/* * 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 #include #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(event_factory->CreateEvent()), rtc::scoped_ptr(event_factory->CreateEvent())) { } VCMReceiver::VCMReceiver(VCMTiming* timing, Clock* clock, rtc::scoped_ptr receiver_event, rtc::scoped_ptr 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(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(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(clock_->TimeInMilliseconds() - start_time_ms); uint16_t new_max_wait_time = static_cast( 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 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(×tamp_start, ×tamp_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