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/*
* Copyright (c) 2014 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 API_VIDEO_ENCODED_IMAGE_H_
#define API_VIDEO_ENCODED_IMAGE_H_
#include <stdint.h>
#include <map>
#include <utility>
#include "absl/types/optional.h"
#include "api/rtp_packet_infos.h"
#include "api/scoped_refptr.h"
#include "api/video/color_space.h"
#include "api/video/video_codec_constants.h"
#include "api/video/video_codec_type.h"
#include "api/video/video_content_type.h"
#include "api/video/video_frame_type.h"
#include "api/video/video_rotation.h"
#include "api/video/video_timing.h"
#include "common_types.h" // NOLINT(build/include)
#include "rtc_base/checks.h"
#include "rtc_base/deprecation.h"
#include "rtc_base/ref_count.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
// Abstract interface for buffer storage. Intended to support buffers owned by
// external encoders with special release requirements, e.g, java encoders with
// releaseOutputBuffer.
class EncodedImageBufferInterface : public rtc::RefCountInterface {
public:
virtual const uint8_t* data() const = 0;
// TODO(bugs.webrtc.org/9378): Make interface essentially read-only, delete
// this non-const data method.
virtual uint8_t* data() = 0;
virtual size_t size() const = 0;
};
// Basic implementation of EncodedImageBufferInterface.
class RTC_EXPORT EncodedImageBuffer : public EncodedImageBufferInterface {
public:
static rtc::scoped_refptr<EncodedImageBuffer> Create() { return Create(0); }
static rtc::scoped_refptr<EncodedImageBuffer> Create(size_t size);
static rtc::scoped_refptr<EncodedImageBuffer> Create(const uint8_t* data,
size_t size);
const uint8_t* data() const override;
uint8_t* data() override;
size_t size() const override;
void Realloc(size_t t);
protected:
explicit EncodedImageBuffer(size_t size);
EncodedImageBuffer(const uint8_t* data, size_t size);
~EncodedImageBuffer();
size_t size_;
uint8_t* buffer_;
};
// TODO(bug.webrtc.org/9378): This is a legacy api class, which is slowly being
// cleaned up. Direct use of its members is strongly discouraged.
class RTC_EXPORT EncodedImage {
public:
EncodedImage();
EncodedImage(EncodedImage&&);
// Discouraged: potentially expensive.
EncodedImage(const EncodedImage&);
EncodedImage(uint8_t* buffer, size_t length, size_t capacity);
~EncodedImage();
EncodedImage& operator=(EncodedImage&&);
// Discouraged: potentially expensive.
EncodedImage& operator=(const EncodedImage&);
// TODO(nisse): Change style to timestamp(), set_timestamp(), for consistency
// with the VideoFrame class.
// Set frame timestamp (90kHz).
void SetTimestamp(uint32_t timestamp) { timestamp_rtp_ = timestamp; }
// Get frame timestamp (90kHz).
uint32_t Timestamp() const { return timestamp_rtp_; }
void SetEncodeTime(int64_t encode_start_ms, int64_t encode_finish_ms);
absl::optional<int> SpatialIndex() const { return spatial_index_; }
void SetSpatialIndex(absl::optional<int> spatial_index) {
RTC_DCHECK_GE(spatial_index.value_or(0), 0);
RTC_DCHECK_LT(spatial_index.value_or(0), kMaxSpatialLayers);
spatial_index_ = spatial_index;
}
// These methods can be used to set/get size of subframe with spatial index
// |spatial_index| on encoded frames that consist of multiple spatial layers.
absl::optional<size_t> SpatialLayerFrameSize(int spatial_index) const;
void SetSpatialLayerFrameSize(int spatial_index, size_t size_bytes);
const webrtc::ColorSpace* ColorSpace() const {
return color_space_ ? &*color_space_ : nullptr;
}
void SetColorSpace(const absl::optional<webrtc::ColorSpace>& color_space) {
color_space_ = color_space;
}
const RtpPacketInfos& PacketInfos() const { return packet_infos_; }
void SetPacketInfos(RtpPacketInfos packet_infos) {
packet_infos_ = std::move(packet_infos);
}
bool RetransmissionAllowed() const { return retransmission_allowed_; }
void SetRetransmissionAllowed(bool retransmission_allowed) {
retransmission_allowed_ = retransmission_allowed;
}
size_t size() const { return size_; }
void set_size(size_t new_size) {
// Allow set_size(0) even if we have no buffer.
RTC_DCHECK_LE(new_size, new_size == 0 ? 0 : capacity());
size_ = new_size;
}
// TODO(nisse): Delete, provide only read-only access to the buffer.
size_t capacity() const {
return buffer_ ? capacity_ : (encoded_data_ ? encoded_data_->size() : 0);
}
void set_buffer(uint8_t* buffer, size_t capacity) {
buffer_ = buffer;
capacity_ = capacity;
}
void SetEncodedData(
rtc::scoped_refptr<EncodedImageBufferInterface> encoded_data) {
encoded_data_ = encoded_data;
size_ = encoded_data->size();
buffer_ = nullptr;
}
void ClearEncodedData() {
encoded_data_ = nullptr;
size_ = 0;
buffer_ = nullptr;
capacity_ = 0;
}
rtc::scoped_refptr<EncodedImageBufferInterface> GetEncodedData() const {
RTC_DCHECK(buffer_ == nullptr);
return encoded_data_;
}
// TODO(nisse): Delete, provide only read-only access to the buffer.
uint8_t* data() {
return buffer_ ? buffer_
: (encoded_data_ ? encoded_data_->data() : nullptr);
}
const uint8_t* data() const {
return buffer_ ? buffer_
: (encoded_data_ ? encoded_data_->data() : nullptr);
}
// TODO(nisse): At some places, code accepts a const ref EncodedImage, but
// still writes to it, to clear padding at the end of the encoded data.
// Padding is required by ffmpeg; the best way to deal with that is likely to
// make this class ensure that buffers always have a few zero padding bytes.
uint8_t* mutable_data() const { return const_cast<uint8_t*>(data()); }
// TODO(bugs.webrtc.org/9378): Delete. Used by code that wants to modify a
// buffer corresponding to a const EncodedImage. Requires an un-owned buffer.
uint8_t* buffer() const { return buffer_; }
// Hack to workaround lack of ownership of the encoded data. If we don't
// already own the underlying data, make an owned copy.
void Retain();
uint32_t _encodedWidth = 0;
uint32_t _encodedHeight = 0;
// NTP time of the capture time in local timebase in milliseconds.
int64_t ntp_time_ms_ = 0;
int64_t capture_time_ms_ = 0;
VideoFrameType _frameType = VideoFrameType::kVideoFrameDelta;
VideoRotation rotation_ = kVideoRotation_0;
VideoContentType content_type_ = VideoContentType::UNSPECIFIED;
bool _completeFrame = false;
int qp_ = -1; // Quantizer value.
// When an application indicates non-zero values here, it is taken as an
// indication that all future frames will be constrained with those limits
// until the application indicates a change again.
PlayoutDelay playout_delay_ = {-1, -1};
struct Timing {
uint8_t flags = VideoSendTiming::kInvalid;
int64_t encode_start_ms = 0;
int64_t encode_finish_ms = 0;
int64_t packetization_finish_ms = 0;
int64_t pacer_exit_ms = 0;
int64_t network_timestamp_ms = 0;
int64_t network2_timestamp_ms = 0;
int64_t receive_start_ms = 0;
int64_t receive_finish_ms = 0;
} timing_;
private:
// TODO(bugs.webrtc.org/9378): We're transitioning to always owning the
// encoded data.
rtc::scoped_refptr<EncodedImageBufferInterface> encoded_data_;
size_t size_; // Size of encoded frame data.
// Non-null when used with an un-owned buffer.
uint8_t* buffer_;
// Allocated size of _buffer; relevant only if it's non-null.
size_t capacity_;
uint32_t timestamp_rtp_ = 0;
absl::optional<int> spatial_index_;
std::map<int, size_t> spatial_layer_frame_size_bytes_;
absl::optional<webrtc::ColorSpace> color_space_;
// Information about packets used to assemble this video frame. This is needed
// by |SourceTracker| when the frame is delivered to the RTCRtpReceiver's
// MediaStreamTrack, in order to implement getContributingSources(). See:
// https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-getcontributingsources
RtpPacketInfos packet_infos_;
bool retransmission_allowed_ = true;
};
} // namespace webrtc
#endif // API_VIDEO_ENCODED_IMAGE_H_