Adds support for running the analysis on 16 kHz inputexp_24k_analysis3

2 files changed, 40 insertions, 5 deletions

diff --git a/src/analysis.c b/src/analysis.c
index 91a9cc84..48e3ec8a 100644
--- a/src/analysis.c
+++ b/src/analysis.c
@@ -180,7 +180,7 @@ void tonality_get_info(TonalityAnalysisState *tonal, AnalysisInfo *info_out, int
    for (;i<DETECT_SIZE;i++)
       psum += tonal->pspeech[i];
    psum = psum*tonal->music_confidence + (1-psum)*tonal->speech_confidence;
-   /*printf("%f %f %f %f %f\n", psum, info_out->music_prob, info_out->vad_prob, info_out->activity_probability, info_out->tonality);*/
+   printf("%f %f %f %f %f\n", psum, info_out->music_prob, info_out->vad_prob, info_out->activity_probability, info_out->tonality);
 
    info_out->music_prob = psum;
 }
@@ -239,6 +239,9 @@ static void tonality_analysis(TonalityAnalysisState *tonal, const CELTMode *celt
        /* len and offset are now at 24 kHz. */
        len/= 2;
        offset /= 2;
+    } else if (tonal->Fs == 16000) {
+       len = 3*len/2;
+       offset = 3*offset/2;
     }
 
     if (tonal->count<4)
diff --git a/src/opus_encoder.c b/src/opus_encoder.c
index 79f80cda..ed487ae6 100644
--- a/src/opus_encoder.c
+++ b/src/opus_encoder.c
@@ -640,6 +640,9 @@ opus_val32 downmix_float(const void *_x, opus_val32 *sub, opus_val32 S[3], int s
    {
       subframe *= 2;
       offset *= 2;
+   } else if (Fs == 16000) {
+      subframe = subframe*2/3;
+      offset = offset*2/3;
    }
    ALLOC(tmp, subframe, opus_val32);
 
@@ -673,8 +676,21 @@ opus_val32 downmix_float(const void *_x, opus_val32 *sub, opus_val32 S[3], int s
    if (Fs == 48000)
    {
       ret = silk_resampler_down2_hp(S, sub, tmp, subframe);
-   } else /*if (Fs == 12000)*/ {
+   } else if (Fs == 24000) {
       OPUS_COPY(sub, tmp, subframe);
+   } else if (Fs == 16000) {
+      VARDECL(opus_val32, tmp3x);
+      ALLOC(tmp3x, 3*subframe, opus_val32);
+      /* Don't do this at home! This resampler is horrible and it's only (barely)
+         usable for the purpose of the analysis because we don't care about all
+         the aliasing between 8 kHz and 12 kHz. */
+      for (j=0;j<subframe;j++)
+      {
+         tmp3x[3*j] = tmp[j];
+         tmp3x[3*j+1] = tmp[j];
+         tmp3x[3*j+2] = tmp[j];
+      }
+      silk_resampler_down2_hp(S, sub, tmp3x, 3*subframe);
    }
    RESTORE_STACK;
    return ret;
@@ -695,6 +711,9 @@ opus_val32 downmix_int(const void *_x, opus_val32 *sub, opus_val32 S[3], int sub
    {
       subframe *= 2;
       offset *= 2;
+   } else if (Fs == 16000) {
+      subframe = subframe*2/3;
+      offset = offset*2/3;
    }
    ALLOC(tmp, subframe, opus_val32);
 
@@ -728,8 +747,21 @@ opus_val32 downmix_int(const void *_x, opus_val32 *sub, opus_val32 S[3], int sub
    if (Fs == 48000)
    {
       ret = silk_resampler_down2_hp(S, sub, tmp, subframe);
-   } else /*if (Fs == 12000)*/ {
+   } else if (Fs == 24000) {
       OPUS_COPY(sub, tmp, subframe);
+   } else if (Fs == 16000) {
+      VARDECL(opus_val32, tmp3x);
+      ALLOC(tmp3x, 3*subframe, opus_val32);
+      /* Don't do this at home! This resampler is horrible and it's only (barely)
+         usable for the purpose of the analysis because we don't care about all
+         the aliasing between 8 kHz and 12 kHz. */
+      for (j=0;j<subframe;j++)
+      {
+         tmp3x[3*j] = tmp[j];
+         tmp3x[3*j+1] = tmp[j];
+         tmp3x[3*j+2] = tmp[j];
+      }
+      silk_resampler_down2_hp(S, sub, tmp3x, 3*subframe);
    }
    RESTORE_STACK;
    return ret;
@@ -1222,9 +1254,9 @@ opus_int32 opus_encode_native(OpusEncoder *st, const opus_val16 *pcm, int frame_
 #ifndef DISABLE_FLOAT_API
     analysis_info.valid = 0;
 #ifdef FIXED_POINT
-    if (st->silk_mode.complexity >= 10 && st->Fs>=24000)
+    if (st->silk_mode.complexity >= 10 && st->Fs>=16000)
 #else
-    if (st->silk_mode.complexity >= 7 && st->Fs>=24000)
+    if (st->silk_mode.complexity >= 7 && st->Fs>=16000)
 #endif
     {
        if (is_digital_silence(pcm, frame_size, st->channels, lsb_depth))