| /* | |
| * 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/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/interface/clock.h" | |
| #include "webrtc/system_wrappers/interface/logging.h" | |
| #include "webrtc/system_wrappers/interface/trace_event.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(×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 |