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Moving src/webrtc into src/.

Browse Source 
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}
This commit is contained in:
Mirko Bonadei
2017-09-15 06:15:48 +02:00
committed by Commit Bot
parent 6674846b4a
commit bb547203bf
4576 changed files with 1092 additions and 1196 deletions
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2
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hbos@webrtc.org
sprang@webrtc.org

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modules/video_coding/codecs/h264/h264.cc Normal file
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/*
* Copyright (c) 2015 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/codecs/h264/include/h264.h"
#if defined(WEBRTC_USE_H264)
#include "webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.h"
#include "webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.h"
#endif
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/logging.h"
namespace webrtc {
namespace {
#if defined(WEBRTC_USE_H264)
bool g_rtc_use_h264 = true;
#endif
} // namespace
void DisableRtcUseH264() {
#if defined(WEBRTC_USE_H264)
g_rtc_use_h264 = false;
#endif
}
// If any H.264 codec is supported (iOS HW or OpenH264/FFmpeg).
bool IsH264CodecSupported() {
#if defined(WEBRTC_USE_H264)
return g_rtc_use_h264;
#else
return false;
#endif
}
H264Encoder* H264Encoder::Create(const cricket::VideoCodec& codec) {
RTC_DCHECK(H264Encoder::IsSupported());
#if defined(WEBRTC_USE_H264)
RTC_CHECK(g_rtc_use_h264);
LOG(LS_INFO) << "Creating H264EncoderImpl.";
return new H264EncoderImpl(codec);
#else
RTC_NOTREACHED();
return nullptr;
#endif
}
bool H264Encoder::IsSupported() {
return IsH264CodecSupported();
}
H264Decoder* H264Decoder::Create() {
RTC_DCHECK(H264Decoder::IsSupported());
#if defined(WEBRTC_USE_H264)
RTC_CHECK(g_rtc_use_h264);
LOG(LS_INFO) << "Creating H264DecoderImpl.";
return new H264DecoderImpl();
#else
RTC_NOTREACHED();
return nullptr;
#endif
}
bool H264Decoder::IsSupported() {
return IsH264CodecSupported();
}
} // namespace webrtc

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/*
* Copyright (c) 2015 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/codecs/h264/h264_decoder_impl.h"
#include <algorithm>
#include <limits>
extern "C" {
#include "third_party/ffmpeg/libavcodec/avcodec.h"
#include "third_party/ffmpeg/libavformat/avformat.h"
#include "third_party/ffmpeg/libavutil/imgutils.h"
} // extern "C"
#include "webrtc/api/video/i420_buffer.h"
#include "webrtc/common_video/include/video_frame_buffer.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/criticalsection.h"
#include "webrtc/rtc_base/keep_ref_until_done.h"
#include "webrtc/rtc_base/logging.h"
#include "webrtc/system_wrappers/include/metrics.h"
namespace webrtc {
namespace {
const AVPixelFormat kPixelFormat = AV_PIX_FMT_YUV420P;
const size_t kYPlaneIndex = 0;
const size_t kUPlaneIndex = 1;
const size_t kVPlaneIndex = 2;
// Used by histograms. Values of entries should not be changed.
enum H264DecoderImplEvent {
kH264DecoderEventInit = 0,
kH264DecoderEventError = 1,
kH264DecoderEventMax = 16,
};
#if defined(WEBRTC_INITIALIZE_FFMPEG)
rtc::CriticalSection ffmpeg_init_lock;
bool ffmpeg_initialized = false;
// Called by FFmpeg to do mutex operations if initialized using
// |InitializeFFmpeg|. Disabling thread safety analysis because void** does not
// play nicely with thread_annotations.h macros.
int LockManagerOperation(void** lock,
AVLockOp op) RTC_NO_THREAD_SAFETY_ANALYSIS {
switch (op) {
case AV_LOCK_CREATE:
*lock = new rtc::CriticalSection();
return 0;
case AV_LOCK_OBTAIN:
static_cast<rtc::CriticalSection*>(*lock)->Enter();
return 0;
case AV_LOCK_RELEASE:
static_cast<rtc::CriticalSection*>(*lock)->Leave();
return 0;
case AV_LOCK_DESTROY:
delete static_cast<rtc::CriticalSection*>(*lock);
*lock = nullptr;
return 0;
}
RTC_NOTREACHED() << "Unrecognized AVLockOp.";
return -1;
}
void InitializeFFmpeg() {
rtc::CritScope cs(&ffmpeg_init_lock);
if (!ffmpeg_initialized) {
if (av_lockmgr_register(LockManagerOperation) < 0) {
RTC_NOTREACHED() << "av_lockmgr_register failed.";
return;
}
av_register_all();
ffmpeg_initialized = true;
}
}
#endif // defined(WEBRTC_INITIALIZE_FFMPEG)
} // namespace
int H264DecoderImpl::AVGetBuffer2(
AVCodecContext* context, AVFrame* av_frame, int flags) {
// Set in |InitDecode|.
H264DecoderImpl* decoder = static_cast<H264DecoderImpl*>(context->opaque);
// DCHECK values set in |InitDecode|.
RTC_DCHECK(decoder);
RTC_DCHECK_EQ(context->pix_fmt, kPixelFormat);
// Necessary capability to be allowed to provide our own buffers.
RTC_DCHECK(context->codec->capabilities | AV_CODEC_CAP_DR1);
// |av_frame->width| and |av_frame->height| are set by FFmpeg. These are the
// actual image's dimensions and may be different from |context->width| and
// |context->coded_width| due to reordering.
int width = av_frame->width;
int height = av_frame->height;
// See |lowres|, if used the decoder scales the image by 1/2^(lowres). This
// has implications on which resolutions are valid, but we don't use it.
RTC_CHECK_EQ(context->lowres, 0);
// Adjust the |width| and |height| to values acceptable by the decoder.
// Without this, FFmpeg may overflow the buffer. If modified, |width| and/or
// |height| are larger than the actual image and the image has to be cropped
// (top-left corner) after decoding to avoid visible borders to the right and
// bottom of the actual image.
avcodec_align_dimensions(context, &width, &height);
RTC_CHECK_GE(width, 0);
RTC_CHECK_GE(height, 0);
int ret = av_image_check_size(static_cast<unsigned int>(width),
static_cast<unsigned int>(height), 0, nullptr);
if (ret < 0) {
LOG(LS_ERROR) << "Invalid picture size " << width << "x" << height;
decoder->ReportError();
return ret;
}
// The video frame is stored in |frame_buffer|. |av_frame| is FFmpeg's version
// of a video frame and will be set up to reference |frame_buffer|'s data.
// FFmpeg expects the initial allocation to be zero-initialized according to
// http://crbug.com/390941. Our pool is set up to zero-initialize new buffers.
// TODO(nisse): Delete that feature from the video pool, instead add
// an explicit call to InitializeData here.
rtc::scoped_refptr<I420Buffer> frame_buffer =
decoder->pool_.CreateBuffer(width, height);
int y_size = width * height;
int uv_size = frame_buffer->ChromaWidth() * frame_buffer->ChromaHeight();
// DCHECK that we have a continuous buffer as is required.
RTC_DCHECK_EQ(frame_buffer->DataU(), frame_buffer->DataY() + y_size);
RTC_DCHECK_EQ(frame_buffer->DataV(), frame_buffer->DataU() + uv_size);
int total_size = y_size + 2 * uv_size;
av_frame->format = context->pix_fmt;
av_frame->reordered_opaque = context->reordered_opaque;
// Set |av_frame| members as required by FFmpeg.
av_frame->data[kYPlaneIndex] = frame_buffer->MutableDataY();
av_frame->linesize[kYPlaneIndex] = frame_buffer->StrideY();
av_frame->data[kUPlaneIndex] = frame_buffer->MutableDataU();
av_frame->linesize[kUPlaneIndex] = frame_buffer->StrideU();
av_frame->data[kVPlaneIndex] = frame_buffer->MutableDataV();
av_frame->linesize[kVPlaneIndex] = frame_buffer->StrideV();
RTC_DCHECK_EQ(av_frame->extended_data, av_frame->data);
// Create a VideoFrame object, to keep a reference to the buffer.
// TODO(nisse): The VideoFrame's timestamp and rotation info is not used.
// Refactor to do not use a VideoFrame object at all.
av_frame->buf[0] = av_buffer_create(
av_frame->data[kYPlaneIndex],
total_size,
AVFreeBuffer2,
static_cast<void*>(new VideoFrame(frame_buffer,
0 /* timestamp */,
0 /* render_time_ms */,
kVideoRotation_0)),
0);
RTC_CHECK(av_frame->buf[0]);
return 0;
}
void H264DecoderImpl::AVFreeBuffer2(void* opaque, uint8_t* data) {
// The buffer pool recycles the buffer used by |video_frame| when there are no
// more references to it. |video_frame| is a thin buffer holder and is not
// recycled.
VideoFrame* video_frame = static_cast<VideoFrame*>(opaque);
delete video_frame;
}
H264DecoderImpl::H264DecoderImpl() : pool_(true),
decoded_image_callback_(nullptr),
has_reported_init_(false),
has_reported_error_(false) {
}
H264DecoderImpl::~H264DecoderImpl() {
Release();
}
int32_t H264DecoderImpl::InitDecode(const VideoCodec* codec_settings,
int32_t number_of_cores) {
ReportInit();
if (codec_settings &&
codec_settings->codecType != kVideoCodecH264) {
ReportError();
return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
}
// FFmpeg must have been initialized (with |av_lockmgr_register| and
// |av_register_all|) before we proceed. |InitializeFFmpeg| does this, which
// makes sense for WebRTC standalone. In other cases, such as Chromium, FFmpeg
// is initialized externally and calling |InitializeFFmpeg| would be
// thread-unsafe and result in FFmpeg being initialized twice, which could
// break other FFmpeg usage. See the |rtc_initialize_ffmpeg| flag.
#if defined(WEBRTC_INITIALIZE_FFMPEG)
// Make sure FFmpeg has been initialized. Subsequent |InitializeFFmpeg| calls
// do nothing.
InitializeFFmpeg();
#endif
// Release necessary in case of re-initializing.
int32_t ret = Release();
if (ret != WEBRTC_VIDEO_CODEC_OK) {
ReportError();
return ret;
}
RTC_DCHECK(!av_context_);
// Initialize AVCodecContext.
av_context_.reset(avcodec_alloc_context3(nullptr));
av_context_->codec_type = AVMEDIA_TYPE_VIDEO;
av_context_->codec_id = AV_CODEC_ID_H264;
if (codec_settings) {
av_context_->coded_width = codec_settings->width;
av_context_->coded_height = codec_settings->height;
}
av_context_->pix_fmt = kPixelFormat;
av_context_->extradata = nullptr;
av_context_->extradata_size = 0;
// If this is ever increased, look at |av_context_->thread_safe_callbacks| and
// make it possible to disable the thread checker in the frame buffer pool.
av_context_->thread_count = 1;
av_context_->thread_type = FF_THREAD_SLICE;
// Function used by FFmpeg to get buffers to store decoded frames in.
av_context_->get_buffer2 = AVGetBuffer2;
// |get_buffer2| is called with the context, there |opaque| can be used to get
// a pointer |this|.
av_context_->opaque = this;
// Use ref counted frames (av_frame_unref).
av_context_->refcounted_frames = 1; // true
AVCodec* codec = avcodec_find_decoder(av_context_->codec_id);
if (!codec) {
// This is an indication that FFmpeg has not been initialized or it has not
// been compiled/initialized with the correct set of codecs.
LOG(LS_ERROR) << "FFmpeg H.264 decoder not found.";
Release();
ReportError();
return WEBRTC_VIDEO_CODEC_ERROR;
}
int res = avcodec_open2(av_context_.get(), codec, nullptr);
if (res < 0) {
LOG(LS_ERROR) << "avcodec_open2 error: " << res;
Release();
ReportError();
return WEBRTC_VIDEO_CODEC_ERROR;
}
av_frame_.reset(av_frame_alloc());
return WEBRTC_VIDEO_CODEC_OK;
}
int32_t H264DecoderImpl::Release() {
av_context_.reset();
av_frame_.reset();
return WEBRTC_VIDEO_CODEC_OK;
}
int32_t H264DecoderImpl::RegisterDecodeCompleteCallback(
DecodedImageCallback* callback) {
decoded_image_callback_ = callback;
return WEBRTC_VIDEO_CODEC_OK;
}
int32_t H264DecoderImpl::Decode(const EncodedImage& input_image,
bool /*missing_frames*/,
const RTPFragmentationHeader* /*fragmentation*/,
const CodecSpecificInfo* codec_specific_info,
int64_t /*render_time_ms*/) {
if (!IsInitialized()) {
ReportError();
return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
}
if (!decoded_image_callback_) {
LOG(LS_WARNING) << "InitDecode() has been called, but a callback function "
"has not been set with RegisterDecodeCompleteCallback()";
ReportError();
return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
}
if (!input_image._buffer || !input_image._length) {
ReportError();
return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
}
if (codec_specific_info &&
codec_specific_info->codecType != kVideoCodecH264) {
ReportError();
return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
}
// FFmpeg requires padding due to some optimized bitstream readers reading 32
// or 64 bits at once and could read over the end. See avcodec_decode_video2.
RTC_CHECK_GE(input_image._size, input_image._length +
EncodedImage::GetBufferPaddingBytes(kVideoCodecH264));
// "If the first 23 bits of the additional bytes are not 0, then damaged MPEG
// bitstreams could cause overread and segfault." See
// AV_INPUT_BUFFER_PADDING_SIZE. We'll zero the entire padding just in case.
memset(input_image._buffer + input_image._length,
0,
EncodedImage::GetBufferPaddingBytes(kVideoCodecH264));
AVPacket packet;
av_init_packet(&packet);
packet.data = input_image._buffer;
if (input_image._length >
static_cast<size_t>(std::numeric_limits<int>::max())) {
ReportError();
return WEBRTC_VIDEO_CODEC_ERROR;
}
packet.size = static_cast<int>(input_image._length);
av_context_->reordered_opaque = input_image.ntp_time_ms_ * 1000; // ms -> μs
int frame_decoded = 0;
int result = avcodec_decode_video2(av_context_.get(),
av_frame_.get(),
&frame_decoded,
&packet);
if (result < 0) {
LOG(LS_ERROR) << "avcodec_decode_video2 error: " << result;
ReportError();
return WEBRTC_VIDEO_CODEC_ERROR;
}
// |result| is number of bytes used, which should be all of them.
if (result != packet.size) {
LOG(LS_ERROR) << "avcodec_decode_video2 consumed " << result << " bytes "
"when " << packet.size << " bytes were expected.";
ReportError();
return WEBRTC_VIDEO_CODEC_ERROR;
}
if (!frame_decoded) {
LOG(LS_WARNING) << "avcodec_decode_video2 successful but no frame was "
"decoded.";
return WEBRTC_VIDEO_CODEC_OK;
}
// Obtain the |video_frame| containing the decoded image.
VideoFrame* video_frame = static_cast<VideoFrame*>(
av_buffer_get_opaque(av_frame_->buf[0]));
RTC_DCHECK(video_frame);
rtc::scoped_refptr<webrtc::I420BufferInterface> i420_buffer =
video_frame->video_frame_buffer()->GetI420();
RTC_CHECK_EQ(av_frame_->data[kYPlaneIndex], i420_buffer->DataY());
RTC_CHECK_EQ(av_frame_->data[kUPlaneIndex], i420_buffer->DataU());
RTC_CHECK_EQ(av_frame_->data[kVPlaneIndex], i420_buffer->DataV());
video_frame->set_timestamp(input_image._timeStamp);
rtc::Optional<uint8_t> qp;
// TODO(sakal): Maybe it is possible to get QP directly from FFmpeg.
h264_bitstream_parser_.ParseBitstream(input_image._buffer,
input_image._length);
int qp_int;
if (h264_bitstream_parser_.GetLastSliceQp(&qp_int)) {
qp.emplace(qp_int);
}
// The decoded image may be larger than what is supposed to be visible, see
// |AVGetBuffer2|'s use of |avcodec_align_dimensions|. This crops the image
// without copying the underlying buffer.
if (av_frame_->width != i420_buffer->width() ||
av_frame_->height != i420_buffer->height()) {
rtc::scoped_refptr<VideoFrameBuffer> cropped_buf(
new rtc::RefCountedObject<WrappedI420Buffer>(
av_frame_->width, av_frame_->height,
i420_buffer->DataY(), i420_buffer->StrideY(),
i420_buffer->DataU(), i420_buffer->StrideU(),
i420_buffer->DataV(), i420_buffer->StrideV(),
rtc::KeepRefUntilDone(i420_buffer)));
VideoFrame cropped_frame(
cropped_buf, video_frame->timestamp(), video_frame->render_time_ms(),
video_frame->rotation());
// TODO(nisse): Timestamp and rotation are all zero here. Change decoder
// interface to pass a VideoFrameBuffer instead of a VideoFrame?
decoded_image_callback_->Decoded(cropped_frame, rtc::Optional<int32_t>(),
qp);
} else {
// Return decoded frame.
decoded_image_callback_->Decoded(*video_frame, rtc::Optional<int32_t>(),
qp);
}
// Stop referencing it, possibly freeing |video_frame|.
av_frame_unref(av_frame_.get());
video_frame = nullptr;
return WEBRTC_VIDEO_CODEC_OK;
}
const char* H264DecoderImpl::ImplementationName() const {
return "FFmpeg";
}
bool H264DecoderImpl::IsInitialized() const {
return av_context_ != nullptr;
}
void H264DecoderImpl::ReportInit() {
if (has_reported_init_)
return;
RTC_HISTOGRAM_ENUMERATION("WebRTC.Video.H264DecoderImpl.Event",
kH264DecoderEventInit,
kH264DecoderEventMax);
has_reported_init_ = true;
}
void H264DecoderImpl::ReportError() {
if (has_reported_error_)
return;
RTC_HISTOGRAM_ENUMERATION("WebRTC.Video.H264DecoderImpl.Event",
kH264DecoderEventError,
kH264DecoderEventMax);
has_reported_error_ = true;
}
} // namespace webrtc

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/*
* Copyright (c) 2015 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 WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_DECODER_IMPL_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_DECODER_IMPL_H_
#include <memory>
#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
extern "C" {
#include "third_party/ffmpeg/libavcodec/avcodec.h"
} // extern "C"
#include "webrtc/common_video/h264/h264_bitstream_parser.h"
#include "webrtc/common_video/include/i420_buffer_pool.h"
namespace webrtc {
struct AVCodecContextDeleter {
void operator()(AVCodecContext* ptr) const { avcodec_free_context(&ptr); }
};
struct AVFrameDeleter {
void operator()(AVFrame* ptr) const { av_frame_free(&ptr); }
};
class H264DecoderImpl : public H264Decoder {
public:
H264DecoderImpl();
~H264DecoderImpl() override;
// If |codec_settings| is NULL it is ignored. If it is not NULL,
// |codec_settings->codecType| must be |kVideoCodecH264|.
int32_t InitDecode(const VideoCodec* codec_settings,
int32_t number_of_cores) override;
int32_t Release() override;
int32_t RegisterDecodeCompleteCallback(
DecodedImageCallback* callback) override;
// |missing_frames|, |fragmentation| and |render_time_ms| are ignored.
int32_t Decode(const EncodedImage& input_image,
bool /*missing_frames*/,
const RTPFragmentationHeader* /*fragmentation*/,
const CodecSpecificInfo* codec_specific_info = nullptr,
int64_t render_time_ms = -1) override;
const char* ImplementationName() const override;
private:
// Called by FFmpeg when it needs a frame buffer to store decoded frames in.
// The |VideoFrame| returned by FFmpeg at |Decode| originate from here. Their
// buffers are reference counted and freed by FFmpeg using |AVFreeBuffer2|.
static int AVGetBuffer2(
AVCodecContext* context, AVFrame* av_frame, int flags);
// Called by FFmpeg when it is done with a video frame, see |AVGetBuffer2|.
static void AVFreeBuffer2(void* opaque, uint8_t* data);
bool IsInitialized() const;
// Reports statistics with histograms.
void ReportInit();
void ReportError();
I420BufferPool pool_;
std::unique_ptr<AVCodecContext, AVCodecContextDeleter> av_context_;
std::unique_ptr<AVFrame, AVFrameDeleter> av_frame_;
DecodedImageCallback* decoded_image_callback_;
bool has_reported_init_;
bool has_reported_error_;
webrtc::H264BitstreamParser h264_bitstream_parser_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_DECODER_IMPL_H_

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/*
* Copyright (c) 2015 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/codecs/h264/h264_encoder_impl.h"
#include <limits>
#include <string>
#include "third_party/openh264/src/codec/api/svc/codec_api.h"
#include "third_party/openh264/src/codec/api/svc/codec_app_def.h"
#include "third_party/openh264/src/codec/api/svc/codec_def.h"
#include "third_party/openh264/src/codec/api/svc/codec_ver.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/logging.h"
#include "webrtc/rtc_base/timeutils.h"
#include "webrtc/system_wrappers/include/metrics.h"
namespace webrtc {
namespace {
const bool kOpenH264EncoderDetailedLogging = false;
// Used by histograms. Values of entries should not be changed.
enum H264EncoderImplEvent {
kH264EncoderEventInit = 0,
kH264EncoderEventError = 1,
kH264EncoderEventMax = 16,
};
int NumberOfThreads(int width, int height, int number_of_cores) {
// TODO(hbos): In Chromium, multiple threads do not work with sandbox on Mac,
// see crbug.com/583348. Until further investigated, only use one thread.
// if (width * height >= 1920 * 1080 && number_of_cores > 8) {
// return 8; // 8 threads for 1080p on high perf machines.
// } else if (width * height > 1280 * 960 && number_of_cores >= 6) {
// return 3; // 3 threads for 1080p.
// } else if (width * height > 640 * 480 && number_of_cores >= 3) {
// return 2; // 2 threads for qHD/HD.
// } else {
// return 1; // 1 thread for VGA or less.
// }
// TODO(sprang): Also check sSliceArgument.uiSliceNum om GetEncoderPrams(),
// before enabling multithreading here.
return 1;
}
FrameType ConvertToVideoFrameType(EVideoFrameType type) {
switch (type) {
case videoFrameTypeIDR:
return kVideoFrameKey;
case videoFrameTypeSkip:
case videoFrameTypeI:
case videoFrameTypeP:
case videoFrameTypeIPMixed:
return kVideoFrameDelta;
case videoFrameTypeInvalid:
break;
}
RTC_NOTREACHED() << "Unexpected/invalid frame type: " << type;
return kEmptyFrame;
}
} // namespace
// Helper method used by H264EncoderImpl::Encode.
// Copies the encoded bytes from |info| to |encoded_image| and updates the
// fragmentation information of |frag_header|. The |encoded_image->_buffer| may
// be deleted and reallocated if a bigger buffer is required.
//
// After OpenH264 encoding, the encoded bytes are stored in |info| spread out
// over a number of layers and "NAL units". Each NAL unit is a fragment starting
// with the four-byte start code {0,0,0,1}. All of this data (including the
// start codes) is copied to the |encoded_image->_buffer| and the |frag_header|
// is updated to point to each fragment, with offsets and lengths set as to
// exclude the start codes.
static void RtpFragmentize(EncodedImage* encoded_image,
std::unique_ptr<uint8_t[]>* encoded_image_buffer,
const VideoFrameBuffer& frame_buffer,
SFrameBSInfo* info,
RTPFragmentationHeader* frag_header) {
// Calculate minimum buffer size required to hold encoded data.
size_t required_size = 0;
size_t fragments_count = 0;
for (int layer = 0; layer < info->iLayerNum; ++layer) {
const SLayerBSInfo& layerInfo = info->sLayerInfo[layer];
for (int nal = 0; nal < layerInfo.iNalCount; ++nal, ++fragments_count) {
RTC_CHECK_GE(layerInfo.pNalLengthInByte[nal], 0);
// Ensure |required_size| will not overflow.
RTC_CHECK_LE(layerInfo.pNalLengthInByte[nal],
std::numeric_limits<size_t>::max() - required_size);
required_size += layerInfo.pNalLengthInByte[nal];
}
}
if (encoded_image->_size < required_size) {
// Increase buffer size. Allocate enough to hold an unencoded image, this
// should be more than enough to hold any encoded data of future frames of
// the same size (avoiding possible future reallocation due to variations in
// required size).
encoded_image->_size = CalcBufferSize(
VideoType::kI420, frame_buffer.width(), frame_buffer.height());
if (encoded_image->_size < required_size) {
// Encoded data > unencoded data. Allocate required bytes.
LOG(LS_WARNING) << "Encoding produced more bytes than the original image "
<< "data! Original bytes: " << encoded_image->_size
<< ", encoded bytes: " << required_size << ".";
encoded_image->_size = required_size;
}
encoded_image->_buffer = new uint8_t[encoded_image->_size];
encoded_image_buffer->reset(encoded_image->_buffer);
}
// Iterate layers and NAL units, note each NAL unit as a fragment and copy
// the data to |encoded_image->_buffer|.
const uint8_t start_code[4] = {0, 0, 0, 1};
frag_header->VerifyAndAllocateFragmentationHeader(fragments_count);
size_t frag = 0;
encoded_image->_length = 0;
for (int layer = 0; layer < info->iLayerNum; ++layer) {
const SLayerBSInfo& layerInfo = info->sLayerInfo[layer];
// Iterate NAL units making up this layer, noting fragments.
size_t layer_len = 0;
for (int nal = 0; nal < layerInfo.iNalCount; ++nal, ++frag) {
// Because the sum of all layer lengths, |required_size|, fits in a
// |size_t|, we know that any indices in-between will not overflow.
RTC_DCHECK_GE(layerInfo.pNalLengthInByte[nal], 4);
RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len+0], start_code[0]);
RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len+1], start_code[1]);
RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len+2], start_code[2]);
RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len+3], start_code[3]);
frag_header->fragmentationOffset[frag] =
encoded_image->_length + layer_len + sizeof(start_code);
frag_header->fragmentationLength[frag] =
layerInfo.pNalLengthInByte[nal] - sizeof(start_code);
layer_len += layerInfo.pNalLengthInByte[nal];
}
// Copy the entire layer's data (including start codes).
memcpy(encoded_image->_buffer + encoded_image->_length,
layerInfo.pBsBuf,
layer_len);
encoded_image->_length += layer_len;
}
}
H264EncoderImpl::H264EncoderImpl(const cricket::VideoCodec& codec)
: openh264_encoder_(nullptr),
width_(0),
height_(0),
max_frame_rate_(0.0f),
target_bps_(0),
max_bps_(0),
mode_(kRealtimeVideo),
frame_dropping_on_(false),
key_frame_interval_(0),
packetization_mode_(H264PacketizationMode::SingleNalUnit),
max_payload_size_(0),
number_of_cores_(0),
encoded_image_callback_(nullptr),
has_reported_init_(false),
has_reported_error_(false) {
RTC_CHECK(cricket::CodecNamesEq(codec.name, cricket::kH264CodecName));
std::string packetization_mode_string;
if (codec.GetParam(cricket::kH264FmtpPacketizationMode,
&packetization_mode_string) &&
packetization_mode_string == "1") {
packetization_mode_ = H264PacketizationMode::NonInterleaved;
}
}
H264EncoderImpl::~H264EncoderImpl() {
Release();
}
int32_t H264EncoderImpl::InitEncode(const VideoCodec* codec_settings,
int32_t number_of_cores,
size_t max_payload_size) {
ReportInit();
if (!codec_settings ||
codec_settings->codecType != kVideoCodecH264) {
ReportError();
return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
}
if (codec_settings->maxFramerate == 0) {
ReportError();
return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
}
if (codec_settings->width < 1 || codec_settings->height < 1) {
ReportError();
return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
}
int32_t release_ret = Release();
if (release_ret != WEBRTC_VIDEO_CODEC_OK) {
ReportError();
return release_ret;
}
RTC_DCHECK(!openh264_encoder_);
// Create encoder.
if (WelsCreateSVCEncoder(&openh264_encoder_) != 0) {
// Failed to create encoder.
LOG(LS_ERROR) << "Failed to create OpenH264 encoder";
RTC_DCHECK(!openh264_encoder_);
ReportError();
return WEBRTC_VIDEO_CODEC_ERROR;
}
RTC_DCHECK(openh264_encoder_);
if (kOpenH264EncoderDetailedLogging) {
int trace_level = WELS_LOG_DETAIL;
openh264_encoder_->SetOption(ENCODER_OPTION_TRACE_LEVEL,
&trace_level);
}
// else WELS_LOG_DEFAULT is used by default.
number_of_cores_ = number_of_cores;
// Set internal settings from codec_settings
width_ = codec_settings->width;
height_ = codec_settings->height;
max_frame_rate_ = static_cast<float>(codec_settings->maxFramerate);
mode_ = codec_settings->mode;
frame_dropping_on_ = codec_settings->H264().frameDroppingOn;
key_frame_interval_ = codec_settings->H264().keyFrameInterval;
max_payload_size_ = max_payload_size;
// Codec_settings uses kbits/second; encoder uses bits/second.
max_bps_ = codec_settings->maxBitrate * 1000;
if (codec_settings->targetBitrate == 0)
target_bps_ = codec_settings->startBitrate * 1000;
else
target_bps_ = codec_settings->targetBitrate * 1000;
SEncParamExt encoder_params = CreateEncoderParams();
// Initialize.
if (openh264_encoder_->InitializeExt(&encoder_params) != 0) {
LOG(LS_ERROR) << "Failed to initialize OpenH264 encoder";
Release();
ReportError();
return WEBRTC_VIDEO_CODEC_ERROR;
}
// TODO(pbos): Base init params on these values before submitting.
int video_format = EVideoFormatType::videoFormatI420;
openh264_encoder_->SetOption(ENCODER_OPTION_DATAFORMAT,
&video_format);
// Initialize encoded image. Default buffer size: size of unencoded data.
encoded_image_._size = CalcBufferSize(VideoType::kI420, codec_settings->width,
codec_settings->height);
encoded_image_._buffer = new uint8_t[encoded_image_._size];
encoded_image_buffer_.reset(encoded_image_._buffer);
encoded_image_._completeFrame = true;
encoded_image_._encodedWidth = 0;
encoded_image_._encodedHeight = 0;
encoded_image_._length = 0;
return WEBRTC_VIDEO_CODEC_OK;
}
int32_t H264EncoderImpl::Release() {
if (openh264_encoder_) {
RTC_CHECK_EQ(0, openh264_encoder_->Uninitialize());
WelsDestroySVCEncoder(openh264_encoder_);
openh264_encoder_ = nullptr;
}
encoded_image_._buffer = nullptr;
encoded_image_buffer_.reset();
return WEBRTC_VIDEO_CODEC_OK;
}
int32_t H264EncoderImpl::RegisterEncodeCompleteCallback(
EncodedImageCallback* callback) {
encoded_image_callback_ = callback;
return WEBRTC_VIDEO_CODEC_OK;
}
int32_t H264EncoderImpl::SetRateAllocation(
const BitrateAllocation& bitrate_allocation,
uint32_t framerate) {
if (bitrate_allocation.get_sum_bps() <= 0 || framerate <= 0)
return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
target_bps_ = bitrate_allocation.get_sum_bps();
max_frame_rate_ = static_cast<float>(framerate);
SBitrateInfo target_bitrate;
memset(&target_bitrate, 0, sizeof(SBitrateInfo));
target_bitrate.iLayer = SPATIAL_LAYER_ALL,
target_bitrate.iBitrate = target_bps_;
openh264_encoder_->SetOption(ENCODER_OPTION_BITRATE,
&target_bitrate);
openh264_encoder_->SetOption(ENCODER_OPTION_FRAME_RATE, &max_frame_rate_);
return WEBRTC_VIDEO_CODEC_OK;
}
int32_t H264EncoderImpl::Encode(const VideoFrame& input_frame,
const CodecSpecificInfo* codec_specific_info,
const std::vector<FrameType>* frame_types) {
if (!IsInitialized()) {
ReportError();
return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
}
if (!encoded_image_callback_) {
LOG(LS_WARNING) << "InitEncode() has been called, but a callback function "
<< "has not been set with RegisterEncodeCompleteCallback()";
ReportError();
return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
}
bool force_key_frame = false;
if (frame_types != nullptr) {
// We only support a single stream.
RTC_DCHECK_EQ(frame_types->size(), 1);
// Skip frame?
if ((*frame_types)[0] == kEmptyFrame) {
return WEBRTC_VIDEO_CODEC_OK;
}
// Force key frame?
force_key_frame = (*frame_types)[0] == kVideoFrameKey;
}
if (force_key_frame) {
// API doc says ForceIntraFrame(false) does nothing, but calling this
// function forces a key frame regardless of the |bIDR| argument's value.
// (If every frame is a key frame we get lag/delays.)
openh264_encoder_->ForceIntraFrame(true);
}
rtc::scoped_refptr<const I420BufferInterface> frame_buffer =
input_frame.video_frame_buffer()->ToI420();
// EncodeFrame input.
SSourcePicture picture;
memset(&picture, 0, sizeof(SSourcePicture));
picture.iPicWidth = frame_buffer->width();
picture.iPicHeight = frame_buffer->height();
picture.iColorFormat = EVideoFormatType::videoFormatI420;
picture.uiTimeStamp = input_frame.ntp_time_ms();
picture.iStride[0] = frame_buffer->StrideY();
picture.iStride[1] = frame_buffer->StrideU();
picture.iStride[2] = frame_buffer->StrideV();
picture.pData[0] = const_cast<uint8_t*>(frame_buffer->DataY());
picture.pData[1] = const_cast<uint8_t*>(frame_buffer->DataU());
picture.pData[2] = const_cast<uint8_t*>(frame_buffer->DataV());
// EncodeFrame output.
SFrameBSInfo info;
memset(&info, 0, sizeof(SFrameBSInfo));
// Encode!
int enc_ret = openh264_encoder_->EncodeFrame(&picture, &info);
if (enc_ret != 0) {
LOG(LS_ERROR) << "OpenH264 frame encoding failed, EncodeFrame returned "
<< enc_ret << ".";
ReportError();
return WEBRTC_VIDEO_CODEC_ERROR;
}
encoded_image_._encodedWidth = frame_buffer->width();
encoded_image_._encodedHeight = frame_buffer->height();
encoded_image_._timeStamp = input_frame.timestamp();
encoded_image_.ntp_time_ms_ = input_frame.ntp_time_ms();
encoded_image_.capture_time_ms_ = input_frame.render_time_ms();
encoded_image_.rotation_ = input_frame.rotation();
encoded_image_.content_type_ = (mode_ == kScreensharing)
? VideoContentType::SCREENSHARE
: VideoContentType::UNSPECIFIED;
encoded_image_.timing_.flags = TimingFrameFlags::kInvalid;
encoded_image_._frameType = ConvertToVideoFrameType(info.eFrameType);
// Split encoded image up into fragments. This also updates |encoded_image_|.
RTPFragmentationHeader frag_header;
RtpFragmentize(&encoded_image_, &encoded_image_buffer_, *frame_buffer, &info,
&frag_header);
// Encoder can skip frames to save bandwidth in which case
// |encoded_image_._length| == 0.
if (encoded_image_._length > 0) {
// Parse QP.
h264_bitstream_parser_.ParseBitstream(encoded_image_._buffer,
encoded_image_._length);
h264_bitstream_parser_.GetLastSliceQp(&encoded_image_.qp_);
// Deliver encoded image.
CodecSpecificInfo codec_specific;
codec_specific.codecType = kVideoCodecH264;
codec_specific.codecSpecific.H264.packetization_mode = packetization_mode_;
encoded_image_callback_->OnEncodedImage(encoded_image_, &codec_specific,
&frag_header);
}
return WEBRTC_VIDEO_CODEC_OK;
}
const char* H264EncoderImpl::ImplementationName() const {
return "OpenH264";
}
bool H264EncoderImpl::IsInitialized() const {
return openh264_encoder_ != nullptr;
}
// Initialization parameters.
// There are two ways to initialize. There is SEncParamBase (cleared with
// memset(&p, 0, sizeof(SEncParamBase)) used in Initialize, and SEncParamExt
// which is a superset of SEncParamBase (cleared with GetDefaultParams) used
// in InitializeExt.
SEncParamExt H264EncoderImpl::CreateEncoderParams() const {
RTC_DCHECK(openh264_encoder_);
SEncParamExt encoder_params;
openh264_encoder_->GetDefaultParams(&encoder_params);
if (mode_ == kRealtimeVideo) {
encoder_params.iUsageType = CAMERA_VIDEO_REAL_TIME;
} else if (mode_ == kScreensharing) {
encoder_params.iUsageType = SCREEN_CONTENT_REAL_TIME;
} else {
RTC_NOTREACHED();
}
encoder_params.iPicWidth = width_;
encoder_params.iPicHeight = height_;
encoder_params.iTargetBitrate = target_bps_;
encoder_params.iMaxBitrate = max_bps_;
// Rate Control mode
encoder_params.iRCMode = RC_BITRATE_MODE;
encoder_params.fMaxFrameRate = max_frame_rate_;
// The following parameters are extension parameters (they're in SEncParamExt,
// not in SEncParamBase).
encoder_params.bEnableFrameSkip = frame_dropping_on_;
// |uiIntraPeriod| - multiple of GOP size
// |keyFrameInterval| - number of frames
encoder_params.uiIntraPeriod = key_frame_interval_;
encoder_params.uiMaxNalSize = 0;
// Threading model: use auto.
// 0: auto (dynamic imp. internal encoder)
// 1: single thread (default value)
// >1: number of threads
encoder_params.iMultipleThreadIdc = NumberOfThreads(
encoder_params.iPicWidth, encoder_params.iPicHeight, number_of_cores_);
// The base spatial layer 0 is the only one we use.
encoder_params.sSpatialLayers[0].iVideoWidth = encoder_params.iPicWidth;
encoder_params.sSpatialLayers[0].iVideoHeight = encoder_params.iPicHeight;
encoder_params.sSpatialLayers[0].fFrameRate = encoder_params.fMaxFrameRate;
encoder_params.sSpatialLayers[0].iSpatialBitrate =
encoder_params.iTargetBitrate;
encoder_params.sSpatialLayers[0].iMaxSpatialBitrate =
encoder_params.iMaxBitrate;
LOG(INFO) << "OpenH264 version is " << OPENH264_MAJOR << "."
<< OPENH264_MINOR;
switch (packetization_mode_) {
case H264PacketizationMode::SingleNalUnit:
// Limit the size of the packets produced.
encoder_params.sSpatialLayers[0].sSliceArgument.uiSliceNum = 1;
encoder_params.sSpatialLayers[0].sSliceArgument.uiSliceMode =
SM_SIZELIMITED_SLICE;
encoder_params.sSpatialLayers[0].sSliceArgument.uiSliceSizeConstraint =
static_cast<unsigned int>(max_payload_size_);
break;
case H264PacketizationMode::NonInterleaved:
// When uiSliceMode = SM_FIXEDSLCNUM_SLICE, uiSliceNum = 0 means auto
// design it with cpu core number.
// TODO(sprang): Set to 0 when we understand why the rate controller borks
// when uiSliceNum > 1.
encoder_params.sSpatialLayers[0].sSliceArgument.uiSliceNum = 1;
encoder_params.sSpatialLayers[0].sSliceArgument.uiSliceMode =
SM_FIXEDSLCNUM_SLICE;
break;
}
return encoder_params;
}
void H264EncoderImpl::ReportInit() {
if (has_reported_init_)
return;
RTC_HISTOGRAM_ENUMERATION("WebRTC.Video.H264EncoderImpl.Event",
kH264EncoderEventInit,
kH264EncoderEventMax);
has_reported_init_ = true;
}
void H264EncoderImpl::ReportError() {
if (has_reported_error_)
return;
RTC_HISTOGRAM_ENUMERATION("WebRTC.Video.H264EncoderImpl.Event",
kH264EncoderEventError,
kH264EncoderEventMax);
has_reported_error_ = true;
}
int32_t H264EncoderImpl::SetChannelParameters(
uint32_t packet_loss, int64_t rtt) {
return WEBRTC_VIDEO_CODEC_OK;
}
int32_t H264EncoderImpl::SetPeriodicKeyFrames(bool enable) {
return WEBRTC_VIDEO_CODEC_OK;
}
VideoEncoder::ScalingSettings H264EncoderImpl::GetScalingSettings() const {
return VideoEncoder::ScalingSettings(true);
}
} // namespace webrtc

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/*
* Copyright (c) 2015 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 WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_ENCODER_IMPL_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_ENCODER_IMPL_H_
#include <memory>
#include <vector>
#include "webrtc/common_video/h264/h264_bitstream_parser.h"
#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
#include "webrtc/modules/video_coding/utility/quality_scaler.h"
#include "third_party/openh264/src/codec/api/svc/codec_app_def.h"
class ISVCEncoder;
namespace webrtc {
class H264EncoderImpl : public H264Encoder {
public:
explicit H264EncoderImpl(const cricket::VideoCodec& codec);
~H264EncoderImpl() override;
// |max_payload_size| is ignored.
// The following members of |codec_settings| are used. The rest are ignored.
// - codecType (must be kVideoCodecH264)
// - targetBitrate
// - maxFramerate
// - width
// - height
int32_t InitEncode(const VideoCodec* codec_settings,
int32_t number_of_cores,
size_t max_payload_size) override;
int32_t Release() override;
int32_t RegisterEncodeCompleteCallback(
EncodedImageCallback* callback) override;
int32_t SetRateAllocation(const BitrateAllocation& bitrate_allocation,
uint32_t framerate) override;
// The result of encoding - an EncodedImage and RTPFragmentationHeader - are
// passed to the encode complete callback.
int32_t Encode(const VideoFrame& frame,
const CodecSpecificInfo* codec_specific_info,
const std::vector<FrameType>* frame_types) override;
const char* ImplementationName() const override;
VideoEncoder::ScalingSettings GetScalingSettings() const override;
// Unsupported / Do nothing.
int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
int32_t SetPeriodicKeyFrames(bool enable) override;
// Exposed for testing.
H264PacketizationMode PacketizationModeForTesting() const {
return packetization_mode_;
}
private:
bool IsInitialized() const;
SEncParamExt CreateEncoderParams() const;
webrtc::H264BitstreamParser h264_bitstream_parser_;
// Reports statistics with histograms.
void ReportInit();
void ReportError();
ISVCEncoder* openh264_encoder_;
// Settings that are used by this encoder.
int width_;
int height_;
float max_frame_rate_;
uint32_t target_bps_;
uint32_t max_bps_;
VideoCodecMode mode_;
// H.264 specifc parameters
bool frame_dropping_on_;
int key_frame_interval_;
H264PacketizationMode packetization_mode_;
size_t max_payload_size_;
int32_t number_of_cores_;
EncodedImage encoded_image_;
std::unique_ptr<uint8_t[]> encoded_image_buffer_;
EncodedImageCallback* encoded_image_callback_;
bool has_reported_init_;
bool has_reported_error_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_ENCODER_IMPL_H_

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/*
* Copyright (c) 2015 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/codecs/h264/h264_encoder_impl.h"
#include "webrtc/test/gtest.h"
namespace webrtc {
namespace {
const int kMaxPayloadSize = 1024;
const int kNumCores = 1;
void SetDefaultSettings(VideoCodec* codec_settings) {
codec_settings->codecType = kVideoCodecH264;
codec_settings->maxFramerate = 60;
codec_settings->width = 640;
codec_settings->height = 480;
// If frame dropping is false, we get a warning that bitrate can't
// be controlled for RC_QUALITY_MODE; RC_BITRATE_MODE and RC_TIMESTAMP_MODE
codec_settings->H264()->frameDroppingOn = true;
codec_settings->targetBitrate = 2000;
codec_settings->maxBitrate = 4000;
}
TEST(H264EncoderImplTest, CanInitializeWithDefaultParameters) {
H264EncoderImpl encoder(cricket::VideoCodec("H264"));
VideoCodec codec_settings;
SetDefaultSettings(&codec_settings);
EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
encoder.InitEncode(&codec_settings, kNumCores, kMaxPayloadSize));
EXPECT_EQ(H264PacketizationMode::NonInterleaved,
encoder.PacketizationModeForTesting());
}
TEST(H264EncoderImplTest, CanInitializeWithNonInterleavedModeExplicitly) {
cricket::VideoCodec codec("H264");
codec.SetParam(cricket::kH264FmtpPacketizationMode, "1");
H264EncoderImpl encoder(codec);
VideoCodec codec_settings;
SetDefaultSettings(&codec_settings);
EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
encoder.InitEncode(&codec_settings, kNumCores, kMaxPayloadSize));
EXPECT_EQ(H264PacketizationMode::NonInterleaved,
encoder.PacketizationModeForTesting());
}
TEST(H264EncoderImplTest, CanInitializeWithSingleNalUnitModeExplicitly) {
cricket::VideoCodec codec("H264");
codec.SetParam(cricket::kH264FmtpPacketizationMode, "0");
H264EncoderImpl encoder(codec);
VideoCodec codec_settings;
SetDefaultSettings(&codec_settings);
EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
encoder.InitEncode(&codec_settings, kNumCores, kMaxPayloadSize));
EXPECT_EQ(H264PacketizationMode::SingleNalUnit,
encoder.PacketizationModeForTesting());
}
TEST(H264EncoderImplTest, CanInitializeWithRemovedParameter) {
cricket::VideoCodec codec("H264");
codec.RemoveParam(cricket::kH264FmtpPacketizationMode);
H264EncoderImpl encoder(codec);
VideoCodec codec_settings;
SetDefaultSettings(&codec_settings);
EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
encoder.InitEncode(&codec_settings, kNumCores, kMaxPayloadSize));
EXPECT_EQ(H264PacketizationMode::SingleNalUnit,
encoder.PacketizationModeForTesting());
}
} // anonymous namespace
} // namespace webrtc

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/*
* Copyright (c) 2015 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 WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_H_
#include "webrtc/media/base/codec.h"
#include "webrtc/modules/video_coding/include/video_codec_interface.h"
namespace webrtc {
// Set to disable the H.264 encoder/decoder implementations that are provided if
// |rtc_use_h264| build flag is true (if false, this function does nothing).
// This function should only be called before or during WebRTC initialization
// and is not thread-safe.
void DisableRtcUseH264();
class H264Encoder : public VideoEncoder {
public:
static H264Encoder* Create(const cricket::VideoCodec& codec);
// If H.264 is supported (any implementation).
static bool IsSupported();
~H264Encoder() override {}
};
class H264Decoder : public VideoDecoder {
public:
static H264Decoder* Create();
static bool IsSupported();
~H264Decoder() override {}
};
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_H_

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/*
* 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.
*/
// This file contains codec dependent definitions that are needed in
// order to compile the WebRTC codebase, even if this codec is not used.
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_GLOBALS_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_GLOBALS_H_
namespace webrtc {
// The packetization types that we support: single, aggregated, and fragmented.
enum H264PacketizationTypes {
kH264SingleNalu, // This packet contains a single NAL unit.
kH264StapA, // This packet contains STAP-A (single time
// aggregation) packets. If this packet has an
// associated NAL unit type, it'll be for the
// first such aggregated packet.
kH264FuA, // This packet contains a FU-A (fragmentation
// unit) packet, meaning it is a part of a frame
// that was too large to fit into a single packet.
};
// Packetization modes are defined in RFC 6184 section 6
// Due to the structure containing this being initialized with zeroes
// in some places, and mode 1 being default, mode 1 needs to have the value
// zero. https://crbug.com/webrtc/6803
enum class H264PacketizationMode {
NonInterleaved = 0, // Mode 1 - STAP-A, FU-A is allowed
SingleNalUnit // Mode 0 - only single NALU allowed
};
// This function is declared inline because it is not clear which
// .cc file it should belong to.
// TODO(hta): Refactor. https://bugs.webrtc.org/6842
inline std::ostream& operator<<(std::ostream& stream,
H264PacketizationMode mode) {
switch (mode) {
case H264PacketizationMode::NonInterleaved:
stream << "NonInterleaved";
break;
case H264PacketizationMode::SingleNalUnit:
stream << "SingleNalUnit";
break;
}
return stream;
}
struct NaluInfo {
uint8_t type;
int sps_id;
int pps_id;
};
const size_t kMaxNalusPerPacket = 10;
struct RTPVideoHeaderH264 {
// The NAL unit type. If this is a header for a
// fragmented packet, it's the NAL unit type of
// the original data. If this is the header for an
// aggregated packet, it's the NAL unit type of
// the first NAL unit in the packet.
uint8_t nalu_type;
// The packetization type of this buffer - single, aggregated or fragmented.
H264PacketizationTypes packetization_type;
NaluInfo nalus[kMaxNalusPerPacket];
size_t nalus_length;
// The packetization mode of this transport. Packetization mode
// determines which packetization types are allowed when packetizing.
H264PacketizationMode packetization_mode;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_GLOBALS_H_

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/*
* Copyright (c) 2017 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/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
#include "webrtc/modules/video_coding/codecs/test/video_codec_test.h"
namespace webrtc {
class TestH264Impl : public VideoCodecTest {
protected:
VideoEncoder* CreateEncoder() override {
return H264Encoder::Create(cricket::VideoCodec(cricket::kH264CodecName));
}
VideoDecoder* CreateDecoder() override { return H264Decoder::Create(); }
VideoCodec codec_settings() override {
VideoCodec codec_inst;
codec_inst.codecType = webrtc::kVideoCodecH264;
// If frame dropping is false, we get a warning that bitrate can't
// be controlled for RC_QUALITY_MODE; RC_BITRATE_MODE and RC_TIMESTAMP_MODE
codec_inst.H264()->frameDroppingOn = true;
return codec_inst;
}
};
#ifdef WEBRTC_USE_H264
#define MAYBE_EncodeDecode EncodeDecode
#define MAYBE_DecodedQpEqualsEncodedQp DecodedQpEqualsEncodedQp
#else
#define MAYBE_EncodeDecode DISABLED_EncodeDecode
#define MAYBE_DecodedQpEqualsEncodedQp DISABLED_DecodedQpEqualsEncodedQp
#endif
TEST_F(TestH264Impl, MAYBE_EncodeDecode) {
EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
encoder_->Encode(*input_frame_, nullptr, nullptr));
EncodedImage encoded_frame;
CodecSpecificInfo codec_specific_info;
ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
// First frame should be a key frame.
encoded_frame._frameType = kVideoFrameKey;
EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
decoder_->Decode(encoded_frame, false, nullptr));
std::unique_ptr<VideoFrame> decoded_frame;
rtc::Optional<uint8_t> decoded_qp;
ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
ASSERT_TRUE(decoded_frame);
EXPECT_GT(I420PSNR(input_frame_.get(), decoded_frame.get()), 36);
}
TEST_F(TestH264Impl, MAYBE_DecodedQpEqualsEncodedQp) {
EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
encoder_->Encode(*input_frame_, nullptr, nullptr));
EncodedImage encoded_frame;
CodecSpecificInfo codec_specific_info;
ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
// First frame should be a key frame.
encoded_frame._frameType = kVideoFrameKey;
EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
decoder_->Decode(encoded_frame, false, nullptr));
std::unique_ptr<VideoFrame> decoded_frame;
rtc::Optional<uint8_t> decoded_qp;
ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
ASSERT_TRUE(decoded_frame);
ASSERT_TRUE(decoded_qp);
EXPECT_EQ(encoded_frame.qp_, *decoded_qp);
}
} // namespace webrtc