Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
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Mirko Bonadei
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303
modules/video_coding/frame_buffer.cc
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303
modules/video_coding/frame_buffer.cc
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/*
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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/frame_buffer.h"
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#include <assert.h>
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#include <string.h>
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#include "webrtc/modules/video_coding/packet.h"
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#include "webrtc/rtc_base/checks.h"
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#include "webrtc/rtc_base/logging.h"
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#include "webrtc/rtc_base/trace_event.h"
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namespace webrtc {
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VCMFrameBuffer::VCMFrameBuffer()
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: _state(kStateEmpty), _nackCount(0), _latestPacketTimeMs(-1) {}
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VCMFrameBuffer::~VCMFrameBuffer() {}
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VCMFrameBuffer::VCMFrameBuffer(const VCMFrameBuffer& rhs)
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: VCMEncodedFrame(rhs),
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_state(rhs._state),
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_sessionInfo(),
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_nackCount(rhs._nackCount),
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_latestPacketTimeMs(rhs._latestPacketTimeMs) {
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_sessionInfo = rhs._sessionInfo;
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_sessionInfo.UpdateDataPointers(rhs._buffer, _buffer);
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}
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webrtc::FrameType VCMFrameBuffer::FrameType() const {
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return _sessionInfo.FrameType();
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}
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int32_t VCMFrameBuffer::GetLowSeqNum() const {
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return _sessionInfo.LowSequenceNumber();
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}
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int32_t VCMFrameBuffer::GetHighSeqNum() const {
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return _sessionInfo.HighSequenceNumber();
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}
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int VCMFrameBuffer::PictureId() const {
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return _sessionInfo.PictureId();
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}
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int VCMFrameBuffer::TemporalId() const {
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return _sessionInfo.TemporalId();
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}
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bool VCMFrameBuffer::LayerSync() const {
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return _sessionInfo.LayerSync();
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}
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int VCMFrameBuffer::Tl0PicId() const {
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return _sessionInfo.Tl0PicId();
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}
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bool VCMFrameBuffer::NonReference() const {
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return _sessionInfo.NonReference();
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}
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std::vector<NaluInfo> VCMFrameBuffer::GetNaluInfos() const {
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return _sessionInfo.GetNaluInfos();
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}
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void VCMFrameBuffer::SetGofInfo(const GofInfoVP9& gof_info, size_t idx) {
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TRACE_EVENT0("webrtc", "VCMFrameBuffer::SetGofInfo");
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_sessionInfo.SetGofInfo(gof_info, idx);
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// TODO(asapersson): Consider adding hdr->VP9.ref_picture_id for testing.
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_codecSpecificInfo.codecSpecific.VP9.temporal_idx =
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gof_info.temporal_idx[idx];
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_codecSpecificInfo.codecSpecific.VP9.temporal_up_switch =
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gof_info.temporal_up_switch[idx];
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}
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bool VCMFrameBuffer::IsSessionComplete() const {
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TRACE_EVENT0("webrtc", "VCMFrameBuffer::IsSessionComplete");
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return _sessionInfo.complete();
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}
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// Insert packet
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VCMFrameBufferEnum VCMFrameBuffer::InsertPacket(
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const VCMPacket& packet,
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int64_t timeInMs,
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VCMDecodeErrorMode decode_error_mode,
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const FrameData& frame_data) {
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TRACE_EVENT0("webrtc", "VCMFrameBuffer::InsertPacket");
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assert(!(NULL == packet.dataPtr && packet.sizeBytes > 0));
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if (packet.dataPtr != NULL) {
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_payloadType = packet.payloadType;
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}
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if (kStateEmpty == _state) {
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// First packet (empty and/or media) inserted into this frame.
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// store some info and set some initial values.
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_timeStamp = packet.timestamp;
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// We only take the ntp timestamp of the first packet of a frame.
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ntp_time_ms_ = packet.ntp_time_ms_;
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_codec = packet.codec;
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if (packet.frameType != kEmptyFrame) {
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// first media packet
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SetState(kStateIncomplete);
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}
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}
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uint32_t requiredSizeBytes =
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Length() + packet.sizeBytes +
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(packet.insertStartCode ? kH264StartCodeLengthBytes : 0) +
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EncodedImage::GetBufferPaddingBytes(packet.codec);
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if (requiredSizeBytes >= _size) {
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const uint8_t* prevBuffer = _buffer;
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const uint32_t increments =
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requiredSizeBytes / kBufferIncStepSizeBytes +
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(requiredSizeBytes % kBufferIncStepSizeBytes > 0);
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const uint32_t newSize = _size + increments * kBufferIncStepSizeBytes;
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if (newSize > kMaxJBFrameSizeBytes) {
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LOG(LS_ERROR) << "Failed to insert packet due to frame being too "
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"big.";
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return kSizeError;
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}
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VerifyAndAllocate(newSize);
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_sessionInfo.UpdateDataPointers(prevBuffer, _buffer);
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}
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if (packet.width > 0 && packet.height > 0) {
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_encodedWidth = packet.width;
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_encodedHeight = packet.height;
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}
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// Don't copy payload specific data for empty packets (e.g padding packets).
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if (packet.sizeBytes > 0)
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CopyCodecSpecific(&packet.video_header);
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int retVal =
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_sessionInfo.InsertPacket(packet, _buffer, decode_error_mode, frame_data);
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if (retVal == -1) {
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return kSizeError;
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} else if (retVal == -2) {
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return kDuplicatePacket;
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} else if (retVal == -3) {
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return kOutOfBoundsPacket;
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}
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// update length
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_length = Length() + static_cast<uint32_t>(retVal);
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_latestPacketTimeMs = timeInMs;
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// http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
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// ts_126114v120700p.pdf Section 7.4.5.
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// The MTSI client shall add the payload bytes as defined in this clause
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// onto the last RTP packet in each group of packets which make up a key
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// frame (I-frame or IDR frame in H.264 (AVC), or an IRAP picture in H.265
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// (HEVC)).
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if (packet.markerBit) {
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RTC_DCHECK(!_rotation_set);
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rotation_ = packet.video_header.rotation;
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_rotation_set = true;
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content_type_ = packet.video_header.content_type;
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if (packet.video_header.video_timing.flags != TimingFrameFlags::kInvalid) {
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timing_.encode_start_ms =
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ntp_time_ms_ + packet.video_header.video_timing.encode_start_delta_ms;
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timing_.encode_finish_ms =
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ntp_time_ms_ +
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packet.video_header.video_timing.encode_finish_delta_ms;
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timing_.packetization_finish_ms =
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ntp_time_ms_ +
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packet.video_header.video_timing.packetization_finish_delta_ms;
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timing_.pacer_exit_ms =
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ntp_time_ms_ + packet.video_header.video_timing.pacer_exit_delta_ms;
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timing_.network_timestamp_ms =
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ntp_time_ms_ +
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packet.video_header.video_timing.network_timstamp_delta_ms;
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timing_.network2_timestamp_ms =
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ntp_time_ms_ +
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packet.video_header.video_timing.network2_timstamp_delta_ms;
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}
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timing_.flags = packet.video_header.video_timing.flags;
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}
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if (packet.is_first_packet_in_frame) {
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playout_delay_ = packet.video_header.playout_delay;
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}
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if (_sessionInfo.complete()) {
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SetState(kStateComplete);
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return kCompleteSession;
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} else if (_sessionInfo.decodable()) {
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SetState(kStateDecodable);
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return kDecodableSession;
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}
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return kIncomplete;
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}
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int64_t VCMFrameBuffer::LatestPacketTimeMs() const {
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TRACE_EVENT0("webrtc", "VCMFrameBuffer::LatestPacketTimeMs");
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return _latestPacketTimeMs;
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}
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void VCMFrameBuffer::IncrementNackCount() {
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TRACE_EVENT0("webrtc", "VCMFrameBuffer::IncrementNackCount");
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_nackCount++;
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}
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int16_t VCMFrameBuffer::GetNackCount() const {
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TRACE_EVENT0("webrtc", "VCMFrameBuffer::GetNackCount");
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return _nackCount;
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}
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bool VCMFrameBuffer::HaveFirstPacket() const {
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TRACE_EVENT0("webrtc", "VCMFrameBuffer::HaveFirstPacket");
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return _sessionInfo.HaveFirstPacket();
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}
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bool VCMFrameBuffer::HaveLastPacket() const {
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TRACE_EVENT0("webrtc", "VCMFrameBuffer::HaveLastPacket");
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return _sessionInfo.HaveLastPacket();
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}
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int VCMFrameBuffer::NumPackets() const {
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TRACE_EVENT0("webrtc", "VCMFrameBuffer::NumPackets");
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return _sessionInfo.NumPackets();
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}
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void VCMFrameBuffer::Reset() {
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TRACE_EVENT0("webrtc", "VCMFrameBuffer::Reset");
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_length = 0;
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_timeStamp = 0;
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_sessionInfo.Reset();
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_payloadType = 0;
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_nackCount = 0;
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_latestPacketTimeMs = -1;
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_state = kStateEmpty;
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VCMEncodedFrame::Reset();
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}
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// Set state of frame
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void VCMFrameBuffer::SetState(VCMFrameBufferStateEnum state) {
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TRACE_EVENT0("webrtc", "VCMFrameBuffer::SetState");
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if (_state == state) {
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return;
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}
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switch (state) {
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case kStateIncomplete:
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// we can go to this state from state kStateEmpty
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assert(_state == kStateEmpty);
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// Do nothing, we received a packet
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break;
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case kStateComplete:
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assert(_state == kStateEmpty || _state == kStateIncomplete ||
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_state == kStateDecodable);
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break;
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case kStateEmpty:
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// Should only be set to empty through Reset().
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assert(false);
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break;
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case kStateDecodable:
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assert(_state == kStateEmpty || _state == kStateIncomplete);
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break;
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}
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_state = state;
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}
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// Get current state of frame
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VCMFrameBufferStateEnum VCMFrameBuffer::GetState() const {
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return _state;
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}
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// Get current state of frame
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VCMFrameBufferStateEnum VCMFrameBuffer::GetState(uint32_t& timeStamp) const {
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TRACE_EVENT0("webrtc", "VCMFrameBuffer::GetState");
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timeStamp = TimeStamp();
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return GetState();
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}
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bool VCMFrameBuffer::IsRetransmitted() const {
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return _sessionInfo.session_nack();
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}
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void VCMFrameBuffer::PrepareForDecode(bool continuous) {
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TRACE_EVENT0("webrtc", "VCMFrameBuffer::PrepareForDecode");
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size_t bytes_removed = _sessionInfo.MakeDecodable();
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_length -= bytes_removed;
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// Transfer frame information to EncodedFrame and create any codec
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// specific information.
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_frameType = _sessionInfo.FrameType();
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_completeFrame = _sessionInfo.complete();
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_missingFrame = !continuous;
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}
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} // namespace webrtc
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