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/* ------------------------------------------------------------------
* 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
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