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interaudio: Fix timestamp, latency and period handling
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parent
8c5a8c76f6
commit
04dbd095a1
2 changed files with 31 additions and 27 deletions
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@ -47,6 +47,9 @@
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#include "gstinteraudiosink.h"
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#include <string.h>
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#define PERIOD 1600
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#define N_PERIODS 10
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GST_DEBUG_CATEGORY_STATIC (gst_inter_audio_sink_debug_category);
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#define GST_CAT_DEFAULT gst_inter_audio_sink_debug_category
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@ -253,11 +256,11 @@ gst_inter_audio_sink_render (GstBaseSink * sink, GstBuffer * buffer)
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g_mutex_lock (&interaudiosink->surface->mutex);
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n = gst_adapter_available (interaudiosink->surface->audio_adapter) / 4;
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#define SIZE 1600
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if (n > SIZE * 3) {
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GST_WARNING_OBJECT (interaudiosink, "flushing %d samples", SIZE / 2);
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while (n > PERIOD * N_PERIODS) {
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GST_WARNING_OBJECT (interaudiosink, "flushing %d samples", PERIOD / 2);
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gst_adapter_flush (interaudiosink->surface->audio_adapter,
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(SIZE / 2) * bpf);
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(PERIOD / 2) * bpf);
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n -= (PERIOD / 2);
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}
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gst_adapter_push (interaudiosink->surface->audio_adapter,
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gst_buffer_ref (buffer));
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@ -50,6 +50,9 @@
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GST_DEBUG_CATEGORY_STATIC (gst_inter_audio_src_debug_category);
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#define GST_CAT_DEFAULT gst_inter_audio_src_debug_category
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#define PERIOD 1600
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#define N_PERIODS 10
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/* prototypes */
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static void gst_inter_audio_src_set_property (GObject * object,
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guint property_id, const GValue * value, GParamSpec * pspec);
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@ -281,8 +284,6 @@ gst_inter_audio_src_get_times (GstBaseSrc * src, GstBuffer * buffer,
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}
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}
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#define SIZE 1600
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static GstFlowReturn
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gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
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GstBuffer ** buf)
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@ -302,7 +303,7 @@ gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
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if (!gst_audio_info_is_equal (&interaudiosrc->surface->audio_info,
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&interaudiosrc->info)) {
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caps = gst_audio_info_to_caps (&interaudiosrc->surface->audio_info);
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interaudiosrc->timestamp_offset =
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interaudiosrc->timestamp_offset +=
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gst_util_uint64_scale_int (interaudiosrc->n_samples, GST_SECOND,
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interaudiosrc->info.rate);
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interaudiosrc->n_samples = 0;
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@ -311,16 +312,19 @@ gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
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bpf = interaudiosrc->surface->audio_info.bpf;
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n = bpf >
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0 ? gst_adapter_available (interaudiosrc->surface->audio_adapter) /
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bpf : 0;
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if (n > SIZE * 3) {
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GST_WARNING_OBJECT (interaudiosrc, "flushing %d samples", SIZE / 2);
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gst_adapter_flush (interaudiosrc->surface->audio_adapter, (SIZE / 2) * bpf);
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n -= (SIZE / 2);
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if (bpf > 0)
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n = gst_adapter_available (interaudiosrc->surface->audio_adapter) / bpf;
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else
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n = 0;
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while (n > PERIOD * 10) {
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GST_WARNING_OBJECT (interaudiosrc, "flushing %d samples", PERIOD / 2);
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gst_adapter_flush (interaudiosrc->surface->audio_adapter,
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(PERIOD / 2) * bpf);
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n -= (PERIOD / 2);
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}
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if (n > SIZE)
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n = SIZE;
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if (n > PERIOD)
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n = PERIOD;
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if (n > 0) {
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buffer = gst_adapter_take_buffer (interaudiosrc->surface->audio_adapter,
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n * bpf);
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@ -341,13 +345,13 @@ gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
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}
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bpf = interaudiosrc->info.bpf;
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if (n < SIZE) {
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if (n < PERIOD) {
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GstMapInfo map;
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GstMemory *mem;
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GST_WARNING_OBJECT (interaudiosrc, "creating %d samples of silence",
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SIZE - n);
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mem = gst_allocator_alloc (NULL, (SIZE - n) * bpf, NULL);
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PERIOD - n);
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mem = gst_allocator_alloc (NULL, (PERIOD - n) * bpf, NULL);
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if (gst_memory_map (mem, &map, GST_MAP_WRITE)) {
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gst_audio_format_fill_silence (interaudiosrc->info.finfo, map.data,
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map.size);
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@ -356,20 +360,18 @@ gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
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buffer = gst_buffer_make_writable (buffer);
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gst_buffer_prepend_memory (buffer, mem);
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}
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n = SIZE;
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n = PERIOD;
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GST_BUFFER_OFFSET (buffer) = interaudiosrc->n_samples;
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GST_BUFFER_OFFSET_END (buffer) = interaudiosrc->n_samples + n;
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GST_BUFFER_TIMESTAMP (buffer) =
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GST_BUFFER_TIMESTAMP (buffer) = interaudiosrc->timestamp_offset +
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gst_util_uint64_scale_int (interaudiosrc->n_samples, GST_SECOND,
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interaudiosrc->info.rate);
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GST_DEBUG_OBJECT (interaudiosrc, "create ts %" GST_TIME_FORMAT,
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GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
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GST_BUFFER_DURATION (buffer) =
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GST_BUFFER_DURATION (buffer) = interaudiosrc->timestamp_offset +
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gst_util_uint64_scale_int (interaudiosrc->n_samples + n, GST_SECOND,
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interaudiosrc->info.rate) - GST_BUFFER_TIMESTAMP (buffer);
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GST_BUFFER_OFFSET (buffer) = interaudiosrc->n_samples;
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GST_BUFFER_OFFSET_END (buffer) = -1;
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GST_BUFFER_FLAG_UNSET (buffer, GST_BUFFER_FLAG_DISCONT);
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if (interaudiosrc->n_samples == 0) {
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GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
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@ -394,10 +396,9 @@ gst_inter_audio_src_query (GstBaseSrc * src, GstQuery * query)
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GstClockTime min_latency, max_latency;
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if (interaudiosrc->info.rate > 0) {
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/* FIXME: Where does the 30 come from? */
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/* 1.5 just as a good measure */
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min_latency =
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30 * gst_util_uint64_scale_int (GST_SECOND, SIZE,
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1.5 * N_PERIODS * gst_util_uint64_scale_int (GST_SECOND, PERIOD,
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interaudiosrc->info.rate);
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max_latency = min_latency;
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