1/*
2 * Copyright (C) 2018 Metrological Group B.V.
3 * Author: Thibault Saunier <tsaunier@igalia.com>
4 * Author: Alejandro G. Castro <alex@igalia.com>
5 *
6 * This library is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Library General Public
8 * License as published by the Free Software Foundation; either
9 * version 2 of the License, or (at your option) any later version.
10 *
11 * This library is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Library General Public License for more details.
15 *
16 * You should have received a copy of the GNU Library General Public License
17 * aint with this library; see the file COPYING.LIB. If not, write to
18 * the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
19 * Boston, MA 02110-1301, USA.
20 */
21
22#include "config.h"
23
24#if ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
25#include "GStreamerAudioCapturer.h"
26
27#include "LibWebRTCAudioFormat.h"
28
29#include <gst/app/gstappsink.h>
30
31namespace WebCore {
32
33GStreamerAudioCapturer::GStreamerAudioCapturer(GStreamerCaptureDevice device)
34 : GStreamerCapturer(device, adoptGRef(gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, LibWebRTCAudioFormat::sampleRate, nullptr)))
35{
36}
37
38GStreamerAudioCapturer::GStreamerAudioCapturer()
39 : GStreamerCapturer("appsrc", adoptGRef(gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, LibWebRTCAudioFormat::sampleRate, nullptr)))
40{
41}
42
43GstElement* GStreamerAudioCapturer::createConverter()
44{
45 auto converter = gst_parse_bin_from_description("audioconvert ! audioresample", TRUE, nullptr);
46
47 ASSERT(converter);
48
49 return converter;
50}
51
52bool GStreamerAudioCapturer::setSampleRate(int sampleRate)
53{
54
55 if (sampleRate <= 0) {
56 GST_INFO_OBJECT(m_pipeline.get(), "Not forcing sample rate");
57
58 return false;
59 }
60
61 GST_INFO_OBJECT(m_pipeline.get(), "Setting SampleRate %d", sampleRate);
62 m_caps = adoptGRef(gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, sampleRate, nullptr));
63
64 if (!m_capsfilter.get())
65 return false;
66
67 g_object_set(m_capsfilter.get(), "caps", m_caps.get(), nullptr);
68
69 return true;
70}
71
72} // namespace WebCore
73
74#endif // ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
75