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completely ignore timestamps when sync=false (gstreamer)

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This commit is contained in:
F. Duncanh
2023-09-03 19:53:03 -04:00
parent 18a19c1557
commit dc52c04ed0
2 changed files with 41 additions and 21 deletions
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38
renderers/audio_renderer_gstreamer.c
View File

@@ -37,6 +37,9 @@ static const gchar *avdec_alac = "avdec_alac";
static gboolean aac = FALSE;
static gboolean alac = FALSE;
static gboolean render_audio = FALSE;
static gboolean async = FALSE;
static gboolean vsync = FALSE;
static gboolean sync = FALSE;
typedef struct audio_renderer_s {
GstElement *appsrc;
@@ -157,18 +160,22 @@ void audio_renderer_init(logger_t *render_logger, const char* audiosink, const b
g_string_append (launch, audiosink);
switch(i) {
case 1: /*ALAC*/
if (*audio_sync) {
if (*audio_sync) {
g_string_append (launch, " sync=true");
} else {
async = TRUE;
} else {
g_string_append (launch, " sync=false");
}
async = FALSE;
}
break;
default:
if (*video_sync) {
g_string_append (launch, " sync=true");
} else {
vsync = TRUE;
} else {
g_string_append (launch, " sync=false");
}
vsync = FALSE;
}
break;
}
renderer_type[i]->pipeline = gst_parse_launch(launch->str, &error);
@@ -240,6 +247,7 @@ static void get_renderer_type(unsigned char *ct, int *id) {
} else {
logger_log(logger, LOGGER_INFO, "*** GStreamer libav plugin feature avdec_aac is missing, cannot decode AAC audio");
}
sync = vsync;
break;
case 1:
if (alac) {
@@ -247,9 +255,11 @@ static void get_renderer_type(unsigned char *ct, int *id) {
} else {
logger_log(logger, LOGGER_INFO, "*** GStreamer libav plugin feature avdec_alac is missing, cannot decode ALAC audio");
}
sync = async;
break;
case 3:
render_audio = TRUE;
sync = FALSE;
break;
default:
break;
@@ -286,12 +296,14 @@ void audio_renderer_render_buffer(unsigned char* data, int *data_len, unsigned s
GstClockTime pts = (GstClockTime) *ntp_time ; /* now in nsecs */
//GstClockTimeDiff latency = GST_CLOCK_DIFF(gst_element_get_current_clock_time (renderer->appsrc), pts);
if (pts >= gst_audio_pipeline_base_time) {
pts -= gst_audio_pipeline_base_time;
} else {
logger_log(logger, LOGGER_ERR, "*** invalid ntp_time < gst_audio_pipeline_base_time\n%8.6f ntp_time\n%8.6f base_time",
((double) *ntp_time) / SECOND_IN_NSECS, ((double) gst_audio_pipeline_base_time) / SECOND_IN_NSECS);
return;
if (sync) {
if (pts >= gst_audio_pipeline_base_time) {
pts -= gst_audio_pipeline_base_time;
} else {
logger_log(logger, LOGGER_ERR, "*** invalid ntp_time < gst_audio_pipeline_base_time\n%8.6f ntp_time\n%8.6f base_time",
((double) *ntp_time) / SECOND_IN_NSECS, ((double) gst_audio_pipeline_base_time) / SECOND_IN_NSECS);
return;
}
}
if (data_len == 0 || renderer == NULL) return;
@@ -305,7 +317,9 @@ void audio_renderer_render_buffer(unsigned char* data, int *data_len, unsigned s
buffer = gst_buffer_new_allocate(NULL, *data_len, NULL);
g_assert(buffer != NULL);
//g_print("audio latency %8.6f\n", (double) latency / SECOND_IN_NSECS);
GST_BUFFER_PTS(buffer) = pts;
if (sync) {
GST_BUFFER_PTS(buffer) = pts;
}
gst_buffer_fill(buffer, 0, data, *data_len);
switch (renderer->ct){
case 8: /*AAC-ELD*/

24
renderers/video_renderer_gstreamer.c
View File

@@ -39,7 +39,7 @@ static GstClockTime gst_video_pipeline_base_time = GST_CLOCK_TIME_NONE;
static logger_t *logger = NULL;
static unsigned short width, height, width_source, height_source; /* not currently used */
static bool first_packet = false;
static bool sync = false;
struct video_renderer_s {
GstElement *appsrc, *pipeline, *sink;
@@ -160,8 +160,10 @@ void video_renderer_init(logger_t *render_logger, const char *server_name, vide
g_string_append(launch, " name=video_sink");
if (*video_sync) {
g_string_append(launch, " sync=true");
sync = true;
} else {
g_string_append(launch, " sync=false");
sync = false;
}
logger_log(logger, LOGGER_DEBUG, "GStreamer video pipeline will be:\n\"%s\"", launch->str);
renderer->pipeline = gst_parse_launch(launch->str, &error);
@@ -233,12 +235,14 @@ void video_renderer_render_buffer(unsigned char* data, int *data_len, int *nal_c
GstBuffer *buffer;
GstClockTime pts = (GstClockTime) *ntp_time; /*now in nsecs */
//GstClockTimeDiff latency = GST_CLOCK_DIFF(gst_element_get_current_clock_time (renderer->appsrc), pts);
if (pts >= gst_video_pipeline_base_time) {
pts -= gst_video_pipeline_base_time;
} else {
logger_log(logger, LOGGER_ERR, "*** invalid ntp_time < gst_video_pipeline_base_time\n%8.6f ntp_time\n%8.6f base_time",
((double) *ntp_time) / SECOND_IN_NSECS, ((double) gst_video_pipeline_base_time) / SECOND_IN_NSECS);
return;
if (sync) {
if (pts >= gst_video_pipeline_base_time) {
pts -= gst_video_pipeline_base_time;
} else {
logger_log(logger, LOGGER_ERR, "*** invalid ntp_time < gst_video_pipeline_base_time\n%8.6f ntp_time\n%8.6f base_time",
((double) *ntp_time) / SECOND_IN_NSECS, ((double) gst_video_pipeline_base_time) / SECOND_IN_NSECS);
return;
}
}
g_assert(data_len != 0);
/* first four bytes of valid h264 video data are 0x00, 0x00, 0x00, 0x01. *
@@ -254,8 +258,10 @@ void video_renderer_render_buffer(unsigned char* data, int *data_len, int *nal_c
}
buffer = gst_buffer_new_allocate(NULL, *data_len, NULL);
g_assert(buffer != NULL);
//g_print("video latency %8.6f\n", (double) latency / SECOND_IN_NSECS);
GST_BUFFER_PTS(buffer) = pts;
//g_print("video latency %8.6f\n", (double) latency / SECOND_IN_NSECS);
if (sync) {
GST_BUFFER_PTS(buffer) = pts;
}
gst_buffer_fill(buffer, 0, data, *data_len);
gst_app_src_push_buffer (GST_APP_SRC(renderer->appsrc), buffer);
#ifdef X_DISPLAY_FIX