1/*
2 * Copyright (C) 2011, 2012 Igalia S.L
3 * Copyright (C) 2014 Sebastian Dröge <sebastian@centricular.com>
4 *
5 * This library is free software; you can redistribute it and/or
6 * modify it under the terms of the GNU Lesser General Public
7 * License as published by the Free Software Foundation; either
8 * version 2 of the License, or (at your option) any later version.
9 *
10 * This library is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * Lesser General Public License for more details.
14 *
15 * You should have received a copy of the GNU Lesser General Public
16 * License along with this library; if not, write to the Free Software
17 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
18 */
19
20#include "config.h"
21
22#if ENABLE(WEB_AUDIO)
23
24#include "AudioDestinationGStreamer.h"
25
26#include "AudioChannel.h"
27#include "AudioSourceProvider.h"
28#include "GRefPtrGStreamer.h"
29#include "Logging.h"
30#include "WebKitWebAudioSourceGStreamer.h"
31#include <gst/audio/gstaudiobasesink.h>
32#include <gst/gst.h>
33#include <wtf/glib/GUniquePtr.h>
34#include <wtf/glib/RunLoopSourcePriority.h>
35
36namespace WebCore {
37
38// Size of the AudioBus for playback. The webkitwebaudiosrc element
39// needs to handle this number of frames per cycle as well.
40const unsigned framesToPull = 128;
41
42gboolean messageCallback(GstBus*, GstMessage* message, AudioDestinationGStreamer* destination)
43{
44 return destination->handleMessage(message);
45}
46
47static void autoAudioSinkChildAddedCallback(GstChildProxy*, GObject* object, gchar*, gpointer)
48{
49 if (GST_IS_AUDIO_BASE_SINK(object))
50 g_object_set(GST_AUDIO_BASE_SINK(object), "buffer-time", static_cast<gint64>(100000), nullptr);
51}
52
53std::unique_ptr<AudioDestination> AudioDestination::create(AudioIOCallback& callback, const String&, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float sampleRate)
54{
55 // FIXME: make use of inputDeviceId as appropriate.
56
57 // FIXME: Add support for local/live audio input.
58 if (numberOfInputChannels)
59 LOG(Media, "AudioDestination::create(%u, %u, %f) - unhandled input channels", numberOfInputChannels, numberOfOutputChannels, sampleRate);
60
61 // FIXME: Add support for multi-channel (> stereo) output.
62 if (numberOfOutputChannels != 2)
63 LOG(Media, "AudioDestination::create(%u, %u, %f) - unhandled output channels", numberOfInputChannels, numberOfOutputChannels, sampleRate);
64
65 return std::make_unique<AudioDestinationGStreamer>(callback, sampleRate);
66}
67
68float AudioDestination::hardwareSampleRate()
69{
70 return 44100;
71}
72
73unsigned long AudioDestination::maxChannelCount()
74{
75 // FIXME: query the default audio hardware device to return the actual number
76 // of channels of the device. Also see corresponding FIXME in create().
77 return 0;
78}
79
80AudioDestinationGStreamer::AudioDestinationGStreamer(AudioIOCallback& callback, float sampleRate)
81 : m_callback(callback)
82 , m_renderBus(AudioBus::create(2, framesToPull, false))
83 , m_sampleRate(sampleRate)
84 , m_isPlaying(false)
85{
86 m_pipeline = gst_pipeline_new("play");
87 GRefPtr<GstBus> bus = adoptGRef(gst_pipeline_get_bus(GST_PIPELINE(m_pipeline)));
88 ASSERT(bus);
89 gst_bus_add_signal_watch_full(bus.get(), RunLoopSourcePriority::RunLoopDispatcher);
90 g_signal_connect(bus.get(), "message", G_CALLBACK(messageCallback), this);
91
92 GstElement* webkitAudioSrc = reinterpret_cast<GstElement*>(g_object_new(WEBKIT_TYPE_WEB_AUDIO_SRC,
93 "rate", sampleRate,
94 "bus", m_renderBus.get(),
95 "provider", &m_callback,
96 "frames", framesToPull, nullptr));
97
98 GRefPtr<GstElement> audioSink = gst_element_factory_make("autoaudiosink", nullptr);
99 m_audioSinkAvailable = audioSink;
100 if (!audioSink) {
101 LOG_ERROR("Failed to create GStreamer autoaudiosink element");
102 return;
103 }
104
105 g_signal_connect(audioSink.get(), "child-added", G_CALLBACK(autoAudioSinkChildAddedCallback), nullptr);
106
107 // Autoaudiosink does the real sink detection in the GST_STATE_NULL->READY transition
108 // so it's best to roll it to READY as soon as possible to ensure the underlying platform
109 // audiosink was loaded correctly.
110 GstStateChangeReturn stateChangeReturn = gst_element_set_state(audioSink.get(), GST_STATE_READY);
111 if (stateChangeReturn == GST_STATE_CHANGE_FAILURE) {
112 LOG_ERROR("Failed to change autoaudiosink element state");
113 gst_element_set_state(audioSink.get(), GST_STATE_NULL);
114 m_audioSinkAvailable = false;
115 return;
116 }
117
118 GstElement* audioConvert = gst_element_factory_make("audioconvert", nullptr);
119 GstElement* audioResample = gst_element_factory_make("audioresample", nullptr);
120 gst_bin_add_many(GST_BIN(m_pipeline), webkitAudioSrc, audioConvert, audioResample, audioSink.get(), nullptr);
121
122 // Link src pads from webkitAudioSrc to audioConvert ! audioResample ! autoaudiosink.
123 gst_element_link_pads_full(webkitAudioSrc, "src", audioConvert, "sink", GST_PAD_LINK_CHECK_NOTHING);
124 gst_element_link_pads_full(audioConvert, "src", audioResample, "sink", GST_PAD_LINK_CHECK_NOTHING);
125 gst_element_link_pads_full(audioResample, "src", audioSink.get(), "sink", GST_PAD_LINK_CHECK_NOTHING);
126}
127
128AudioDestinationGStreamer::~AudioDestinationGStreamer()
129{
130 GRefPtr<GstBus> bus = adoptGRef(gst_pipeline_get_bus(GST_PIPELINE(m_pipeline)));
131 ASSERT(bus);
132 g_signal_handlers_disconnect_by_func(bus.get(), reinterpret_cast<gpointer>(messageCallback), this);
133 gst_bus_remove_signal_watch(bus.get());
134
135 gst_element_set_state(m_pipeline, GST_STATE_NULL);
136 gst_object_unref(m_pipeline);
137}
138
139gboolean AudioDestinationGStreamer::handleMessage(GstMessage* message)
140{
141 GUniqueOutPtr<GError> error;
142 GUniqueOutPtr<gchar> debug;
143
144 switch (GST_MESSAGE_TYPE(message)) {
145 case GST_MESSAGE_WARNING:
146 gst_message_parse_warning(message, &error.outPtr(), &debug.outPtr());
147 g_warning("Warning: %d, %s. Debug output: %s", error->code, error->message, debug.get());
148 break;
149 case GST_MESSAGE_ERROR:
150 gst_message_parse_error(message, &error.outPtr(), &debug.outPtr());
151 g_warning("Error: %d, %s. Debug output: %s", error->code, error->message, debug.get());
152 gst_element_set_state(m_pipeline, GST_STATE_NULL);
153 m_isPlaying = false;
154 break;
155 default:
156 break;
157 }
158 return TRUE;
159}
160
161void AudioDestinationGStreamer::start()
162{
163 ASSERT(m_audioSinkAvailable);
164 if (!m_audioSinkAvailable)
165 return;
166
167 if (gst_element_set_state(m_pipeline, GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE) {
168 g_warning("Error: Failed to set pipeline to playing");
169 m_isPlaying = false;
170 return;
171 }
172
173 m_isPlaying = true;
174}
175
176void AudioDestinationGStreamer::stop()
177{
178 ASSERT(m_audioSinkAvailable);
179 if (!m_audioSinkAvailable)
180 return;
181
182 gst_element_set_state(m_pipeline, GST_STATE_PAUSED);
183 m_isPlaying = false;
184}
185
186} // namespace WebCore
187
188#endif // ENABLE(WEB_AUDIO)
189