| // Copyright 2013 The Chromium Authors. All rights reserved. |
| // Use of this source code is governed by a BSD-style license that can be |
| // found in the LICENSE file. |
| |
| #include <algorithm> |
| #include <climits> |
| #include <cstdarg> |
| #include <cstdio> |
| #include <deque> |
| #include <map> |
| #include <string> |
| #include <utility> |
| |
| #include "base/at_exit.h" |
| #include "base/command_line.h" |
| #include "base/logging.h" |
| #include "base/memory/ref_counted.h" |
| #include "base/memory/scoped_ptr.h" |
| #include "base/message_loop/message_loop.h" |
| #include "base/synchronization/lock.h" |
| #include "base/synchronization/waitable_event.h" |
| #include "base/threading/thread.h" |
| #include "base/time/default_tick_clock.h" |
| #include "base/timer/timer.h" |
| #include "media/audio/audio_io.h" |
| #include "media/audio/audio_manager.h" |
| #include "media/audio/audio_parameters.h" |
| #include "media/audio/fake_audio_log_factory.h" |
| #include "media/base/audio_bus.h" |
| #include "media/base/channel_layout.h" |
| #include "media/base/video_frame.h" |
| #include "media/cast/cast_config.h" |
| #include "media/cast/cast_environment.h" |
| #include "media/cast/cast_receiver.h" |
| #include "media/cast/logging/logging_defines.h" |
| #include "media/cast/net/udp_transport.h" |
| #include "media/cast/test/utility/audio_utility.h" |
| #include "media/cast/test/utility/barcode.h" |
| #include "media/cast/test/utility/default_config.h" |
| #include "media/cast/test/utility/in_process_receiver.h" |
| #include "media/cast/test/utility/input_builder.h" |
| #include "media/cast/test/utility/standalone_cast_environment.h" |
| #include "net/base/net_util.h" |
| |
| #if defined(USE_X11) |
| #include "media/cast/test/linux_output_window.h" |
| #endif // defined(USE_X11) |
| |
| namespace media { |
| namespace cast { |
| |
| // Settings chosen to match default sender settings. |
| #define DEFAULT_SEND_PORT "0" |
| #define DEFAULT_RECEIVE_PORT "2344" |
| #define DEFAULT_SEND_IP "0.0.0.0" |
| #define DEFAULT_AUDIO_FEEDBACK_SSRC "2" |
| #define DEFAULT_AUDIO_INCOMING_SSRC "1" |
| #define DEFAULT_AUDIO_PAYLOAD_TYPE "127" |
| #define DEFAULT_VIDEO_FEEDBACK_SSRC "12" |
| #define DEFAULT_VIDEO_INCOMING_SSRC "11" |
| #define DEFAULT_VIDEO_PAYLOAD_TYPE "96" |
| |
| #if defined(USE_X11) |
| const char* kVideoWindowWidth = "1280"; |
| const char* kVideoWindowHeight = "720"; |
| #endif // defined(USE_X11) |
| |
| void GetPorts(int* tx_port, int* rx_port) { |
| test::InputBuilder tx_input( |
| "Enter send port.", DEFAULT_SEND_PORT, 1, INT_MAX); |
| *tx_port = tx_input.GetIntInput(); |
| |
| test::InputBuilder rx_input( |
| "Enter receive port.", DEFAULT_RECEIVE_PORT, 1, INT_MAX); |
| *rx_port = rx_input.GetIntInput(); |
| } |
| |
| std::string GetIpAddress(const std::string display_text) { |
| test::InputBuilder input(display_text, DEFAULT_SEND_IP, INT_MIN, INT_MAX); |
| std::string ip_address = input.GetStringInput(); |
| // Ensure IP address is either the default value or in correct form. |
| while (ip_address != DEFAULT_SEND_IP && |
| std::count(ip_address.begin(), ip_address.end(), '.') != 3) { |
| ip_address = input.GetStringInput(); |
| } |
| return ip_address; |
| } |
| |
| void GetAudioSsrcs(FrameReceiverConfig* audio_config) { |
| test::InputBuilder input_tx( |
| "Choose audio sender SSRC.", DEFAULT_AUDIO_FEEDBACK_SSRC, 1, INT_MAX); |
| audio_config->feedback_ssrc = input_tx.GetIntInput(); |
| |
| test::InputBuilder input_rx( |
| "Choose audio receiver SSRC.", DEFAULT_AUDIO_INCOMING_SSRC, 1, INT_MAX); |
| audio_config->incoming_ssrc = input_rx.GetIntInput(); |
| } |
| |
| void GetVideoSsrcs(FrameReceiverConfig* video_config) { |
| test::InputBuilder input_tx( |
| "Choose video sender SSRC.", DEFAULT_VIDEO_FEEDBACK_SSRC, 1, INT_MAX); |
| video_config->feedback_ssrc = input_tx.GetIntInput(); |
| |
| test::InputBuilder input_rx( |
| "Choose video receiver SSRC.", DEFAULT_VIDEO_INCOMING_SSRC, 1, INT_MAX); |
| video_config->incoming_ssrc = input_rx.GetIntInput(); |
| } |
| |
| #if defined(USE_X11) |
| void GetWindowSize(int* width, int* height) { |
| // Resolution values based on sender settings |
| test::InputBuilder input_w( |
| "Choose window width.", kVideoWindowWidth, 144, 1920); |
| *width = input_w.GetIntInput(); |
| |
| test::InputBuilder input_h( |
| "Choose window height.", kVideoWindowHeight, 176, 1080); |
| *height = input_h.GetIntInput(); |
| } |
| #endif // defined(USE_X11) |
| |
| void GetAudioPayloadtype(FrameReceiverConfig* audio_config) { |
| test::InputBuilder input("Choose audio receiver payload type.", |
| DEFAULT_AUDIO_PAYLOAD_TYPE, |
| 96, |
| 127); |
| audio_config->rtp_payload_type = input.GetIntInput(); |
| } |
| |
| FrameReceiverConfig GetAudioReceiverConfig() { |
| FrameReceiverConfig audio_config = GetDefaultAudioReceiverConfig(); |
| GetAudioSsrcs(&audio_config); |
| GetAudioPayloadtype(&audio_config); |
| audio_config.rtp_max_delay_ms = 300; |
| return audio_config; |
| } |
| |
| void GetVideoPayloadtype(FrameReceiverConfig* video_config) { |
| test::InputBuilder input("Choose video receiver payload type.", |
| DEFAULT_VIDEO_PAYLOAD_TYPE, |
| 96, |
| 127); |
| video_config->rtp_payload_type = input.GetIntInput(); |
| } |
| |
| FrameReceiverConfig GetVideoReceiverConfig() { |
| FrameReceiverConfig video_config = GetDefaultVideoReceiverConfig(); |
| GetVideoSsrcs(&video_config); |
| GetVideoPayloadtype(&video_config); |
| video_config.rtp_max_delay_ms = 300; |
| return video_config; |
| } |
| |
| AudioParameters ToAudioParameters(const FrameReceiverConfig& config) { |
| const int samples_in_10ms = config.frequency / 100; |
| return AudioParameters(AudioParameters::AUDIO_PCM_LOW_LATENCY, |
| GuessChannelLayout(config.channels), |
| config.frequency, 32, samples_in_10ms); |
| } |
| |
| // An InProcessReceiver that renders video frames to a LinuxOutputWindow and |
| // audio frames via Chromium's audio stack. |
| // |
| // InProcessReceiver pushes audio and video frames to this subclass, and these |
| // frames are pushed into a queue. Then, for audio, the Chromium audio stack |
| // will make polling calls on a separate, unknown thread whereby audio frames |
| // are pulled out of the audio queue as needed. For video, however, NaivePlayer |
| // is responsible for scheduling updates to the screen itself. For both, the |
| // queues are pruned (i.e., received frames are skipped) when the system is not |
| // able to play back as fast as frames are entering the queue. |
| // |
| // This is NOT a good reference implementation for a Cast receiver player since: |
| // 1. It only skips frames to handle slower-than-expected playout, or halts |
| // playback to handle frame underruns. |
| // 2. It makes no attempt to synchronize the timing of playout of the video |
| // frames with the audio frames. |
| // 3. It does nothing to smooth or hide discontinuities in playback due to |
| // timing issues or missing frames. |
| class NaivePlayer : public InProcessReceiver, |
| public AudioOutputStream::AudioSourceCallback { |
| public: |
| NaivePlayer(const scoped_refptr<CastEnvironment>& cast_environment, |
| const net::IPEndPoint& local_end_point, |
| const net::IPEndPoint& remote_end_point, |
| const FrameReceiverConfig& audio_config, |
| const FrameReceiverConfig& video_config, |
| int window_width, |
| int window_height) |
| : InProcessReceiver(cast_environment, |
| local_end_point, |
| remote_end_point, |
| audio_config, |
| video_config), |
| // Maximum age is the duration of 3 video frames. 3 was chosen |
| // arbitrarily, but seems to work well. |
| max_frame_age_(base::TimeDelta::FromSeconds(1) * 3 / |
| video_config.max_frame_rate), |
| #if defined(USE_X11) |
| render_(0, 0, window_width, window_height, "Cast_receiver"), |
| #endif // defined(USE_X11) |
| num_video_frames_processed_(0), |
| num_audio_frames_processed_(0), |
| currently_playing_audio_frame_start_(-1) {} |
| |
| ~NaivePlayer() override {} |
| |
| void Start() override { |
| AudioManager::Get()->GetTaskRunner()->PostTask( |
| FROM_HERE, |
| base::Bind(&NaivePlayer::StartAudioOutputOnAudioManagerThread, |
| base::Unretained(this))); |
| // Note: No need to wait for audio polling to start since the push-and-pull |
| // mechanism is synchronized via the |audio_playout_queue_|. |
| InProcessReceiver::Start(); |
| } |
| |
| void Stop() override { |
| // First, stop audio output to the Chromium audio stack. |
| base::WaitableEvent done(false, false); |
| DCHECK(!AudioManager::Get()->GetTaskRunner()->BelongsToCurrentThread()); |
| AudioManager::Get()->GetTaskRunner()->PostTask( |
| FROM_HERE, |
| base::Bind(&NaivePlayer::StopAudioOutputOnAudioManagerThread, |
| base::Unretained(this), |
| &done)); |
| done.Wait(); |
| |
| // Now, stop receiving new frames. |
| InProcessReceiver::Stop(); |
| |
| // Finally, clear out any frames remaining in the queues. |
| while (!audio_playout_queue_.empty()) { |
| const scoped_ptr<AudioBus> to_be_deleted( |
| audio_playout_queue_.front().second); |
| audio_playout_queue_.pop_front(); |
| } |
| video_playout_queue_.clear(); |
| } |
| |
| private: |
| void StartAudioOutputOnAudioManagerThread() { |
| DCHECK(AudioManager::Get()->GetTaskRunner()->BelongsToCurrentThread()); |
| DCHECK(!audio_output_stream_); |
| audio_output_stream_.reset(AudioManager::Get()->MakeAudioOutputStreamProxy( |
| ToAudioParameters(audio_config()), "")); |
| if (audio_output_stream_.get() && audio_output_stream_->Open()) { |
| audio_output_stream_->Start(this); |
| } else { |
| LOG(ERROR) << "Failed to open an audio output stream. " |
| << "Audio playback disabled."; |
| audio_output_stream_.reset(); |
| } |
| } |
| |
| void StopAudioOutputOnAudioManagerThread(base::WaitableEvent* done) { |
| DCHECK(AudioManager::Get()->GetTaskRunner()->BelongsToCurrentThread()); |
| if (audio_output_stream_.get()) { |
| audio_output_stream_->Stop(); |
| audio_output_stream_->Close(); |
| audio_output_stream_.reset(); |
| } |
| done->Signal(); |
| } |
| |
| //////////////////////////////////////////////////////////////////// |
| // InProcessReceiver overrides. |
| |
| void OnVideoFrame(const scoped_refptr<VideoFrame>& video_frame, |
| const base::TimeTicks& playout_time, |
| bool is_continuous) override { |
| DCHECK(cast_env()->CurrentlyOn(CastEnvironment::MAIN)); |
| LOG_IF(WARNING, !is_continuous) |
| << "Video: Discontinuity in received frames."; |
| video_playout_queue_.push_back(std::make_pair(playout_time, video_frame)); |
| ScheduleVideoPlayout(); |
| uint16 frame_no; |
| if (media::cast::test::DecodeBarcode(video_frame, &frame_no)) { |
| video_play_times_.insert( |
| std::pair<uint16, base::TimeTicks>(frame_no, playout_time)); |
| } else { |
| VLOG(2) << "Barcode decode failed!"; |
| } |
| } |
| |
| void OnAudioFrame(scoped_ptr<AudioBus> audio_frame, |
| const base::TimeTicks& playout_time, |
| bool is_continuous) override { |
| DCHECK(cast_env()->CurrentlyOn(CastEnvironment::MAIN)); |
| LOG_IF(WARNING, !is_continuous) |
| << "Audio: Discontinuity in received frames."; |
| base::AutoLock auto_lock(audio_lock_); |
| uint16 frame_no; |
| if (media::cast::DecodeTimestamp(audio_frame->channel(0), |
| audio_frame->frames(), |
| &frame_no)) { |
| // Since there are lots of audio packets with the same frame_no, |
| // we really want to make sure that we get the playout_time from |
| // the first one. If is_continous is true, then it's possible |
| // that we already missed the first one. |
| if (is_continuous && frame_no == last_audio_frame_no_ + 1) { |
| audio_play_times_.insert( |
| std::pair<uint16, base::TimeTicks>(frame_no, playout_time)); |
| } |
| last_audio_frame_no_ = frame_no; |
| } else { |
| VLOG(2) << "Audio decode failed!"; |
| last_audio_frame_no_ = -2; |
| } |
| audio_playout_queue_.push_back( |
| std::make_pair(playout_time, audio_frame.release())); |
| } |
| |
| // End of InProcessReceiver overrides. |
| //////////////////////////////////////////////////////////////////// |
| |
| //////////////////////////////////////////////////////////////////// |
| // AudioSourceCallback implementation. |
| |
| int OnMoreData(AudioBus* dest, uint32 total_bytes_delay) override { |
| // Note: This method is being invoked by a separate thread unknown to us |
| // (i.e., outside of CastEnvironment). |
| |
| int samples_remaining = dest->frames(); |
| |
| while (samples_remaining > 0) { |
| // Get next audio frame ready for playout. |
| if (!currently_playing_audio_frame_.get()) { |
| base::AutoLock auto_lock(audio_lock_); |
| |
| // Prune the queue, skipping entries that are too old. |
| // TODO(miu): Use |total_bytes_delay| to account for audio buffering |
| // delays upstream. |
| const base::TimeTicks earliest_time_to_play = |
| cast_env()->Clock()->NowTicks() - max_frame_age_; |
| while (!audio_playout_queue_.empty() && |
| audio_playout_queue_.front().first < earliest_time_to_play) { |
| PopOneAudioFrame(true); |
| } |
| if (audio_playout_queue_.empty()) |
| break; |
| |
| currently_playing_audio_frame_ = PopOneAudioFrame(false).Pass(); |
| currently_playing_audio_frame_start_ = 0; |
| } |
| |
| // Copy some or all of the samples in |currently_playing_audio_frame_| to |
| // |dest|. Once all samples in |currently_playing_audio_frame_| have been |
| // consumed, release it. |
| const int num_samples_to_copy = |
| std::min(samples_remaining, |
| currently_playing_audio_frame_->frames() - |
| currently_playing_audio_frame_start_); |
| currently_playing_audio_frame_->CopyPartialFramesTo( |
| currently_playing_audio_frame_start_, |
| num_samples_to_copy, |
| 0, |
| dest); |
| samples_remaining -= num_samples_to_copy; |
| currently_playing_audio_frame_start_ += num_samples_to_copy; |
| if (currently_playing_audio_frame_start_ == |
| currently_playing_audio_frame_->frames()) { |
| currently_playing_audio_frame_.reset(); |
| } |
| } |
| |
| // If |dest| has not been fully filled, then an underrun has occurred; and |
| // fill the remainder of |dest| with zeros. |
| if (samples_remaining > 0) { |
| // Note: Only logging underruns after the first frame has been received. |
| LOG_IF(WARNING, currently_playing_audio_frame_start_ != -1) |
| << "Audio: Playback underrun of " << samples_remaining << " samples!"; |
| dest->ZeroFramesPartial(dest->frames() - samples_remaining, |
| samples_remaining); |
| } |
| |
| return dest->frames(); |
| } |
| |
| void OnError(AudioOutputStream* stream) override { |
| LOG(ERROR) << "AudioOutputStream reports an error. " |
| << "Playback is unlikely to continue."; |
| } |
| |
| // End of AudioSourceCallback implementation. |
| //////////////////////////////////////////////////////////////////// |
| |
| void ScheduleVideoPlayout() { |
| DCHECK(cast_env()->CurrentlyOn(CastEnvironment::MAIN)); |
| |
| // Prune the queue, skipping entries that are too old. |
| const base::TimeTicks now = cast_env()->Clock()->NowTicks(); |
| const base::TimeTicks earliest_time_to_play = now - max_frame_age_; |
| while (!video_playout_queue_.empty() && |
| video_playout_queue_.front().first < earliest_time_to_play) { |
| PopOneVideoFrame(true); |
| } |
| |
| // If the queue is not empty, schedule playout of its first frame. |
| if (video_playout_queue_.empty()) { |
| video_playout_timer_.Stop(); |
| } else { |
| video_playout_timer_.Start( |
| FROM_HERE, |
| video_playout_queue_.front().first - now, |
| base::Bind(&NaivePlayer::PlayNextVideoFrame, |
| base::Unretained(this))); |
| } |
| } |
| |
| void PlayNextVideoFrame() { |
| DCHECK(cast_env()->CurrentlyOn(CastEnvironment::MAIN)); |
| if (!video_playout_queue_.empty()) { |
| const scoped_refptr<VideoFrame> video_frame = PopOneVideoFrame(false); |
| #if defined(USE_X11) |
| render_.RenderFrame(video_frame); |
| #endif // defined(USE_X11) |
| } |
| ScheduleVideoPlayout(); |
| CheckAVSync(); |
| } |
| |
| scoped_refptr<VideoFrame> PopOneVideoFrame(bool is_being_skipped) { |
| DCHECK(cast_env()->CurrentlyOn(CastEnvironment::MAIN)); |
| |
| if (is_being_skipped) { |
| VLOG(1) << "VideoFrame[" << num_video_frames_processed_ |
| << " (dt=" << (video_playout_queue_.front().first - |
| last_popped_video_playout_time_).InMicroseconds() |
| << " usec)]: Skipped."; |
| } else { |
| VLOG(1) << "VideoFrame[" << num_video_frames_processed_ |
| << " (dt=" << (video_playout_queue_.front().first - |
| last_popped_video_playout_time_).InMicroseconds() |
| << " usec)]: Playing " |
| << (cast_env()->Clock()->NowTicks() - |
| video_playout_queue_.front().first).InMicroseconds() |
| << " usec later than intended."; |
| } |
| |
| last_popped_video_playout_time_ = video_playout_queue_.front().first; |
| const scoped_refptr<VideoFrame> ret = video_playout_queue_.front().second; |
| video_playout_queue_.pop_front(); |
| ++num_video_frames_processed_; |
| return ret; |
| } |
| |
| scoped_ptr<AudioBus> PopOneAudioFrame(bool was_skipped) { |
| audio_lock_.AssertAcquired(); |
| |
| if (was_skipped) { |
| VLOG(1) << "AudioFrame[" << num_audio_frames_processed_ |
| << " (dt=" << (audio_playout_queue_.front().first - |
| last_popped_audio_playout_time_).InMicroseconds() |
| << " usec)]: Skipped."; |
| } else { |
| VLOG(1) << "AudioFrame[" << num_audio_frames_processed_ |
| << " (dt=" << (audio_playout_queue_.front().first - |
| last_popped_audio_playout_time_).InMicroseconds() |
| << " usec)]: Playing " |
| << (cast_env()->Clock()->NowTicks() - |
| audio_playout_queue_.front().first).InMicroseconds() |
| << " usec later than intended."; |
| } |
| |
| last_popped_audio_playout_time_ = audio_playout_queue_.front().first; |
| scoped_ptr<AudioBus> ret(audio_playout_queue_.front().second); |
| audio_playout_queue_.pop_front(); |
| ++num_audio_frames_processed_; |
| return ret.Pass(); |
| } |
| |
| void CheckAVSync() { |
| if (video_play_times_.size() > 30 && |
| audio_play_times_.size() > 30) { |
| size_t num_events = 0; |
| base::TimeDelta delta; |
| std::map<uint16, base::TimeTicks>::iterator audio_iter, video_iter; |
| for (video_iter = video_play_times_.begin(); |
| video_iter != video_play_times_.end(); |
| ++video_iter) { |
| audio_iter = audio_play_times_.find(video_iter->first); |
| if (audio_iter != audio_play_times_.end()) { |
| num_events++; |
| // Positive values means audio is running behind video. |
| delta += audio_iter->second - video_iter->second; |
| } |
| } |
| |
| if (num_events > 30) { |
| VLOG(0) << "Audio behind by: " |
| << (delta / num_events).InMilliseconds() |
| << "ms"; |
| video_play_times_.clear(); |
| audio_play_times_.clear(); |
| } |
| } else if (video_play_times_.size() + audio_play_times_.size() > 500) { |
| // We are decoding audio or video timestamps, but not both, clear it out. |
| video_play_times_.clear(); |
| audio_play_times_.clear(); |
| } |
| } |
| |
| // Frames in the queue older than this (relative to NowTicks()) will be |
| // dropped (i.e., playback is falling behind). |
| const base::TimeDelta max_frame_age_; |
| |
| // Outputs created, started, and destroyed by this NaivePlayer. |
| #if defined(USE_X11) |
| test::LinuxOutputWindow render_; |
| #endif // defined(USE_X11) |
| scoped_ptr<AudioOutputStream> audio_output_stream_; |
| |
| // Video playout queue. |
| typedef std::pair<base::TimeTicks, scoped_refptr<VideoFrame> > |
| VideoQueueEntry; |
| std::deque<VideoQueueEntry> video_playout_queue_; |
| base::TimeTicks last_popped_video_playout_time_; |
| int64 num_video_frames_processed_; |
| |
| base::OneShotTimer<NaivePlayer> video_playout_timer_; |
| |
| // Audio playout queue, synchronized by |audio_lock_|. |
| base::Lock audio_lock_; |
| typedef std::pair<base::TimeTicks, AudioBus*> AudioQueueEntry; |
| std::deque<AudioQueueEntry> audio_playout_queue_; |
| base::TimeTicks last_popped_audio_playout_time_; |
| int64 num_audio_frames_processed_; |
| |
| // These must only be used on the audio thread calling OnMoreData(). |
| scoped_ptr<AudioBus> currently_playing_audio_frame_; |
| int currently_playing_audio_frame_start_; |
| |
| std::map<uint16, base::TimeTicks> audio_play_times_; |
| std::map<uint16, base::TimeTicks> video_play_times_; |
| int32 last_audio_frame_no_; |
| }; |
| |
| } // namespace cast |
| } // namespace media |
| |
| int main(int argc, char** argv) { |
| base::AtExitManager at_exit; |
| CommandLine::Init(argc, argv); |
| InitLogging(logging::LoggingSettings()); |
| |
| scoped_refptr<media::cast::CastEnvironment> cast_environment( |
| new media::cast::StandaloneCastEnvironment); |
| |
| // Start up Chromium audio system. |
| media::FakeAudioLogFactory fake_audio_log_factory_; |
| const scoped_ptr<media::AudioManager> audio_manager( |
| media::AudioManager::Create(&fake_audio_log_factory_)); |
| CHECK(media::AudioManager::Get()); |
| |
| media::cast::FrameReceiverConfig audio_config = |
| media::cast::GetAudioReceiverConfig(); |
| media::cast::FrameReceiverConfig video_config = |
| media::cast::GetVideoReceiverConfig(); |
| |
| // Determine local and remote endpoints. |
| int remote_port, local_port; |
| media::cast::GetPorts(&remote_port, &local_port); |
| if (!local_port) { |
| LOG(ERROR) << "Invalid local port."; |
| return 1; |
| } |
| std::string remote_ip_address = media::cast::GetIpAddress("Enter remote IP."); |
| std::string local_ip_address = media::cast::GetIpAddress("Enter local IP."); |
| net::IPAddressNumber remote_ip_number; |
| net::IPAddressNumber local_ip_number; |
| if (!net::ParseIPLiteralToNumber(remote_ip_address, &remote_ip_number)) { |
| LOG(ERROR) << "Invalid remote IP address."; |
| return 1; |
| } |
| if (!net::ParseIPLiteralToNumber(local_ip_address, &local_ip_number)) { |
| LOG(ERROR) << "Invalid local IP address."; |
| return 1; |
| } |
| net::IPEndPoint remote_end_point(remote_ip_number, remote_port); |
| net::IPEndPoint local_end_point(local_ip_number, local_port); |
| |
| // Create and start the player. |
| int window_width = 0; |
| int window_height = 0; |
| #if defined(USE_X11) |
| media::cast::GetWindowSize(&window_width, &window_height); |
| #endif // defined(USE_X11) |
| media::cast::NaivePlayer player(cast_environment, |
| local_end_point, |
| remote_end_point, |
| audio_config, |
| video_config, |
| window_width, |
| window_height); |
| player.Start(); |
| |
| base::MessageLoop().Run(); // Run forever (i.e., until SIGTERM). |
| NOTREACHED(); |
| return 0; |
| } |
