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/*
* Copyright (C) 2017-2022 Igalia S.L. All rights reserved.
* Copyright (C) 2022 Metrological Group B.V.
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "config.h"
#include "RealtimeOutgoingAudioSourceGStreamer.h"
#if USE(GSTREAMER_WEBRTC)
#include "GStreamerRegistryScanner.h"
#include "MediaStreamTrack.h"
#include <wtf/text/StringToIntegerConversion.h>
GST_DEBUG_CATEGORY_EXTERN(webkit_webrtc_endpoint_debug);
#define GST_CAT_DEFAULT webkit_webrtc_endpoint_debug
namespace WebCore {
RealtimeOutgoingAudioSourceGStreamer::RealtimeOutgoingAudioSourceGStreamer(const String& mediaStreamId, MediaStreamTrack& track)
: RealtimeOutgoingMediaSourceGStreamer(mediaStreamId, track)
{
static Atomic<uint64_t> sourceCounter = 0;
gst_element_set_name(m_bin.get(), makeString("outgoing-audio-source-", sourceCounter.exchangeAdd(1)).ascii().data());
m_audioconvert = makeGStreamerElement("audioconvert", nullptr);
m_audioresample = makeGStreamerElement("audioresample", nullptr);
gst_bin_add_many(GST_BIN_CAST(m_bin.get()), m_audioconvert.get(), m_audioresample.get(), nullptr);
}
RTCRtpCapabilities RealtimeOutgoingAudioSourceGStreamer::rtpCapabilities() const
{
auto& registryScanner = GStreamerRegistryScanner::singleton();
return registryScanner.audioRtpCapabilities(GStreamerRegistryScanner::Configuration::Encoding);
}
bool RealtimeOutgoingAudioSourceGStreamer::setPayloadType(const GRefPtr<GstCaps>& caps)
{
GST_DEBUG_OBJECT(m_bin.get(), "Setting payload caps: %" GST_PTR_FORMAT, caps.get());
auto* structure = gst_caps_get_structure(caps.get(), 0);
const char* encodingName = gst_structure_get_string(structure, "encoding-name");
if (!encodingName) {
GST_ERROR_OBJECT(m_bin.get(), "encoding-name not found");
return false;
}
auto encoding = makeString(encodingName).convertToASCIILowercase();
m_payloader = makeGStreamerElement(makeString("rtp"_s, encoding, "pay"_s).ascii().data(), nullptr);
if (UNLIKELY(!m_payloader)) {
GST_ERROR_OBJECT(m_bin.get(), "RTP payloader not found for encoding %s", encodingName);
return false;
}
if (encoding == "opus"_s) {
m_encoder = makeGStreamerElement("opusenc", nullptr);
if (!m_encoder)
return false;
// FIXME: Enable dtx too?
gst_util_set_object_arg(G_OBJECT(m_encoder.get()), "audio-type", "voice");
if (const char* useInbandFec = gst_structure_get_string(structure, "useinbandfec")) {
if (!strcmp(useInbandFec, "1"))
g_object_set(m_encoder.get(), "inband-fec", true, nullptr);
}
} else if (encoding == "g722"_s)
m_encoder = makeGStreamerElement("avenc_g722", nullptr);
else if (encoding == "pcma"_s)
m_encoder = makeGStreamerElement("alawenc", nullptr);
else if (encoding == "pcmu"_s)
m_encoder = makeGStreamerElement("mulawenc", nullptr);
else if (encoding == "isac"_s)
m_encoder = makeGStreamerElement("isacenc", nullptr);
else {
GST_ERROR_OBJECT(m_bin.get(), "Unsupported outgoing audio encoding: %s", encodingName);
return false;
}
if (!m_encoder) {
GST_ERROR_OBJECT(m_bin.get(), "Encoder not found for encoding %s", encodingName);
return false;
}
// FIXME: Re-enable auto-header-extension. Currently triggers caps negotiation error.
// Align MTU with libwebrtc implementation, also helping to reduce packet fragmentation.
g_object_set(m_payloader.get(), "auto-header-extension", FALSE, "mtu", 1200, nullptr);
if (const char* minPTime = gst_structure_get_string(structure, "minptime")) {
auto time = String::fromLatin1(minPTime);
if (auto value = parseIntegerAllowingTrailingJunk<int64_t>(time))
g_object_set(m_payloader.get(), "min-ptime", *value * GST_MSECOND, nullptr);
}
int payloadType;
if (gst_structure_get_int(structure, "payload", &payloadType))
g_object_set(m_payloader.get(), "pt", payloadType, nullptr);
g_object_set(m_capsFilter.get(), "caps", caps.get(), nullptr);
gst_bin_add_many(GST_BIN_CAST(m_bin.get()), m_payloader.get(), m_encoder.get(), nullptr);
auto preEncoderSinkPad = adoptGRef(gst_element_get_static_pad(m_preEncoderQueue.get(), "sink"));
if (!gst_pad_is_linked(preEncoderSinkPad.get()))
gst_element_link_many(m_outgoingSource.get(), m_valve.get(), m_audioconvert.get(), m_audioresample.get(), m_preEncoderQueue.get(), nullptr);
gst_element_link_many(m_preEncoderQueue.get(), m_encoder.get(), m_payloader.get(), m_postEncoderQueue.get(), nullptr);
gst_bin_sync_children_states(GST_BIN_CAST(m_bin.get()));
return true;
}
void RealtimeOutgoingAudioSourceGStreamer::codecPreferencesChanged(const GRefPtr<GstCaps>& codecPreferences)
{
gst_element_set_locked_state(m_bin.get(), TRUE);
if (m_payloader) {
gst_element_set_locked_state(m_payloader.get(), TRUE);
gst_element_set_locked_state(m_encoder.get(), TRUE);
gst_element_set_state(m_payloader.get(), GST_STATE_NULL);
gst_element_set_state(m_encoder.get(), GST_STATE_NULL);
gst_element_unlink_many(m_preEncoderQueue.get(), m_encoder.get(), m_payloader.get(), m_postEncoderQueue.get(), nullptr);
gst_bin_remove_many(GST_BIN_CAST(m_bin.get()), m_payloader.get(), m_encoder.get(), nullptr);
m_payloader.clear();
m_encoder.clear();
}
setPayloadType(codecPreferences);
gst_element_set_locked_state(m_bin.get(), FALSE);
gst_element_sync_state_with_parent(m_bin.get());
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS(GST_BIN_CAST(m_bin.get()), GST_DEBUG_GRAPH_SHOW_ALL, "outgoing-audio-new-codec-prefs");
}
} // namespace WebCore
#endif // USE(GSTREAMER_WEBRTC)
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