Telegram-Android/TMessagesProj/jni/voip/webrtc/pc/channel_manager.cc

/*
 *  Copyright 2004 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/channel_manager.h"

#include <algorithm>
#include <utility>

#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
#include "absl/strings/match.h"
#include "api/sequence_checker.h"
#include "media/base/media_constants.h"
#include "rtc_base/checks.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/trace_event.h"

namespace cricket {

// static
std::unique_ptr<ChannelManager> ChannelManager::Create(
    std::unique_ptr<MediaEngineInterface> media_engine,
    bool enable_rtx,
    rtc::Thread* worker_thread,
    rtc::Thread* network_thread) {
  RTC_DCHECK_RUN_ON(worker_thread);
  RTC_DCHECK(network_thread);
  RTC_DCHECK(worker_thread);

  if (media_engine)
    media_engine->Init();

  return absl::WrapUnique(new ChannelManager(
      std::move(media_engine), enable_rtx, worker_thread, network_thread));
}

ChannelManager::ChannelManager(
    std::unique_ptr<MediaEngineInterface> media_engine,
    bool enable_rtx,
    rtc::Thread* worker_thread,
    rtc::Thread* network_thread)
    : media_engine_(std::move(media_engine)),
      worker_thread_(worker_thread),
      network_thread_(network_thread),
      enable_rtx_(enable_rtx) {
  RTC_DCHECK(worker_thread_);
  RTC_DCHECK(network_thread_);
  RTC_DCHECK_RUN_ON(worker_thread_);
}

ChannelManager::~ChannelManager() {
  RTC_DCHECK_RUN_ON(worker_thread_);
}

void ChannelManager::GetSupportedAudioSendCodecs(
    std::vector<AudioCodec>* codecs) const {
  if (!media_engine_) {
    return;
  }
  *codecs = media_engine_->voice().send_codecs();
}

void ChannelManager::GetSupportedAudioReceiveCodecs(
    std::vector<AudioCodec>* codecs) const {
  if (!media_engine_) {
    return;
  }
  *codecs = media_engine_->voice().recv_codecs();
}

void ChannelManager::GetSupportedVideoSendCodecs(
    std::vector<VideoCodec>* codecs) const {
  if (!media_engine_) {
    return;
  }
  codecs->clear();

  std::vector<VideoCodec> video_codecs = media_engine_->video().send_codecs();
  for (const auto& video_codec : video_codecs) {
    if (!enable_rtx_ &&
        absl::EqualsIgnoreCase(kRtxCodecName, video_codec.name)) {
      continue;
    }
    codecs->push_back(video_codec);
  }
}

void ChannelManager::GetSupportedVideoReceiveCodecs(
    std::vector<VideoCodec>* codecs) const {
  if (!media_engine_) {
    return;
  }
  codecs->clear();

  std::vector<VideoCodec> video_codecs = media_engine_->video().recv_codecs();
  for (const auto& video_codec : video_codecs) {
    if (!enable_rtx_ &&
        absl::EqualsIgnoreCase(kRtxCodecName, video_codec.name)) {
      continue;
    }
    codecs->push_back(video_codec);
  }
}

RtpHeaderExtensions ChannelManager::GetDefaultEnabledAudioRtpHeaderExtensions()
    const {
  if (!media_engine_)
    return {};
  return GetDefaultEnabledRtpHeaderExtensions(media_engine_->voice());
}

std::vector<webrtc::RtpHeaderExtensionCapability>
ChannelManager::GetSupportedAudioRtpHeaderExtensions() const {
  if (!media_engine_)
    return {};
  return media_engine_->voice().GetRtpHeaderExtensions();
}

RtpHeaderExtensions ChannelManager::GetDefaultEnabledVideoRtpHeaderExtensions()
    const {
  if (!media_engine_)
    return {};
  return GetDefaultEnabledRtpHeaderExtensions(media_engine_->video());
}

std::vector<webrtc::RtpHeaderExtensionCapability>
ChannelManager::GetSupportedVideoRtpHeaderExtensions() const {
  if (!media_engine_)
    return {};
  return media_engine_->video().GetRtpHeaderExtensions();
}

VoiceChannel* ChannelManager::CreateVoiceChannel(
    webrtc::Call* call,
    const MediaConfig& media_config,
    webrtc::RtpTransportInternal* rtp_transport,
    rtc::Thread* signaling_thread,
    const std::string& content_name,
    bool srtp_required,
    const webrtc::CryptoOptions& crypto_options,
    rtc::UniqueRandomIdGenerator* ssrc_generator,
    const AudioOptions& options) {
  RTC_DCHECK(call);
  RTC_DCHECK(media_engine_);
  // TODO(bugs.webrtc.org/11992): Remove this workaround after updates in
  // PeerConnection and add the expectation that we're already on the right
  // thread.
  if (!worker_thread_->IsCurrent()) {
    return worker_thread_->Invoke<VoiceChannel*>(RTC_FROM_HERE, [&] {
      return CreateVoiceChannel(call, media_config, rtp_transport,
                                signaling_thread, content_name, srtp_required,
                                crypto_options, ssrc_generator, options);
    });
  }

  RTC_DCHECK_RUN_ON(worker_thread_);

  VoiceMediaChannel* media_channel = media_engine_->voice().CreateMediaChannel(
      call, media_config, options, crypto_options);
  if (!media_channel) {
    return nullptr;
  }

  auto voice_channel = std::make_unique<VoiceChannel>(
      worker_thread_, network_thread_, signaling_thread,
      absl::WrapUnique(media_channel), content_name, srtp_required,
      crypto_options, ssrc_generator);

  voice_channel->Init_w(rtp_transport);

  VoiceChannel* voice_channel_ptr = voice_channel.get();
  voice_channels_.push_back(std::move(voice_channel));
  return voice_channel_ptr;
}

void ChannelManager::DestroyVoiceChannel(VoiceChannel* voice_channel) {
  TRACE_EVENT0("webrtc", "ChannelManager::DestroyVoiceChannel");
  RTC_DCHECK(voice_channel);

  if (!worker_thread_->IsCurrent()) {
    worker_thread_->Invoke<void>(RTC_FROM_HERE,
                                 [&] { DestroyVoiceChannel(voice_channel); });
    return;
  }

  RTC_DCHECK_RUN_ON(worker_thread_);

  voice_channels_.erase(absl::c_find_if(
      voice_channels_, [&](const std::unique_ptr<VoiceChannel>& p) {
        return p.get() == voice_channel;
      }));
}

VideoChannel* ChannelManager::CreateVideoChannel(
    webrtc::Call* call,
    const MediaConfig& media_config,
    webrtc::RtpTransportInternal* rtp_transport,
    rtc::Thread* signaling_thread,
    const std::string& content_name,
    bool srtp_required,
    const webrtc::CryptoOptions& crypto_options,
    rtc::UniqueRandomIdGenerator* ssrc_generator,
    const VideoOptions& options,
    webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory) {
  RTC_DCHECK(call);
  RTC_DCHECK(media_engine_);
  // TODO(bugs.webrtc.org/11992): Remove this workaround after updates in
  // PeerConnection and add the expectation that we're already on the right
  // thread.
  if (!worker_thread_->IsCurrent()) {
    return worker_thread_->Invoke<VideoChannel*>(RTC_FROM_HERE, [&] {
      return CreateVideoChannel(call, media_config, rtp_transport,
                                signaling_thread, content_name, srtp_required,
                                crypto_options, ssrc_generator, options,
                                video_bitrate_allocator_factory);
    });
  }

  RTC_DCHECK_RUN_ON(worker_thread_);

  VideoMediaChannel* media_channel = media_engine_->video().CreateMediaChannel(
      call, media_config, options, crypto_options,
      video_bitrate_allocator_factory);
  if (!media_channel) {
    return nullptr;
  }

  auto video_channel = std::make_unique<VideoChannel>(
      worker_thread_, network_thread_, signaling_thread,
      absl::WrapUnique(media_channel), content_name, srtp_required,
      crypto_options, ssrc_generator);

  video_channel->Init_w(rtp_transport);

  VideoChannel* video_channel_ptr = video_channel.get();
  video_channels_.push_back(std::move(video_channel));
  return video_channel_ptr;
}

void ChannelManager::DestroyVideoChannel(VideoChannel* video_channel) {
  TRACE_EVENT0("webrtc", "ChannelManager::DestroyVideoChannel");
  RTC_DCHECK(video_channel);

  if (!worker_thread_->IsCurrent()) {
    worker_thread_->Invoke<void>(RTC_FROM_HERE,
                                 [&] { DestroyVideoChannel(video_channel); });
    return;
  }
  RTC_DCHECK_RUN_ON(worker_thread_);

  video_channels_.erase(absl::c_find_if(
      video_channels_, [&](const std::unique_ptr<VideoChannel>& p) {
        return p.get() == video_channel;
      }));
}

bool ChannelManager::StartAecDump(webrtc::FileWrapper file,
                                  int64_t max_size_bytes) {
  RTC_DCHECK_RUN_ON(worker_thread_);
  return media_engine_->voice().StartAecDump(std::move(file), max_size_bytes);
}

void ChannelManager::StopAecDump() {
  RTC_DCHECK_RUN_ON(worker_thread_);
  media_engine_->voice().StopAecDump();
}

}  // namespace cricket