b4e33748375d1a2ed916490b5cb5fa8e810b..21966456a0f12b98f11e1bd5582a427b05ea9db0 100644 (file)
--- a/libavcodec/amrnbdec.c
+++ b/libavcodec/amrnbdec.c
@@ -43,10 +43,10 @@
 #include <string.h>
 #include <math.h>
 
+#include "libavutil/audioconvert.h"
 #include "avcodec.h"
-#include "get_bits.h"
+#include "dsputil.h"
 #include "libavutil/common.h"
-#include "celp_math.h"
 #include "celp_filters.h"
 #include "acelp_filters.h"
 #include "acelp_vectors.h"
@@ -155,7 +155,15 @@ static av_cold int amrnb_decode_init(AVCodecContext *avctx)
     AMRContext *p = avctx->priv_data;
     int i;
 
-    avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+    if (avctx->channels > 1) {
+        av_log_missing_feature(avctx, "multi-channel AMR", 0);
+        return AVERROR_PATCHWELCOME;
+    }
+
+    avctx->channels       = 1;
+    avctx->channel_layout = AV_CH_LAYOUT_MONO;
+    avctx->sample_rate    = 8000;
+    avctx->sample_fmt     = AV_SAMPLE_FMT_FLT;
 
     // p->excitation always points to the same position in p->excitation_buf
     p->excitation = &p->excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1];
@@ -189,16 +197,11 @@ static av_cold int amrnb_decode_init(AVCodecContext *avctx)
 static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf,
                                   int buf_size)
 {
-    GetBitContext gb;
     enum Mode mode;
 
-    init_get_bits(&gb, buf, buf_size * 8);
-
     // Decode the first octet.
-    skip_bits(&gb, 1);                        // padding bit
-    mode = get_bits(&gb, 4);                  // frame type
-    p->bad_frame_indicator = !get_bits1(&gb); // quality bit
-    skip_bits(&gb, 2);                        // two padding bits
+    mode = buf[0] >> 3 & 0x0F;                      // frame type
+    p->bad_frame_indicator = (buf[0] & 0x4) != 0x4; // quality bit
 
     if (mode >= N_MODES || buf_size < frame_sizes_nb[mode] + 1) {
         return NO_DATA;
@@ -790,8 +793,8 @@ static int synthesis(AMRContext *p, float *lpc,
 
     // emphasize pitch vector contribution
     if (p->pitch_gain[4] > 0.5 && !overflow) {
-        float energy = ff_dot_productf(excitation, excitation,
-                                       AMR_SUBFRAME_SIZE);
+        float energy = ff_scalarproduct_float_c(excitation, excitation,
+                                                AMR_SUBFRAME_SIZE);
         float pitch_factor =
             p->pitch_gain[4] *
             (p->cur_frame_mode == MODE_12k2 ?
@@ -867,8 +870,8 @@ static float tilt_factor(float *lpc_n, float *lpc_d)
     ff_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 = ff_scalarproduct_float_c(hf, hf,     AMR_TILT_RESPONSE);
+    rh1 = ff_scalarproduct_float_c(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.
@@ -888,8 +891,8 @@ 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,
-                                              AMR_SUBFRAME_SIZE);
+    float speech_gain       = ff_scalarproduct_float_c(samples, samples,
+                                                       AMR_SUBFRAME_SIZE);
 
     float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER];  // Output of pole filter
     const float *gamma_n, *gamma_d;                       // Formant filter factor table
@@ -957,7 +960,7 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data,
     }
     if (p->cur_frame_mode == MODE_DTX) {
         av_log_missing_feature(avctx, "dtx mode", 1);
-        return -1;
+        return AVERROR_PATCHWELCOME;
     }
 
     if (p->cur_frame_mode == MODE_12k2) {
@@ -994,8 +997,10 @@ 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,
-                                       AMR_SUBFRAME_SIZE)/AMR_SUBFRAME_SIZE,
+                                  ff_scalarproduct_float_c(p->fixed_vector,
+                                                           p->fixed_vector,
+                                                           AMR_SUBFRAME_SIZE) /
+                                  AMR_SUBFRAME_SIZE,
                        p->prediction_error,
                        energy_mean[p->cur_frame_mode], energy_pred_fac);
 
@@ -1063,11 +1068,12 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data,
 AVCodec ff_amrnb_decoder = {
     .name           = "amrnb",
     .type           = AVMEDIA_TYPE_AUDIO,
-    .id             = CODEC_ID_AMR_NB,
+    .id             = AV_CODEC_ID_AMR_NB,
     .priv_data_size = sizeof(AMRContext),
     .init           = amrnb_decode_init,
     .decode         = amrnb_decode_frame,
     .capabilities   = CODEC_CAP_DR1,
-    .long_name      = NULL_IF_CONFIG_SMALL("Adaptive Multi-Rate NarrowBand"),
-    .sample_fmts    = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
+    .long_name      = NULL_IF_CONFIG_SMALL("AMR-NB (Adaptive Multi-Rate NarrowBand)"),
+    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
+                                                     AV_SAMPLE_FMT_NONE },
 };