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Diffstat (limited to 'src/GSTInput.cpp')
| -rw-r--r-- | src/GSTInput.cpp | 188 |
1 files changed, 188 insertions, 0 deletions
diff --git a/src/GSTInput.cpp b/src/GSTInput.cpp new file mode 100644 index 0000000..41fbfc0 --- /dev/null +++ b/src/GSTInput.cpp @@ -0,0 +1,188 @@ +/* ------------------------------------------------------------------ + * 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 <string> +#include <chrono> +#include <algorithm> +#include <functional> +#include <cstddef> +#include <cstdint> +#include <cstdio> +#include <cstring> + +#include <gst/audio/audio.h> + +#include "GSTInput.h" + +#include "config.h" + +#if HAVE_GST + +using namespace std; + +GSTData::GSTData(SampleQueue<uint8_t>& samplequeue) : + samplequeue(samplequeue) +{ } + +GSTInput::GSTInput(const std::string& uri, + int rate, + unsigned channels, + SampleQueue<uint8_t>& queue) : + m_uri(uri), + m_channels(channels), + m_rate(rate), + m_gst_data(queue), + m_samplequeue(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); + + g_main_loop_quit(data->main_loop); +} + +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) { + GstSample *sample; + /* Retrieve the buffer */ + g_signal_emit_by_name(sink, "pull-sample", &sample); + 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); + + 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.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); + + 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); + + gst_bin_add_many(GST_BIN(m_gst_data.pipeline), + m_gst_data.uridecodebin, + m_gst_data.audio_convert, + m_gst_data.caps_filter, + m_gst_data.app_sink, NULL); + + if (gst_element_link_many( + m_gst_data.audio_convert, + 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); +} + +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); + + if (msg) { + gst_message_unref(msg); + return false; + } + return true; +} + +GSTInput::~GSTInput() +{ + fprintf(stderr, "<<<<<<<<<<<<<<<<<<<< DTOR\n"); + + if (m_gst_data.bus) { + gst_object_unref(m_gst_data.bus); + } + + if (m_gst_data.pipeline) { + gst_element_set_state(m_gst_data.pipeline, GST_STATE_NULL); + gst_object_unref(m_gst_data.pipeline); + } +} + +#endif // HAVE_GST |