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authorMatthias P. Braendli <matthias.braendli@mpb.li>2020-03-31 17:07:38 +0200
committerMatthias P. Braendli <matthias.braendli@mpb.li>2020-03-31 17:07:38 +0200
commitec75b9e317baf249d67295300bc5308b7c33f4ac (patch)
tree6f43693530b463fc913f7c7153a3f54a43ebd04b /src/GSTInput.cpp
parenta1eb6cf861d3c1cbd4e6c016be3cbd2a1e3d797d (diff)
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Fix GStreamer input, rework ICY-Text write
Diffstat (limited to 'src/GSTInput.cpp')
-rw-r--r--src/GSTInput.cpp112
1 files changed, 99 insertions, 13 deletions
diff --git a/src/GSTInput.cpp b/src/GSTInput.cpp
index 41fbfc0..bc7d44b 100644
--- a/src/GSTInput.cpp
+++ b/src/GSTInput.cpp
@@ -26,6 +26,7 @@
#include <cstring>
#include <gst/audio/audio.h>
+#include <gst/app/gstappsink.h>
#include "GSTInput.h"
@@ -46,8 +47,7 @@ GSTInput::GSTInput(const std::string& uri,
m_uri(uri),
m_channels(channels),
m_rate(rate),
- m_gst_data(queue),
- m_samplequeue(queue)
+ m_gst_data(queue)
{ }
static void error_cb(GstBus *bus, GstMessage *msg, GSTData *data)
@@ -61,8 +61,6 @@ static void error_cb(GstBus *bus, GstMessage *msg, GSTData *data)
g_printerr("Debugging information: %s\n", debug_info ? debug_info : "none");
g_clear_error(&err);
g_free(debug_info);
-
- g_main_loop_quit(data->main_loop);
}
static void cb_newpad(GstElement *decodebin, GstPad *pad, GSTData *data)
@@ -89,10 +87,9 @@ static void cb_newpad(GstElement *decodebin, GstPad *pad, GSTData *data)
g_object_unref(audiopad);
}
-static GstFlowReturn new_sample (GstElement *sink, GSTData *data) {
- GstSample *sample;
+static GstFlowReturn new_sample(GstElement *sink, GSTData *data) {
/* Retrieve the buffer */
- g_signal_emit_by_name(sink, "pull-sample", &sample);
+ GstSample* sample = gst_app_sink_pull_sample(GST_APP_SINK(sink));
if (sample) {
GstBuffer* buffer = gst_sample_get_buffer(sample);
@@ -121,6 +118,14 @@ void GSTInput::prepare()
m_gst_data.audio_convert = gst_element_factory_make("audioconvert", "audio_convert");
assert(m_gst_data.audio_convert != nullptr);
+ m_gst_data.audio_resample = gst_element_factory_make("audioresample", "audio_resample");
+ assert(m_gst_data.audio_resample != nullptr);
+ g_object_set(m_gst_data.audio_resample,
+ "sinc-filter-mode", GST_AUDIO_RESAMPLER_FILTER_MODE_FULL,
+ "quality", 6, // between 0 and 10, 10 being best
+ /* default audio-resampler-method: GST_AUDIO_RESAMPLER_METHOD_KAISER */
+ NULL);
+
m_gst_data.caps_filter = gst_element_factory_make("capsfilter", "caps_filter");
assert(m_gst_data.caps_filter != nullptr);
@@ -135,6 +140,7 @@ void GSTInput::prepare()
m_gst_data.pipeline = gst_pipeline_new("pipeline");
assert(m_gst_data.pipeline != nullptr);
+ // TODO also set max-buffers
g_object_set(m_gst_data.app_sink, "emit-signals", TRUE, "caps", audio_caps, NULL);
g_signal_connect(m_gst_data.app_sink, "new-sample", G_CALLBACK(new_sample), &m_gst_data);
gst_caps_unref(audio_caps);
@@ -142,11 +148,13 @@ void GSTInput::prepare()
gst_bin_add_many(GST_BIN(m_gst_data.pipeline),
m_gst_data.uridecodebin,
m_gst_data.audio_convert,
+ m_gst_data.audio_resample,
m_gst_data.caps_filter,
m_gst_data.app_sink, NULL);
if (gst_element_link_many(
m_gst_data.audio_convert,
+ m_gst_data.audio_resample,
m_gst_data.caps_filter,
m_gst_data.app_sink, NULL) != true) {
throw runtime_error("Could not link GST elements");
@@ -157,23 +165,101 @@ void GSTInput::prepare()
g_signal_connect(G_OBJECT(m_gst_data.bus), "message::error", (GCallback)error_cb, &m_gst_data);
gst_element_set_state(m_gst_data.pipeline, GST_STATE_PLAYING);
+
+ m_running = true;
+ m_thread = std::thread(&GSTInput::process, this);
}
bool GSTInput::read_source(size_t num_bytes)
{
- // Reading done in glib main loop
- GstMessage *msg = gst_bus_pop_filtered(m_gst_data.bus, GST_MESSAGE_EOS);
+ return m_running;
+}
- if (msg) {
+ICY_TEXT_t GSTInput::get_icy_text() const
+{
+ ICY_TEXT_t now_playing;
+ {
+ std::lock_guard<std::mutex> lock(m_nowplaying_mutex);
+ now_playing = m_nowplaying;
+ }
+
+ return now_playing;
+}
+
+void GSTInput::process()
+{
+ while (m_running) {
+ GstMessage *msg = gst_bus_timed_pop(m_gst_data.bus, 100000);
+
+ if (not msg) {
+ continue;
+ }
+
+ switch (GST_MESSAGE_TYPE(msg)) {
+ case GST_MESSAGE_BUFFERING:
+ {
+ gint percent = 0;
+ gst_message_parse_buffering(msg, &percent);
+ //fprintf(stderr, "GST buffering %d\n", percent);
+ break;
+ }
+ case GST_MESSAGE_TAG:
+ {
+ GstTagList *tags = nullptr;
+ gst_message_parse_tag(msg, &tags);
+ //fprintf(stderr, "Got tags from element %s\n", GST_OBJECT_NAME(msg->src));
+
+ string new_title;
+
+ auto extract_title = [](const GstTagList *list, const gchar *tag, void *user_data) {
+ GValue val = { 0, };
+
+ auto new_title = (string*)user_data;
+
+ gst_tag_list_copy_value(&val, list, tag);
+
+ if (strcmp(tag, "title") == 0 and G_VALUE_HOLDS_STRING(&val)) {
+ *new_title = g_value_dup_string(&val);
+ }
+
+ g_value_unset(&val);
+ };
+
+ gst_tag_list_foreach(tags, extract_title, &new_title);
+
+ gst_tag_list_unref(tags);
+ {
+ std::lock_guard<std::mutex> lock(m_nowplaying_mutex);
+ m_nowplaying.useNowPlaying(new_title);
+ }
+ break;
+ }
+ case GST_MESSAGE_ERROR:
+ {
+ GError *err = nullptr;
+ gst_message_parse_error(msg, &err, nullptr);
+ fprintf(stderr, "GST error: %s\n", err->message);
+ g_error_free(err);
+ m_fault = true;
+ break;
+ }
+ case GST_MESSAGE_EOS:
+ m_fault = true;
+ break;
+ default:
+ //fprintf(stderr, "GST message %s\n", gst_message_type_get_name(GST_MESSAGE_TYPE(msg)));
+ break;
+ }
gst_message_unref(msg);
- return false;
}
- return true;
}
GSTInput::~GSTInput()
{
- fprintf(stderr, "<<<<<<<<<<<<<<<<<<<< DTOR\n");
+ m_running = false;
+ if (m_thread.joinable()) {
+ m_thread.join();
+ }
if (m_gst_data.bus) {
gst_object_unref(m_gst_data.bus);