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authorChih-hung Hsieh <chh@google.com>2015-12-01 17:07:48 +0000
committerandroid-build-merger <android-build-merger@google.com>2015-12-01 17:07:48 +0000
commita4acd9d6bc9b3b033d7d274316e75ee067df8d20 (patch)
tree672a185b294789cf991f385c3e395dd63bea9063 /webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.cc
parent3681b90ba4fe7a27232dd3e27897d5d7ed9d651c (diff)
parentfe8b4a657979b49e1701bd92f6d5814a99e0b2be (diff)
downloadwebrtc-a4acd9d6bc9b3b033d7d274316e75ee067df8d20.tar.gz
Merge changes I7bbf776e,I1b827825
am: fe8b4a6579 * commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits) WIP: Changes after merge commit 'cb3f9bd' Make the nonlinear beamformer steerable Utilize bitrate above codec max to protect video. Enable VP9 internal resize by default. Filter overlapping RTP header extensions. Make VCMEncodedFrameCallback const. MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback. Remove redudant encoder rate calls. Create isolate files for nonparallel tests. Register header extensions in RtpRtcpObserver to avoid log spam. Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table ACM: Move NACK functionality inside NetEq Fix chromium-style warnings in webrtc/sound/. Create a 'webrtc_nonparallel_tests' target. Update scalability structure data according to updates in the RTP payload profile. audio_coding: rename interface -> include Rewrote perform_action_on_all_files to be parallell. Update reference indices according to updates in the RTP payload profile. Disable P2PTransport...TestFailoverControlledSide on Memcheck pass clangcl compile options to ignore warnings in gflags.cc ...
Diffstat (limited to 'webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.cc')
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diff --git a/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.cc b/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.cc
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+++ b/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.cc
@@ -0,0 +1,510 @@
+/*
+ * Copyright (c) 2014 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/vp8/simulcast_encoder_adapter.h"
+
+#include <algorithm>
+
+// NOTE(ajm): Path provided by gyp.
+#include "libyuv/scale.h" // NOLINT
+
+#include "webrtc/base/checks.h"
+#include "webrtc/common.h"
+#include "webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h"
+
+namespace {
+
+const unsigned int kDefaultMinQp = 2;
+const unsigned int kDefaultMaxQp = 56;
+// Max qp for lowest spatial resolution when doing simulcast.
+const unsigned int kLowestResMaxQp = 45;
+
+uint32_t SumStreamTargetBitrate(int streams, const webrtc::VideoCodec& codec) {
+ uint32_t bitrate_sum = 0;
+ for (int i = 0; i < streams; ++i) {
+ bitrate_sum += codec.simulcastStream[i].targetBitrate;
+ }
+ return bitrate_sum;
+}
+
+uint32_t SumStreamMaxBitrate(int streams, const webrtc::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 webrtc::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 webrtc::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;
+ }
+ }
+ return true;
+}
+
+int VerifyCodec(const webrtc::VideoCodec* inst) {
+ 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 (inst->codecSpecific.VP8.feedbackModeOn &&
+ inst->numberOfSimulcastStreams > 1) {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+ if (inst->codecSpecific.VP8.automaticResizeOn &&
+ inst->numberOfSimulcastStreams > 1) {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+// TL1 FrameDropper's max time to drop frames.
+const float kTl1MaxTimeToDropFrames = 20.0f;
+
+struct ScreenshareTemporalLayersFactory : webrtc::TemporalLayers::Factory {
+ ScreenshareTemporalLayersFactory()
+ : tl1_frame_dropper_(kTl1MaxTimeToDropFrames) {}
+
+ virtual ~ScreenshareTemporalLayersFactory() {}
+
+ virtual webrtc::TemporalLayers* Create(int num_temporal_layers,
+ uint8_t initial_tl0_pic_idx) const {
+ return new webrtc::ScreenshareLayers(num_temporal_layers, rand());
+ }
+
+ mutable webrtc::FrameDropper tl0_frame_dropper_;
+ mutable webrtc::FrameDropper tl1_frame_dropper_;
+};
+
+// An EncodedImageCallback implementation that forwards on calls to a
+// SimulcastEncoderAdapter, but with the stream index it's registered with as
+// the first parameter to Encoded.
+class AdapterEncodedImageCallback : public webrtc::EncodedImageCallback {
+ public:
+ AdapterEncodedImageCallback(webrtc::SimulcastEncoderAdapter* adapter,
+ size_t stream_idx)
+ : adapter_(adapter), stream_idx_(stream_idx) {}
+
+ int32_t Encoded(
+ const webrtc::EncodedImage& encodedImage,
+ const webrtc::CodecSpecificInfo* codecSpecificInfo = NULL,
+ const webrtc::RTPFragmentationHeader* fragmentation = NULL) override {
+ return adapter_->Encoded(stream_idx_, encodedImage, codecSpecificInfo,
+ fragmentation);
+ }
+
+ private:
+ webrtc::SimulcastEncoderAdapter* const adapter_;
+ const size_t stream_idx_;
+};
+
+} // namespace
+
+namespace webrtc {
+
+SimulcastEncoderAdapter::SimulcastEncoderAdapter(VideoEncoderFactory* factory)
+ : factory_(factory), encoded_complete_callback_(NULL) {
+ memset(&codec_, 0, sizeof(webrtc::VideoCodec));
+}
+
+SimulcastEncoderAdapter::~SimulcastEncoderAdapter() {
+ Release();
+}
+
+int SimulcastEncoderAdapter::Release() {
+ // TODO(pbos): Keep the last encoder instance but call ::Release() on it, then
+ // re-use this instance in ::InitEncode(). This means that changing
+ // resolutions doesn't require reallocation of the first encoder, but only
+ // reinitialization, which makes sense. Then Destroy this instance instead in
+ // ~SimulcastEncoderAdapter().
+ while (!streaminfos_.empty()) {
+ VideoEncoder* encoder = streaminfos_.back().encoder;
+ EncodedImageCallback* callback = streaminfos_.back().callback;
+ factory_->Destroy(encoder);
+ delete callback;
+ streaminfos_.pop_back();
+ }
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int SimulcastEncoderAdapter::InitEncode(const VideoCodec* inst,
+ int number_of_cores,
+ size_t max_payload_size) {
+ if (number_of_cores < 1) {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+
+ int ret = VerifyCodec(inst);
+ if (ret < 0) {
+ return ret;
+ }
+
+ ret = Release();
+ if (ret < 0) {
+ return ret;
+ }
+
+ int number_of_streams = NumberOfStreams(*inst);
+ bool doing_simulcast = (number_of_streams > 1);
+
+ if (doing_simulcast && !ValidSimulcastResolutions(*inst, number_of_streams)) {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+
+ codec_ = *inst;
+
+ // Special mode when screensharing on a single stream.
+ if (number_of_streams == 1 && inst->mode == kScreensharing) {
+ screensharing_extra_options_.reset(new Config());
+ screensharing_extra_options_->Set<TemporalLayers::Factory>(
+ new ScreenshareTemporalLayersFactory());
+ codec_.extra_options = screensharing_extra_options_.get();
+ }
+
+ // Create |number_of_streams| of encoder instances and init them.
+ for (int i = 0; i < number_of_streams; ++i) {
+ VideoCodec stream_codec;
+ bool send_stream = true;
+ if (!doing_simulcast) {
+ stream_codec = codec_;
+ stream_codec.numberOfSimulcastStreams = 1;
+ } else {
+ bool highest_resolution_stream = (i == (number_of_streams - 1));
+ PopulateStreamCodec(&codec_, i, number_of_streams,
+ highest_resolution_stream, &stream_codec,
+ &send_stream);
+ }
+
+ // TODO(ronghuawu): Remove once this is handled in VP8EncoderImpl.
+ if (stream_codec.qpMax < kDefaultMinQp) {
+ stream_codec.qpMax = kDefaultMaxQp;
+ }
+
+ VideoEncoder* encoder = factory_->Create();
+ ret = encoder->InitEncode(&stream_codec,
+ number_of_cores,
+ max_payload_size);
+ if (ret < 0) {
+ Release();
+ return ret;
+ }
+ EncodedImageCallback* callback = new AdapterEncodedImageCallback(this, i);
+ encoder->RegisterEncodeCompleteCallback(callback);
+ streaminfos_.push_back(StreamInfo(encoder, callback, stream_codec.width,
+ stream_codec.height, send_stream));
+ }
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int SimulcastEncoderAdapter::Encode(
+ const VideoFrame& input_image,
+ const CodecSpecificInfo* codec_specific_info,
+ const std::vector<FrameType>* frame_types) {
+ if (!Initialized()) {
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+ }
+ if (encoded_complete_callback_ == NULL) {
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+ }
+
+ // All active streams should generate a key frame if
+ // a key frame is requested by any stream.
+ bool send_key_frame = false;
+ if (frame_types) {
+ for (size_t i = 0; i < frame_types->size(); ++i) {
+ if (frame_types->at(i) == kVideoFrameKey) {
+ send_key_frame = true;
+ break;
+ }
+ }
+ }
+ for (size_t stream_idx = 0; stream_idx < streaminfos_.size(); ++stream_idx) {
+ if (streaminfos_[stream_idx].key_frame_request &&
+ streaminfos_[stream_idx].send_stream) {
+ send_key_frame = true;
+ break;
+ }
+ }
+
+ int src_width = input_image.width();
+ int src_height = input_image.height();
+ for (size_t stream_idx = 0; stream_idx < streaminfos_.size(); ++stream_idx) {
+ // Don't encode frames in resolutions that we don't intend to send.
+ if (!streaminfos_[stream_idx].send_stream)
+ continue;
+
+ std::vector<FrameType> stream_frame_types;
+ if (send_key_frame) {
+ stream_frame_types.push_back(kVideoFrameKey);
+ streaminfos_[stream_idx].key_frame_request = false;
+ } else {
+ stream_frame_types.push_back(kVideoFrameDelta);
+ }
+
+ int dst_width = streaminfos_[stream_idx].width;
+ int dst_height = streaminfos_[stream_idx].height;
+ // If scaling isn't required, because the input resolution
+ // matches the destination or the input image is empty (e.g.
+ // a keyframe request for encoders with internal camera
+ // sources), pass the image on directly. Otherwise, we'll
+ // scale it to match what the encoder expects (below).
+ if ((dst_width == src_width && dst_height == src_height) ||
+ input_image.IsZeroSize()) {
+ streaminfos_[stream_idx].encoder->Encode(input_image,
+ codec_specific_info,
+ &stream_frame_types);
+ } else {
+ VideoFrame dst_frame;
+ // Making sure that destination frame is of sufficient size.
+ // Aligning stride values based on width.
+ dst_frame.CreateEmptyFrame(dst_width, dst_height,
+ dst_width, (dst_width + 1) / 2,
+ (dst_width + 1) / 2);
+ libyuv::I420Scale(input_image.buffer(kYPlane),
+ input_image.stride(kYPlane),
+ input_image.buffer(kUPlane),
+ input_image.stride(kUPlane),
+ input_image.buffer(kVPlane),
+ input_image.stride(kVPlane),
+ src_width, src_height,
+ dst_frame.buffer(kYPlane),
+ dst_frame.stride(kYPlane),
+ dst_frame.buffer(kUPlane),
+ dst_frame.stride(kUPlane),
+ dst_frame.buffer(kVPlane),
+ dst_frame.stride(kVPlane),
+ dst_width, dst_height,
+ libyuv::kFilterBilinear);
+ dst_frame.set_timestamp(input_image.timestamp());
+ dst_frame.set_render_time_ms(input_image.render_time_ms());
+ streaminfos_[stream_idx].encoder->Encode(dst_frame,
+ codec_specific_info,
+ &stream_frame_types);
+ }
+ }
+
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int SimulcastEncoderAdapter::RegisterEncodeCompleteCallback(
+ EncodedImageCallback* callback) {
+ encoded_complete_callback_ = callback;
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int SimulcastEncoderAdapter::SetChannelParameters(uint32_t packet_loss,
+ int64_t rtt) {
+ for (size_t stream_idx = 0; stream_idx < streaminfos_.size(); ++stream_idx) {
+ streaminfos_[stream_idx].encoder->SetChannelParameters(packet_loss, rtt);
+ }
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int SimulcastEncoderAdapter::SetRates(uint32_t new_bitrate_kbit,
+ uint32_t new_framerate) {
+ if (!Initialized()) {
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+ }
+ if (new_framerate < 1) {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+ if (codec_.maxBitrate > 0 && new_bitrate_kbit > codec_.maxBitrate) {
+ new_bitrate_kbit = codec_.maxBitrate;
+ }
+ if (new_bitrate_kbit < codec_.minBitrate) {
+ new_bitrate_kbit = codec_.minBitrate;
+ }
+ if (codec_.numberOfSimulcastStreams > 0 &&
+ new_bitrate_kbit < codec_.simulcastStream[0].minBitrate) {
+ new_bitrate_kbit = codec_.simulcastStream[0].minBitrate;
+ }
+ codec_.maxFramerate = new_framerate;
+
+ bool send_stream = true;
+ uint32_t stream_bitrate = 0;
+ for (size_t stream_idx = 0; stream_idx < streaminfos_.size(); ++stream_idx) {
+ stream_bitrate = GetStreamBitrate(stream_idx, streaminfos_.size(),
+ new_bitrate_kbit, &send_stream);
+ // Need a key frame if we have not sent this stream before.
+ if (send_stream && !streaminfos_[stream_idx].send_stream) {
+ streaminfos_[stream_idx].key_frame_request = true;
+ }
+ streaminfos_[stream_idx].send_stream = send_stream;
+
+ // TODO(holmer): This is a temporary hack for screensharing, where we
+ // interpret the startBitrate as the encoder target bitrate. This is
+ // to allow for a different max bitrate, so if the codec can't meet
+ // the target we still allow it to overshoot up to the max before dropping
+ // frames. This hack should be improved.
+ if (codec_.targetBitrate > 0 &&
+ (codec_.codecSpecific.VP8.numberOfTemporalLayers == 2 ||
+ codec_.simulcastStream[0].numberOfTemporalLayers == 2)) {
+ stream_bitrate = std::min(codec_.maxBitrate, stream_bitrate);
+ // TODO(ronghuawu): Can't change max bitrate via the VideoEncoder
+ // interface. And VP8EncoderImpl doesn't take negative framerate.
+ // max_bitrate = std::min(codec_.maxBitrate, stream_bitrate);
+ // new_framerate = -1;
+ }
+
+ streaminfos_[stream_idx].encoder->SetRates(stream_bitrate, new_framerate);
+ }
+
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t SimulcastEncoderAdapter::Encoded(
+ size_t stream_idx,
+ const EncodedImage& encodedImage,
+ const CodecSpecificInfo* codecSpecificInfo,
+ const RTPFragmentationHeader* fragmentation) {
+ CodecSpecificInfo stream_codec_specific = *codecSpecificInfo;
+ CodecSpecificInfoVP8* vp8Info = &(stream_codec_specific.codecSpecific.VP8);
+ vp8Info->simulcastIdx = stream_idx;
+
+ return encoded_complete_callback_->Encoded(
+ encodedImage, &stream_codec_specific, fragmentation);
+}
+
+uint32_t SimulcastEncoderAdapter::GetStreamBitrate(
+ int stream_idx,
+ size_t total_number_of_streams,
+ uint32_t new_bitrate_kbit,
+ bool* send_stream) const {
+ if (total_number_of_streams == 1) {
+ *send_stream = true;
+ return new_bitrate_kbit;
+ }
+
+ // The bitrate needed to start sending this stream is given by the
+ // minimum bitrate allowed for encoding this stream, plus the sum target
+ // rates of all lower streams.
+ uint32_t sum_target_lower_streams =
+ SumStreamTargetBitrate(stream_idx, codec_);
+ uint32_t bitrate_to_send_this_layer =
+ codec_.simulcastStream[stream_idx].minBitrate + sum_target_lower_streams;
+ if (new_bitrate_kbit >= bitrate_to_send_this_layer) {
+ // We have enough bandwidth to send this stream.
+ *send_stream = true;
+ // Bitrate for this stream is the new bitrate (|new_bitrate_kbit|) minus the
+ // sum target rates of the lower streams, and capped to a maximum bitrate.
+ // The maximum cap depends on whether we send the next higher stream.
+ // If we will be sending the next higher stream, |max_rate| is given by
+ // current stream's |targetBitrate|, otherwise it's capped by |maxBitrate|.
+ if (stream_idx < codec_.numberOfSimulcastStreams - 1) {
+ unsigned int max_rate = codec_.simulcastStream[stream_idx].maxBitrate;
+ if (new_bitrate_kbit >= SumStreamTargetBitrate(stream_idx + 1, codec_) +
+ codec_.simulcastStream[stream_idx + 1].minBitrate) {
+ max_rate = codec_.simulcastStream[stream_idx].targetBitrate;
+ }
+ return std::min(new_bitrate_kbit - sum_target_lower_streams, max_rate);
+ } else {
+ // For the highest stream (highest resolution), the |targetBitRate| and
+ // |maxBitrate| are not used. Any excess bitrate (above the targets of
+ // all lower streams) is given to this (highest resolution) stream.
+ return new_bitrate_kbit - sum_target_lower_streams;
+ }
+ } else {
+ // Not enough bitrate for this stream.
+ // Return our max bitrate of |stream_idx| - 1, but we don't send it. We need
+ // to keep this resolution coding in order for the multi-encoder to work.
+ *send_stream = false;
+ return codec_.simulcastStream[stream_idx - 1].maxBitrate;
+ }
+}
+
+void SimulcastEncoderAdapter::PopulateStreamCodec(
+ const webrtc::VideoCodec* inst,
+ int stream_index,
+ size_t total_number_of_streams,
+ bool highest_resolution_stream,
+ webrtc::VideoCodec* stream_codec,
+ bool* send_stream) {
+ *stream_codec = *inst;
+
+ // Stream specific settings.
+ stream_codec->codecSpecific.VP8.numberOfTemporalLayers =
+ inst->simulcastStream[stream_index].numberOfTemporalLayers;
+ stream_codec->numberOfSimulcastStreams = 0;
+ stream_codec->width = inst->simulcastStream[stream_index].width;
+ stream_codec->height = inst->simulcastStream[stream_index].height;
+ stream_codec->maxBitrate = inst->simulcastStream[stream_index].maxBitrate;
+ stream_codec->minBitrate = inst->simulcastStream[stream_index].minBitrate;
+ stream_codec->qpMax = inst->simulcastStream[stream_index].qpMax;
+ // Settings that are based on stream/resolution.
+ if (stream_index == 0) {
+ // Settings for lowest spatial resolutions.
+ stream_codec->qpMax = kLowestResMaxQp;
+ }
+ if (!highest_resolution_stream) {
+ // For resolutions below CIF, set the codec |complexity| parameter to
+ // kComplexityHigher, which maps to cpu_used = -4.
+ int pixels_per_frame = stream_codec->width * stream_codec->height;
+ if (pixels_per_frame < 352 * 288) {
+ stream_codec->codecSpecific.VP8.complexity = webrtc::kComplexityHigher;
+ }
+ // Turn off denoising for all streams but the highest resolution.
+ stream_codec->codecSpecific.VP8.denoisingOn = false;
+ }
+ // TODO(ronghuawu): what to do with targetBitrate.
+
+ int stream_bitrate = GetStreamBitrate(stream_index, total_number_of_streams,
+ inst->startBitrate, send_stream);
+ stream_codec->startBitrate = stream_bitrate;
+}
+
+bool SimulcastEncoderAdapter::Initialized() const {
+ return !streaminfos_.empty();
+}
+
+void SimulcastEncoderAdapter::OnDroppedFrame() {
+ streaminfos_[0].encoder->OnDroppedFrame();
+}
+
+int SimulcastEncoderAdapter::GetTargetFramerate() {
+ return streaminfos_[0].encoder->GetTargetFramerate();
+}
+
+bool SimulcastEncoderAdapter::SupportsNativeHandle() const {
+ // We should not be calling this method before streaminfos_ are configured.
+ RTC_DCHECK(!streaminfos_.empty());
+ // TODO(pbos): Support textures when using more than one encoder.
+ if (streaminfos_.size() != 1)
+ return false;
+ return streaminfos_[0].encoder->SupportsNativeHandle();
+}
+
+} // namespace webrtc