mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-10-02 00:32:43 +00:00
Code Issues 1 Projects Releases Wiki Activity

interaudio: Fix timestamp, latency and period handling

Browse source
This commit is contained in:
Sebastian Dröge 2014-10-22 19:08:39 +02:00
parent 8c5a8c76f6
commit 04dbd095a1
2 changed files with 31 additions and 27 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

11
gst/inter/gstinteraudiosink.c
View file

@ -47,6 +47,9 @@
#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
@ -253,11 +256,11 @@ gst_inter_audio_sink_render (GstBaseSink * sink, GstBuffer * buffer)
g_mutex_lock (&interaudiosink->surface->mutex);
n = gst_adapter_available (interaudiosink->surface->audio_adapter) / 4;
#define SIZE 1600
if (n > SIZE * 3) {
GST_WARNING_OBJECT (interaudiosink, "flushing %d samples", SIZE / 2);
while (n > PERIOD * N_PERIODS) {
GST_WARNING_OBJECT (interaudiosink, "flushing %d samples", PERIOD / 2);
gst_adapter_flush (interaudiosink->surface->audio_adapter,
(SIZE / 2) * bpf);
(PERIOD / 2) * bpf);
n -= (PERIOD / 2);
}
gst_adapter_push (interaudiosink->surface->audio_adapter,
gst_buffer_ref (buffer));

47
gst/inter/gstinteraudiosrc.c
View file

@ -50,6 +50,9 @@
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);
@ -281,8 +284,6 @@ gst_inter_audio_src_get_times (GstBaseSrc * src, GstBuffer * buffer,
}
}
#define SIZE 1600
static GstFlowReturn
gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
GstBuffer ** buf)
@ -302,7 +303,7 @@ gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
if (!gst_audio_info_is_equal (&interaudiosrc->surface->audio_info,
&interaudiosrc->info)) {
caps = gst_audio_info_to_caps (&interaudiosrc->surface->audio_info);
interaudiosrc->timestamp_offset =
interaudiosrc->timestamp_offset +=
gst_util_uint64_scale_int (interaudiosrc->n_samples, GST_SECOND,
interaudiosrc->info.rate);
interaudiosrc->n_samples = 0;
@ -311,16 +312,19 @@ gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
bpf = interaudiosrc->surface->audio_info.bpf;
n = bpf >
0 ? gst_adapter_available (interaudiosrc->surface->audio_adapter) /
bpf : 0;
if (n > SIZE * 3) {
GST_WARNING_OBJECT (interaudiosrc, "flushing %d samples", SIZE / 2);
gst_adapter_flush (interaudiosrc->surface->audio_adapter, (SIZE / 2) * bpf);
n -= (SIZE / 2);
if (bpf > 0)
n = gst_adapter_available (interaudiosrc->surface->audio_adapter) / bpf;
else
n = 0;
while (n > PERIOD * 10) {
GST_WARNING_OBJECT (interaudiosrc, "flushing %d samples", PERIOD / 2);
gst_adapter_flush (interaudiosrc->surface->audio_adapter,
(PERIOD / 2) * bpf);
n -= (PERIOD / 2);
}
if (n > SIZE)
n = SIZE;
if (n > PERIOD)
n = PERIOD;
if (n > 0) {
buffer = gst_adapter_take_buffer (interaudiosrc->surface->audio_adapter,
n * bpf);
@ -341,13 +345,13 @@ gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
}
bpf = interaudiosrc->info.bpf;
if (n < SIZE) {
if (n < PERIOD) {
GstMapInfo map;
GstMemory *mem;
GST_WARNING_OBJECT (interaudiosrc, "creating %d samples of silence",
SIZE - n);
mem = gst_allocator_alloc (NULL, (SIZE - n) * bpf, NULL);
PERIOD - n);
mem = gst_allocator_alloc (NULL, (PERIOD - n) * bpf, NULL);
if (gst_memory_map (mem, &map, GST_MAP_WRITE)) {
gst_audio_format_fill_silence (interaudiosrc->info.finfo, map.data,
map.size);
@ -356,20 +360,18 @@ gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
buffer = gst_buffer_make_writable (buffer);
gst_buffer_prepend_memory (buffer, mem);
}
n = SIZE;
n = PERIOD;
GST_BUFFER_OFFSET (buffer) = interaudiosrc->n_samples;
GST_BUFFER_OFFSET_END (buffer) = interaudiosrc->n_samples + n;
GST_BUFFER_TIMESTAMP (buffer) =
GST_BUFFER_TIMESTAMP (buffer) = interaudiosrc->timestamp_offset +
gst_util_uint64_scale_int (interaudiosrc->n_samples, GST_SECOND,
interaudiosrc->info.rate);
GST_DEBUG_OBJECT (interaudiosrc, "create ts %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
GST_BUFFER_DURATION (buffer) =
GST_BUFFER_DURATION (buffer) = interaudiosrc->timestamp_offset +
gst_util_uint64_scale_int (interaudiosrc->n_samples + n, GST_SECOND,
interaudiosrc->info.rate) - GST_BUFFER_TIMESTAMP (buffer);
GST_BUFFER_OFFSET (buffer) = interaudiosrc->n_samples;
GST_BUFFER_OFFSET_END (buffer) = -1;
GST_BUFFER_FLAG_UNSET (buffer, GST_BUFFER_FLAG_DISCONT);
if (interaudiosrc->n_samples == 0) {
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
@ -394,10 +396,9 @@ gst_inter_audio_src_query (GstBaseSrc * src, GstQuery * query)
GstClockTime min_latency, max_latency;
if (interaudiosrc->info.rate > 0) {
/* FIXME: Where does the 30 come from? */
/* 1.5 just as a good measure */
min_latency =
30 * gst_util_uint64_scale_int (GST_SECOND, SIZE,
1.5 * N_PERIODS * gst_util_uint64_scale_int (GST_SECOND, PERIOD,
interaudiosrc->info.rate);
max_latency = min_latency;