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authorChristoph Reiter <reiter.christoph@gmail.com>2018-05-24 11:04:08 +0200
committerNirbheek Chauhan <nirbheek@centricular.com>2018-05-25 19:06:16 +0530
commit0aed64426bc2ec443ff637b4b3089174b309eaa8 (patch)
treef74c5df3644980cd437f89660643b0ffa6fb0908 /sys
parentffb8476a38ec5c56a19decb3d9dfe35b82190d3b (diff)
downloadgstreamer-plugins-bad-0aed64426bc2ec443ff637b4b3089174b309eaa8.tar.gz
wasapisink: fix a rounding error when calculating the buffer frame count
The calculation for the frame count in the non-aligned case resulted in a one too low buffer frame count. This resulted in: 1) exclusive mode not working as the frame count has to match exactly there. 2) Buffer underruns in shared mode as the current write() code doesn't handle catching up to low buffer levels (fixed in the next commit) To fix just use the wasapi API to get the buffer size which will always be correct. https://bugzilla.gnome.org/show_bug.cgi?id=796354
Diffstat (limited to 'sys')
-rw-r--r--sys/wasapi/gstwasapiutil.c8
1 files changed, 5 insertions, 3 deletions
diff --git a/sys/wasapi/gstwasapiutil.c b/sys/wasapi/gstwasapiutil.c
index 824b3a77b..e1ac7e568 100644
--- a/sys/wasapi/gstwasapiutil.c
+++ b/sys/wasapi/gstwasapiutil.c
@@ -845,6 +845,7 @@ gst_wasapi_util_initialize_audioclient (GstElement * self,
REFERENCE_TIME default_period, min_period;
REFERENCE_TIME device_period, device_buffer_duration;
guint rate, stream_flags;
+ guint32 n_frames;
HRESULT hr;
hr = IAudioClient_GetDevicePeriod (client, &default_period, &min_period);
@@ -881,8 +882,6 @@ gst_wasapi_util_initialize_audioclient (GstElement * self,
if (hr == AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED &&
sharemode == AUDCLNT_SHAREMODE_EXCLUSIVE) {
- guint32 n_frames;
-
GST_WARNING_OBJECT (self, "initialize failed due to unaligned period %i",
(int) device_period);
@@ -900,7 +899,10 @@ gst_wasapi_util_initialize_audioclient (GstElement * self,
}
HR_FAILED_RET (hr, IAudioClient::Initialize, FALSE);
- *ret_devicep_frames = (rate * device_period * 100) / GST_SECOND;
+ hr = IAudioClient_GetBufferSize (client, &n_frames);
+ HR_FAILED_RET (hr, IAudioClient::GetBufferSize, FALSE);
+
+ *ret_devicep_frames = n_frames;
return TRUE;
}