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/*
* Copyright (c) 2011 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.
*/
#ifndef MODULES_VIDEO_CODING_TIMING_H_
#define MODULES_VIDEO_CODING_TIMING_H_
#include <memory>
#include "absl/types/optional.h"
#include "api/units/time_delta.h"
#include "api/video/video_timing.h"
#include "modules/video_coding/codec_timer.h"
#include "rtc_base/experiments/field_trial_parser.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/time/timestamp_extrapolator.h"
namespace webrtc {
class Clock;
class TimestampExtrapolator;
class VCMTiming {
public:
explicit VCMTiming(Clock* clock);
virtual ~VCMTiming() = default;
// Resets the timing to the initial state.
void Reset();
// Set the amount of time needed to render an image. Defaults to 10 ms.
void set_render_delay(int render_delay_ms);
// Set the minimum time the video must be delayed on the receiver to
// get the desired jitter buffer level.
void SetJitterDelay(int required_delay_ms);
// Set/get the minimum playout delay from capture to render in ms.
void set_min_playout_delay(int min_playout_delay_ms);
int min_playout_delay();
// Set/get the maximum playout delay from capture to render in ms.
void set_max_playout_delay(int max_playout_delay_ms);
int max_playout_delay();
// Increases or decreases the current delay to get closer to the target delay.
// Calculates how long it has been since the previous call to this function,
// and increases/decreases the delay in proportion to the time difference.
void UpdateCurrentDelay(uint32_t frame_timestamp);
// Increases or decreases the current delay to get closer to the target delay.
// Given the actual decode time in ms and the render time in ms for a frame,
// this function calculates how late the frame is and increases the delay
// accordingly.
void UpdateCurrentDelay(int64_t render_time_ms,
int64_t actual_decode_time_ms);
// Stops the decoder timer, should be called when the decoder returns a frame
// or when the decoded frame callback is called.
void StopDecodeTimer(int32_t decode_time_ms, int64_t now_ms);
// TODO(kron): Remove once downstream projects has been changed to use the
// above function.
void StopDecodeTimer(uint32_t time_stamp,
int32_t decode_time_ms,
int64_t now_ms,
int64_t render_time_ms);
// Used to report that a frame is passed to decoding. Updates the timestamp
// filter which is used to map between timestamps and receiver system time.
void IncomingTimestamp(uint32_t time_stamp, int64_t last_packet_time_ms);
// Returns the receiver system time when the frame with timestamp
// `frame_timestamp` should be rendered, assuming that the system time
// currently is `now_ms`.
virtual int64_t RenderTimeMs(uint32_t frame_timestamp, int64_t now_ms) const;
// Returns the maximum time in ms that we can wait for a frame to become
// complete before we must pass it to the decoder.
virtual int64_t MaxWaitingTime(int64_t render_time_ms, int64_t now_ms) const;
// Returns the current target delay which is required delay + decode time +
// render delay.
int TargetVideoDelay() const;
// Return current timing information. Returns true if the first frame has been
// decoded, false otherwise.
virtual bool GetTimings(int* max_decode_ms,
int* current_delay_ms,
int* target_delay_ms,
int* jitter_buffer_ms,
int* min_playout_delay_ms,
int* render_delay_ms) const;
void SetTimingFrameInfo(const TimingFrameInfo& info);
absl::optional<TimingFrameInfo> GetTimingFrameInfo();
void SetMaxCompositionDelayInFrames(
absl::optional<int> max_composition_delay_in_frames);
absl::optional<int> MaxCompositionDelayInFrames() const;
// Updates the last time a frame was scheduled for decoding.
void SetLastDecodeScheduledTimestamp(int64_t last_decode_scheduled_ts);
enum { kDefaultRenderDelayMs = 10 };
enum { kDelayMaxChangeMsPerS = 100 };
protected:
int RequiredDecodeTimeMs() const RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
int64_t RenderTimeMsInternal(uint32_t frame_timestamp, int64_t now_ms) const
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
int TargetDelayInternal() const RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
private:
mutable Mutex mutex_;
Clock* const clock_;
const std::unique_ptr<TimestampExtrapolator> ts_extrapolator_
RTC_PT_GUARDED_BY(mutex_);
std::unique_ptr<VCMCodecTimer> codec_timer_ RTC_GUARDED_BY(mutex_)
RTC_PT_GUARDED_BY(mutex_);
int render_delay_ms_ RTC_GUARDED_BY(mutex_);
// Best-effort playout delay range for frames from capture to render.
// The receiver tries to keep the delay between `min_playout_delay_ms_`
// and `max_playout_delay_ms_` taking the network jitter into account.
// A special case is where min_playout_delay_ms_ = max_playout_delay_ms_ = 0,
// in which case the receiver tries to play the frames as they arrive.
int min_playout_delay_ms_ RTC_GUARDED_BY(mutex_);
int max_playout_delay_ms_ RTC_GUARDED_BY(mutex_);
int jitter_delay_ms_ RTC_GUARDED_BY(mutex_);
int current_delay_ms_ RTC_GUARDED_BY(mutex_);
uint32_t prev_frame_timestamp_ RTC_GUARDED_BY(mutex_);
absl::optional<TimingFrameInfo> timing_frame_info_ RTC_GUARDED_BY(mutex_);
size_t num_decoded_frames_ RTC_GUARDED_BY(mutex_);
// Set by the field trial WebRTC-LowLatencyRenderer. The parameter enabled
// determines if the low-latency renderer algorithm should be used for the
// case min playout delay=0 and max playout delay>0.
FieldTrialParameter<bool> low_latency_renderer_enabled_
RTC_GUARDED_BY(mutex_);
absl::optional<int> max_composition_delay_in_frames_ RTC_GUARDED_BY(mutex_);
// Set by the field trial WebRTC-ZeroPlayoutDelay. The parameter min_pacing
// determines the minimum delay between frames scheduled for decoding that is
// used when min playout delay=0 and max playout delay>=0.
FieldTrialParameter<TimeDelta> zero_playout_delay_min_pacing_
RTC_GUARDED_BY(mutex_);
// Timestamp at which the last frame was scheduled to be sent to the decoder.
// Used only when the RTP header extension playout delay is set to min=0 ms
// which is indicated by a render time set to 0.
int64_t last_decode_scheduled_ts_ RTC_GUARDED_BY(mutex_);
};
} // namespace webrtc
#endif // MODULES_VIDEO_CODING_TIMING_H_