/* ------------------------------------------------------------------ * Copyright (C) 2019 Matthias P. Braendli * * Licensed under the Apache License, Version 2.0 (the "License"); * you may not use this file except in compliance with the License. * You may obtain a copy of the License at * * http://www.apache.org/licenses/LICENSE-2.0 * * Unless required by applicable law or agreed to in writing, software * distributed under the License is distributed on an "AS IS" BASIS, * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either * express or implied. * See the License for the specific language governing permissions * and limitations under the License. * ------------------------------------------------------------------- */ #include #include #include #include #include #include #include #include #include "GSTInput.h" #include "config.h" #if HAVE_GST #include #include using namespace std; GSTData::GSTData(SampleQueue& samplequeue) : samplequeue(samplequeue) { } GSTInput::GSTInput(const std::string& uri, const std::string& pipeline, int rate, unsigned channels, SampleQueue& queue) : m_uri(uri), m_pipeline(pipeline), m_channels(channels), m_rate(rate), m_gst_data(queue) { } static void error_cb(GstBus *bus, GstMessage *msg, GSTData *data) { GError *err; gchar *debug_info; /* Print error details on the screen */ gst_message_parse_error(msg, &err, &debug_info); g_printerr("Error received from element %s: %s\n", GST_OBJECT_NAME (msg->src), err->message); g_printerr("Debugging information: %s\n", debug_info ? debug_info : "none"); g_clear_error(&err); g_free(debug_info); } static void cb_newpad(GstElement *decodebin, GstPad *pad, GSTData *data) { /* only link once */ GstPad *audiopad = gst_element_get_static_pad(data->audio_convert, "sink"); if (GST_PAD_IS_LINKED(audiopad)) { g_object_unref(audiopad); return; } /* check media type */ GstCaps *caps = gst_pad_query_caps(pad, NULL); GstStructure *str = gst_caps_get_structure(caps, 0); if (!g_strrstr(gst_structure_get_name(str), "audio")) { gst_caps_unref(caps); gst_object_unref(audiopad); return; } gst_caps_unref(caps); gst_pad_link(pad, audiopad); g_object_unref(audiopad); } static GstFlowReturn new_sample(GstElement *sink, GSTData *data) { /* Retrieve the buffer */ GstSample* sample = gst_app_sink_pull_sample(GST_APP_SINK(sink)); if (sample) { GstBuffer* buffer = gst_sample_get_buffer(sample); GstMapInfo map; gst_buffer_map(buffer, &map, GST_MAP_READ); data->samplequeue.push(map.data, map.size); gst_buffer_unmap(buffer, &map); gst_sample_unref(sample); return GST_FLOW_OK; } return GST_FLOW_ERROR; } void GSTInput::prepare() { gst_init(nullptr, nullptr); if (not m_uri.empty()) { m_gst_data.uridecodebin = gst_element_factory_make("uridecodebin", "uridecodebin"); assert(m_gst_data.uridecodebin != nullptr); g_object_set(m_gst_data.uridecodebin, "uri", m_uri.c_str(), nullptr); g_signal_connect(m_gst_data.uridecodebin, "pad-added", G_CALLBACK(cb_newpad), &m_gst_data); 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, #if (GST_VERSION_MAJOR == 1 && GST_VERSION_MINOR >= 10) || GST_VERSION_MAJOR > 1 "sinc-filter-mode", GST_AUDIO_RESAMPLER_FILTER_MODE_FULL, #else #warning "GStreamer version is too old to set GST_AUDIO_RESAMPLER_FILTER_MODE_FULL" GST_VERSION_MAJOR #endif "quality", 6, // between 0 and 10, 10 being best /* default audio-resampler-method: GST_AUDIO_RESAMPLER_METHOD_KAISER */ NULL); } else if (not m_pipeline.empty()) { m_gst_data.custom_bin = gst_parse_bin_from_description(m_pipeline.c_str(), true, nullptr); if (m_gst_data.custom_bin == nullptr) { throw runtime_error("Could not instantiate pipeline"); } } m_gst_data.caps_filter = gst_element_factory_make("capsfilter", "caps_filter"); assert(m_gst_data.caps_filter != nullptr); GstAudioInfo info; gst_audio_info_set_format(&info, GST_AUDIO_FORMAT_S16, m_rate, m_channels, NULL); GstCaps *audio_caps = gst_audio_info_to_caps(&info); g_object_set(m_gst_data.caps_filter, "caps", audio_caps, NULL); m_gst_data.app_sink = gst_element_factory_make("appsink", "app_sink"); assert(m_gst_data.app_sink != nullptr); 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); if (not m_uri.empty()) { 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"); } } else if (not m_pipeline.empty()) { gst_bin_add_many(GST_BIN(m_gst_data.pipeline), m_gst_data.custom_bin, m_gst_data.caps_filter, m_gst_data.app_sink, NULL); if (gst_element_link_many( m_gst_data.custom_bin, m_gst_data.caps_filter, m_gst_data.app_sink, NULL) != true) { throw runtime_error("Could not link GST elements"); } } m_gst_data.bus = gst_element_get_bus(m_gst_data.pipeline); gst_bus_add_signal_watch(m_gst_data.bus); 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) { return m_running; } ICY_TEXT_t GSTInput::get_icy_text() const { ICY_TEXT_t now_playing; { std::lock_guard 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)); struct user_data_t { string new_artist; string new_title; } user_data; auto extract_title = [](const GstTagList *list, const gchar *tag, void *user_data) { GValue val = { 0, }; auto data = (user_data_t*)user_data; gst_tag_list_copy_value(&val, list, tag); if (G_VALUE_HOLDS_STRING(&val)) { if (strcmp(tag, "title") == 0) { data->new_title = g_value_get_string(&val); } else if (strcmp(tag, "artist") == 0) { data->new_artist = g_value_get_string(&val); } } g_value_unset(&val); }; gst_tag_list_foreach(tags, extract_title, &user_data); gst_tag_list_unref(tags); { std::lock_guard lock(m_nowplaying_mutex); if (user_data.new_artist.empty()) { m_nowplaying.useNowPlaying(user_data.new_title); } else { m_nowplaying.useArtistTitle(user_data.new_artist, user_data.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); } } GSTInput::~GSTInput() { if (m_gst_data.pipeline) { gst_element_set_state(m_gst_data.pipeline, GST_STATE_NULL); } m_running = false; // Ensures push() doesn't get blocked m_gst_data.samplequeue.clear(); if (m_thread.joinable()) { m_thread.join(); } if (m_gst_data.bus) { gst_object_unref(m_gst_data.bus); } if (m_gst_data.pipeline) { gst_object_unref(m_gst_data.pipeline); } } #endif // HAVE_GST