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authorJohn Glover <j@johnglover.net>2012-10-03 12:32:54 +0200
committerJohn Glover <j@johnglover.net>2012-10-03 12:32:54 +0200
commit98d77b676e13eb362ce85146035bc2f5d098e4f6 (patch)
tree76ea3271569e64b05c8519f3c48d6825cb35411d /examples
parentcbce075cd66355ff0709bf7fd94a9553946e38c0 (diff)
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[examples] Bug fix: Use original sampling rate
when writing synthesised audio files.
Diffstat (limited to 'examples')
-rw-r--r--examples/harmonicsynthesis.py4
-rw-r--r--examples/residualsynthesis.py4
-rw-r--r--examples/timescale.py4
3 files changed, 6 insertions, 6 deletions
diff --git a/examples/harmonicsynthesis.py b/examples/harmonicsynthesis.py
index 99d6e39..401025d 100644
--- a/examples/harmonicsynthesis.py
+++ b/examples/harmonicsynthesis.py
@@ -8,7 +8,7 @@ if len(sys.argv) != 3:
print usage
sys.exit(1)
-audio = simpl.read_wav(sys.argv[1])[0]
+audio, sampling_rate = simpl.read_wav(sys.argv[1])
output_file = sys.argv[2]
pd = simpl.LorisPeakDetection()
@@ -18,4 +18,4 @@ partials = pt.find_partials(peaks)
synth = simpl.LorisSynthesis()
audio_out = synth.synth(partials)
audio_out = np.asarray(audio_out * 32768, np.int16)
-wav.write(output_file, 44100, audio_out)
+wav.write(output_file, sampling_rate, audio_out)
diff --git a/examples/residualsynthesis.py b/examples/residualsynthesis.py
index cd41dd4..f3559c8 100644
--- a/examples/residualsynthesis.py
+++ b/examples/residualsynthesis.py
@@ -8,10 +8,10 @@ if len(sys.argv) != 3:
print usage
sys.exit(1)
-audio = simpl.read_wav(sys.argv[1])[0]
+audio, sampling_rate = simpl.read_wav(sys.argv[1])
output_file = sys.argv[2]
r = simpl.SMSResidual()
audio_out = r.synth(audio)
audio_out = np.asarray(audio_out * 32768, np.int16)
-wav.write(output_file, 44100, audio_out)
+wav.write(output_file, sampling_rate, audio_out)
diff --git a/examples/timescale.py b/examples/timescale.py
index a83916b..2b061d7 100644
--- a/examples/timescale.py
+++ b/examples/timescale.py
@@ -10,7 +10,7 @@ if len(sys.argv) != 4:
print usage
sys.exit(1)
-audio = simpl.read_wav(sys.argv[1])[0]
+audio, sampling_rate = simpl.read_wav(sys.argv[1])
time_scale_factor = float(sys.argv[2])
output_file = sys.argv[3]
@@ -31,4 +31,4 @@ while current_frame < len(partials):
current_frame += step_size
audio_out = np.asarray(audio_out * 32768, np.int16)
-wav.write(output_file, 44100, audio_out)
+wav.write(output_file, sampling_rate, audio_out)