
Also correct the logging of incoming key frame packets. BUG=1814 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1537004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4090 4adac7df-926f-26a2-2b94-8c16560cd09d
382 lines
13 KiB
C++
382 lines
13 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/video_coding/main/source/receiver.h"
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#include <assert.h>
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#include "webrtc/modules/video_coding/main/interface/video_coding.h"
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#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
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#include "webrtc/modules/video_coding/main/source/internal_defines.h"
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#include "webrtc/modules/video_coding/main/source/media_opt_util.h"
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#include "webrtc/system_wrappers/interface/clock.h"
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#include "webrtc/system_wrappers/interface/trace.h"
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#include "webrtc/system_wrappers/interface/trace_event.h"
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namespace webrtc {
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enum { kMaxReceiverDelayMs = 10000 };
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VCMReceiver::VCMReceiver(VCMTiming* timing,
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Clock* clock,
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EventFactory* event_factory,
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int32_t vcm_id,
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int32_t receiver_id,
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bool master)
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: crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
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vcm_id_(vcm_id),
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clock_(clock),
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receiver_id_(receiver_id),
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master_(master),
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jitter_buffer_(clock_, event_factory, vcm_id, receiver_id, master),
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timing_(timing),
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render_wait_event_(event_factory->CreateEvent()),
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state_(kPassive),
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max_video_delay_ms_(kMaxVideoDelayMs) {}
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VCMReceiver::~VCMReceiver() {
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render_wait_event_->Set();
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delete crit_sect_;
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}
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void VCMReceiver::Reset() {
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CriticalSectionScoped cs(crit_sect_);
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if (!jitter_buffer_.Running()) {
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jitter_buffer_.Start();
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} else {
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jitter_buffer_.Flush();
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}
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render_wait_event_->Reset();
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if (master_) {
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state_ = kReceiving;
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} else {
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state_ = kPassive;
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}
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}
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int32_t VCMReceiver::Initialize() {
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CriticalSectionScoped cs(crit_sect_);
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Reset();
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if (!master_) {
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SetNackMode(kNoNack, -1, -1);
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}
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return VCM_OK;
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}
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void VCMReceiver::UpdateRtt(uint32_t rtt) {
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jitter_buffer_.UpdateRtt(rtt);
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}
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int32_t VCMReceiver::InsertPacket(const VCMPacket& packet,
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uint16_t frame_width,
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uint16_t frame_height) {
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if (packet.frameType == kVideoFrameKey) {
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WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCoding,
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VCMId(vcm_id_, receiver_id_),
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"Inserting key frame packet seqnum=%u, timestamp=%u",
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packet.seqNum, packet.timestamp);
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}
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// Insert the packet into the jitter buffer. The packet can either be empty or
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// contain media at this point.
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bool retransmitted = false;
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const VCMFrameBufferEnum ret = jitter_buffer_.InsertPacket(packet,
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&retransmitted);
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if (ret == kOldPacket) {
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return VCM_OK;
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} else if (ret == kFlushIndicator) {
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return VCM_FLUSH_INDICATOR;
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} else if (ret < 0) {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding,
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VCMId(vcm_id_, receiver_id_),
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"Error inserting packet seqnum=%u, timestamp=%u",
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packet.seqNum, packet.timestamp);
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return VCM_JITTER_BUFFER_ERROR;
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}
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if (ret == kCompleteSession && !retransmitted) {
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// We don't want to include timestamps which have suffered from
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// retransmission here, since we compensate with extra retransmission
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// delay within the jitter estimate.
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timing_->IncomingTimestamp(packet.timestamp, clock_->TimeInMilliseconds());
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}
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if (master_) {
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// Only trace the primary receiver to make it possible to parse and plot
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// the trace file.
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WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
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VCMId(vcm_id_, receiver_id_),
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"Packet seqnum=%u timestamp=%u inserted at %u",
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packet.seqNum, packet.timestamp,
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MaskWord64ToUWord32(clock_->TimeInMilliseconds()));
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}
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return VCM_OK;
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}
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VCMEncodedFrame* VCMReceiver::FrameForDecoding(
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uint16_t max_wait_time_ms,
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int64_t& next_render_time_ms,
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bool render_timing,
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VCMReceiver* dual_receiver) {
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TRACE_EVENT0("webrtc", "Recv::FrameForDecoding");
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const int64_t start_time_ms = clock_->TimeInMilliseconds();
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uint32_t frame_timestamp = 0;
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// Exhaust wait time to get a complete frame for decoding.
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bool found_frame = jitter_buffer_.NextCompleteTimestamp(
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max_wait_time_ms, &frame_timestamp);
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if (!found_frame) {
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// Get an incomplete frame when enabled.
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const bool dual_receiver_enabled_and_passive = (dual_receiver != NULL &&
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dual_receiver->State() == kPassive &&
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dual_receiver->NackMode() == kNack);
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if (dual_receiver_enabled_and_passive &&
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!jitter_buffer_.CompleteSequenceWithNextFrame()) {
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// Jitter buffer state might get corrupt with this frame.
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dual_receiver->CopyJitterBufferStateFromReceiver(*this);
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}
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found_frame = jitter_buffer_.NextMaybeIncompleteTimestamp(
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&frame_timestamp);
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}
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if (!found_frame) {
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return NULL;
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}
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// We have a frame - Set timing and render timestamp.
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timing_->SetRequiredDelay(jitter_buffer_.EstimatedJitterMs());
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const int64_t now_ms = clock_->TimeInMilliseconds();
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timing_->UpdateCurrentDelay(frame_timestamp);
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next_render_time_ms = timing_->RenderTimeMs(frame_timestamp, now_ms);
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// Check render timing.
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bool timing_error = false;
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// Assume that render timing errors are due to changes in the video stream.
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if (next_render_time_ms < 0) {
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timing_error = true;
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} else if (next_render_time_ms < now_ms - max_video_delay_ms_) {
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WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding,
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VCMId(vcm_id_, receiver_id_),
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"This frame should have been rendered more than %u ms ago."
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"Flushing jitter buffer and resetting timing.",
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max_video_delay_ms_);
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timing_error = true;
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} else if (static_cast<int>(timing_->TargetVideoDelay()) >
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max_video_delay_ms_) {
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WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding,
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VCMId(vcm_id_, receiver_id_),
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"More than %u ms target delay. Flushing jitter buffer and"
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"resetting timing.", max_video_delay_ms_);
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timing_error = true;
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}
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if (timing_error) {
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// Timing error => reset timing and flush the jitter buffer.
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jitter_buffer_.Flush();
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timing_->Reset();
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return NULL;
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}
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if (!render_timing) {
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// Decode frame as close as possible to the render timestamp.
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TRACE_EVENT0("webrtc", "FrameForRendering");
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const int32_t available_wait_time = max_wait_time_ms -
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static_cast<int32_t>(clock_->TimeInMilliseconds() - start_time_ms);
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uint16_t new_max_wait_time = static_cast<uint16_t>(
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VCM_MAX(available_wait_time, 0));
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uint32_t wait_time_ms = timing_->MaxWaitingTime(
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next_render_time_ms, clock_->TimeInMilliseconds());
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if (new_max_wait_time < wait_time_ms) {
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// We're not allowed to wait until the frame is supposed to be rendered,
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// waiting as long as we're allowed to avoid busy looping, and then return
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// NULL. Next call to this function might return the frame.
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render_wait_event_->Wait(max_wait_time_ms);
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return NULL;
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}
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// Wait until it's time to render.
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render_wait_event_->Wait(wait_time_ms);
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}
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// Extract the frame from the jitter buffer and set the render time.
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VCMEncodedFrame* frame = jitter_buffer_.ExtractAndSetDecode(frame_timestamp);
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if (frame == NULL) {
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return NULL;
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}
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frame->SetRenderTime(next_render_time_ms);
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if (dual_receiver != NULL) {
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dual_receiver->UpdateState(*frame);
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}
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if (!frame->Complete()) {
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// Update stats for incomplete frames.
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bool retransmitted = false;
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const int64_t last_packet_time_ms =
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jitter_buffer_.LastPacketTime(frame, &retransmitted);
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if (last_packet_time_ms >= 0 && !retransmitted) {
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// We don't want to include timestamps which have suffered from
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// retransmission here, since we compensate with extra retransmission
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// delay within the jitter estimate.
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timing_->IncomingTimestamp(frame_timestamp, last_packet_time_ms);
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}
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}
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return frame;
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}
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void VCMReceiver::ReleaseFrame(VCMEncodedFrame* frame) {
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jitter_buffer_.ReleaseFrame(frame);
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}
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void VCMReceiver::ReceiveStatistics(uint32_t* bitrate,
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uint32_t* framerate) {
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assert(bitrate);
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assert(framerate);
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jitter_buffer_.IncomingRateStatistics(framerate, bitrate);
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}
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void VCMReceiver::ReceivedFrameCount(VCMFrameCount* frame_count) const {
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assert(frame_count);
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jitter_buffer_.FrameStatistics(&frame_count->numDeltaFrames,
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&frame_count->numKeyFrames);
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}
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uint32_t VCMReceiver::DiscardedPackets() const {
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return jitter_buffer_.num_discarded_packets();
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}
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void VCMReceiver::SetNackMode(VCMNackMode nackMode,
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int low_rtt_nack_threshold_ms,
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int high_rtt_nack_threshold_ms) {
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CriticalSectionScoped cs(crit_sect_);
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// Default to always having NACK enabled in hybrid mode.
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jitter_buffer_.SetNackMode(nackMode, low_rtt_nack_threshold_ms,
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high_rtt_nack_threshold_ms);
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if (!master_) {
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state_ = kPassive; // The dual decoder defaults to passive.
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}
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}
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void VCMReceiver::SetNackSettings(size_t max_nack_list_size,
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int max_packet_age_to_nack,
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int max_incomplete_time_ms) {
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jitter_buffer_.SetNackSettings(max_nack_list_size,
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max_packet_age_to_nack,
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max_incomplete_time_ms);
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}
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VCMNackMode VCMReceiver::NackMode() const {
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CriticalSectionScoped cs(crit_sect_);
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return jitter_buffer_.nack_mode();
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}
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VCMNackStatus VCMReceiver::NackList(uint16_t* nack_list,
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uint16_t size,
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uint16_t* nack_list_length) {
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bool request_key_frame = false;
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uint16_t* internal_nack_list = jitter_buffer_.GetNackList(
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nack_list_length, &request_key_frame);
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if (*nack_list_length > size) {
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*nack_list_length = 0;
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return kNackNeedMoreMemory;
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}
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if (internal_nack_list != NULL && *nack_list_length > 0) {
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memcpy(nack_list, internal_nack_list, *nack_list_length * sizeof(uint16_t));
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}
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if (request_key_frame) {
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return kNackKeyFrameRequest;
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}
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return kNackOk;
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}
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// Decide whether we should change decoder state. This should be done if the
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// dual decoder has caught up with the decoder decoding with packet losses.
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bool VCMReceiver::DualDecoderCaughtUp(VCMEncodedFrame* dual_frame,
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VCMReceiver& dual_receiver) const {
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if (dual_frame == NULL) {
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return false;
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}
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if (jitter_buffer_.LastDecodedTimestamp() == dual_frame->TimeStamp()) {
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dual_receiver.UpdateState(kWaitForPrimaryDecode);
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return true;
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}
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return false;
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}
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void VCMReceiver::CopyJitterBufferStateFromReceiver(
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const VCMReceiver& receiver) {
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jitter_buffer_.CopyFrom(receiver.jitter_buffer_);
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}
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VCMReceiverState VCMReceiver::State() const {
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CriticalSectionScoped cs(crit_sect_);
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return state_;
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}
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void VCMReceiver::SetDecodeWithErrors(bool enable){
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CriticalSectionScoped cs(crit_sect_);
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jitter_buffer_.DecodeWithErrors(enable);
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}
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bool VCMReceiver::DecodeWithErrors() const {
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CriticalSectionScoped cs(crit_sect_);
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return jitter_buffer_.decode_with_errors();
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}
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int VCMReceiver::SetMinReceiverDelay(int desired_delay_ms) {
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CriticalSectionScoped cs(crit_sect_);
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if (desired_delay_ms < 0 || desired_delay_ms > kMaxReceiverDelayMs) {
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return -1;
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}
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jitter_buffer_.SetMaxJitterEstimate(desired_delay_ms > 0);
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max_video_delay_ms_ = desired_delay_ms + kMaxVideoDelayMs;
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// Initializing timing to the desired delay.
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timing_->SetMinimumTotalDelay(desired_delay_ms);
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return 0;
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}
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int VCMReceiver::RenderBufferSizeMs() {
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uint32_t timestamp_start = 0u;
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uint32_t timestamp_end = 0u;
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// Render timestamps are computed just prior to decoding. Therefore this is
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// only an estimate based on frames' timestamps and current timing state.
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jitter_buffer_.RenderBufferSize(×tamp_start, ×tamp_end);
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if (timestamp_start == timestamp_end) {
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return 0;
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}
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// Update timing.
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const int64_t now_ms = clock_->TimeInMilliseconds();
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timing_->SetRequiredDelay(jitter_buffer_.EstimatedJitterMs());
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// Get render timestamps.
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uint32_t render_start = timing_->RenderTimeMs(timestamp_start, now_ms);
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uint32_t render_end = timing_->RenderTimeMs(timestamp_end, now_ms);
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return render_end - render_start;
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}
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void VCMReceiver::UpdateState(VCMReceiverState new_state) {
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CriticalSectionScoped cs(crit_sect_);
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assert(!(state_ == kPassive && new_state == kWaitForPrimaryDecode));
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state_ = new_state;
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}
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void VCMReceiver::UpdateState(const VCMEncodedFrame& frame) {
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if (jitter_buffer_.nack_mode() == kNoNack) {
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// Dual decoder mode has not been enabled.
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return;
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}
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// Update the dual receiver state.
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if (frame.Complete() && frame.FrameType() == kVideoFrameKey) {
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UpdateState(kPassive);
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}
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if (State() == kWaitForPrimaryDecode &&
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frame.Complete() && !frame.MissingFrame()) {
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UpdateState(kPassive);
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}
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if (frame.MissingFrame() || !frame.Complete()) {
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// State was corrupted, enable dual receiver.
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UpdateState(kReceiving);
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}
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}
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} // namespace webrtc
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