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wasapi: Unprepare when src/sink_prepare fails

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unprepare() is not called automatically on failure.

https://bugzilla.gnome.org/show_bug.cgi?id=793289
This commit is contained in:
Nirbheek Chauhan 2018-02-08 14:27:43 +05:30
parent cbe2fc40a4
commit 69b90224fa
2 changed files with 17 additions and 28 deletions
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16
sys/wasapi/gstwasapisink.c
View file

@ -418,7 +418,6 @@ gst_wasapi_sink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec)
GstWasapiSink *self = GST_WASAPI_SINK (asink);
gboolean res = FALSE;
REFERENCE_TIME latency_rt;
IAudioRenderClient *render_client = NULL;
REFERENCE_TIME default_period, min_period;
REFERENCE_TIME device_period, device_buffer_duration;
guint bpf, rate;
@ -533,7 +532,7 @@ gst_wasapi_sink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec)
/* Get render sink client and start it up */
if (!gst_wasapi_util_get_render_client (GST_ELEMENT (self), self->client,
&render_client)) {
&self->render_client)) {
goto beach;
}
@ -555,7 +554,7 @@ gst_wasapi_sink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec)
len = n_frames * self->mix_format->nBlockAlign;
hr = IAudioRenderClient_GetBuffer (render_client, n_frames,
hr = IAudioRenderClient_GetBuffer (self->render_client, n_frames,
(BYTE **) & dst);
if (hr != S_OK) {
gchar *msg = gst_wasapi_util_hresult_to_string (hr);
@ -567,7 +566,7 @@ gst_wasapi_sink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec)
GST_DEBUG_OBJECT (self, "pre-wrote %i bytes of silence", len);
hr = IAudioRenderClient_ReleaseBuffer (render_client, n_frames,
hr = IAudioRenderClient_ReleaseBuffer (self->render_client, n_frames,
AUDCLNT_BUFFERFLAGS_SILENT);
if (hr != S_OK) {
gchar *msg = gst_wasapi_util_hresult_to_string (hr);
@ -584,9 +583,6 @@ gst_wasapi_sink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec)
goto beach;
}
self->render_client = render_client;
render_client = NULL;
gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SINK
(self)->ringbuffer, self->positions);
@ -602,8 +598,10 @@ gst_wasapi_sink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec)
res = TRUE;
beach:
if (render_client != NULL)
IUnknown_Release (render_client);
/* unprepare() is not called if prepare() fails, but we want it to be, so call
* it manually when needed */
if (!res)
gst_wasapi_sink_unprepare (asink);
return res;
}

29
sys/wasapi/gstwasapisrc.c
View file

@ -380,9 +380,6 @@ gst_wasapi_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
{
GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
gboolean res = FALSE;
IAudioClock *client_clock = NULL;
guint64 client_clock_freq = 0;
IAudioCaptureClient *capture_client = NULL;
REFERENCE_TIME latency_rt, default_period, min_period;
REFERENCE_TIME device_period, device_buffer_duration;
guint bpf, rate, buffer_frames;
@ -392,7 +389,7 @@ gst_wasapi_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
&min_period);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::GetDevicePeriod failed");
goto beach;
return FALSE;
}
GST_INFO_OBJECT (self, "wasapi default period: %" G_GINT64_FORMAT
", min period: %" G_GINT64_FORMAT, default_period, min_period);
@ -468,11 +465,11 @@ gst_wasapi_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
/* Get the clock and the clock freq */
if (!gst_wasapi_util_get_clock (GST_ELEMENT (self), self->client,
&client_clock)) {
&self->client_clock)) {
goto beach;
}
hr = IAudioClock_GetFrequency (client_clock, &client_clock_freq);
hr = IAudioClock_GetFrequency (self->client_clock, &self->client_clock_freq);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClock::GetFrequency failed");
goto beach;
@ -480,7 +477,7 @@ gst_wasapi_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
/* Get capture source client and start it up */
if (!gst_wasapi_util_get_capture_client (GST_ELEMENT (self), self->client,
&capture_client)) {
&self->capture_client)) {
goto beach;
}
@ -490,10 +487,6 @@ gst_wasapi_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
goto beach;
}
self->client_clock = client_clock;
self->client_clock_freq = client_clock_freq;
self->capture_client = capture_client;
gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SRC
(self)->ringbuffer, self->positions);
@ -507,15 +500,11 @@ gst_wasapi_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
#endif
res = TRUE;
beach:
if (!res) {
if (capture_client != NULL)
IUnknown_Release (capture_client);
if (client_clock != NULL)
IUnknown_Release (client_clock);
}
/* unprepare() is not called if prepare() fails, but we want it to be, so call
* it manually when needed */
if (!res)
gst_wasapi_src_unprepare (asrc);
return res;
}
@ -549,6 +538,8 @@ gst_wasapi_src_unprepare (GstAudioSrc * asrc)
self->client_clock = NULL;
}
self->client_clock_freq = 0;
return TRUE;
}