H.264 video codec support using OpenH264/FFmpeg

webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc

+/*
+ *  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>
+
+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/base/checks.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/logging.h"
+
+namespace webrtc {
+
+namespace {
+
+static const AVPixelFormat kPixelFormat = AV_PIX_FMT_YUV420P;
+static const size_t kYPlaneIndex = 0;
+static const size_t kUPlaneIndex = 1;
+static const size_t kVPlaneIndex = 2;
+
+#if !defined(WEBRTC_CHROMIUM_BUILD)
+
+static bool ffmpeg_initialized = false;
+
+// Called by FFmpeg to do mutex operations if init using InitializeFFmpeg.
+static int LockManagerOperation(void** lock, AVLockOp op)
+    EXCLUSIVE_LOCK_FUNCTION() UNLOCK_FUNCTION() {
+  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;
+  }
+  return 1;
+}
+
+// TODO(hbos): Assumed to be called on a single thread. Should DCHECK that
+// InitializeFFmpeg is only called on one thread or make it thread safe.
+static bool InitializeFFmpeg() {
+  if (!ffmpeg_initialized) {
+    if (av_lockmgr_register(LockManagerOperation) < 0) {
+      LOG(LS_ERROR) << "av_lockmgr_register failed.";
+      return false;
+    }
+    av_register_all();
+    ffmpeg_initialized = true;
+  }
+  return true;
+}
+
+#endif  // !defined(WEBRTC_CHROMIUM_BUILD)
+
+static int NumberOfThreads(int width, int height, int number_of_cores) {
+  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) {
+    // 3 threads for 1080p.
+    return 3;

[stefan-webrtc, 2016/01/08 09:25:04]

Comment on the same line as above? Same below.

[hbos, 2016/01/11 16:21:53]

Done.

+  } else if (width * height > 640 * 480 && number_of_cores >= 3) {
+    // 2 threads for qHD/HD.
+    return 2;
+  } else {
+    // 1 thread for VGA or less.
+    return 1;
+  }
+}
+
+// Called by FFmpeg when it is done with a frame buffer, see AVGetBuffer2.
+static void AVFreeBuffer2(void* opaque, uint8_t* data) {
+  VideoFrame* video_frame = static_cast<VideoFrame*>(opaque);
+  delete video_frame;
+}
+
+// Called by FFmpeg when it needs a frame buffer to store decoded frames in.
+// The VideoFrames returned by FFmpeg at Decode originate from here. They are
+// reference counted and freed by FFmpeg using AVFreeBuffer2.
+// TODO(hbos): Use a frame pool for better performance instead of create/free.
+// Could be owned by decoder, static_cast<H264DecoderImpl*>(context->opaque).
+static int AVGetBuffer2(AVCodecContext* context, AVFrame* av_frame, int flags) {
+  RTC_CHECK_EQ(context->pix_fmt, kPixelFormat);  // Same as in InitDecode.
+  // width/height and coded_width/coded_height can be different due to cropping
+  // or |lowres|.
+  int width = std::max(context->width, context->coded_width);
+  int height = std::max(context->height, context->coded_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);
+
+  RTC_CHECK_GE(width, 0);
+  RTC_CHECK_GE(height, 0);
+  int ret = av_image_check_size(width, height, 0, nullptr);
+  if (ret < 0) {
+    LOG(LS_ERROR) << "Invalid picture size " << width << "x" << height;
+    return ret;
+  }
+
+  // The video frame is stored in |video_frame|. |av_frame| is FFmpeg's version
+  // of a video frame and will be set up to reference |video_frame|'s buffers.
+  VideoFrame* video_frame = new VideoFrame();
+  int stride_y = width;
+  int stride_uv = (width + 1) / 2;
+  RTC_CHECK_EQ(0, video_frame->CreateEmptyFrame(
+      width, height, stride_y, stride_uv, stride_uv));
+  size_t total_size = video_frame->allocated_size(kYPlane) +
+                      video_frame->allocated_size(kUPlane) +
+                      video_frame->allocated_size(kVPlane);
+  RTC_DCHECK_EQ(total_size, static_cast<size_t>(stride_y * height +
+                (stride_uv + stride_uv) * ((height + 1) / 2)));
+  // FFmpeg note: "Each data plane must be aligned to the maximum required by
+  // the target CPU." See get_buffer2.
+  // TODO(hbos): Memory alignment on a per-plane basis. CreateEmptyFrame only
+  // guarantees that the buffer of all planes is memory aligned, not each
+  // individual plane. Or does "data plane" here refer to one data[] entry or
+  // one allocation?
+
+  // FFmpeg expects the initial allocation to be zero-initialized according to
+  // http://crbug.com/390941.
+  // Using a single |av_frame->buf| - YUV is required to be a continuous blob of
+  // memory. We can zero-initialize with one memset operation for all planes.
+  RTC_DCHECK_EQ(video_frame->buffer(kUPlane),
+      video_frame->buffer(kYPlane) + video_frame->allocated_size(kYPlane));
+  RTC_DCHECK_EQ(video_frame->buffer(kVPlane),
+      video_frame->buffer(kUPlane) + video_frame->allocated_size(kUPlane));
+  memset(video_frame->buffer(kYPlane), 0, total_size);
+
+  RTC_DCHECK_EQ(av_frame->width, width);
+  RTC_DCHECK_EQ(av_frame->height, height);
+  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] = video_frame->buffer(kYPlane);
+  av_frame->linesize[kYPlaneIndex] = video_frame->stride(kYPlane);
+  av_frame->data[kUPlaneIndex] = video_frame->buffer(kUPlane);
+  av_frame->linesize[kUPlaneIndex] = video_frame->stride(kUPlane);
+  av_frame->data[kVPlaneIndex] = video_frame->buffer(kVPlane);
+  av_frame->linesize[kVPlaneIndex] = video_frame->stride(kVPlane);
+  RTC_DCHECK_EQ(av_frame->extended_data, av_frame->data);
+
+  av_frame->buf[0] = av_buffer_create(av_frame->data[kYPlaneIndex],
+                                      total_size,
+                                      AVFreeBuffer2,
+                                      static_cast<void*>(video_frame),
+                                      0);
+  RTC_CHECK(av_frame->buf[0]);
+  return 0;
+}
+
+}  // namespace
+
+H264DecoderImpl::H264DecoderImpl()
+    : decoded_image_callback_(nullptr) {
+}
+
+H264DecoderImpl::~H264DecoderImpl() {
+  Release();
+}
+
+int32_t H264DecoderImpl::InitDecode(const VideoCodec* codec_settings,
+                                    int32_t number_of_cores) {
+  if (codec_settings &&
+      codec_settings->codecType != kVideoCodecH264) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+
+  // In Chromium FFmpeg will be initialized outside of WebRTC and we should not
+  // attempt to do so ourselves or it will be initialized twice.
+  // TODO(hbos): Put behind a different flag in case non-chromium project wants
+  // to initialize externally.
+#if !defined(WEBRTC_CHROMIUM_BUILD)
+  // Make sure FFmpeg has been initialized.
+  InitializeFFmpeg();
+#endif
+
+  // Release necessary in case of re-initializing.
+  int32_t ret = Release();
+  if (ret != WEBRTC_VIDEO_CODEC_OK)
+    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;
+
+  av_context_->thread_count = NumberOfThreads(av_context_->coded_width,
+                                              av_context_->coded_height,
+                                              number_of_cores);
+  av_context_->thread_type = FF_THREAD_SLICE;
+
+  // FFmpeg will get video buffers from our AVGetBuffer2, memory managed by us.
+  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();
+    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();
+    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::Reset() {
+  if (!IsInitialized())
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  InitDecode(nullptr, 1);
+  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())
+    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()";
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  }
+  if (!input_image._buffer || !input_image._length)
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  if (codec_specific_info &&
+      codec_specific_info->codecType != kVideoCodecH264) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+
+  AVPacket packet;
+  av_init_packet(&packet);
+  // TODO(hbos): "The input buffer must be AV_INPUT_BUFFER_PADDING_SIZE larger
+  // than the actual read bytes because some optimized bitstream readers read 32
+  // or 64 bits at once and could read over the end." See avcodec_decode_video2.
+  // - Is this an issue? Do we have to make sure EncodedImage is allocated with
+  // additional bytes or do we have to do an otherwise unnecessary copy? Might
+  // only be a problem with non-mul-32 frame widths?
+  // ("If the first 23 bits of the additional bytes are not 0, then damaged MPEG
+  // bitstreams could cause overread and segfault.")
+  packet.data = input_image._buffer;
+  packet.size = 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;
+    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.";
+    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_CHECK_EQ(av_frame_->data[kYPlane], video_frame->buffer(kYPlane));
+  RTC_CHECK_EQ(av_frame_->data[kUPlane], video_frame->buffer(kUPlane));
+  RTC_CHECK_EQ(av_frame_->data[kVPlane], video_frame->buffer(kVPlane));
+  video_frame->set_timestamp(input_image._timeStamp);
+
+  // Return decoded frame.
+  int32_t ret = decoded_image_callback_->Decoded(*video_frame);
+  // Stop referencing it, possibly freeing |video_frame|.
+  av_frame_unref(av_frame_.get());
+  video_frame = nullptr;
+
+  if (ret) {
+    LOG(LS_WARNING) << "DecodedImageCallback::Decoded returned " << ret;
+    return ret;
+  }
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+bool H264DecoderImpl::IsInitialized() const {
+  return av_context_ != nullptr;
+}
+
+}  // namespace webrtc