From a58805216a293f789f7d3570e5a3f56b4779c033 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Tue, 21 Jun 2011 18:17:59 +0200 Subject: [PATCH] audio: clean up headers --- gst-libs/gst/audio/gstaudioclock.c | 18 +++++++++--------- gst-libs/gst/audio/gstaudioclock.h | 17 ++++++----------- gst-libs/gst/audio/gstbaseaudiosink.c | 9 ++++----- gst-libs/gst/audio/gstbaseaudiosink.h | 15 +++++---------- 4 files changed, 24 insertions(+), 35 deletions(-) diff --git a/gst-libs/gst/audio/gstaudioclock.c b/gst-libs/gst/audio/gstaudioclock.c index 9d6e485..44879e3 100644 --- a/gst-libs/gst/audio/gstaudioclock.c +++ b/gst-libs/gst/audio/gstaudioclock.c @@ -102,7 +102,7 @@ gst_audio_clock_init (GstAudioClock * clock) { GST_DEBUG_OBJECT (clock, "init"); clock->last_time = 0; - clock->abidata.ABI.time_offset = 0; + clock->time_offset = 0; GST_OBJECT_FLAG_SET (clock, GST_CLOCK_FLAG_CAN_SET_MASTER); } @@ -111,9 +111,9 @@ gst_audio_clock_dispose (GObject * object) { GstAudioClock *clock = GST_AUDIO_CLOCK (object); - if (clock->abidata.ABI.destroy_notify && clock->user_data) - clock->abidata.ABI.destroy_notify (clock->user_data); - clock->abidata.ABI.destroy_notify = NULL; + if (clock->destroy_notify && clock->user_data) + clock->destroy_notify (clock->user_data); + clock->destroy_notify = NULL; clock->user_data = NULL; G_OBJECT_CLASS (parent_class)->dispose (object); @@ -168,7 +168,7 @@ gst_audio_clock_new_full (const gchar * name, GstAudioClockGetTimeFunc func, aclock->func = func; aclock->user_data = user_data; - aclock->abidata.ABI.destroy_notify = destroy_notify; + aclock->destroy_notify = destroy_notify; return (GstClock *) aclock; } @@ -193,7 +193,7 @@ gst_audio_clock_reset (GstAudioClock * clock, GstClockTime time) else time_offset = -(time - clock->last_time); - clock->abidata.ABI.time_offset = time_offset; + clock->time_offset = time_offset; GST_DEBUG_OBJECT (clock, "reset clock to %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT ", offset %" @@ -219,7 +219,7 @@ gst_audio_clock_get_internal_time (GstClock * clock) if (result == GST_CLOCK_TIME_NONE) { result = aclock->last_time; } else { - result += aclock->abidata.ABI.time_offset; + result += aclock->time_offset; /* clock must be increasing */ if (aclock->last_time < result) aclock->last_time = result; @@ -256,7 +256,7 @@ gst_audio_clock_get_time (GstClock * clock) result = aclock->func (clock, aclock->user_data); if (result == GST_CLOCK_TIME_NONE) { GST_DEBUG_OBJECT (clock, "no time, reuse last"); - result = aclock->last_time - aclock->abidata.ABI.time_offset; + result = aclock->last_time - aclock->time_offset; } GST_DEBUG_OBJECT (clock, @@ -285,7 +285,7 @@ gst_audio_clock_adjust (GstClock * clock, GstClockTime time) aclock = GST_AUDIO_CLOCK_CAST (clock); - result = time + aclock->abidata.ABI.time_offset; + result = time + aclock->time_offset; return result; } diff --git a/gst-libs/gst/audio/gstaudioclock.h b/gst-libs/gst/audio/gstaudioclock.h index 906c802..cd96dc0 100644 --- a/gst-libs/gst/audio/gstaudioclock.h +++ b/gst-libs/gst/audio/gstaudioclock.h @@ -68,19 +68,14 @@ struct _GstAudioClock { /*< protected >*/ GstAudioClockGetTimeFunc func; - gpointer user_data; - - GstClockTime last_time; + gpointer user_data; + GDestroyNotify destroy_notify; /*< private >*/ - union { - struct { - GstClockTimeDiff time_offset; - GDestroyNotify destroy_notify; - } ABI; - /* adding + 0 to mark ABI change to be undone later */ - gpointer _gst_reserved[GST_PADDING + 0]; - } abidata; + GstClockTime last_time; + GstClockTimeDiff time_offset; + + gpointer _gst_reserved[GST_PADDING]; }; struct _GstAudioClockClass { diff --git a/gst-libs/gst/audio/gstbaseaudiosink.c b/gst-libs/gst/audio/gstbaseaudiosink.c index 6d09d12..cd42528 100644 --- a/gst-libs/gst/audio/gstbaseaudiosink.c +++ b/gst-libs/gst/audio/gstbaseaudiosink.c @@ -826,7 +826,7 @@ gst_base_audio_sink_drain (GstBaseAudioSink * sink) /* if PLAYING is interrupted, * arrange to have clock running when going to PLAYING again */ - g_atomic_int_set (&sink->abidata.ABI.eos_rendering, 1); + g_atomic_int_set (&sink->eos_rendering, 1); /* need to start playback before we can drain, but only when * we have successfully negotiated a format and thus acquired the @@ -845,7 +845,7 @@ gst_base_audio_sink_drain (GstBaseAudioSink * sink) GST_DEBUG_OBJECT (sink, "drained audio"); } - g_atomic_int_set (&sink->abidata.ABI.eos_rendering, 0); + g_atomic_int_set (&sink->eos_rendering, 0); return TRUE; } @@ -1570,8 +1570,7 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf) GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop)); /* bring to position in the ringbuffer */ - time_offset = - GST_AUDIO_CLOCK_CAST (sink->provided_clock)->abidata.ABI.time_offset; + time_offset = GST_AUDIO_CLOCK_CAST (sink->provided_clock)->time_offset; GST_DEBUG_OBJECT (sink, "time offset %" GST_TIME_FORMAT, GST_TIME_ARGS (time_offset)); if (render_start > time_offset) @@ -1953,7 +1952,7 @@ gst_base_audio_sink_change_state (GstElement * element, gst_ring_buffer_may_start (sink->ringbuffer, TRUE); if (GST_BASE_SINK_CAST (sink)->pad_mode == GST_ACTIVATE_PULL || - g_atomic_int_get (&sink->abidata.ABI.eos_rendering) || eos) { + g_atomic_int_get (&sink->eos_rendering) || eos) { /* we always start the ringbuffer in pull mode immediatly */ /* sync rendering on eos needs running clock, * and others need running clock when finished rendering eos */ diff --git a/gst-libs/gst/audio/gstbaseaudiosink.h b/gst-libs/gst/audio/gstbaseaudiosink.h index 95c0774..0511a0f 100644 --- a/gst-libs/gst/audio/gstbaseaudiosink.h +++ b/gst-libs/gst/audio/gstbaseaudiosink.h @@ -124,17 +124,13 @@ struct _GstBaseAudioSink { gboolean provide_clock; GstClock *provided_clock; + /* with g_atomic_; currently rendering eos */ + gboolean eos_rendering; + /*< private >*/ GstBaseAudioSinkPrivate *priv; - union { - struct { - /*< protected >*/ - /* with g_atomic_; currently rendering eos */ - gboolean eos_rendering; - } ABI; - gpointer _gst_reserved[GST_PADDING - 1]; - } abidata; + gpointer _gst_reserved[GST_PADDING]; }; /** @@ -158,9 +154,8 @@ struct _GstBaseAudioSinkClass { /* subclass payloader */ GstBuffer* (*payload) (GstBaseAudioSink *sink, GstBuffer *buffer); - /*< private >*/ - gpointer _gst_reserved[GST_PADDING - 1]; + gpointer _gst_reserved[GST_PADDING]; }; GType gst_base_audio_sink_get_type(void); -- 2.7.4