webrtc_m130/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc

Ignoring revisions in .git-blame-ignore-revs. Click here to bypass and see the normal blame view.

979 lines
37 KiB
C++
Raw Normal View History

/*
Revert "Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8." This reverts commit 07efe436c9002e139845f62486e3ee4e29f0d85b. Reason for revert: Breaks downstream project. cricket::GetSimulcastConfig method signature has been updated. I think you can get away with a default value for temporal_layers_supported (and then you can remove it after a few days when projects will be updated). Original change's description: > Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8. > > * Move SimulcastEncoderAdapter out under modules/video_coding > * Move SimulcastRateAllocator back out to modules/video_coding/utility > * Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility > * Move any VP8 specific code - such as temporal layer bitrate budgeting - > under codec type dependent conditionals. > * Plumb the simulcast index for H264 in the codec specific and RTP format data structures. > > Bug: webrtc:5840 > Change-Id: Ieced8a00e38f273c1a6cfd0f5431a87d07b8f44e > Reviewed-on: https://webrtc-review.googlesource.com/64100 > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23705} TBR=sprang@webrtc.org,stefan@webrtc.org,mflodman@webrtc.org,hta@webrtc.org,sergio.garcia.murillo@gmail.com,titovartem@webrtc.org,agouaillard@gmail.com Change-Id: Ic9d3b1eeaf195bb5ec2063954421f5e77866d663 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:5840 Reviewed-on: https://webrtc-review.googlesource.com/84760 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23710}
2018-06-21 13:41:01 +00:00
* Copyright (c) 2018 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 <algorithm>
#include <string>
#include <vector>
#include "common_video/libyuv/include/webrtc_libyuv.h"
#include "modules/video_coding/codecs/vp8/libvpx_vp8_encoder.h"
Revert "Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8." This reverts commit 07efe436c9002e139845f62486e3ee4e29f0d85b. Reason for revert: Breaks downstream project. cricket::GetSimulcastConfig method signature has been updated. I think you can get away with a default value for temporal_layers_supported (and then you can remove it after a few days when projects will be updated). Original change's description: > Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8. > > * Move SimulcastEncoderAdapter out under modules/video_coding > * Move SimulcastRateAllocator back out to modules/video_coding/utility > * Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility > * Move any VP8 specific code - such as temporal layer bitrate budgeting - > under codec type dependent conditionals. > * Plumb the simulcast index for H264 in the codec specific and RTP format data structures. > > Bug: webrtc:5840 > Change-Id: Ieced8a00e38f273c1a6cfd0f5431a87d07b8f44e > Reviewed-on: https://webrtc-review.googlesource.com/64100 > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23705} TBR=sprang@webrtc.org,stefan@webrtc.org,mflodman@webrtc.org,hta@webrtc.org,sergio.garcia.murillo@gmail.com,titovartem@webrtc.org,agouaillard@gmail.com Change-Id: Ic9d3b1eeaf195bb5ec2063954421f5e77866d663 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:5840 Reviewed-on: https://webrtc-review.googlesource.com/84760 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23710}
2018-06-21 13:41:01 +00:00
#include "modules/video_coding/codecs/vp8/simulcast_rate_allocator.h"
#include "rtc_base/checks.h"
Reland "Update internal SW codecs to return unique_ptrs" This reverts commit 34c8e6bce8af0c31f2b0b31d691a6a931fa3cb7b. Reason for revert: Fix Android compilation Original change's description: > Revert "Update internal SW codecs to return unique_ptrs" > > This reverts commit 4fe6adc06a8524ac25f85260bfe392eb31dae6b4. > > Reason for revert: Breaks android compile. > > Original change's description: > > Update internal SW codecs to return unique_ptrs > > > > TBR=stefan@webrtc.org > > > > Bug: webrtc:7925 > > Change-Id: I84239b071a2608d928f09b06809090eec5eafb14 > > Reviewed-on: https://webrtc-review.googlesource.com/21165 > > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20650} > > TBR=magjed@webrtc.org,sprang@webrtc.org,stefan@webrtc.org > > Change-Id: If33c3a0ee0dfce63d105558a2897a472f0633306 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:7925 > Reviewed-on: https://webrtc-review.googlesource.com/22540 > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20652} TBR=magjed@webrtc.org,sprang@webrtc.org,stefan@webrtc.org Change-Id: Ic8551af4687e927c9b605060155abdd5bc6208b2 Bug: webrtc:7925 Reviewed-on: https://webrtc-review.googlesource.com/22541 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20655}
2017-11-13 14:10:02 +01:00
#include "rtc_base/ptr_util.h"
#include "rtc_base/timeutils.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/field_trial.h"
#include "third_party/libyuv/include/libyuv/convert.h"
#include "third_party/libyuv/include/libyuv/scale.h"
namespace webrtc {
namespace {
const char kVp8GfBoostFieldTrial[] = "WebRTC-VP8-GfBoost";
// QP is obtained from VP8-bitstream for HW, so the QP corresponds to the
// bitstream range of [0, 127] and not the user-level range of [0,63].
constexpr int kLowVp8QpThreshold = 29;
constexpr int kHighVp8QpThreshold = 95;
constexpr int kTokenPartitions = VP8_ONE_TOKENPARTITION;
constexpr uint32_t kVp832ByteAlign = 32u;
// VP8 denoiser states.
enum denoiserState {
kDenoiserOff,
kDenoiserOnYOnly,
kDenoiserOnYUV,
kDenoiserOnYUVAggressive,
// Adaptive mode defaults to kDenoiserOnYUV on key frame, but may switch
// to kDenoiserOnYUVAggressive based on a computed noise metric.
kDenoiserOnAdaptive
};
// Greatest common divisior
Revert "Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8." This reverts commit 07efe436c9002e139845f62486e3ee4e29f0d85b. Reason for revert: Breaks downstream project. cricket::GetSimulcastConfig method signature has been updated. I think you can get away with a default value for temporal_layers_supported (and then you can remove it after a few days when projects will be updated). Original change's description: > Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8. > > * Move SimulcastEncoderAdapter out under modules/video_coding > * Move SimulcastRateAllocator back out to modules/video_coding/utility > * Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility > * Move any VP8 specific code - such as temporal layer bitrate budgeting - > under codec type dependent conditionals. > * Plumb the simulcast index for H264 in the codec specific and RTP format data structures. > > Bug: webrtc:5840 > Change-Id: Ieced8a00e38f273c1a6cfd0f5431a87d07b8f44e > Reviewed-on: https://webrtc-review.googlesource.com/64100 > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23705} TBR=sprang@webrtc.org,stefan@webrtc.org,mflodman@webrtc.org,hta@webrtc.org,sergio.garcia.murillo@gmail.com,titovartem@webrtc.org,agouaillard@gmail.com Change-Id: Ic9d3b1eeaf195bb5ec2063954421f5e77866d663 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:5840 Reviewed-on: https://webrtc-review.googlesource.com/84760 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23710}
2018-06-21 13:41:01 +00:00
int GCD(int a, int b) {
int c = a % b;
while (c != 0) {
a = b;
b = c;
c = a % b;
}
return b;
}
Revert "Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8." This reverts commit 07efe436c9002e139845f62486e3ee4e29f0d85b. Reason for revert: Breaks downstream project. cricket::GetSimulcastConfig method signature has been updated. I think you can get away with a default value for temporal_layers_supported (and then you can remove it after a few days when projects will be updated). Original change's description: > Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8. > > * Move SimulcastEncoderAdapter out under modules/video_coding > * Move SimulcastRateAllocator back out to modules/video_coding/utility > * Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility > * Move any VP8 specific code - such as temporal layer bitrate budgeting - > under codec type dependent conditionals. > * Plumb the simulcast index for H264 in the codec specific and RTP format data structures. > > Bug: webrtc:5840 > Change-Id: Ieced8a00e38f273c1a6cfd0f5431a87d07b8f44e > Reviewed-on: https://webrtc-review.googlesource.com/64100 > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23705} TBR=sprang@webrtc.org,stefan@webrtc.org,mflodman@webrtc.org,hta@webrtc.org,sergio.garcia.murillo@gmail.com,titovartem@webrtc.org,agouaillard@gmail.com Change-Id: Ic9d3b1eeaf195bb5ec2063954421f5e77866d663 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:5840 Reviewed-on: https://webrtc-review.googlesource.com/84760 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23710}
2018-06-21 13:41:01 +00:00
uint32_t SumStreamMaxBitrate(int streams, const VideoCodec& codec) {
uint32_t bitrate_sum = 0;
for (int i = 0; i < streams; ++i) {
bitrate_sum += codec.simulcastStream[i].maxBitrate;
}
return bitrate_sum;
}
int NumberOfStreams(const VideoCodec& codec) {
int streams =
codec.numberOfSimulcastStreams < 1 ? 1 : codec.numberOfSimulcastStreams;
uint32_t simulcast_max_bitrate = SumStreamMaxBitrate(streams, codec);
if (simulcast_max_bitrate == 0) {
streams = 1;
}
return streams;
}
bool ValidSimulcastResolutions(const VideoCodec& codec, int num_streams) {
if (codec.width != codec.simulcastStream[num_streams - 1].width ||
codec.height != codec.simulcastStream[num_streams - 1].height) {
return false;
}
for (int i = 0; i < num_streams; ++i) {
if (codec.width * codec.simulcastStream[i].height !=
codec.height * codec.simulcastStream[i].width) {
return false;
}
}
for (int i = 1; i < num_streams; ++i) {
if (codec.simulcastStream[i].width !=
codec.simulcastStream[i - 1].width * 2) {
return false;
}
}
return true;
}
bool ValidSimulcastTemporalLayers(const VideoCodec& codec, int num_streams) {
for (int i = 0; i < num_streams - 1; ++i) {
if (codec.simulcastStream[i].numberOfTemporalLayers !=
codec.simulcastStream[i + 1].numberOfTemporalLayers)
return false;
}
return true;
}
bool GetGfBoostPercentageFromFieldTrialGroup(int* boost_percentage) {
std::string group = webrtc::field_trial::FindFullName(kVp8GfBoostFieldTrial);
if (group.empty())
return false;
if (sscanf(group.c_str(), "Enabled-%d", boost_percentage) != 1)
return false;
if (*boost_percentage < 0 || *boost_percentage > 100)
return false;
return true;
}
static_assert(
VP8_TS_MAX_PERIODICITY == VPX_TS_MAX_PERIODICITY,
"VP8_TS_MAX_PERIODICITY must be kept in sync with the constant in libvpx.");
static_assert(
VP8_TS_MAX_LAYERS == VPX_TS_MAX_LAYERS,
"VP8_TS_MAX_LAYERS must be kept in sync with the constant in libvpx.");
static Vp8EncoderConfig GetEncoderConfig(vpx_codec_enc_cfg* vpx_config) {
Vp8EncoderConfig config;
config.ts_number_layers = vpx_config->ts_number_layers;
memcpy(config.ts_target_bitrate, vpx_config->ts_target_bitrate,
sizeof(unsigned int) * VP8_TS_MAX_LAYERS);
memcpy(config.ts_rate_decimator, vpx_config->ts_rate_decimator,
sizeof(unsigned int) * VP8_TS_MAX_LAYERS);
config.ts_periodicity = vpx_config->ts_periodicity;
memcpy(config.ts_layer_id, vpx_config->ts_layer_id,
sizeof(unsigned int) * VP8_TS_MAX_PERIODICITY);
config.rc_target_bitrate = vpx_config->rc_target_bitrate;
config.rc_min_quantizer = vpx_config->rc_min_quantizer;
config.rc_max_quantizer = vpx_config->rc_max_quantizer;
return config;
}
static void FillInEncoderConfig(vpx_codec_enc_cfg* vpx_config,
const Vp8EncoderConfig& config) {
vpx_config->ts_number_layers = config.ts_number_layers;
memcpy(vpx_config->ts_target_bitrate, config.ts_target_bitrate,
sizeof(unsigned int) * VP8_TS_MAX_LAYERS);
memcpy(vpx_config->ts_rate_decimator, config.ts_rate_decimator,
sizeof(unsigned int) * VP8_TS_MAX_LAYERS);
vpx_config->ts_periodicity = config.ts_periodicity;
memcpy(vpx_config->ts_layer_id, config.ts_layer_id,
sizeof(unsigned int) * VP8_TS_MAX_PERIODICITY);
vpx_config->rc_target_bitrate = config.rc_target_bitrate;
vpx_config->rc_min_quantizer = config.rc_min_quantizer;
vpx_config->rc_max_quantizer = config.rc_max_quantizer;
}
bool UpdateVpxConfiguration(TemporalLayers* temporal_layers,
vpx_codec_enc_cfg_t* cfg) {
Vp8EncoderConfig config = GetEncoderConfig(cfg);
const bool res = temporal_layers->UpdateConfiguration(&config);
if (res)
FillInEncoderConfig(cfg, config);
return res;
}
} // namespace
Reland "Update internal SW codecs to return unique_ptrs" This reverts commit 34c8e6bce8af0c31f2b0b31d691a6a931fa3cb7b. Reason for revert: Fix Android compilation Original change's description: > Revert "Update internal SW codecs to return unique_ptrs" > > This reverts commit 4fe6adc06a8524ac25f85260bfe392eb31dae6b4. > > Reason for revert: Breaks android compile. > > Original change's description: > > Update internal SW codecs to return unique_ptrs > > > > TBR=stefan@webrtc.org > > > > Bug: webrtc:7925 > > Change-Id: I84239b071a2608d928f09b06809090eec5eafb14 > > Reviewed-on: https://webrtc-review.googlesource.com/21165 > > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20650} > > TBR=magjed@webrtc.org,sprang@webrtc.org,stefan@webrtc.org > > Change-Id: If33c3a0ee0dfce63d105558a2897a472f0633306 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:7925 > Reviewed-on: https://webrtc-review.googlesource.com/22540 > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20652} TBR=magjed@webrtc.org,sprang@webrtc.org,stefan@webrtc.org Change-Id: Ic8551af4687e927c9b605060155abdd5bc6208b2 Bug: webrtc:7925 Reviewed-on: https://webrtc-review.googlesource.com/22541 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20655}
2017-11-13 14:10:02 +01:00
std::unique_ptr<VP8Encoder> VP8Encoder::Create() {
return rtc::MakeUnique<LibvpxVp8Encoder>();
}
vpx_enc_frame_flags_t LibvpxVp8Encoder::EncodeFlags(
const TemporalLayers::FrameConfig& references) {
RTC_DCHECK(!references.drop_frame);
vpx_enc_frame_flags_t flags = 0;
if ((references.last_buffer_flags & TemporalLayers::kReference) == 0)
flags |= VP8_EFLAG_NO_REF_LAST;
if ((references.last_buffer_flags & TemporalLayers::kUpdate) == 0)
flags |= VP8_EFLAG_NO_UPD_LAST;
if ((references.golden_buffer_flags & TemporalLayers::kReference) == 0)
flags |= VP8_EFLAG_NO_REF_GF;
if ((references.golden_buffer_flags & TemporalLayers::kUpdate) == 0)
flags |= VP8_EFLAG_NO_UPD_GF;
if ((references.arf_buffer_flags & TemporalLayers::kReference) == 0)
flags |= VP8_EFLAG_NO_REF_ARF;
if ((references.arf_buffer_flags & TemporalLayers::kUpdate) == 0)
flags |= VP8_EFLAG_NO_UPD_ARF;
if (references.freeze_entropy)
flags |= VP8_EFLAG_NO_UPD_ENTROPY;
return flags;
}
LibvpxVp8Encoder::LibvpxVp8Encoder()
: use_gf_boost_(webrtc::field_trial::IsEnabled(kVp8GfBoostFieldTrial)),
encoded_complete_callback_(nullptr),
inited_(false),
timestamp_(0),
qp_max_(56), // Setting for max quantizer.
cpu_speed_default_(-6),
number_of_cores_(0),
rc_max_intra_target_(0),
key_frame_request_(kMaxSimulcastStreams, false) {
temporal_layers_.reserve(kMaxSimulcastStreams);
temporal_layers_checkers_.reserve(kMaxSimulcastStreams);
raw_images_.reserve(kMaxSimulcastStreams);
encoded_images_.reserve(kMaxSimulcastStreams);
send_stream_.reserve(kMaxSimulcastStreams);
cpu_speed_.assign(kMaxSimulcastStreams, cpu_speed_default_);
encoders_.reserve(kMaxSimulcastStreams);
configurations_.reserve(kMaxSimulcastStreams);
downsampling_factors_.reserve(kMaxSimulcastStreams);
}
LibvpxVp8Encoder::~LibvpxVp8Encoder() {
Release();
}
int LibvpxVp8Encoder::Release() {
int ret_val = WEBRTC_VIDEO_CODEC_OK;
while (!encoded_images_.empty()) {
EncodedImage& image = encoded_images_.back();
delete[] image._buffer;
encoded_images_.pop_back();
}
while (!encoders_.empty()) {
vpx_codec_ctx_t& encoder = encoders_.back();
if (inited_) {
if (vpx_codec_destroy(&encoder)) {
ret_val = WEBRTC_VIDEO_CODEC_MEMORY;
}
}
encoders_.pop_back();
}
configurations_.clear();
send_stream_.clear();
cpu_speed_.clear();
while (!raw_images_.empty()) {
vpx_img_free(&raw_images_.back());
raw_images_.pop_back();
}
temporal_layers_.clear();
temporal_layers_checkers_.clear();
inited_ = false;
return ret_val;
}
int LibvpxVp8Encoder::SetRateAllocation(const VideoBitrateAllocation& bitrate,
uint32_t new_framerate) {
Reland #2 of Issue 2434073003: Extract bitrate allocation ... This is yet another reland of https://codereview.webrtc.org/2434073003/ including two fixes: 1. SimulcastRateAllocator did not handle the screenshare settings properly for numSimulcastStreams = 1. Additional test case was added for that. 2. In VideoSender, when rate allocation is updated after setting a new VideoCodec config, only update the state of the EncoderParameters, but don't actually run SetRateAllocation on the encoder itself. This caused some problems upstreams. Please review only the changes after patch set 1. Original description: Extract bitrate allocation of spatial/temporal layers out of codec impl. This CL makes a number of intervowen changes: * Add BitrateAllocation struct, that contains a codec independent view of how the target bitrate is distributed over spatial and temporal layers. * Adds the BitrateAllocator interface, which takes a bitrate and frame rate and produces a BitrateAllocation. * A default (non layered) implementation is added, and SimulcastRateAllocator is extended to fully handle VP8 allocation. This includes capturing TemporalLayer instances created by the encoder. * ViEEncoder now owns both the bitrate allocator and the temporal layer factories for VP8. This allows allocation to happen fully outside of the encoder implementation. This refactoring will make it possible for ViEEncoder to signal the full picture of target bitrates to the RTCP module. BUG=webrtc:6301 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/2510583002 . Cr-Commit-Position: refs/heads/master@{#15105}
2016-11-16 16:41:30 +01:00
if (!inited_)
return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
Reland #2 of Issue 2434073003: Extract bitrate allocation ... This is yet another reland of https://codereview.webrtc.org/2434073003/ including two fixes: 1. SimulcastRateAllocator did not handle the screenshare settings properly for numSimulcastStreams = 1. Additional test case was added for that. 2. In VideoSender, when rate allocation is updated after setting a new VideoCodec config, only update the state of the EncoderParameters, but don't actually run SetRateAllocation on the encoder itself. This caused some problems upstreams. Please review only the changes after patch set 1. Original description: Extract bitrate allocation of spatial/temporal layers out of codec impl. This CL makes a number of intervowen changes: * Add BitrateAllocation struct, that contains a codec independent view of how the target bitrate is distributed over spatial and temporal layers. * Adds the BitrateAllocator interface, which takes a bitrate and frame rate and produces a BitrateAllocation. * A default (non layered) implementation is added, and SimulcastRateAllocator is extended to fully handle VP8 allocation. This includes capturing TemporalLayer instances created by the encoder. * ViEEncoder now owns both the bitrate allocator and the temporal layer factories for VP8. This allows allocation to happen fully outside of the encoder implementation. This refactoring will make it possible for ViEEncoder to signal the full picture of target bitrates to the RTCP module. BUG=webrtc:6301 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/2510583002 . Cr-Commit-Position: refs/heads/master@{#15105}
2016-11-16 16:41:30 +01:00
if (encoders_[0].err)
return WEBRTC_VIDEO_CODEC_ERROR;
Reland #2 of Issue 2434073003: Extract bitrate allocation ... This is yet another reland of https://codereview.webrtc.org/2434073003/ including two fixes: 1. SimulcastRateAllocator did not handle the screenshare settings properly for numSimulcastStreams = 1. Additional test case was added for that. 2. In VideoSender, when rate allocation is updated after setting a new VideoCodec config, only update the state of the EncoderParameters, but don't actually run SetRateAllocation on the encoder itself. This caused some problems upstreams. Please review only the changes after patch set 1. Original description: Extract bitrate allocation of spatial/temporal layers out of codec impl. This CL makes a number of intervowen changes: * Add BitrateAllocation struct, that contains a codec independent view of how the target bitrate is distributed over spatial and temporal layers. * Adds the BitrateAllocator interface, which takes a bitrate and frame rate and produces a BitrateAllocation. * A default (non layered) implementation is added, and SimulcastRateAllocator is extended to fully handle VP8 allocation. This includes capturing TemporalLayer instances created by the encoder. * ViEEncoder now owns both the bitrate allocator and the temporal layer factories for VP8. This allows allocation to happen fully outside of the encoder implementation. This refactoring will make it possible for ViEEncoder to signal the full picture of target bitrates to the RTCP module. BUG=webrtc:6301 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/2510583002 . Cr-Commit-Position: refs/heads/master@{#15105}
2016-11-16 16:41:30 +01:00
if (new_framerate < 1)
return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
Reland #2 of Issue 2434073003: Extract bitrate allocation ... This is yet another reland of https://codereview.webrtc.org/2434073003/ including two fixes: 1. SimulcastRateAllocator did not handle the screenshare settings properly for numSimulcastStreams = 1. Additional test case was added for that. 2. In VideoSender, when rate allocation is updated after setting a new VideoCodec config, only update the state of the EncoderParameters, but don't actually run SetRateAllocation on the encoder itself. This caused some problems upstreams. Please review only the changes after patch set 1. Original description: Extract bitrate allocation of spatial/temporal layers out of codec impl. This CL makes a number of intervowen changes: * Add BitrateAllocation struct, that contains a codec independent view of how the target bitrate is distributed over spatial and temporal layers. * Adds the BitrateAllocator interface, which takes a bitrate and frame rate and produces a BitrateAllocation. * A default (non layered) implementation is added, and SimulcastRateAllocator is extended to fully handle VP8 allocation. This includes capturing TemporalLayer instances created by the encoder. * ViEEncoder now owns both the bitrate allocator and the temporal layer factories for VP8. This allows allocation to happen fully outside of the encoder implementation. This refactoring will make it possible for ViEEncoder to signal the full picture of target bitrates to the RTCP module. BUG=webrtc:6301 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/2510583002 . Cr-Commit-Position: refs/heads/master@{#15105}
2016-11-16 16:41:30 +01:00
if (bitrate.get_sum_bps() == 0) {
// Encoder paused, turn off all encoding.
const int num_streams = static_cast<size_t>(encoders_.size());
for (int i = 0; i < num_streams; ++i)
SetStreamState(false, i);
return WEBRTC_VIDEO_CODEC_OK;
}
Reland #2 of Issue 2434073003: Extract bitrate allocation ... This is yet another reland of https://codereview.webrtc.org/2434073003/ including two fixes: 1. SimulcastRateAllocator did not handle the screenshare settings properly for numSimulcastStreams = 1. Additional test case was added for that. 2. In VideoSender, when rate allocation is updated after setting a new VideoCodec config, only update the state of the EncoderParameters, but don't actually run SetRateAllocation on the encoder itself. This caused some problems upstreams. Please review only the changes after patch set 1. Original description: Extract bitrate allocation of spatial/temporal layers out of codec impl. This CL makes a number of intervowen changes: * Add BitrateAllocation struct, that contains a codec independent view of how the target bitrate is distributed over spatial and temporal layers. * Adds the BitrateAllocator interface, which takes a bitrate and frame rate and produces a BitrateAllocation. * A default (non layered) implementation is added, and SimulcastRateAllocator is extended to fully handle VP8 allocation. This includes capturing TemporalLayer instances created by the encoder. * ViEEncoder now owns both the bitrate allocator and the temporal layer factories for VP8. This allows allocation to happen fully outside of the encoder implementation. This refactoring will make it possible for ViEEncoder to signal the full picture of target bitrates to the RTCP module. BUG=webrtc:6301 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/2510583002 . Cr-Commit-Position: refs/heads/master@{#15105}
2016-11-16 16:41:30 +01:00
// At this point, bitrate allocation should already match codec settings.
if (codec_.maxBitrate > 0)
RTC_DCHECK_LE(bitrate.get_sum_kbps(), codec_.maxBitrate);
RTC_DCHECK_GE(bitrate.get_sum_kbps(), codec_.minBitrate);
if (codec_.numberOfSimulcastStreams > 0)
RTC_DCHECK_GE(bitrate.get_sum_kbps(), codec_.simulcastStream[0].minBitrate);
codec_.maxFramerate = new_framerate;
if (encoders_.size() > 1) {
// If we have more than 1 stream, reduce the qp_max for the low resolution
// stream if frame rate is not too low. The trade-off with lower qp_max is
// possibly more dropped frames, so we only do this if the frame rate is
// above some threshold (base temporal layer is down to 1/4 for 3 layers).
// We may want to condition this on bitrate later.
if (new_framerate > 20) {
configurations_[encoders_.size() - 1].rc_max_quantizer = 45;
} else {
// Go back to default value set in InitEncode.
configurations_[encoders_.size() - 1].rc_max_quantizer = qp_max_;
}
}
size_t stream_idx = encoders_.size() - 1;
for (size_t i = 0; i < encoders_.size(); ++i, --stream_idx) {
Reland #2 of Issue 2434073003: Extract bitrate allocation ... This is yet another reland of https://codereview.webrtc.org/2434073003/ including two fixes: 1. SimulcastRateAllocator did not handle the screenshare settings properly for numSimulcastStreams = 1. Additional test case was added for that. 2. In VideoSender, when rate allocation is updated after setting a new VideoCodec config, only update the state of the EncoderParameters, but don't actually run SetRateAllocation on the encoder itself. This caused some problems upstreams. Please review only the changes after patch set 1. Original description: Extract bitrate allocation of spatial/temporal layers out of codec impl. This CL makes a number of intervowen changes: * Add BitrateAllocation struct, that contains a codec independent view of how the target bitrate is distributed over spatial and temporal layers. * Adds the BitrateAllocator interface, which takes a bitrate and frame rate and produces a BitrateAllocation. * A default (non layered) implementation is added, and SimulcastRateAllocator is extended to fully handle VP8 allocation. This includes capturing TemporalLayer instances created by the encoder. * ViEEncoder now owns both the bitrate allocator and the temporal layer factories for VP8. This allows allocation to happen fully outside of the encoder implementation. This refactoring will make it possible for ViEEncoder to signal the full picture of target bitrates to the RTCP module. BUG=webrtc:6301 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/2510583002 . Cr-Commit-Position: refs/heads/master@{#15105}
2016-11-16 16:41:30 +01:00
unsigned int target_bitrate_kbps =
bitrate.GetSpatialLayerSum(stream_idx) / 1000;
bool send_stream = target_bitrate_kbps > 0;
if (send_stream || encoders_.size() > 1)
SetStreamState(send_stream, stream_idx);
configurations_[i].rc_target_bitrate = target_bitrate_kbps;
if (send_stream) {
temporal_layers_[stream_idx]->OnRatesUpdated(
bitrate.GetTemporalLayerAllocation(stream_idx), new_framerate);
}
UpdateVpxConfiguration(temporal_layers_[stream_idx].get(),
&configurations_[i]);
Reland #2 of Issue 2434073003: Extract bitrate allocation ... This is yet another reland of https://codereview.webrtc.org/2434073003/ including two fixes: 1. SimulcastRateAllocator did not handle the screenshare settings properly for numSimulcastStreams = 1. Additional test case was added for that. 2. In VideoSender, when rate allocation is updated after setting a new VideoCodec config, only update the state of the EncoderParameters, but don't actually run SetRateAllocation on the encoder itself. This caused some problems upstreams. Please review only the changes after patch set 1. Original description: Extract bitrate allocation of spatial/temporal layers out of codec impl. This CL makes a number of intervowen changes: * Add BitrateAllocation struct, that contains a codec independent view of how the target bitrate is distributed over spatial and temporal layers. * Adds the BitrateAllocator interface, which takes a bitrate and frame rate and produces a BitrateAllocation. * A default (non layered) implementation is added, and SimulcastRateAllocator is extended to fully handle VP8 allocation. This includes capturing TemporalLayer instances created by the encoder. * ViEEncoder now owns both the bitrate allocator and the temporal layer factories for VP8. This allows allocation to happen fully outside of the encoder implementation. This refactoring will make it possible for ViEEncoder to signal the full picture of target bitrates to the RTCP module. BUG=webrtc:6301 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/2510583002 . Cr-Commit-Position: refs/heads/master@{#15105}
2016-11-16 16:41:30 +01:00
if (vpx_codec_enc_config_set(&encoders_[i], &configurations_[i])) {
return WEBRTC_VIDEO_CODEC_ERROR;
}
}
return WEBRTC_VIDEO_CODEC_OK;
}
const char* LibvpxVp8Encoder::ImplementationName() const {
return "libvpx";
}
void LibvpxVp8Encoder::SetStreamState(bool send_stream, int stream_idx) {
if (send_stream && !send_stream_[stream_idx]) {
// Need a key frame if we have not sent this stream before.
key_frame_request_[stream_idx] = true;
}
send_stream_[stream_idx] = send_stream;
}
void LibvpxVp8Encoder::SetupTemporalLayers(int num_streams,
int num_temporal_layers,
const VideoCodec& codec) {
RTC_DCHECK(temporal_layers_.empty());
for (int i = 0; i < num_streams; ++i) {
temporal_layers_.emplace_back(
TemporalLayers::CreateTemporalLayers(codec, i));
temporal_layers_checkers_.emplace_back(
TemporalLayers::CreateTemporalLayersChecker(codec, i));
}
}
int LibvpxVp8Encoder::InitEncode(const VideoCodec* inst,
int number_of_cores,
size_t /*maxPayloadSize */) {
if (inst == NULL) {
return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
}
if (inst->maxFramerate < 1) {
return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
}
// allow zero to represent an unspecified maxBitRate
if (inst->maxBitrate > 0 && inst->startBitrate > inst->maxBitrate) {
return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
}
if (inst->width <= 1 || inst->height <= 1) {
return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
}
if (number_of_cores < 1) {
return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
}
if (inst->VP8().automaticResizeOn && inst->numberOfSimulcastStreams > 1) {
return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
}
int retVal = Release();
if (retVal < 0) {
return retVal;
}
Revert "Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8." This reverts commit 07efe436c9002e139845f62486e3ee4e29f0d85b. Reason for revert: Breaks downstream project. cricket::GetSimulcastConfig method signature has been updated. I think you can get away with a default value for temporal_layers_supported (and then you can remove it after a few days when projects will be updated). Original change's description: > Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8. > > * Move SimulcastEncoderAdapter out under modules/video_coding > * Move SimulcastRateAllocator back out to modules/video_coding/utility > * Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility > * Move any VP8 specific code - such as temporal layer bitrate budgeting - > under codec type dependent conditionals. > * Plumb the simulcast index for H264 in the codec specific and RTP format data structures. > > Bug: webrtc:5840 > Change-Id: Ieced8a00e38f273c1a6cfd0f5431a87d07b8f44e > Reviewed-on: https://webrtc-review.googlesource.com/64100 > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23705} TBR=sprang@webrtc.org,stefan@webrtc.org,mflodman@webrtc.org,hta@webrtc.org,sergio.garcia.murillo@gmail.com,titovartem@webrtc.org,agouaillard@gmail.com Change-Id: Ic9d3b1eeaf195bb5ec2063954421f5e77866d663 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:5840 Reviewed-on: https://webrtc-review.googlesource.com/84760 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23710}
2018-06-21 13:41:01 +00:00
int number_of_streams = NumberOfStreams(*inst);
bool doing_simulcast = (number_of_streams > 1);
Revert "Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8." This reverts commit 07efe436c9002e139845f62486e3ee4e29f0d85b. Reason for revert: Breaks downstream project. cricket::GetSimulcastConfig method signature has been updated. I think you can get away with a default value for temporal_layers_supported (and then you can remove it after a few days when projects will be updated). Original change's description: > Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8. > > * Move SimulcastEncoderAdapter out under modules/video_coding > * Move SimulcastRateAllocator back out to modules/video_coding/utility > * Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility > * Move any VP8 specific code - such as temporal layer bitrate budgeting - > under codec type dependent conditionals. > * Plumb the simulcast index for H264 in the codec specific and RTP format data structures. > > Bug: webrtc:5840 > Change-Id: Ieced8a00e38f273c1a6cfd0f5431a87d07b8f44e > Reviewed-on: https://webrtc-review.googlesource.com/64100 > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23705} TBR=sprang@webrtc.org,stefan@webrtc.org,mflodman@webrtc.org,hta@webrtc.org,sergio.garcia.murillo@gmail.com,titovartem@webrtc.org,agouaillard@gmail.com Change-Id: Ic9d3b1eeaf195bb5ec2063954421f5e77866d663 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:5840 Reviewed-on: https://webrtc-review.googlesource.com/84760 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23710}
2018-06-21 13:41:01 +00:00
if (doing_simulcast &&
(!ValidSimulcastResolutions(*inst, number_of_streams) ||
!ValidSimulcastTemporalLayers(*inst, number_of_streams))) {
return WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED;
}
int num_temporal_layers =
doing_simulcast ? inst->simulcastStream[0].numberOfTemporalLayers
: inst->VP8().numberOfTemporalLayers;
Reland #2 of Issue 2434073003: Extract bitrate allocation ... This is yet another reland of https://codereview.webrtc.org/2434073003/ including two fixes: 1. SimulcastRateAllocator did not handle the screenshare settings properly for numSimulcastStreams = 1. Additional test case was added for that. 2. In VideoSender, when rate allocation is updated after setting a new VideoCodec config, only update the state of the EncoderParameters, but don't actually run SetRateAllocation on the encoder itself. This caused some problems upstreams. Please review only the changes after patch set 1. Original description: Extract bitrate allocation of spatial/temporal layers out of codec impl. This CL makes a number of intervowen changes: * Add BitrateAllocation struct, that contains a codec independent view of how the target bitrate is distributed over spatial and temporal layers. * Adds the BitrateAllocator interface, which takes a bitrate and frame rate and produces a BitrateAllocation. * A default (non layered) implementation is added, and SimulcastRateAllocator is extended to fully handle VP8 allocation. This includes capturing TemporalLayer instances created by the encoder. * ViEEncoder now owns both the bitrate allocator and the temporal layer factories for VP8. This allows allocation to happen fully outside of the encoder implementation. This refactoring will make it possible for ViEEncoder to signal the full picture of target bitrates to the RTCP module. BUG=webrtc:6301 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/2510583002 . Cr-Commit-Position: refs/heads/master@{#15105}
2016-11-16 16:41:30 +01:00
RTC_DCHECK_GT(num_temporal_layers, 0);
SetupTemporalLayers(number_of_streams, num_temporal_layers, *inst);
number_of_cores_ = number_of_cores;
timestamp_ = 0;
codec_ = *inst;
// Code expects simulcastStream resolutions to be correct, make sure they are
// filled even when there are no simulcast layers.
if (codec_.numberOfSimulcastStreams == 0) {
codec_.simulcastStream[0].width = codec_.width;
codec_.simulcastStream[0].height = codec_.height;
}
encoded_images_.resize(number_of_streams);
encoders_.resize(number_of_streams);
configurations_.resize(number_of_streams);
downsampling_factors_.resize(number_of_streams);
raw_images_.resize(number_of_streams);
send_stream_.resize(number_of_streams);
send_stream_[0] = true; // For non-simulcast case.
cpu_speed_.resize(number_of_streams);
std::fill(key_frame_request_.begin(), key_frame_request_.end(), false);
int idx = number_of_streams - 1;
for (int i = 0; i < (number_of_streams - 1); ++i, --idx) {
int gcd = GCD(inst->simulcastStream[idx].width,
inst->simulcastStream[idx - 1].width);
downsampling_factors_[i].num = inst->simulcastStream[idx].width / gcd;
downsampling_factors_[i].den = inst->simulcastStream[idx - 1].width / gcd;
send_stream_[i] = false;
}
if (number_of_streams > 1) {
send_stream_[number_of_streams - 1] = false;
downsampling_factors_[number_of_streams - 1].num = 1;
downsampling_factors_[number_of_streams - 1].den = 1;
}
for (int i = 0; i < number_of_streams; ++i) {
// allocate memory for encoded image
if (encoded_images_[i]._buffer != NULL) {
delete[] encoded_images_[i]._buffer;
}
encoded_images_[i]._size =
CalcBufferSize(VideoType::kI420, codec_.width, codec_.height);
encoded_images_[i]._buffer = new uint8_t[encoded_images_[i]._size];
encoded_images_[i]._completeFrame = true;
}
// populate encoder configuration with default values
if (vpx_codec_enc_config_default(vpx_codec_vp8_cx(), &configurations_[0],
0)) {
return WEBRTC_VIDEO_CODEC_ERROR;
}
// setting the time base of the codec
configurations_[0].g_timebase.num = 1;
configurations_[0].g_timebase.den = 90000;
configurations_[0].g_lag_in_frames = 0; // 0- no frame lagging
// Set the error resilience mode for temporal layers (but not simulcast).
configurations_[0].g_error_resilient =
(num_temporal_layers > 1) ? VPX_ERROR_RESILIENT_DEFAULT : 0;
// rate control settings
configurations_[0].rc_dropframe_thresh = inst->VP8().frameDroppingOn ? 30 : 0;
configurations_[0].rc_end_usage = VPX_CBR;
configurations_[0].g_pass = VPX_RC_ONE_PASS;
// Handle resizing outside of libvpx.
configurations_[0].rc_resize_allowed = 0;
configurations_[0].rc_min_quantizer = 2;
if (inst->qpMax >= configurations_[0].rc_min_quantizer) {
qp_max_ = inst->qpMax;
}
configurations_[0].rc_max_quantizer = qp_max_;
configurations_[0].rc_undershoot_pct = 100;
configurations_[0].rc_overshoot_pct = 15;
configurations_[0].rc_buf_initial_sz = 500;
configurations_[0].rc_buf_optimal_sz = 600;
configurations_[0].rc_buf_sz = 1000;
// Set the maximum target size of any key-frame.
rc_max_intra_target_ = MaxIntraTarget(configurations_[0].rc_buf_optimal_sz);
if (inst->VP8().keyFrameInterval > 0) {
configurations_[0].kf_mode = VPX_KF_AUTO;
configurations_[0].kf_max_dist = inst->VP8().keyFrameInterval;
} else {
configurations_[0].kf_mode = VPX_KF_DISABLED;
}
// Allow the user to set the complexity for the base stream.
switch (inst->VP8().complexity) {
case VideoCodecComplexity::kComplexityHigh:
cpu_speed_[0] = -5;
break;
case VideoCodecComplexity::kComplexityHigher:
cpu_speed_[0] = -4;
break;
case VideoCodecComplexity::kComplexityMax:
cpu_speed_[0] = -3;
break;
default:
cpu_speed_[0] = -6;
break;
}
cpu_speed_default_ = cpu_speed_[0];
// Set encoding complexity (cpu_speed) based on resolution and/or platform.
cpu_speed_[0] = SetCpuSpeed(inst->width, inst->height);
for (int i = 1; i < number_of_streams; ++i) {
cpu_speed_[i] =
SetCpuSpeed(inst->simulcastStream[number_of_streams - 1 - i].width,
inst->simulcastStream[number_of_streams - 1 - i].height);
}
configurations_[0].g_w = inst->width;
configurations_[0].g_h = inst->height;
// Determine number of threads based on the image size and #cores.
// TODO(fbarchard): Consider number of Simulcast layers.
configurations_[0].g_threads = NumberOfThreads(
configurations_[0].g_w, configurations_[0].g_h, number_of_cores);
// Creating a wrapper to the image - setting image data to NULL.
// Actual pointer will be set in encode. Setting align to 1, as it
// is meaningless (no memory allocation is done here).
vpx_img_wrap(&raw_images_[0], VPX_IMG_FMT_I420, inst->width, inst->height, 1,
NULL);
Reland #2 of Issue 2434073003: Extract bitrate allocation ... This is yet another reland of https://codereview.webrtc.org/2434073003/ including two fixes: 1. SimulcastRateAllocator did not handle the screenshare settings properly for numSimulcastStreams = 1. Additional test case was added for that. 2. In VideoSender, when rate allocation is updated after setting a new VideoCodec config, only update the state of the EncoderParameters, but don't actually run SetRateAllocation on the encoder itself. This caused some problems upstreams. Please review only the changes after patch set 1. Original description: Extract bitrate allocation of spatial/temporal layers out of codec impl. This CL makes a number of intervowen changes: * Add BitrateAllocation struct, that contains a codec independent view of how the target bitrate is distributed over spatial and temporal layers. * Adds the BitrateAllocator interface, which takes a bitrate and frame rate and produces a BitrateAllocation. * A default (non layered) implementation is added, and SimulcastRateAllocator is extended to fully handle VP8 allocation. This includes capturing TemporalLayer instances created by the encoder. * ViEEncoder now owns both the bitrate allocator and the temporal layer factories for VP8. This allows allocation to happen fully outside of the encoder implementation. This refactoring will make it possible for ViEEncoder to signal the full picture of target bitrates to the RTCP module. BUG=webrtc:6301 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/2510583002 . Cr-Commit-Position: refs/heads/master@{#15105}
2016-11-16 16:41:30 +01:00
// Note the order we use is different from webm, we have lowest resolution
// at position 0 and they have highest resolution at position 0.
int stream_idx = encoders_.size() - 1;
SimulcastRateAllocator init_allocator(codec_);
VideoBitrateAllocation allocation = init_allocator.GetAllocation(
Reland #2 of Issue 2434073003: Extract bitrate allocation ... This is yet another reland of https://codereview.webrtc.org/2434073003/ including two fixes: 1. SimulcastRateAllocator did not handle the screenshare settings properly for numSimulcastStreams = 1. Additional test case was added for that. 2. In VideoSender, when rate allocation is updated after setting a new VideoCodec config, only update the state of the EncoderParameters, but don't actually run SetRateAllocation on the encoder itself. This caused some problems upstreams. Please review only the changes after patch set 1. Original description: Extract bitrate allocation of spatial/temporal layers out of codec impl. This CL makes a number of intervowen changes: * Add BitrateAllocation struct, that contains a codec independent view of how the target bitrate is distributed over spatial and temporal layers. * Adds the BitrateAllocator interface, which takes a bitrate and frame rate and produces a BitrateAllocation. * A default (non layered) implementation is added, and SimulcastRateAllocator is extended to fully handle VP8 allocation. This includes capturing TemporalLayer instances created by the encoder. * ViEEncoder now owns both the bitrate allocator and the temporal layer factories for VP8. This allows allocation to happen fully outside of the encoder implementation. This refactoring will make it possible for ViEEncoder to signal the full picture of target bitrates to the RTCP module. BUG=webrtc:6301 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/2510583002 . Cr-Commit-Position: refs/heads/master@{#15105}
2016-11-16 16:41:30 +01:00
inst->startBitrate * 1000, inst->maxFramerate);
std::vector<uint32_t> stream_bitrates;
for (int i = 0; i == 0 || i < inst->numberOfSimulcastStreams; ++i) {
uint32_t bitrate = allocation.GetSpatialLayerSum(i) / 1000;
stream_bitrates.push_back(bitrate);
}
configurations_[0].rc_target_bitrate = stream_bitrates[stream_idx];
if (stream_bitrates[stream_idx] > 0) {
temporal_layers_[stream_idx]->OnRatesUpdated(
allocation.GetTemporalLayerAllocation(stream_idx), inst->maxFramerate);
}
UpdateVpxConfiguration(temporal_layers_[stream_idx].get(),
&configurations_[0]);
Reland #2 of Issue 2434073003: Extract bitrate allocation ... This is yet another reland of https://codereview.webrtc.org/2434073003/ including two fixes: 1. SimulcastRateAllocator did not handle the screenshare settings properly for numSimulcastStreams = 1. Additional test case was added for that. 2. In VideoSender, when rate allocation is updated after setting a new VideoCodec config, only update the state of the EncoderParameters, but don't actually run SetRateAllocation on the encoder itself. This caused some problems upstreams. Please review only the changes after patch set 1. Original description: Extract bitrate allocation of spatial/temporal layers out of codec impl. This CL makes a number of intervowen changes: * Add BitrateAllocation struct, that contains a codec independent view of how the target bitrate is distributed over spatial and temporal layers. * Adds the BitrateAllocator interface, which takes a bitrate and frame rate and produces a BitrateAllocation. * A default (non layered) implementation is added, and SimulcastRateAllocator is extended to fully handle VP8 allocation. This includes capturing TemporalLayer instances created by the encoder. * ViEEncoder now owns both the bitrate allocator and the temporal layer factories for VP8. This allows allocation to happen fully outside of the encoder implementation. This refactoring will make it possible for ViEEncoder to signal the full picture of target bitrates to the RTCP module. BUG=webrtc:6301 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/2510583002 . Cr-Commit-Position: refs/heads/master@{#15105}
2016-11-16 16:41:30 +01:00
--stream_idx;
for (size_t i = 1; i < encoders_.size(); ++i, --stream_idx) {
memcpy(&configurations_[i], &configurations_[0],
sizeof(configurations_[0]));
configurations_[i].g_w = inst->simulcastStream[stream_idx].width;
configurations_[i].g_h = inst->simulcastStream[stream_idx].height;
// Use 1 thread for lower resolutions.
configurations_[i].g_threads = 1;
// Setting alignment to 32 - as that ensures at least 16 for all
// planes (32 for Y, 16 for U,V). Libvpx sets the requested stride for
// the y plane, but only half of it to the u and v planes.
vpx_img_alloc(&raw_images_[i], VPX_IMG_FMT_I420,
inst->simulcastStream[stream_idx].width,
inst->simulcastStream[stream_idx].height, kVp832ByteAlign);
SetStreamState(stream_bitrates[stream_idx] > 0, stream_idx);
Reland #2 of Issue 2434073003: Extract bitrate allocation ... This is yet another reland of https://codereview.webrtc.org/2434073003/ including two fixes: 1. SimulcastRateAllocator did not handle the screenshare settings properly for numSimulcastStreams = 1. Additional test case was added for that. 2. In VideoSender, when rate allocation is updated after setting a new VideoCodec config, only update the state of the EncoderParameters, but don't actually run SetRateAllocation on the encoder itself. This caused some problems upstreams. Please review only the changes after patch set 1. Original description: Extract bitrate allocation of spatial/temporal layers out of codec impl. This CL makes a number of intervowen changes: * Add BitrateAllocation struct, that contains a codec independent view of how the target bitrate is distributed over spatial and temporal layers. * Adds the BitrateAllocator interface, which takes a bitrate and frame rate and produces a BitrateAllocation. * A default (non layered) implementation is added, and SimulcastRateAllocator is extended to fully handle VP8 allocation. This includes capturing TemporalLayer instances created by the encoder. * ViEEncoder now owns both the bitrate allocator and the temporal layer factories for VP8. This allows allocation to happen fully outside of the encoder implementation. This refactoring will make it possible for ViEEncoder to signal the full picture of target bitrates to the RTCP module. BUG=webrtc:6301 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/2510583002 . Cr-Commit-Position: refs/heads/master@{#15105}
2016-11-16 16:41:30 +01:00
configurations_[i].rc_target_bitrate = stream_bitrates[stream_idx];
if (stream_bitrates[stream_idx] > 0) {
temporal_layers_[stream_idx]->OnRatesUpdated(
allocation.GetTemporalLayerAllocation(stream_idx),
inst->maxFramerate);
}
UpdateVpxConfiguration(temporal_layers_[stream_idx].get(),
&configurations_[i]);
}
return InitAndSetControlSettings();
}
int LibvpxVp8Encoder::SetCpuSpeed(int width, int height) {
#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || \
defined(WEBRTC_ANDROID)
// On mobile platform, use a lower speed setting for lower resolutions for
// CPUs with 4 or more cores.
RTC_DCHECK_GT(number_of_cores_, 0);
if (number_of_cores_ <= 3)
return -12;
if (width * height <= 352 * 288)
return -8;
else if (width * height <= 640 * 480)
return -10;
else
return -12;
#else
// For non-ARM, increase encoding complexity (i.e., use lower speed setting)
// if resolution is below CIF. Otherwise, keep the default/user setting
// (|cpu_speed_default_|) set on InitEncode via VP8().complexity.
if (width * height < 352 * 288)
return (cpu_speed_default_ < -4) ? -4 : cpu_speed_default_;
else
return cpu_speed_default_;
#endif
}
int LibvpxVp8Encoder::NumberOfThreads(int width, int height, int cpus) {
#if defined(WEBRTC_ANDROID)
if (width * height >= 320 * 180) {
if (cpus >= 4) {
// 3 threads for CPUs with 4 and more cores since most of times only 4
// cores will be active.
return 3;
} else if (cpus == 3 || cpus == 2) {
return 2;
} else {
return 1;
}
}
return 1;
#else
if (width * height >= 1920 * 1080 && cpus > 8) {
return 8; // 8 threads for 1080p on high perf machines.
} else if (width * height > 1280 * 960 && cpus >= 6) {
// 3 threads for 1080p.
return 3;
} else if (width * height > 640 * 480 && cpus >= 3) {
// Default 2 threads for qHD/HD, but allow 3 if core count is high enough,
// as this will allow more margin for high-core/low clock machines or if
// not built with highest optimization.
if (cpus >= 6) {
return 3;
}
return 2;
} else {
// 1 thread for VGA or less.
return 1;
}
#endif
}
int LibvpxVp8Encoder::InitAndSetControlSettings() {
vpx_codec_flags_t flags = 0;
flags |= VPX_CODEC_USE_OUTPUT_PARTITION;
if (encoders_.size() > 1) {
int error = vpx_codec_enc_init_multi(&encoders_[0], vpx_codec_vp8_cx(),
&configurations_[0], encoders_.size(),
flags, &downsampling_factors_[0]);
if (error) {
return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
}
} else {
if (vpx_codec_enc_init(&encoders_[0], vpx_codec_vp8_cx(),
&configurations_[0], flags)) {
return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
}
}
// Enable denoising for the highest resolution stream, and for
// the second highest resolution if we are doing more than 2
// spatial layers/streams.
// TODO(holmer): Investigate possibility of adding a libvpx API
// for getting the denoised frame from the encoder and using that
// when encoding lower resolution streams. Would it work with the
// multi-res encoding feature?
denoiserState denoiser_state = kDenoiserOnYOnly;
#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || \
defined(WEBRTC_ANDROID)
denoiser_state = kDenoiserOnYOnly;
#else
denoiser_state = kDenoiserOnAdaptive;
#endif
vpx_codec_control(&encoders_[0], VP8E_SET_NOISE_SENSITIVITY,
codec_.VP8()->denoisingOn ? denoiser_state : kDenoiserOff);
if (encoders_.size() > 2) {
vpx_codec_control(
&encoders_[1], VP8E_SET_NOISE_SENSITIVITY,
codec_.VP8()->denoisingOn ? denoiser_state : kDenoiserOff);
}
for (size_t i = 0; i < encoders_.size(); ++i) {
// Allow more screen content to be detected as static.
vpx_codec_control(&(encoders_[i]), VP8E_SET_STATIC_THRESHOLD,
codec_.mode == VideoCodecMode::kScreensharing ? 300 : 1);
vpx_codec_control(&(encoders_[i]), VP8E_SET_CPUUSED, cpu_speed_[i]);
vpx_codec_control(&(encoders_[i]), VP8E_SET_TOKEN_PARTITIONS,
static_cast<vp8e_token_partitions>(kTokenPartitions));
vpx_codec_control(&(encoders_[i]), VP8E_SET_MAX_INTRA_BITRATE_PCT,
rc_max_intra_target_);
// VP8E_SET_SCREEN_CONTENT_MODE 2 = screen content with more aggressive
// rate control (drop frames on large target bitrate overshoot)
vpx_codec_control(&(encoders_[i]), VP8E_SET_SCREEN_CONTENT_MODE,
codec_.mode == VideoCodecMode::kScreensharing ? 2 : 0);
// Apply boost on golden frames (has only effect when resilience is off).
if (use_gf_boost_ && configurations_[0].g_error_resilient == 0) {
int gf_boost_percent;
if (GetGfBoostPercentageFromFieldTrialGroup(&gf_boost_percent)) {
vpx_codec_control(&(encoders_[i]), VP8E_SET_GF_CBR_BOOST_PCT,
gf_boost_percent);
}
}
}
inited_ = true;
return WEBRTC_VIDEO_CODEC_OK;
}
uint32_t LibvpxVp8Encoder::MaxIntraTarget(uint32_t optimalBuffersize) {
// Set max to the optimal buffer level (normalized by target BR),
// and scaled by a scalePar.
// Max target size = scalePar * optimalBufferSize * targetBR[Kbps].
// This values is presented in percentage of perFrameBw:
// perFrameBw = targetBR[Kbps] * 1000 / frameRate.
// The target in % is as follows:
float scalePar = 0.5;
uint32_t targetPct = optimalBuffersize * scalePar * codec_.maxFramerate / 10;
// Don't go below 3 times the per frame bandwidth.
const uint32_t minIntraTh = 300;
return (targetPct < minIntraTh) ? minIntraTh : targetPct;
}
int LibvpxVp8Encoder::Encode(const VideoFrame& frame,
const CodecSpecificInfo* codec_specific_info,
const std::vector<FrameType>* frame_types) {
RTC_DCHECK_EQ(frame.width(), codec_.width);
RTC_DCHECK_EQ(frame.height(), codec_.height);
if (!inited_)
return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
if (encoded_complete_callback_ == NULL)
return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
Revert "Revert "Update video_coding/codecs to new VideoFrameBuffer interface"" This reverts commit 88f94fa36aa61f7904d30251205c544ada2c4301. Chromium code has been updated. Original change's description: > Revert "Update video_coding/codecs to new VideoFrameBuffer interface" > > This reverts commit 20ebf4ede803cd4f628ef9378700f60b72f2eab0. > > Reason for revert: > > Suspect of breaking FYI bots. > See https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win7%20Tester/builds/9036 and others. > > Sample logs: > Backtrace: > [5024:1036:0607/173649.857:FATAL:webrtc_video_frame_adapter.cc(98)] Check failed: false. > Backtrace: > base::debug::StackTrace::StackTrace [0x02D04A37+55] > base::debug::StackTrace::StackTrace [0x02CCBB8A+10] > content::WebRtcVideoFrameAdapter::NativeToI420Buffer [0x0508AD71+305] > webrtc::VideoFrameBuffer::ToI420 [0x0230BF67+39] > webrtc::H264EncoderImpl::Encode [0x057E8D0B+267] > webrtc::VCMGenericEncoder::Encode [0x057E0E34+333] > webrtc::vcm::VideoSender::AddVideoFrame [0x057DED9B+796] > webrtc::ViEEncoder::EncodeVideoFrame [0x057C00F6+884] > webrtc::ViEEncoder::EncodeTask::Run [0x057C12D7+215] > rtc::TaskQueue::PostTask [0x03EE5CFB+194] > base::internal::Invoker<base::internal::BindState<enum extensions::`anonymous namespace'::VerificationResult (__cdecl*)(std::unique_ptr<extensions::NetworkingCastPrivateDelegate::Credentials,std::default_delete<extensions::NetworkingCastPrivateDelegate::C [0x02DDCAA5+31] > base::internal::Invoker<base::internal::BindState<enum extensions::`anonymous namespace'::VerificationResult (__cdecl*)(std::unique_ptr<extensions::NetworkingCastPrivateDelegate::Credentials,std::default_delete<extensions::NetworkingCastPrivateDelegate::C [0x02DDEE86+22] > base::debug::TaskAnnotator::RunTask [0x02D08289+409] > base::MessageLoop::RunTask [0x02C8CEC1+1233] > base::MessageLoop::DoWork [0x02C8C1AD+765] > base::MessagePumpDefault::Run [0x02D0A20B+219] > base::MessageLoop::Run [0x02C8C9DB+107] > base::RunLoop::Run [0x02C89583+147] > base::Thread::Run [0x02CBEFCD+173] > base::Thread::ThreadMain [0x02CBFADE+622] > base::PlatformThread::Sleep [0x02C9E1A2+290] > BaseThreadInitThunk [0x75C3338A+18] > RtlInitializeExceptionChain [0x773A9902+99] > RtlInitializeExceptionChain [0x773A98D5+54] > > Original change's description: > > Update video_coding/codecs to new VideoFrameBuffer interface > > > > This is a follow-up cleanup for CL > > https://codereview.webrtc.org/2847383002/. > > > > Bug: webrtc:7632 > > Change-Id: I47861d779968f2fee94db9c017102a8e87e67fb7 > > Reviewed-on: https://chromium-review.googlesource.com/524163 > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#18477} > > TBR=magjed@webrtc.org,nisse@webrtc.org,brandtr@webrtc.org > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:7632 > > Change-Id: I3b73fc7d16ff19ceba196e964dcb36a36510912c > Reviewed-on: https://chromium-review.googlesource.com/527793 > Reviewed-by: Guido Urdaneta <guidou@chromium.org> > Commit-Queue: Guido Urdaneta <guidou@chromium.org> > Cr-Commit-Position: refs/heads/master@{#18489} TBR=tterriberry@mozilla.com,mflodman@webrtc.org,magjed@webrtc.org,stefan@webrtc.org,guidou@chromium.org,nisse@webrtc.org,brandtr@webrtc.org,webrtc-reviews@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. No-Presubmit: true Bug: webrtc:7632 Change-Id: I0962a704e8a9939d4364ce9069c863c9951654c9 Reviewed-on: https://chromium-review.googlesource.com/530684 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#18527}
2017-06-10 17:03:37 +00:00
rtc::scoped_refptr<I420BufferInterface> input_image =
frame.video_frame_buffer()->ToI420();
// Since we are extracting raw pointers from |input_image| to
// |raw_images_[0]|, the resolution of these frames must match.
RTC_DCHECK_EQ(input_image->width(), raw_images_[0].d_w);
RTC_DCHECK_EQ(input_image->height(), raw_images_[0].d_h);
// Image in vpx_image_t format.
// Input image is const. VP8's raw image is not defined as const.
raw_images_[0].planes[VPX_PLANE_Y] =
const_cast<uint8_t*>(input_image->DataY());
raw_images_[0].planes[VPX_PLANE_U] =
const_cast<uint8_t*>(input_image->DataU());
raw_images_[0].planes[VPX_PLANE_V] =
const_cast<uint8_t*>(input_image->DataV());
raw_images_[0].stride[VPX_PLANE_Y] = input_image->StrideY();
raw_images_[0].stride[VPX_PLANE_U] = input_image->StrideU();
raw_images_[0].stride[VPX_PLANE_V] = input_image->StrideV();
for (size_t i = 1; i < encoders_.size(); ++i) {
// Scale the image down a number of times by downsampling factor
libyuv::I420Scale(
raw_images_[i - 1].planes[VPX_PLANE_Y],
raw_images_[i - 1].stride[VPX_PLANE_Y],
raw_images_[i - 1].planes[VPX_PLANE_U],
raw_images_[i - 1].stride[VPX_PLANE_U],
raw_images_[i - 1].planes[VPX_PLANE_V],
raw_images_[i - 1].stride[VPX_PLANE_V], raw_images_[i - 1].d_w,
raw_images_[i - 1].d_h, raw_images_[i].planes[VPX_PLANE_Y],
raw_images_[i].stride[VPX_PLANE_Y], raw_images_[i].planes[VPX_PLANE_U],
raw_images_[i].stride[VPX_PLANE_U], raw_images_[i].planes[VPX_PLANE_V],
raw_images_[i].stride[VPX_PLANE_V], raw_images_[i].d_w,
raw_images_[i].d_h, libyuv::kFilterBilinear);
}
bool send_key_frame = false;
for (size_t i = 0; i < key_frame_request_.size() && i < send_stream_.size();
++i) {
if (key_frame_request_[i] && send_stream_[i]) {
send_key_frame = true;
break;
}
}
if (!send_key_frame && frame_types) {
for (size_t i = 0; i < frame_types->size() && i < send_stream_.size();
++i) {
if ((*frame_types)[i] == kVideoFrameKey && send_stream_[i]) {
send_key_frame = true;
break;
}
}
}
vpx_enc_frame_flags_t flags[kMaxSimulcastStreams];
TemporalLayers::FrameConfig tl_configs[kMaxSimulcastStreams];
for (size_t i = 0; i < encoders_.size(); ++i) {
tl_configs[i] = temporal_layers_[i]->UpdateLayerConfig(frame.timestamp());
RTC_DCHECK(temporal_layers_checkers_[i]->CheckTemporalConfig(
send_key_frame, tl_configs[i]));
if (tl_configs[i].drop_frame) {
// Drop this frame.
return WEBRTC_VIDEO_CODEC_OK;
}
flags[i] = EncodeFlags(tl_configs[i]);
}
if (send_key_frame) {
// Adapt the size of the key frame when in screenshare with 1 temporal
// layer.
if (encoders_.size() == 1 &&
codec_.mode == VideoCodecMode::kScreensharing &&
codec_.VP8()->numberOfTemporalLayers <= 1) {
const uint32_t forceKeyFrameIntraTh = 100;
vpx_codec_control(&(encoders_[0]), VP8E_SET_MAX_INTRA_BITRATE_PCT,
forceKeyFrameIntraTh);
}
// Key frame request from caller.
// Will update both golden and alt-ref.
for (size_t i = 0; i < encoders_.size(); ++i) {
flags[i] = VPX_EFLAG_FORCE_KF;
}
std::fill(key_frame_request_.begin(), key_frame_request_.end(), false);
}
// Set the encoder frame flags and temporal layer_id for each spatial stream.
// Note that |temporal_layers_| are defined starting from lowest resolution at
// position 0 to highest resolution at position |encoders_.size() - 1|,
// whereas |encoder_| is from highest to lowest resolution.
size_t stream_idx = encoders_.size() - 1;
for (size_t i = 0; i < encoders_.size(); ++i, --stream_idx) {
// Allow the layers adapter to temporarily modify the configuration. This
// change isn't stored in configurations_ so change will be discarded at
// the next update.
vpx_codec_enc_cfg_t temp_config;
memcpy(&temp_config, &configurations_[i], sizeof(vpx_codec_enc_cfg_t));
if (UpdateVpxConfiguration(temporal_layers_[stream_idx].get(),
&temp_config)) {
if (vpx_codec_enc_config_set(&encoders_[i], &temp_config))
return WEBRTC_VIDEO_CODEC_ERROR;
}
vpx_codec_control(&encoders_[i], VP8E_SET_FRAME_FLAGS, flags[stream_idx]);
vpx_codec_control(&encoders_[i], VP8E_SET_TEMPORAL_LAYER_ID,
tl_configs[i].encoder_layer_id);
}
// TODO(holmer): Ideally the duration should be the timestamp diff of this
// frame and the next frame to be encoded, which we don't have. Instead we
// would like to use the duration of the previous frame. Unfortunately the
// rate control seems to be off with that setup. Using the average input
// frame rate to calculate an average duration for now.
assert(codec_.maxFramerate > 0);
uint32_t duration = 90000 / codec_.maxFramerate;
int error = WEBRTC_VIDEO_CODEC_OK;
int num_tries = 0;
// If the first try returns WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT
// the frame must be reencoded with the same parameters again because
// target bitrate is exceeded and encoder state has been reset.
while (num_tries == 0 ||
(num_tries == 1 &&
error == WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT)) {
++num_tries;
// Note we must pass 0 for |flags| field in encode call below since they are
// set above in |vpx_codec_control| function for each encoder/spatial layer.
error = vpx_codec_encode(&encoders_[0], &raw_images_[0], timestamp_,
duration, 0, VPX_DL_REALTIME);
// Reset specific intra frame thresholds, following the key frame.
if (send_key_frame) {
vpx_codec_control(&(encoders_[0]), VP8E_SET_MAX_INTRA_BITRATE_PCT,
rc_max_intra_target_);
}
if (error)
return WEBRTC_VIDEO_CODEC_ERROR;
timestamp_ += duration;
// Examines frame timestamps only.
error = GetEncodedPartitions(tl_configs, frame);
}
return error;
}
void LibvpxVp8Encoder::PopulateCodecSpecific(
CodecSpecificInfo* codec_specific,
const TemporalLayers::FrameConfig& tl_config,
const vpx_codec_cx_pkt_t& pkt,
int stream_idx,
uint32_t timestamp) {
assert(codec_specific != NULL);
codec_specific->codecType = kVideoCodecVP8;
codec_specific->codec_name = ImplementationName();
CodecSpecificInfoVP8* vp8Info = &(codec_specific->codecSpecific.VP8);
vp8Info->simulcastIdx = stream_idx;
vp8Info->keyIdx = kNoKeyIdx; // TODO(hlundin) populate this
vp8Info->nonReference = (pkt.data.frame.flags & VPX_FRAME_IS_DROPPABLE) != 0;
temporal_layers_[stream_idx]->PopulateCodecSpecific(
(pkt.data.frame.flags & VPX_FRAME_IS_KEY) != 0, tl_config, vp8Info,
timestamp);
}
int LibvpxVp8Encoder::GetEncodedPartitions(
const TemporalLayers::FrameConfig tl_configs[],
const VideoFrame& input_image) {
int stream_idx = static_cast<int>(encoders_.size()) - 1;
int result = WEBRTC_VIDEO_CODEC_OK;
for (size_t encoder_idx = 0; encoder_idx < encoders_.size();
++encoder_idx, --stream_idx) {
vpx_codec_iter_t iter = NULL;
int part_idx = 0;
encoded_images_[encoder_idx]._length = 0;
encoded_images_[encoder_idx]._frameType = kVideoFrameDelta;
RTPFragmentationHeader frag_info;
// kTokenPartitions is number of bits used.
frag_info.VerifyAndAllocateFragmentationHeader((1 << kTokenPartitions) + 1);
CodecSpecificInfo codec_specific;
const vpx_codec_cx_pkt_t* pkt = NULL;
while ((pkt = vpx_codec_get_cx_data(&encoders_[encoder_idx], &iter)) !=
NULL) {
switch (pkt->kind) {
case VPX_CODEC_CX_FRAME_PKT: {
size_t length = encoded_images_[encoder_idx]._length;
if (pkt->data.frame.sz + length >
encoded_images_[encoder_idx]._size) {
uint8_t* buffer = new uint8_t[pkt->data.frame.sz + length];
memcpy(buffer, encoded_images_[encoder_idx]._buffer, length);
delete[] encoded_images_[encoder_idx]._buffer;
encoded_images_[encoder_idx]._buffer = buffer;
encoded_images_[encoder_idx]._size = pkt->data.frame.sz + length;
}
memcpy(&encoded_images_[encoder_idx]._buffer[length],
pkt->data.frame.buf, pkt->data.frame.sz);
frag_info.fragmentationOffset[part_idx] = length;
frag_info.fragmentationLength[part_idx] = pkt->data.frame.sz;
frag_info.fragmentationPlType[part_idx] = 0; // not known here
frag_info.fragmentationTimeDiff[part_idx] = 0;
encoded_images_[encoder_idx]._length += pkt->data.frame.sz;
assert(length <= encoded_images_[encoder_idx]._size);
++part_idx;
break;
}
default:
break;
}
// End of frame
if ((pkt->data.frame.flags & VPX_FRAME_IS_FRAGMENT) == 0) {
// check if encoded frame is a key frame
if (pkt->data.frame.flags & VPX_FRAME_IS_KEY) {
encoded_images_[encoder_idx]._frameType = kVideoFrameKey;
}
PopulateCodecSpecific(&codec_specific, tl_configs[stream_idx], *pkt,
stream_idx, input_image.timestamp());
break;
}
}
encoded_images_[encoder_idx]._timeStamp = input_image.timestamp();
encoded_images_[encoder_idx].capture_time_ms_ =
input_image.render_time_ms();
encoded_images_[encoder_idx].rotation_ = input_image.rotation();
encoded_images_[encoder_idx].content_type_ =
(codec_.mode == VideoCodecMode::kScreensharing)
? VideoContentType::SCREENSHARE
: VideoContentType::UNSPECIFIED;
encoded_images_[encoder_idx].timing_.flags = VideoSendTiming::kInvalid;
int qp = -1;
vpx_codec_control(&encoders_[encoder_idx], VP8E_GET_LAST_QUANTIZER_64, &qp);
temporal_layers_[stream_idx]->FrameEncoded(
encoded_images_[encoder_idx]._length, qp);
if (send_stream_[stream_idx]) {
if (encoded_images_[encoder_idx]._length > 0) {
TRACE_COUNTER_ID1("webrtc", "EncodedFrameSize", encoder_idx,
encoded_images_[encoder_idx]._length);
encoded_images_[encoder_idx]._encodedHeight =
codec_.simulcastStream[stream_idx].height;
encoded_images_[encoder_idx]._encodedWidth =
codec_.simulcastStream[stream_idx].width;
int qp_128 = -1;
vpx_codec_control(&encoders_[encoder_idx], VP8E_GET_LAST_QUANTIZER,
&qp_128);
encoded_images_[encoder_idx].qp_ = qp_128;
encoded_complete_callback_->OnEncodedImage(encoded_images_[encoder_idx],
&codec_specific, &frag_info);
} else if (codec_.mode == VideoCodecMode::kScreensharing) {
result = WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT;
}
}
}
return result;
}
VideoEncoder::ScalingSettings LibvpxVp8Encoder::GetScalingSettings() const {
const bool enable_scaling = encoders_.size() == 1 &&
configurations_[0].rc_dropframe_thresh > 0 &&
codec_.VP8().automaticResizeOn;
return enable_scaling ? VideoEncoder::ScalingSettings(kLowVp8QpThreshold,
kHighVp8QpThreshold)
: VideoEncoder::ScalingSettings::kOff;
}
int LibvpxVp8Encoder::SetChannelParameters(uint32_t packetLoss, int64_t rtt) {
return WEBRTC_VIDEO_CODEC_OK;
}
int LibvpxVp8Encoder::RegisterEncodeCompleteCallback(
EncodedImageCallback* callback) {
encoded_complete_callback_ = callback;
return WEBRTC_VIDEO_CODEC_OK;
}
} // namespace webrtc