alsasink: Fix stall for transition of alsasink from PAUSED to READY with USB speakerphone

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4699>

diff --git a/subprojects/gst-plugins-base/ext/alsa/gstalsasink.c b/subprojects/gst-plugins-base/ext/alsa/gstalsasink.c
index 6165b4f..7ed5b30 100644 (file)
--- a/subprojects/gst-plugins-base/ext/alsa/gstalsasink.c
+++ b/subprojects/gst-plugins-base/ext/alsa/gstalsasink.c
@@ -259,7 +259,6 @@ gst_alsasink_init (GstAlsaSink * alsasink)
   alsasink->is_paused = FALSE;
   alsasink->after_paused = FALSE;
   alsasink->hw_support_pause = FALSE;
-  alsasink->stop_streaming_threads = TRUE;
   g_mutex_init (&alsasink->alsa_lock);
   g_mutex_init (&alsasink->delay_lock);
 
@@ -943,7 +942,6 @@ gst_alsasink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec)
       alsa->channels, GST_AUDIO_BASE_SINK (alsa)->ringbuffer);
 #endif /* SND_CHMAP_API_VERSION */
 
-  alsa->stop_streaming_threads = FALSE;
   return TRUE;
 
   /* ERRORS */
@@ -1063,8 +1061,6 @@ gst_alsasink_write (GstAudioSink * asink, gpointer data, guint length)
 
   GST_ALSA_SINK_LOCK (asink);
   while (cptr > 0) {
-    if (alsa->stop_streaming_threads)
-      goto write_error;
     /* start by doing a blocking wait for free space. Set the timeout
      * to 4 times the period time */
     err = snd_pcm_wait (alsa->handle, (4 * alsa->period_time / 1000));
@@ -1173,7 +1169,6 @@ gst_alsasink_pause (GstAudioSink * asink)
     CHECK (snd_pcm_pause (alsa->handle, 1), pause_error);
     GST_DEBUG_OBJECT (alsa, "pause done");
     alsa->is_paused = TRUE;
-    alsa->stop_streaming_threads = TRUE;
     GST_ALSA_SINK_UNLOCK (asink);
   } else {
     gst_alsasink_stop (asink);
@@ -1186,6 +1181,7 @@ pause_error:
     GST_ERROR_OBJECT (alsa, "alsa-pause: pcm pause error: %s",
         snd_strerror (err));
     GST_ALSA_SINK_UNLOCK (asink);
+    gst_alsasink_stop (asink);
     return;
   }
 }
@@ -1205,7 +1201,6 @@ gst_alsasink_resume (GstAudioSink * asink)
     GST_ALSA_SINK_UNLOCK (asink);
   }
 
-  alsa->stop_streaming_threads = FALSE;
   return;
 
 resume_error:
@@ -1231,7 +1226,6 @@ gst_alsasink_stop (GstAudioSink * asink)
   GST_DEBUG_OBJECT (alsa, "prepare");
   CHECK (snd_pcm_prepare (alsa->handle), prepare_error);
   GST_DEBUG_OBJECT (alsa, "stop done");
-  alsa->stop_streaming_threads = TRUE;
   GST_ALSA_SINK_UNLOCK (asink);
 
   return;

diff --git a/subprojects/gst-plugins-base/ext/alsa/gstalsasink.h b/subprojects/gst-plugins-base/ext/alsa/gstalsasink.h
index e444311..35efa88 100644 (file)
--- a/subprojects/gst-plugins-base/ext/alsa/gstalsasink.h
+++ b/subprojects/gst-plugins-base/ext/alsa/gstalsasink.h
@@ -77,7 +77,6 @@ struct _GstAlsaSink {
   gboolean is_paused;
   gboolean after_paused;
   gboolean hw_support_pause;
-  gboolean stop_streaming_threads;
   snd_pcm_sframes_t pos_in_buffer;
 
   GMutex alsa_lock;