| /* |
| * 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/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; |
| |
| 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_); |
| }; |
| } // namespace webrtc |
| |
| #endif // MODULES_VIDEO_CODING_TIMING_H_ |