blob: 9efd4cd8336c41c321daf7cbff1dd5236f2a79af [file] [log] [blame]
/*
* 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)
: crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
clock_(clock),
jitter_buffer_(clock_, event_factory),
timing_(timing),
render_wait_event_(event_factory->CreateEvent()),
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(max_wait_time_ms);
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();
}
VCMNackStatus VCMReceiver::NackList(uint16_t* nack_list,
uint16_t size,
uint16_t* nack_list_length) {
bool request_key_frame = false;
uint16_t* internal_nack_list = jitter_buffer_.GetNackList(
nack_list_length, &request_key_frame);
assert(*nack_list_length <= size);
if (*nack_list_length > size) {
*nack_list_length = size;
}
if (internal_nack_list != NULL && *nack_list_length > 0) {
memcpy(nack_list, internal_nack_list, *nack_list_length * sizeof(uint16_t));
}
if (request_key_frame) {
return kNackKeyFrameRequest;
}
return kNackOk;
}
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