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author | Matthias P. Braendli <matthias.braendli@mpb.li> | 2020-03-31 17:07:38 +0200 |
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committer | Matthias P. Braendli <matthias.braendli@mpb.li> | 2020-03-31 17:07:38 +0200 |
commit | ec75b9e317baf249d67295300bc5308b7c33f4ac (patch) | |
tree | 6f43693530b463fc913f7c7153a3f54a43ebd04b /src/GSTInput.cpp | |
parent | a1eb6cf861d3c1cbd4e6c016be3cbd2a1e3d797d (diff) | |
download | ODR-AudioEnc-ec75b9e317baf249d67295300bc5308b7c33f4ac.tar.gz ODR-AudioEnc-ec75b9e317baf249d67295300bc5308b7c33f4ac.tar.bz2 ODR-AudioEnc-ec75b9e317baf249d67295300bc5308b7c33f4ac.zip |
Fix GStreamer input, rework ICY-Text write
Diffstat (limited to 'src/GSTInput.cpp')
-rw-r--r-- | src/GSTInput.cpp | 112 |
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); |