mp3, vorbis: Remove sample clamping. (#434)

Remove the [-1.0, 1.0] sample clamping from MP3 and Vorbis decoders
to align with FFmpeg behavior and enable accurate peak detection
for audio analysis tools.

Background:
- FFmpeg (the de facto standard) does not clamp decoder output
- The AAC decoder in Symphonia already does not clamp
- Clamping prevents accurate peak detection for audio analysis
  tools that need to detect clipping (samples exceeding 0 dBFS)

Fixes: #433

Co-authored-by: m_igashi <@M_Igashi>

Author: M-Igashi

symphonia-bundle-mp3/src/synthesis.rs

@@ -322,11 +322,11 @@ pub fn synthesis(state: &mut SynthesisState, n_frames: usize, in_samples: &[f32]
             }
         }
 
-        // Clamp and copy the PCM samples from o_vec to the output buffer.
+        // Copy the PCM samples from o_vec to the output buffer.
         let offset = b << 5;
 
         for (o, s) in out[offset..offset + 32].iter_mut().zip(&o_vec) {
-            *o = s.clamp(-1.0, 1.0);
+            *o = *s;
         }
 
         // Shift the v_vec FIFO. The value v_front is the index of the 64 sample slot in v_vec

symphonia-codec-vorbis/src/dsp.rs

@@ -131,10 +131,6 @@ impl DspChannel {
                 buf[self.bs0 / 2..].copy_from_slice(&self.imdct[end..self.bs1 / 2]);
             }
 
-            // Clamp the output samples.
-            for s in buf.iter_mut() {
-                *s = s.clamp(-1.0, 1.0);
-            }
         }
 
         // Save right-half of IMDCT buffer for later.