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interaudio: Make buffer size and latency handling more explicit and add properties for them

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This now makes audio work more reliable without disconts.
This commit is contained in:
Sebastian Dröge 2014-11-04 14:56:55 +01:00
parent 2939337b61
commit 5c7d0a1553
5 changed files with 181 additions and 49 deletions

94
gst/inter/gstinteraudiosink.c
View file

@ -47,9 +47,6 @@
#include "gstinteraudiosink.h"
#include <string.h>
#define PERIOD 1600
#define N_PERIODS 10
GST_DEBUG_CATEGORY_STATIC (gst_inter_audio_sink_debug_category);
#define GST_CAT_DEFAULT gst_inter_audio_sink_debug_category
@ -68,6 +65,8 @@ static gboolean gst_inter_audio_sink_set_caps (GstBaseSink * sink,
GstCaps * caps);
static GstFlowReturn gst_inter_audio_sink_render (GstBaseSink * sink,
GstBuffer * buffer);
static gboolean gst_inter_audio_sink_query (GstBaseSink * sink,
GstQuery * query);
enum
{
@ -83,7 +82,6 @@ GST_STATIC_PAD_TEMPLATE ("sink",
GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL))
);
/* class initialization */
G_DEFINE_TYPE (GstInterAudioSink, gst_inter_audio_sink, GST_TYPE_BASE_SINK);
@ -114,6 +112,7 @@ gst_inter_audio_sink_class_init (GstInterAudioSinkClass * klass)
base_sink_class->stop = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_stop);
base_sink_class->set_caps = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_set_caps);
base_sink_class->render = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_render);
base_sink_class->query = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_query);
g_object_class_install_property (gobject_class, PROP_CHANNEL,
g_param_spec_string ("channel", "Channel",
@ -201,6 +200,11 @@ gst_inter_audio_sink_start (GstBaseSink * sink)
interaudiosink->surface = gst_inter_surface_get (interaudiosink->channel);
g_mutex_lock (&interaudiosink->surface->mutex);
memset (&interaudiosink->surface->audio_info, 0, sizeof (GstAudioInfo));
/* We want to write latency-time before syncing has happened */
/* FIXME: The other side can change this value when it starts */
gst_base_sink_set_render_delay (sink,
interaudiosink->surface->audio_latency_time);
g_mutex_unlock (&interaudiosink->surface->mutex);
return TRUE;
@ -249,19 +253,32 @@ static GstFlowReturn
gst_inter_audio_sink_render (GstBaseSink * sink, GstBuffer * buffer)
{
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
int n, bpf;
guint n, bpf;
guint64 period_time, buffer_time;
guint64 period_samples, buffer_samples;
GST_DEBUG_OBJECT (interaudiosink, "render %" G_GSIZE_FORMAT,
gst_buffer_get_size (buffer));
bpf = interaudiosink->info.bpf;
g_mutex_lock (&interaudiosink->surface->mutex);
buffer_time = interaudiosink->surface->audio_buffer_time;
period_time = interaudiosink->surface->audio_period_time;
buffer_samples =
gst_util_uint64_scale (buffer_time, interaudiosink->info.rate,
GST_SECOND);
period_samples =
gst_util_uint64_scale (period_time, interaudiosink->info.rate,
GST_SECOND);
n = gst_adapter_available (interaudiosink->surface->audio_adapter) / bpf;
while (n > PERIOD * N_PERIODS) {
GST_WARNING_OBJECT (interaudiosink, "flushing %d samples", PERIOD / 2);
while (n > buffer_samples) {
GST_WARNING_OBJECT (interaudiosink, "flushing %" GST_TIME_FORMAT,
GST_TIME_ARGS (period_time));
gst_adapter_flush (interaudiosink->surface->audio_adapter,
(PERIOD / 2) * bpf);
n -= (PERIOD / 2);
period_samples * bpf);
n -= period_samples;
}
gst_adapter_push (interaudiosink->surface->audio_adapter,
gst_buffer_ref (buffer));
@ -269,3 +286,62 @@ gst_inter_audio_sink_render (GstBaseSink * sink, GstBuffer * buffer)
return GST_FLOW_OK;
}
static gboolean
gst_inter_audio_sink_query (GstBaseSink * sink, GstQuery * query)
{
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
gboolean ret;
GST_DEBUG_OBJECT (sink, "query");
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:{
gboolean live, us_live;
GstClockTime min_l, max_l;
GST_DEBUG_OBJECT (sink, "latency query");
if ((ret =
gst_base_sink_query_latency (GST_BASE_SINK_CAST (sink), &live,
&us_live, &min_l, &max_l))) {
GstClockTime base_latency, min_latency, max_latency;
/* we and upstream are both live, adjust the min_latency */
if (live && us_live) {
/* FIXME: The other side can change this value when it starts */
base_latency = interaudiosink->surface->audio_latency_time;
/* we cannot go lower than the buffer size and the min peer latency */
min_latency = base_latency + min_l;
/* the max latency is the max of the peer, we can delay an infinite
* amount of time. */
max_latency = (max_l == -1) ? -1 : (base_latency + max_l);
GST_DEBUG_OBJECT (sink,
"peer min %" GST_TIME_FORMAT ", our min latency: %"
GST_TIME_FORMAT, GST_TIME_ARGS (min_l),
GST_TIME_ARGS (min_latency));
GST_DEBUG_OBJECT (sink,
"peer max %" GST_TIME_FORMAT ", our max latency: %"
GST_TIME_FORMAT, GST_TIME_ARGS (max_l),
GST_TIME_ARGS (max_latency));
} else {
GST_DEBUG_OBJECT (sink,
"peer or we are not live, don't care about latency");
min_latency = min_l;
max_latency = max_l;
}
gst_query_set_latency (query, live, min_latency, max_latency);
}
break;
}
default:
ret =
GST_BASE_SINK_CLASS (gst_inter_audio_sink_parent_class)->query (sink,
query);
break;
}
return ret;
}

125
gst/inter/gstinteraudiosrc.c
View file

@ -50,9 +50,6 @@
GST_DEBUG_CATEGORY_STATIC (gst_inter_audio_src_debug_category);
#define GST_CAT_DEFAULT gst_inter_audio_src_debug_category
#define PERIOD 1600
#define N_PERIODS 10
/* prototypes */
static void gst_inter_audio_src_set_property (GObject * object,
guint property_id, const GValue * value, GParamSpec * pspec);
@ -77,7 +74,10 @@ static GstCaps *gst_inter_audio_src_fixate (GstBaseSrc * src, GstCaps * caps);
enum
{
PROP_0,
PROP_CHANNEL
PROP_CHANNEL,
PROP_BUFFER_TIME,
PROP_LATENCY_TIME,
PROP_PERIOD_TIME
};
/* pad templates */
@ -128,6 +128,23 @@ gst_inter_audio_src_class_init (GstInterAudioSrcClass * klass)
g_param_spec_string ("channel", "Channel",
"Channel name to match inter src and sink elements",
"default", G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_BUFFER_TIME,
g_param_spec_uint64 ("buffer-time", "Buffer Time",
"Size of audio buffer", 1, G_MAXUINT64, DEFAULT_AUDIO_BUFFER_TIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_LATENCY_TIME,
g_param_spec_uint64 ("latency-time", "Latency Time",
"Latency as reported by the source",
1, G_MAXUINT64, DEFAULT_AUDIO_LATENCY_TIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_PERIOD_TIME,
g_param_spec_uint64 ("period-time", "Period Time",
"The minimum amount of data to read in each iteration",
1, G_MAXUINT64, DEFAULT_AUDIO_PERIOD_TIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
static void
@ -138,6 +155,9 @@ gst_inter_audio_src_init (GstInterAudioSrc * interaudiosrc)
gst_base_src_set_blocksize (GST_BASE_SRC (interaudiosrc), -1);
interaudiosrc->channel = g_strdup ("default");
interaudiosrc->buffer_time = DEFAULT_AUDIO_BUFFER_TIME;
interaudiosrc->latency_time = DEFAULT_AUDIO_LATENCY_TIME;
interaudiosrc->period_time = DEFAULT_AUDIO_PERIOD_TIME;
}
void
@ -151,6 +171,15 @@ gst_inter_audio_src_set_property (GObject * object, guint property_id,
g_free (interaudiosrc->channel);
interaudiosrc->channel = g_value_dup_string (value);
break;
case PROP_BUFFER_TIME:
interaudiosrc->buffer_time = g_value_get_uint64 (value);
break;
case PROP_LATENCY_TIME:
interaudiosrc->latency_time = g_value_get_uint64 (value);
break;
case PROP_PERIOD_TIME:
interaudiosrc->period_time = g_value_get_uint64 (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
@ -167,6 +196,15 @@ gst_inter_audio_src_get_property (GObject * object, guint property_id,
case PROP_CHANNEL:
g_value_set_string (value, interaudiosrc->channel);
break;
case PROP_BUFFER_TIME:
g_value_set_uint64 (value, interaudiosrc->buffer_time);
break;
case PROP_LATENCY_TIME:
g_value_set_uint64 (value, interaudiosrc->latency_time);
break;
case PROP_PERIOD_TIME:
g_value_set_uint64 (value, interaudiosrc->period_time);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
@ -242,6 +280,12 @@ gst_inter_audio_src_start (GstBaseSrc * src)
interaudiosrc->timestamp_offset = 0;
interaudiosrc->n_samples = 0;
g_mutex_lock (&interaudiosrc->surface->mutex);
interaudiosrc->surface->audio_buffer_time = interaudiosrc->buffer_time;
interaudiosrc->surface->audio_latency_time = interaudiosrc->latency_time;
interaudiosrc->surface->audio_period_time = interaudiosrc->period_time;
g_mutex_unlock (&interaudiosrc->surface->mutex);
return TRUE;
}
@ -262,24 +306,24 @@ static void
gst_inter_audio_src_get_times (GstBaseSrc * src, GstBuffer * buffer,
GstClockTime * start, GstClockTime * end)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
GST_DEBUG_OBJECT (src, "get_times");
/* for live sources, sync on the timestamp of the buffer */
if (gst_base_src_is_live (src)) {
GstClockTime timestamp = GST_BUFFER_TIMESTAMP (buffer);
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
/* get duration to calculate end time */
GstClockTime duration = GST_BUFFER_DURATION (buffer);
if (GST_CLOCK_TIME_IS_VALID (duration)) {
*end = timestamp + duration;
if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
*start = GST_BUFFER_TIMESTAMP (buffer);
if (GST_BUFFER_DURATION_IS_VALID (buffer)) {
*end = *start + GST_BUFFER_DURATION (buffer);
} else {
if (interaudiosrc->info.rate > 0) {
*end = *start +
gst_util_uint64_scale_int (gst_buffer_get_size (buffer),
GST_SECOND, interaudiosrc->info.rate * interaudiosrc->info.bpf);
}
}
*start = timestamp;
}
} else {
*start = -1;
*end = -1;
}
}
@ -290,7 +334,9 @@ gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
GstCaps *caps;
GstBuffer *buffer;
int n, bpf;
guint n, bpf;
guint64 period_time, buffer_time;
guint64 period_samples, buffer_samples;
GST_DEBUG_OBJECT (interaudiosrc, "create");
@ -310,14 +356,20 @@ gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
}
bpf = interaudiosrc->surface->audio_info.bpf;
buffer_time = interaudiosrc->surface->audio_buffer_time;
period_time = interaudiosrc->surface->audio_period_time;
buffer_samples =
gst_util_uint64_scale (buffer_time, interaudiosrc->info.rate, GST_SECOND);
period_samples =
gst_util_uint64_scale (period_time, interaudiosrc->info.rate, GST_SECOND);
if (bpf > 0)
n = gst_adapter_available (interaudiosrc->surface->audio_adapter) / bpf;
else
n = 0;
if (n > PERIOD)
n = PERIOD;
if (n > period_samples)
n = period_samples;
if (n > 0) {
buffer = gst_adapter_take_buffer (interaudiosrc->surface->audio_adapter,
n * bpf);
@ -338,13 +390,14 @@ gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
}
bpf = interaudiosrc->info.bpf;
if (n < PERIOD) {
if (n < period_samples) {
GstMapInfo map;
GstMemory *mem;
GST_WARNING_OBJECT (interaudiosrc, "creating %d samples of silence",
PERIOD - n);
mem = gst_allocator_alloc (NULL, (PERIOD - n) * bpf, NULL);
GST_WARNING_OBJECT (interaudiosrc,
"creating %" G_GUINT64_FORMAT " samples of silence",
period_samples - n);
mem = gst_allocator_alloc (NULL, (period_samples - n) * bpf, NULL);
if (gst_memory_map (mem, &map, GST_MAP_WRITE)) {
gst_audio_format_fill_silence (interaudiosrc->info.finfo, map.data,
map.size);
@ -353,7 +406,7 @@ gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
buffer = gst_buffer_make_writable (buffer);
gst_buffer_prepend_memory (buffer, mem);
}
n = PERIOD;
n = period_samples;
GST_BUFFER_OFFSET (buffer) = interaudiosrc->n_samples;
GST_BUFFER_OFFSET_END (buffer) = interaudiosrc->n_samples + n;
@ -388,25 +441,17 @@ gst_inter_audio_src_query (GstBaseSrc * src, GstQuery * query)
case GST_QUERY_LATENCY:{
GstClockTime min_latency, max_latency;
if (interaudiosrc->info.rate > 0) {
/* 1.5 just as a good measure */
min_latency =
1.5 * N_PERIODS * gst_util_uint64_scale_int (GST_SECOND, PERIOD,
interaudiosrc->info.rate);
min_latency = interaudiosrc->latency_time;
max_latency = min_latency;
max_latency = min_latency;
GST_DEBUG_OBJECT (src,
"report latency min %" GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
GST_DEBUG_OBJECT (src,
"report latency min %" GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
gst_query_set_latency (query,
gst_base_src_is_live (src), min_latency, max_latency);
gst_query_set_latency (query,
gst_base_src_is_live (src), min_latency, max_latency);
ret = TRUE;
} else {
ret = FALSE;
}
ret = TRUE;
break;
}
default:

1
gst/inter/gstinteraudiosrc.h
View file

@ -45,6 +45,7 @@ struct _GstInterAudioSrc
guint64 n_samples;
GstClockTime timestamp_offset;
GstAudioInfo info;
guint64 buffer_time, latency_time, period_time;
};
struct _GstInterAudioSrcClass

3
gst/inter/gstintersurface.c
View file

@ -49,6 +49,9 @@ gst_inter_surface_get (const char *name)
surface->name = g_strdup (name);
g_mutex_init (&surface->mutex);
surface->audio_adapter = gst_adapter_new ();
surface->audio_buffer_time = DEFAULT_AUDIO_BUFFER_TIME;
surface->audio_latency_time = DEFAULT_AUDIO_LATENCY_TIME;
surface->audio_period_time = DEFAULT_AUDIO_PERIOD_TIME;
list = g_list_append (list, surface);
g_mutex_unlock (&mutex);

7
gst/inter/gstintersurface.h
View file

@ -41,12 +41,19 @@ struct _GstInterSurface
/* audio */
GstAudioInfo audio_info;
guint64 audio_buffer_time;
guint64 audio_latency_time;
guint64 audio_period_time;
GstBuffer *video_buffer;
GstBuffer *sub_buffer;
GstAdapter *audio_adapter;
};
#define DEFAULT_AUDIO_BUFFER_TIME (GST_SECOND)
#define DEFAULT_AUDIO_LATENCY_TIME (100 * GST_MSECOND)
#define DEFAULT_AUDIO_PERIOD_TIME (25 * GST_MSECOND)
GstInterSurface * gst_inter_surface_get (const char *name);
void gst_inter_surface_unref (GstInterSurface *surface);