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Implement source timestamp synchronization

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Enhance audio quality
This commit is contained in:
antimof
2020-05-02 17:16:05 +03:00
committed by GitHub
parent 0b97ab0757
commit 0c81570b9b
1 changed files with 8 additions and 3 deletions
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11
renderers/audio_renderer_gstreamer.c
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@@ -44,7 +44,8 @@ audio_renderer_t *audio_renderer_init(logger_t *logger, video_renderer_t *video_
}
renderer->logger = logger;
renderer->pipeline = gst_parse_launch("appsrc name=audio_source is-live=true ! queue ! decodebin ! audioconvert ! audiorate ! volume name=volume volume=6 ! queue ! autoaudiosink sync=false", &error);
renderer->pipeline = gst_parse_launch("appsrc name=audio_source stream-type=0 format=GST_FORMAT_TIME is-live=true ! queue ! decodebin !"
"audioconvert ! volume name=volume ! level ! autoaudiosink sync=false", &error);
g_assert (renderer->pipeline);
renderer->appsrc = gst_bin_get_by_name (GST_BIN (renderer->pipeline), "audio_source");
@@ -89,14 +90,18 @@ void audio_renderer_render_buffer(audio_renderer_t *renderer, raop_ntp_t *ntp, u
buffer = gst_buffer_new_and_alloc(data_len);
assert(buffer != NULL);
GST_BUFFER_PTS (buffer) = GST_BUFFER_TIMESTAMP (buffer);
gst_buffer_fill(buffer, 0, data, data_len);
gst_app_src_push_buffer(GST_APP_SRC(renderer->appsrc), buffer);
}
void audio_renderer_set_volume(audio_renderer_t *renderer, float volume) {
//g_object_set(renderer->volume, "volume", volume, NULL);
float avol;
if (fabs(volume) < 28) {
avol=floorf(((28-fabs(volume))/28)*10)/10;
g_object_set(renderer->volume, "volume", avol, NULL);
}
}
void audio_renderer_flush(audio_renderer_t *renderer) {