Nagram/TMessagesProj/jni/voip/webrtc/pc/rtp_receiver.cc
2020-09-30 16:48:47 +03:00

63 lines
2.0 KiB
C++

/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "pc/rtp_receiver.h"
#include <stddef.h>
#include <utility>
#include <vector>
#include "api/media_stream_proxy.h"
#include "api/media_stream_track_proxy.h"
#include "pc/media_stream.h"
#include "rtc_base/checks.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/trace_event.h"
namespace webrtc {
// This function is only expected to be called on the signalling thread.
int RtpReceiverInternal::GenerateUniqueId() {
static int g_unique_id = 0;
return ++g_unique_id;
}
std::vector<rtc::scoped_refptr<MediaStreamInterface>>
RtpReceiverInternal::CreateStreamsFromIds(std::vector<std::string> stream_ids) {
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams(
stream_ids.size());
for (size_t i = 0; i < stream_ids.size(); ++i) {
streams[i] = MediaStreamProxy::Create(
rtc::Thread::Current(), MediaStream::Create(std::move(stream_ids[i])));
}
return streams;
}
// Attempt to attach the frame decryptor to the current media channel on the
// correct worker thread only if both the media channel exists and a ssrc has
// been allocated to the stream.
void RtpReceiverInternal::MaybeAttachFrameDecryptorToMediaChannel(
const absl::optional<uint32_t>& ssrc,
rtc::Thread* worker_thread,
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor,
cricket::MediaChannel* media_channel,
bool stopped) {
if (media_channel && frame_decryptor && ssrc.has_value() && !stopped) {
worker_thread->Invoke<void>(RTC_FROM_HERE, [&] {
media_channel->SetFrameDecryptor(*ssrc, frame_decryptor);
});
}
}
} // namespace webrtc