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authorMax Kellermann <max@duempel.org>2009-11-10 17:11:34 +0100
committerMax Kellermann <max@duempel.org>2009-12-02 22:29:50 +0100
commitc412d6251e9cd3abe735b7622af4003502e54f72 (patch)
tree7344c13f62e4cc788c830c05d21bb7b5b47f5866 /test
parent68c2cfbb4067b2292e1ff1d4e7716ff370903f84 (diff)
downloadmpd-c412d6251e9cd3abe735b7622af4003502e54f72.tar.gz
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audio_format: changed "bits" to "enum sample_format"
This patch prepares support for floating point samples (and probably other formats). It changes the meaning of the "bits" attribute from a bit count to a symbolic value.
Diffstat (limited to 'test')
-rw-r--r--test/run_encoder.c2
-rw-r--r--test/run_filter.c2
-rw-r--r--test/run_output.c2
-rw-r--r--test/software_volume.c2
4 files changed, 4 insertions, 4 deletions
diff --git a/test/run_encoder.c b/test/run_encoder.c
index aaafd2e8f..bb73e7b6e 100644
--- a/test/run_encoder.c
+++ b/test/run_encoder.c
@@ -63,7 +63,7 @@ int main(int argc, char **argv)
else
encoder_name = "vorbis";
- audio_format_init(&audio_format, 44100, 16, 2);
+ audio_format_init(&audio_format, 44100, SAMPLE_FORMAT_S16, 2);
/* create the encoder */
diff --git a/test/run_filter.c b/test/run_filter.c
index 30495b1fd..62cafdbc1 100644
--- a/test/run_filter.c
+++ b/test/run_filter.c
@@ -85,7 +85,7 @@ int main(int argc, char **argv)
return 1;
}
- audio_format_init(&audio_format, 44100, 16, 2);
+ audio_format_init(&audio_format, 44100, SAMPLE_FORMAT_S16, 2);
g_thread_init(NULL);
diff --git a/test/run_output.c b/test/run_output.c
index 1acc7b377..9ae4e4c4b 100644
--- a/test/run_output.c
+++ b/test/run_output.c
@@ -119,7 +119,7 @@ int main(int argc, char **argv)
return 1;
}
- audio_format_init(&audio_format, 44100, 16, 2);
+ audio_format_init(&audio_format, 44100, SAMPLE_FORMAT_S16, 2);
g_thread_init(NULL);
diff --git a/test/software_volume.c b/test/software_volume.c
index 3b4c5f0fb..11345764d 100644
--- a/test/software_volume.c
+++ b/test/software_volume.c
@@ -55,7 +55,7 @@ int main(int argc, char **argv)
return 1;
}
} else
- audio_format_init(&audio_format, 48000, 16, 2);
+ audio_format_init(&audio_format, 48000, SAMPLE_FORMAT_S16, 2);
while ((nbytes = read(0, buffer, sizeof(buffer))) > 0) {
if (!pcm_volume(buffer, nbytes, &audio_format,