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decklinkaudiosink: Implement resampling/buffer reversing for trick-modes

Browse source 
https://bugzilla.gnome.org/show_bug.cgi?id=790114
This commit is contained in:
Sebastian Dröge 2017-11-22 16:37:12 +02:00
parent 86888d9918
commit 118b2967e7
2 changed files with 144 additions and 10 deletions
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152
sys/decklink/gstdecklinkaudiosink.cpp
View file

@ -70,6 +70,8 @@ static void gst_decklink_audio_sink_get_times (GstBaseSink * bsink,
GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
static gboolean gst_decklink_audio_sink_query (GstBaseSink * bsink,
GstQuery * query);
static gboolean gst_decklink_audio_sink_event (GstBaseSink * bsink,
GstEvent * event);
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
@ -112,6 +114,7 @@ gst_decklink_audio_sink_class_init (GstDecklinkAudioSinkClass * klass)
basesink_class->get_times =
GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_get_times);
basesink_class->query = GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_query);
basesink_class->event = GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_event);
g_object_class_install_property (gobject_class, PROP_DEVICE_NUMBER,
g_param_spec_int ("device-number", "Device number",
@ -292,6 +295,11 @@ gst_decklink_audio_sink_set_caps (GstBaseSink * bsink, GstCaps * caps)
self->output->audio_enabled = TRUE;
self->info = info;
// Create a new resampler as needed
if (self->resampler)
gst_audio_resampler_free (self->resampler);
self->resampler = NULL;
return TRUE;
}
@ -425,6 +433,43 @@ done:
return res;
}
static gboolean
gst_decklink_audio_sink_event (GstBaseSink * bsink, GstEvent * event)
{
GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (bsink);
if (GST_EVENT_TYPE (event) == GST_EVENT_SEGMENT) {
const GstSegment *new_segment;
gst_event_parse_segment (event, &new_segment);
if (ABS (new_segment->rate) != 1.0) {
guint out_rate = self->info.rate / ABS (new_segment->rate);
if (self->resampler && (self->resampler_out_rate != out_rate
|| self->resampler_in_rate != (guint) self->info.rate))
gst_audio_resampler_update (self->resampler, self->info.rate, out_rate,
NULL);
else if (!self->resampler)
self->resampler =
gst_audio_resampler_new (GST_AUDIO_RESAMPLER_METHOD_LINEAR,
GST_AUDIO_RESAMPLER_FLAG_NONE, self->info.finfo->format,
self->info.channels, self->info.rate, out_rate, NULL);
self->resampler_in_rate = self->info.rate;
self->resampler_out_rate = out_rate;
} else if (self->resampler) {
gst_audio_resampler_free (self->resampler);
self->resampler = NULL;
}
if (new_segment->rate < 0)
gst_audio_stream_align_set_rate (self->stream_align, -48000);
}
return GST_BASE_SINK_CLASS (parent_class)->event (bsink, event);
}
static GstFlowReturn
gst_decklink_audio_sink_render (GstBaseSink * bsink, GstBuffer * buffer)
{
@ -440,6 +485,7 @@ gst_decklink_audio_sink_render (GstBaseSink * bsink, GstBuffer * buffer)
GstMapInfo map_info;
const guint8 *data;
gsize len, written_all;
gboolean discont;
GST_DEBUG_OBJECT (self, "Rendering buffer %p", buffer);
@ -457,11 +503,77 @@ gst_decklink_audio_sink_render (GstBaseSink * bsink, GstBuffer * buffer)
timestamp = GST_BUFFER_TIMESTAMP (buffer);
duration = GST_BUFFER_DURATION (buffer);
gst_audio_stream_align_process (self->stream_align,
discont = gst_audio_stream_align_process (self->stream_align,
GST_BUFFER_IS_DISCONT (buffer), timestamp,
gst_buffer_get_size (buffer) / self->info.bpf, &timestamp, &duration,
NULL);
if (discont && self->resampler)
gst_audio_resampler_reset (self->resampler);
if (GST_BASE_SINK_CAST (self)->segment.rate < 0.0) {
GstMapInfo out_map;
gint out_frames = gst_buffer_get_size (buffer) / self->info.bpf;
buffer = gst_buffer_make_writable (gst_buffer_ref (buffer));
gst_buffer_map (buffer, &out_map, GST_MAP_READWRITE);
if (self->info.finfo->format == GST_AUDIO_FORMAT_S16) {
gint16 *swap_data = (gint16 *) out_map.data;
gint16 *swap_data_end =
swap_data + (out_frames - 1) * self->info.channels;
gint16 swap_tmp[16];
while (out_frames > 0) {
memcpy (&swap_tmp, swap_data, self->info.bpf);
memcpy (swap_data, swap_data_end, self->info.bpf);
memcpy (swap_data_end, &swap_tmp, self->info.bpf);
swap_data += self->info.channels;
swap_data_end -= self->info.channels;
out_frames -= 2;
}
} else {
gint32 *swap_data = (gint32 *) out_map.data;
gint32 *swap_data_end =
swap_data + (out_frames - 1) * self->info.channels;
gint32 swap_tmp[16];
while (out_frames > 0) {
memcpy (&swap_tmp, swap_data, self->info.bpf);
memcpy (swap_data, swap_data_end, self->info.bpf);
memcpy (swap_data_end, &swap_tmp, self->info.bpf);
swap_data += self->info.channels;
swap_data_end -= self->info.channels;
out_frames -= 2;
}
}
gst_buffer_unmap (buffer, &out_map);
} else {
gst_buffer_ref (buffer);
}
if (self->resampler) {
gint in_frames = gst_buffer_get_size (buffer) / self->info.bpf;
gint out_frames =
gst_audio_resampler_get_out_frames (self->resampler, in_frames);
GstBuffer *out_buf = gst_buffer_new_and_alloc (out_frames * self->info.bpf);
GstMapInfo out_map;
gst_buffer_map (buffer, &map_info, GST_MAP_READ);
gst_buffer_map (out_buf, &out_map, GST_MAP_READWRITE);
gst_audio_resampler_resample (self->resampler, (gpointer *) & map_info.data,
in_frames, (gpointer *) & out_map.data, out_frames);
gst_buffer_unmap (out_buf, &out_map);
gst_buffer_unmap (buffer, &map_info);
buffer = out_buf;
}
gst_buffer_map (buffer, &map_info, GST_MAP_READ);
data = map_info.data;
len = map_info.size / self->info.bpf;
@ -513,11 +625,10 @@ gst_decklink_audio_sink_render (GstBaseSink * bsink, GstBuffer * buffer)
buffered_time =
gst_util_uint64_scale (buffered_samples, GST_SECOND, self->info.rate);
buffered_time /= ABS (GST_BASE_SINK_CAST (self)->segment.rate);
GST_DEBUG_OBJECT (self,
"Buffered %" GST_TIME_FORMAT " in the driver (%u samples)",
GST_TIME_ARGS (buffered_time), buffered_samples);
buffered_time /= GST_BASE_SINK_CAST (self)->segment.rate;
// We start waiting once we have more than buffer-time buffered
if (buffered_time > self->buffer_time) {
GstClockReturn clock_ret;
@ -562,15 +673,27 @@ gst_decklink_audio_sink_render (GstBaseSink * bsink, GstBuffer * buffer)
self->output->output->WriteAudioSamplesSync ((void *) data, len,
&written);
if (ret != S_OK) {
GST_ELEMENT_WARNING (self, STREAM, FAILED,
(NULL), ("Failed to write audio frame synchronously: 0x%08lx",
(unsigned long) ret));
ret = S_OK;
break;
bool is_running = true;
self->output->output->IsScheduledPlaybackRunning (&is_running);
if (is_running && !GST_BASE_SINK_CAST (self)->flushing && self->output->started) {
GST_ELEMENT_WARNING (self, STREAM, FAILED,
(NULL), ("Failed to write audio frame synchronously: 0x%08lx",
(unsigned long) ret));
flow_ret = GST_FLOW_ERROR;
break;
} else {
flow_ret = GST_FLOW_FLUSHING;
break;
}
}
len -= written;
data += written * self->info.bpf;
written_all += written;
if (self->resampler)
written_all += written * ABS (GST_BASE_SINK_CAST (self)->segment.rate);
else
written_all += written;
} else {
guint32 written = 0;
@ -598,13 +721,17 @@ gst_decklink_audio_sink_render (GstBaseSink * bsink, GstBuffer * buffer)
len -= written;
data += written * self->info.bpf;
written_all += written;
if (self->resampler)
written_all += written * ABS (GST_BASE_SINK_CAST (self)->segment.rate);
else
written_all += written;
}
flow_ret = GST_FLOW_OK;
} while (len > 0);
gst_buffer_unmap (buffer, &map_info);
gst_buffer_unref (buffer);
return flow_ret;
}
@ -666,6 +793,11 @@ gst_decklink_audio_sink_stop (GstDecklinkAudioSink * self)
self->output->output->DisableAudioOutput ();
}
if (self->resampler) {
gst_audio_resampler_free (self->resampler);
self->resampler = NULL;
}
return TRUE;
}

2
sys/decklink/gstdecklinkaudiosink.h
View file

@ -56,6 +56,8 @@ struct _GstDecklinkAudioSink
GstDecklinkOutput *output;
GstAudioInfo info;
GstAudioStreamAlign *stream_align;
GstAudioResampler *resampler;
guint resampler_in_rate, resampler_out_rate;
};
struct _GstDecklinkAudioSinkClass