webrtc/modules/video_coding/main/source/receiver.cc - src - Git at Google

 /*
  *  Copyright (c) 2012 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 "webrtc/modules/video_coding/main/source/receiver.h"

 #include <assert.h>

 #include <cstdlib>

 #include "webrtc/base/trace_event.h"
 #include "webrtc/modules/video_coding/main/source/encoded_frame.h"
 #include "webrtc/modules/video_coding/main/source/internal_defines.h"
 #include "webrtc/modules/video_coding/main/source/media_opt_util.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/logging.h"

 namespace webrtc {

 enum { kMaxReceiverDelayMs = 10000 };

 VCMReceiver::VCMReceiver(VCMTiming* timing,
                          Clock* clock,
                          EventFactory* event_factory)
     : VCMReceiver(timing,
                   clock,
                   rtc::scoped_ptr<EventWrapper>(event_factory->CreateEvent()),
                   rtc::scoped_ptr<EventWrapper>(event_factory->CreateEvent())) {
 }

 VCMReceiver::VCMReceiver(VCMTiming* timing,
                          Clock* clock,
                          rtc::scoped_ptr<EventWrapper> receiver_event,
                          rtc::scoped_ptr<EventWrapper> jitter_buffer_event)
     : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
       clock_(clock),
       jitter_buffer_(clock_, jitter_buffer_event.Pass()),
       timing_(timing),
       render_wait_event_(receiver_event.Pass()),
       max_video_delay_ms_(kMaxVideoDelayMs) {
   Reset();
 }

 VCMReceiver::~VCMReceiver() {
   render_wait_event_->Set();
   delete crit_sect_;
 }

 void VCMReceiver::Reset() {
   CriticalSectionScoped cs(crit_sect_);
   if (!jitter_buffer_.Running()) {
     jitter_buffer_.Start();
   } else {
     jitter_buffer_.Flush();
   }
 }

 void VCMReceiver::UpdateRtt(int64_t rtt) {
   jitter_buffer_.UpdateRtt(rtt);
 }

 int32_t VCMReceiver::InsertPacket(const VCMPacket& packet,
                                   uint16_t frame_width,
                                   uint16_t frame_height) {
   // Insert the packet into the jitter buffer. The packet can either be empty or
   // contain media at this point.
   bool retransmitted = false;
   const VCMFrameBufferEnum ret = jitter_buffer_.InsertPacket(packet,
                                                              &retransmitted);
   if (ret == kOldPacket) {
     return VCM_OK;
   } else if (ret == kFlushIndicator) {
     return VCM_FLUSH_INDICATOR;
   } else if (ret < 0) {
     return VCM_JITTER_BUFFER_ERROR;
   }
   if (ret == kCompleteSession && !retransmitted) {
     // We don't want to include timestamps which have suffered from
     // retransmission here, since we compensate with extra retransmission
     // delay within the jitter estimate.
     timing_->IncomingTimestamp(packet.timestamp, clock_->TimeInMilliseconds());
   }
   return VCM_OK;
 }

 void VCMReceiver::TriggerDecoderShutdown() {
   jitter_buffer_.Stop();
   render_wait_event_->Set();
 }

 VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
                                                int64_t& next_render_time_ms,
                                                bool render_timing) {
   const int64_t start_time_ms = clock_->TimeInMilliseconds();
   uint32_t frame_timestamp = 0;
   // Exhaust wait time to get a complete frame for decoding.
   bool found_frame = jitter_buffer_.NextCompleteTimestamp(
       max_wait_time_ms, &frame_timestamp);

   if (!found_frame)
     found_frame = jitter_buffer_.NextMaybeIncompleteTimestamp(&frame_timestamp);

   if (!found_frame)
     return NULL;

   // We have a frame - Set timing and render timestamp.
   timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
   const int64_t now_ms = clock_->TimeInMilliseconds();
   timing_->UpdateCurrentDelay(frame_timestamp);
   next_render_time_ms = timing_->RenderTimeMs(frame_timestamp, now_ms);
   // Check render timing.
   bool timing_error = false;
   // Assume that render timing errors are due to changes in the video stream.
   if (next_render_time_ms < 0) {
     timing_error = true;
   } else if (std::abs(next_render_time_ms - now_ms) > max_video_delay_ms_) {
     int frame_delay = static_cast<int>(std::abs(next_render_time_ms - now_ms));
     LOG(LS_WARNING) << "A frame about to be decoded is out of the configured "
                     << "delay bounds (" << frame_delay << " > "
                     << max_video_delay_ms_
                     << "). Resetting the video jitter buffer.";
     timing_error = true;
   } else if (static_cast<int>(timing_->TargetVideoDelay()) >
              max_video_delay_ms_) {
     LOG(LS_WARNING) << "The video target delay has grown larger than "
                     << max_video_delay_ms_ << " ms. Resetting jitter buffer.";
     timing_error = true;
   }

   if (timing_error) {
     // Timing error => reset timing and flush the jitter buffer.
     jitter_buffer_.Flush();
     timing_->Reset();
     return NULL;
   }

   if (!render_timing) {
     // Decode frame as close as possible to the render timestamp.
     const int32_t available_wait_time = max_wait_time_ms -
         static_cast<int32_t>(clock_->TimeInMilliseconds() - start_time_ms);
     uint16_t new_max_wait_time = static_cast<uint16_t>(
         VCM_MAX(available_wait_time, 0));
     uint32_t wait_time_ms = timing_->MaxWaitingTime(
         next_render_time_ms, clock_->TimeInMilliseconds());
     if (new_max_wait_time < wait_time_ms) {
       // We're not allowed to wait until the frame is supposed to be rendered,
       // waiting as long as we're allowed to avoid busy looping, and then return
       // NULL. Next call to this function might return the frame.
       render_wait_event_->Wait(new_max_wait_time);
       return NULL;
     }
     // Wait until it's time to render.
     render_wait_event_->Wait(wait_time_ms);
   }

   // Extract the frame from the jitter buffer and set the render time.
   VCMEncodedFrame* frame = jitter_buffer_.ExtractAndSetDecode(frame_timestamp);
   if (frame == NULL) {
     return NULL;
   }
   frame->SetRenderTime(next_render_time_ms);
   TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", frame->TimeStamp(),
                           "SetRenderTS", "render_time", next_render_time_ms);
   if (!frame->Complete()) {
     // Update stats for incomplete frames.
     bool retransmitted = false;
     const int64_t last_packet_time_ms =
         jitter_buffer_.LastPacketTime(frame, &retransmitted);
     if (last_packet_time_ms >= 0 && !retransmitted) {
       // We don't want to include timestamps which have suffered from
       // retransmission here, since we compensate with extra retransmission
       // delay within the jitter estimate.
       timing_->IncomingTimestamp(frame_timestamp, last_packet_time_ms);
     }
   }
   return frame;
 }

 void VCMReceiver::ReleaseFrame(VCMEncodedFrame* frame) {
   jitter_buffer_.ReleaseFrame(frame);
 }

 void VCMReceiver::ReceiveStatistics(uint32_t* bitrate,
                                     uint32_t* framerate) {
   assert(bitrate);
   assert(framerate);
   jitter_buffer_.IncomingRateStatistics(framerate, bitrate);
 }

 uint32_t VCMReceiver::DiscardedPackets() const {
   return jitter_buffer_.num_discarded_packets();
 }

 void VCMReceiver::SetNackMode(VCMNackMode nackMode,
                               int64_t low_rtt_nack_threshold_ms,
                               int64_t high_rtt_nack_threshold_ms) {
   CriticalSectionScoped cs(crit_sect_);
   // Default to always having NACK enabled in hybrid mode.
   jitter_buffer_.SetNackMode(nackMode, low_rtt_nack_threshold_ms,
                              high_rtt_nack_threshold_ms);
 }

 void VCMReceiver::SetNackSettings(size_t max_nack_list_size,
                                   int max_packet_age_to_nack,
                                   int max_incomplete_time_ms) {
   jitter_buffer_.SetNackSettings(max_nack_list_size,
                                  max_packet_age_to_nack,
                                  max_incomplete_time_ms);
 }

 VCMNackMode VCMReceiver::NackMode() const {
   CriticalSectionScoped cs(crit_sect_);
   return jitter_buffer_.nack_mode();
 }

 std::vector<uint16_t> VCMReceiver::NackList(bool* request_key_frame) {
   return jitter_buffer_.GetNackList(request_key_frame);
 }

 void VCMReceiver::SetDecodeErrorMode(VCMDecodeErrorMode decode_error_mode) {
   jitter_buffer_.SetDecodeErrorMode(decode_error_mode);
 }

 VCMDecodeErrorMode VCMReceiver::DecodeErrorMode() const {
   return jitter_buffer_.decode_error_mode();
 }

 int VCMReceiver::SetMinReceiverDelay(int desired_delay_ms) {
   CriticalSectionScoped cs(crit_sect_);
   if (desired_delay_ms < 0 || desired_delay_ms > kMaxReceiverDelayMs) {
     return -1;
   }
   max_video_delay_ms_ = desired_delay_ms + kMaxVideoDelayMs;
   // Initializing timing to the desired delay.
   timing_->set_min_playout_delay(desired_delay_ms);
   return 0;
 }

 int VCMReceiver::RenderBufferSizeMs() {
   uint32_t timestamp_start = 0u;
   uint32_t timestamp_end = 0u;
   // Render timestamps are computed just prior to decoding. Therefore this is
   // only an estimate based on frames' timestamps and current timing state.
   jitter_buffer_.RenderBufferSize(&timestamp_start, &timestamp_end);
   if (timestamp_start == timestamp_end) {
     return 0;
   }
   // Update timing.
   const int64_t now_ms = clock_->TimeInMilliseconds();
   timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
   // Get render timestamps.
   uint32_t render_start = timing_->RenderTimeMs(timestamp_start, now_ms);
   uint32_t render_end = timing_->RenderTimeMs(timestamp_end, now_ms);
   return render_end - render_start;
 }

 void VCMReceiver::RegisterStatsCallback(
     VCMReceiveStatisticsCallback* callback) {
   jitter_buffer_.RegisterStatsCallback(callback);
 }

 }  // namespace webrtc
	/*
	* Copyright (c) 2012 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 "webrtc/modules/video_coding/main/source/receiver.h"

	#include <assert.h>

	#include <cstdlib>

	#include "webrtc/base/trace_event.h"
	#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
	#include "webrtc/modules/video_coding/main/source/internal_defines.h"
	#include "webrtc/modules/video_coding/main/source/media_opt_util.h"
	#include "webrtc/system_wrappers/include/clock.h"
	#include "webrtc/system_wrappers/include/logging.h"

	namespace webrtc {

	enum { kMaxReceiverDelayMs = 10000 };

	VCMReceiver::VCMReceiver(VCMTiming* timing,
	Clock* clock,
	EventFactory* event_factory)
	: VCMReceiver(timing,
	clock,
	rtc::scoped_ptr<EventWrapper>(event_factory->CreateEvent()),
	rtc::scoped_ptr<EventWrapper>(event_factory->CreateEvent())) {
	}

	VCMReceiver::VCMReceiver(VCMTiming* timing,
	Clock* clock,
	rtc::scoped_ptr<EventWrapper> receiver_event,
	rtc::scoped_ptr<EventWrapper> jitter_buffer_event)
	: crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
	clock_(clock),
	jitter_buffer_(clock_, jitter_buffer_event.Pass()),
	timing_(timing),
	render_wait_event_(receiver_event.Pass()),
	max_video_delay_ms_(kMaxVideoDelayMs) {
	Reset();
	}

	VCMReceiver::~VCMReceiver() {
	render_wait_event_->Set();
	delete crit_sect_;
	}

	void VCMReceiver::Reset() {
	CriticalSectionScoped cs(crit_sect_);
	if (!jitter_buffer_.Running()) {
	jitter_buffer_.Start();
	} else {
	jitter_buffer_.Flush();
	}
	}

	void VCMReceiver::UpdateRtt(int64_t rtt) {
	jitter_buffer_.UpdateRtt(rtt);
	}

	int32_t VCMReceiver::InsertPacket(const VCMPacket& packet,
	uint16_t frame_width,
	uint16_t frame_height) {
	// Insert the packet into the jitter buffer. The packet can either be empty or
	// contain media at this point.
	bool retransmitted = false;
	const VCMFrameBufferEnum ret = jitter_buffer_.InsertPacket(packet,
	&retransmitted);
	if (ret == kOldPacket) {
	return VCM_OK;
	} else if (ret == kFlushIndicator) {
	return VCM_FLUSH_INDICATOR;
	} else if (ret < 0) {
	return VCM_JITTER_BUFFER_ERROR;
	}
	if (ret == kCompleteSession && !retransmitted) {
	// We don't want to include timestamps which have suffered from
	// retransmission here, since we compensate with extra retransmission
	// delay within the jitter estimate.
	timing_->IncomingTimestamp(packet.timestamp, clock_->TimeInMilliseconds());
	}
	return VCM_OK;
	}

	void VCMReceiver::TriggerDecoderShutdown() {
	jitter_buffer_.Stop();
	render_wait_event_->Set();
	}

	VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
	int64_t& next_render_time_ms,
	bool render_timing) {
	const int64_t start_time_ms = clock_->TimeInMilliseconds();
	uint32_t frame_timestamp = 0;
	// Exhaust wait time to get a complete frame for decoding.
	bool found_frame = jitter_buffer_.NextCompleteTimestamp(
	max_wait_time_ms, &frame_timestamp);

	if (!found_frame)
	found_frame = jitter_buffer_.NextMaybeIncompleteTimestamp(&frame_timestamp);

	if (!found_frame)
	return NULL;

	// We have a frame - Set timing and render timestamp.
	timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
	const int64_t now_ms = clock_->TimeInMilliseconds();
	timing_->UpdateCurrentDelay(frame_timestamp);
	next_render_time_ms = timing_->RenderTimeMs(frame_timestamp, now_ms);
	// Check render timing.
	bool timing_error = false;
	// Assume that render timing errors are due to changes in the video stream.
	if (next_render_time_ms < 0) {
	timing_error = true;
	} else if (std::abs(next_render_time_ms - now_ms) > max_video_delay_ms_) {
	int frame_delay = static_cast<int>(std::abs(next_render_time_ms - now_ms));
	LOG(LS_WARNING) << "A frame about to be decoded is out of the configured "
	<< "delay bounds (" << frame_delay << " > "
	<< max_video_delay_ms_
	<< "). Resetting the video jitter buffer.";
	timing_error = true;
	} else if (static_cast<int>(timing_->TargetVideoDelay()) >
	max_video_delay_ms_) {
	LOG(LS_WARNING) << "The video target delay has grown larger than "
	<< max_video_delay_ms_ << " ms. Resetting jitter buffer.";
	timing_error = true;
	}

	if (timing_error) {
	// Timing error => reset timing and flush the jitter buffer.
	jitter_buffer_.Flush();
	timing_->Reset();
	return NULL;
	}

	if (!render_timing) {
	// Decode frame as close as possible to the render timestamp.
	const int32_t available_wait_time = max_wait_time_ms -
	static_cast<int32_t>(clock_->TimeInMilliseconds() - start_time_ms);
	uint16_t new_max_wait_time = static_cast<uint16_t>(
	VCM_MAX(available_wait_time, 0));
	uint32_t wait_time_ms = timing_->MaxWaitingTime(
	next_render_time_ms, clock_->TimeInMilliseconds());
	if (new_max_wait_time < wait_time_ms) {
	// We're not allowed to wait until the frame is supposed to be rendered,
	// waiting as long as we're allowed to avoid busy looping, and then return
	// NULL. Next call to this function might return the frame.
	render_wait_event_->Wait(new_max_wait_time);
	return NULL;
	}
	// Wait until it's time to render.
	render_wait_event_->Wait(wait_time_ms);
	}

	// Extract the frame from the jitter buffer and set the render time.
	VCMEncodedFrame* frame = jitter_buffer_.ExtractAndSetDecode(frame_timestamp);
	if (frame == NULL) {
	return NULL;
	}
	frame->SetRenderTime(next_render_time_ms);
	TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", frame->TimeStamp(),
	"SetRenderTS", "render_time", next_render_time_ms);
	if (!frame->Complete()) {
	// Update stats for incomplete frames.
	bool retransmitted = false;
	const int64_t last_packet_time_ms =
	jitter_buffer_.LastPacketTime(frame, &retransmitted);
	if (last_packet_time_ms >= 0 && !retransmitted) {
	// We don't want to include timestamps which have suffered from
	// retransmission here, since we compensate with extra retransmission
	// delay within the jitter estimate.
	timing_->IncomingTimestamp(frame_timestamp, last_packet_time_ms);
	}
	}
	return frame;
	}

	void VCMReceiver::ReleaseFrame(VCMEncodedFrame* frame) {
	jitter_buffer_.ReleaseFrame(frame);
	}

	void VCMReceiver::ReceiveStatistics(uint32_t* bitrate,
	uint32_t* framerate) {
	assert(bitrate);
	assert(framerate);
	jitter_buffer_.IncomingRateStatistics(framerate, bitrate);
	}

	uint32_t VCMReceiver::DiscardedPackets() const {
	return jitter_buffer_.num_discarded_packets();
	}

	void VCMReceiver::SetNackMode(VCMNackMode nackMode,
	int64_t low_rtt_nack_threshold_ms,
	int64_t high_rtt_nack_threshold_ms) {
	CriticalSectionScoped cs(crit_sect_);
	// Default to always having NACK enabled in hybrid mode.
	jitter_buffer_.SetNackMode(nackMode, low_rtt_nack_threshold_ms,
	high_rtt_nack_threshold_ms);
	}

	void VCMReceiver::SetNackSettings(size_t max_nack_list_size,
	int max_packet_age_to_nack,
	int max_incomplete_time_ms) {
	jitter_buffer_.SetNackSettings(max_nack_list_size,
	max_packet_age_to_nack,
	max_incomplete_time_ms);
	}

	VCMNackMode VCMReceiver::NackMode() const {
	CriticalSectionScoped cs(crit_sect_);
	return jitter_buffer_.nack_mode();
	}

	std::vector<uint16_t> VCMReceiver::NackList(bool* request_key_frame) {
	return jitter_buffer_.GetNackList(request_key_frame);
	}

	void VCMReceiver::SetDecodeErrorMode(VCMDecodeErrorMode decode_error_mode) {
	jitter_buffer_.SetDecodeErrorMode(decode_error_mode);
	}

	VCMDecodeErrorMode VCMReceiver::DecodeErrorMode() const {
	return jitter_buffer_.decode_error_mode();
	}

	int VCMReceiver::SetMinReceiverDelay(int desired_delay_ms) {
	CriticalSectionScoped cs(crit_sect_);
	if (desired_delay_ms < 0 \|\| desired_delay_ms > kMaxReceiverDelayMs) {
	return -1;
	}
	max_video_delay_ms_ = desired_delay_ms + kMaxVideoDelayMs;
	// Initializing timing to the desired delay.
	timing_->set_min_playout_delay(desired_delay_ms);
	return 0;
	}

	int VCMReceiver::RenderBufferSizeMs() {
	uint32_t timestamp_start = 0u;
	uint32_t timestamp_end = 0u;
	// Render timestamps are computed just prior to decoding. Therefore this is
	// only an estimate based on frames' timestamps and current timing state.
	jitter_buffer_.RenderBufferSize(&timestamp_start, &timestamp_end);
	if (timestamp_start == timestamp_end) {
	return 0;
	}
	// Update timing.
	const int64_t now_ms = clock_->TimeInMilliseconds();
	timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
	// Get render timestamps.
	uint32_t render_start = timing_->RenderTimeMs(timestamp_start, now_ms);
	uint32_t render_end = timing_->RenderTimeMs(timestamp_end, now_ms);
	return render_end - render_start;
	}

	void VCMReceiver::RegisterStatsCallback(
	VCMReceiveStatisticsCallback* callback) {
	jitter_buffer_.RegisterStatsCallback(callback);
	}

	} // namespace webrtc