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Integrate QualityScaler into VideoToolboxEncoder.

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Improves visual quality and framerate during low bitrates by
dropping resolution.

BUG=webrtc:5678
R=tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/2001153003 .

Cr-Commit-Position: refs/heads/master@{#12867}
This commit is contained in:
Peter Boström
2016-05-24 12:21:58 +02:00
parent cc1543abf3
commit e23e760643
2 changed files with 58 additions and 12 deletions
Download Patch File Download Diff File Expand all files Collapse all files

61
webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.cc
View File

@ -225,10 +225,18 @@ int H264VideoToolboxEncoder::InitEncode(const VideoCodec* codec_settings,
size_t max_payload_size) {
RTC_DCHECK(codec_settings);
RTC_DCHECK_EQ(codec_settings->codecType, kVideoCodecH264);
// TODO(tkchin): We may need to enforce width/height dimension restrictions
// to match what the encoder supports.
width_ = codec_settings->width;
height_ = codec_settings->height;
{
rtc::CritScope lock(&quality_scaler_crit_);
quality_scaler_.Init(QualityScaler::kLowH264QpThreshold,
QualityScaler::kBadH264QpThreshold,
codec_settings->startBitrate, codec_settings->width,
codec_settings->height, codec_settings->maxFramerate);
QualityScaler::Resolution res = quality_scaler_.GetScaledResolution();
// TODO(tkchin): We may need to enforce width/height dimension restrictions
// to match what the encoder supports.
width_ = res.width;
height_ = res.height;
}
// We can only set average bitrate on the HW encoder.
target_bitrate_bps_ = codec_settings->startBitrate;
bitrate_adjuster_.SetTargetBitrateBps(target_bitrate_bps_);
@ -239,17 +247,18 @@ int H264VideoToolboxEncoder::InitEncode(const VideoCodec* codec_settings,
return ResetCompressionSession();
}
const VideoFrame& H264VideoToolboxEncoder::GetScaledFrameOnEncode(
const VideoFrame& frame) {
rtc::CritScope lock(&quality_scaler_crit_);
quality_scaler_.OnEncodeFrame(frame);
return quality_scaler_.GetScaledFrame(frame);
}
int H264VideoToolboxEncoder::Encode(
const VideoFrame& input_image,
const VideoFrame& frame,
const CodecSpecificInfo* codec_specific_info,
const std::vector<FrameType>* frame_types) {
if (input_image.IsZeroSize()) {
// It's possible to get zero sizes as a signal to produce keyframes (this
// happens for internal sources). But this shouldn't happen in
// webrtcvideoengine2.
RTC_NOTREACHED();
return WEBRTC_VIDEO_CODEC_OK;
}
RTC_DCHECK(!frame.IsZeroSize());
if (!callback_ || !compression_session_) {
return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
}
@ -261,6 +270,16 @@ int H264VideoToolboxEncoder::Encode(
}
#endif
bool is_keyframe_required = false;
const VideoFrame& input_image = GetScaledFrameOnEncode(frame);
if (input_image.width() != width_ || input_image.height() != height_) {
width_ = input_image.width();
height_ = input_image.height();
int ret = ResetCompressionSession();
if (ret < 0)
return ret;
}
// Get a pixel buffer from the pool and copy frame data over.
CVPixelBufferPoolRef pixel_buffer_pool =
VTCompressionSessionGetPixelBufferPool(compression_session_);
@ -345,6 +364,11 @@ int H264VideoToolboxEncoder::RegisterEncodeCompleteCallback(
return WEBRTC_VIDEO_CODEC_OK;
}
void H264VideoToolboxEncoder::OnDroppedFrame() {
rtc::CritScope lock(&quality_scaler_crit_);
quality_scaler_.ReportDroppedFrame();
}
int H264VideoToolboxEncoder::SetChannelParameters(uint32_t packet_loss,
int64_t rtt) {
// Encoder doesn't know anything about packet loss or rtt so just return.
@ -357,6 +381,9 @@ int H264VideoToolboxEncoder::SetRates(uint32_t new_bitrate_kbit,
bitrate_adjuster_.SetTargetBitrateBps(target_bitrate_bps_);
SetBitrateBps(bitrate_adjuster_.GetAdjustedBitrateBps());
rtc::CritScope lock(&quality_scaler_crit_);
quality_scaler_.ReportFramerate(frame_rate);
return WEBRTC_VIDEO_CODEC_OK;
}
@ -491,6 +518,9 @@ void H264VideoToolboxEncoder::OnEncodedFrame(
}
if (info_flags & kVTEncodeInfo_FrameDropped) {
LOG(LS_INFO) << "H264 encode dropped frame.";
rtc::CritScope lock(&quality_scaler_crit_);
quality_scaler_.ReportDroppedFrame();
return;
}
bool is_keyframe = false;
@ -526,6 +556,13 @@ void H264VideoToolboxEncoder::OnEncodedFrame(
frame._timeStamp = timestamp;
frame.rotation_ = rotation;
h264_bitstream_parser_.ParseBitstream(buffer->data(), buffer->size());
int qp;
if (h264_bitstream_parser_.GetLastSliceQp(&qp)) {
rtc::CritScope lock(&quality_scaler_crit_);
quality_scaler_.ReportQP(qp);
}
int result = callback_->Encoded(frame, &codec_specific_info, header.get());
if (result != 0) {
LOG(LS_ERROR) << "Encode callback failed: " << result;

9
webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.h
View File

@ -12,9 +12,12 @@
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_VIDEO_TOOLBOX_ENCODER_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_VIDEO_TOOLBOX_ENCODER_H_
#include "webrtc/base/criticalsection.h"
#include "webrtc/common_video/include/bitrate_adjuster.h"
#include "webrtc/common_video/rotation.h"
#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
#include "webrtc/modules/video_coding/utility/h264_bitstream_parser.h"
#include "webrtc/modules/video_coding/utility/quality_scaler.h"
#if defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED)
@ -44,6 +47,7 @@ class H264VideoToolboxEncoder : public H264Encoder {
int RegisterEncodeCompleteCallback(EncodedImageCallback* callback) override;
void OnDroppedFrame() override;
int SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
int SetRates(uint32_t new_bitrate_kbit, uint32_t frame_rate) override;
@ -66,6 +70,7 @@ class H264VideoToolboxEncoder : public H264Encoder {
int ResetCompressionSession();
void ConfigureCompressionSession();
void DestroyCompressionSession();
const VideoFrame& GetScaledFrameOnEncode(const VideoFrame& frame);
void SetBitrateBps(uint32_t bitrate_bps);
void SetEncoderBitrateBps(uint32_t bitrate_bps);
@ -76,6 +81,10 @@ class H264VideoToolboxEncoder : public H264Encoder {
uint32_t encoder_bitrate_bps_;
int32_t width_;
int32_t height_;
rtc::CriticalSection quality_scaler_crit_;
QualityScaler quality_scaler_ GUARDED_BY(quality_scaler_crit_);
H264BitstreamParser h264_bitstream_parser_;
}; // H264VideoToolboxEncoder
} // namespace webrtc