// 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/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 "media/cast/transport/transport/udp_transport.h"
#include "net/base/net_util.h"
#if defined(OS_LINUX)
#include "media/cast/test/linux_output_window.h"
#endif // OS_LINUX
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(OS_LINUX)
const char* kVideoWindowWidth = "1280";
const char* kVideoWindowHeight = "720";
#endif // OS_LINUX
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(OS_LINUX)
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 // OS_LINUX
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(OS_LINUX)
render_(0, 0, window_width, window_height, "Cast_receiver"),
#endif // OS_LINUX
num_video_frames_processed_(0),
num_audio_frames_processed_(0),
currently_playing_audio_frame_start_(-1) {}
virtual ~NaivePlayer() {}
virtual 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();
}
virtual 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.
virtual 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!";
}
}
virtual 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.
virtual int OnMoreData(AudioBus* dest, AudioBuffersState buffers_state)
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 |buffers_state| 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();
}
virtual 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);
#ifdef OS_LINUX
render_.RenderFrame(video_frame);
#endif // OS_LINUX
}
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.
#ifdef OS_LINUX
test::LinuxOutputWindow render_;
#endif // OS_LINUX
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(OS_LINUX)
media::cast::GetWindowSize(&window_width, &window_height);
#endif // OS_LINUX
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;
}