
scoped_array is deprecated. This was done using a Chromium clang tool: http://src.chromium.org/viewvc/chrome/trunk/src/tools/clang/rewrite_scoped_ar... except for the few not-built-on-Linux files which were updated manually. TESTED=trybots BUG=2515 R=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12429004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5985 4adac7df-926f-26a2-2b94-8c16560cd09d
975 lines
32 KiB
C++
975 lines
32 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/video_engine/vie_encoder.h"
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#include <assert.h>
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#include <algorithm>
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#include "webrtc/common_video/interface/video_image.h"
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#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
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#include "webrtc/modules/pacing/include/paced_sender.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
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#include "webrtc/modules/utility/interface/process_thread.h"
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#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
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#include "webrtc/modules/video_coding/main/interface/video_coding.h"
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#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
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#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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#include "webrtc/system_wrappers/interface/logging.h"
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#include "webrtc/system_wrappers/interface/tick_util.h"
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#include "webrtc/system_wrappers/interface/trace_event.h"
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#include "webrtc/video_engine/include/vie_codec.h"
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#include "webrtc/video_engine/include/vie_image_process.h"
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#include "webrtc/frame_callback.h"
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#include "webrtc/video_engine/vie_defines.h"
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namespace webrtc {
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// Pace in kbits/s until we receive first estimate.
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static const int kInitialPace = 2000;
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// Pacing-rate relative to our target send rate.
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// Multiplicative factor that is applied to the target bitrate to calculate the
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// number of bytes that can be transmitted per interval.
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// Increasing this factor will result in lower delays in cases of bitrate
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// overshoots from the encoder.
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static const float kPaceMultiplier = 2.5f;
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// Margin on when we pause the encoder when the pacing buffer overflows relative
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// to the configured buffer delay.
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static const float kEncoderPausePacerMargin = 2.0f;
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// Don't stop the encoder unless the delay is above this configured value.
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static const int kMinPacingDelayMs = 200;
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// Allow packets to be transmitted in up to 2 times max video bitrate if the
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// bandwidth estimate allows it.
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// TODO(holmer): Expose transmission start, min and max bitrates in the
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// VideoEngine API and remove the kTransmissionMaxBitrateMultiplier.
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static const int kTransmissionMaxBitrateMultiplier = 2;
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static const float kStopPaddingThresholdMs = 2000;
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std::vector<uint32_t> AllocateStreamBitrates(
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uint32_t total_bitrate,
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const SimulcastStream* stream_configs,
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size_t number_of_streams) {
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if (number_of_streams == 0) {
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std::vector<uint32_t> stream_bitrates(1, 0);
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stream_bitrates[0] = total_bitrate;
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return stream_bitrates;
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}
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std::vector<uint32_t> stream_bitrates(number_of_streams, 0);
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uint32_t bitrate_remainder = total_bitrate;
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for (size_t i = 0; i < stream_bitrates.size() && bitrate_remainder > 0; ++i) {
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if (stream_configs[i].maxBitrate * 1000 > bitrate_remainder) {
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stream_bitrates[i] = bitrate_remainder;
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} else {
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stream_bitrates[i] = stream_configs[i].maxBitrate * 1000;
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}
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bitrate_remainder -= stream_bitrates[i];
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}
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return stream_bitrates;
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}
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class QMVideoSettingsCallback : public VCMQMSettingsCallback {
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public:
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explicit QMVideoSettingsCallback(VideoProcessingModule* vpm);
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~QMVideoSettingsCallback();
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// Update VPM with QM (quality modes: frame size & frame rate) settings.
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int32_t SetVideoQMSettings(const uint32_t frame_rate,
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const uint32_t width,
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const uint32_t height);
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private:
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VideoProcessingModule* vpm_;
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};
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class ViEBitrateObserver : public BitrateObserver {
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public:
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explicit ViEBitrateObserver(ViEEncoder* owner)
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: owner_(owner) {
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}
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virtual ~ViEBitrateObserver() {}
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// Implements BitrateObserver.
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virtual void OnNetworkChanged(const uint32_t bitrate_bps,
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const uint8_t fraction_lost,
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const uint32_t rtt) {
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owner_->OnNetworkChanged(bitrate_bps, fraction_lost, rtt);
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}
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private:
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ViEEncoder* owner_;
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};
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class ViEPacedSenderCallback : public PacedSender::Callback {
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public:
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explicit ViEPacedSenderCallback(ViEEncoder* owner)
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: owner_(owner) {
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}
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virtual ~ViEPacedSenderCallback() {}
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virtual bool TimeToSendPacket(uint32_t ssrc, uint16_t sequence_number,
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int64_t capture_time_ms, bool retransmission) {
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return owner_->TimeToSendPacket(ssrc, sequence_number, capture_time_ms,
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retransmission);
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}
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virtual int TimeToSendPadding(int bytes) {
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return owner_->TimeToSendPadding(bytes);
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}
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private:
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ViEEncoder* owner_;
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};
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ViEEncoder::ViEEncoder(int32_t engine_id,
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int32_t channel_id,
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uint32_t number_of_cores,
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const Config& config,
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ProcessThread& module_process_thread,
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BitrateController* bitrate_controller)
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: engine_id_(engine_id),
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channel_id_(channel_id),
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number_of_cores_(number_of_cores),
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vcm_(*webrtc::VideoCodingModule::Create()),
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vpm_(*webrtc::VideoProcessingModule::Create(ViEModuleId(engine_id,
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channel_id))),
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callback_cs_(CriticalSectionWrapper::CreateCriticalSection()),
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data_cs_(CriticalSectionWrapper::CreateCriticalSection()),
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bitrate_controller_(bitrate_controller),
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time_of_last_incoming_frame_ms_(0),
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send_padding_(false),
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min_transmit_bitrate_kbps_(0),
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target_delay_ms_(0),
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network_is_transmitting_(true),
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encoder_paused_(false),
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encoder_paused_and_dropped_frame_(false),
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fec_enabled_(false),
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nack_enabled_(false),
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codec_observer_(NULL),
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effect_filter_(NULL),
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module_process_thread_(module_process_thread),
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has_received_sli_(false),
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picture_id_sli_(0),
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has_received_rpsi_(false),
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picture_id_rpsi_(0),
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qm_callback_(NULL),
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video_suspended_(false),
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pre_encode_callback_(NULL) {
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RtpRtcp::Configuration configuration;
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configuration.id = ViEModuleId(engine_id_, channel_id_);
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configuration.audio = false; // Video.
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default_rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(configuration));
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bitrate_observer_.reset(new ViEBitrateObserver(this));
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pacing_callback_.reset(new ViEPacedSenderCallback(this));
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paced_sender_.reset(
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new PacedSender(pacing_callback_.get(), kInitialPace, kPaceMultiplier));
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}
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bool ViEEncoder::Init() {
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if (vcm_.InitializeSender() != 0) {
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return false;
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}
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vpm_.EnableTemporalDecimation(true);
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// Enable/disable content analysis: off by default for now.
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vpm_.EnableContentAnalysis(false);
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if (module_process_thread_.RegisterModule(&vcm_) != 0 ||
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module_process_thread_.RegisterModule(default_rtp_rtcp_.get()) != 0 ||
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module_process_thread_.RegisterModule(paced_sender_.get()) != 0) {
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return false;
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}
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if (qm_callback_) {
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delete qm_callback_;
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}
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qm_callback_ = new QMVideoSettingsCallback(&vpm_);
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#ifdef VIDEOCODEC_VP8
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VideoCodec video_codec;
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if (vcm_.Codec(webrtc::kVideoCodecVP8, &video_codec) != VCM_OK) {
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return false;
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}
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{
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CriticalSectionScoped cs(data_cs_.get());
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send_padding_ = video_codec.numberOfSimulcastStreams > 1;
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}
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if (vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
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default_rtp_rtcp_->MaxDataPayloadLength()) != 0) {
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return false;
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}
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if (default_rtp_rtcp_->RegisterSendPayload(video_codec) != 0) {
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return false;
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}
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#else
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VideoCodec video_codec;
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if (vcm_.Codec(webrtc::kVideoCodecI420, &video_codec) == VCM_OK) {
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{
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CriticalSectionScoped cs(data_cs_.get());
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send_padding_ = video_codec.numberOfSimulcastStreams > 1;
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}
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vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
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default_rtp_rtcp_->MaxDataPayloadLength());
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default_rtp_rtcp_->RegisterSendPayload(video_codec);
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} else {
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return false;
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}
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#endif
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if (vcm_.RegisterTransportCallback(this) != 0) {
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return false;
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}
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if (vcm_.RegisterSendStatisticsCallback(this) != 0) {
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return false;
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}
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if (vcm_.RegisterVideoQMCallback(qm_callback_) != 0) {
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return false;
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}
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return true;
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}
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ViEEncoder::~ViEEncoder() {
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if (bitrate_controller_) {
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bitrate_controller_->RemoveBitrateObserver(bitrate_observer_.get());
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}
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module_process_thread_.DeRegisterModule(&vcm_);
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module_process_thread_.DeRegisterModule(&vpm_);
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module_process_thread_.DeRegisterModule(default_rtp_rtcp_.get());
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module_process_thread_.DeRegisterModule(paced_sender_.get());
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VideoCodingModule::Destroy(&vcm_);
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VideoProcessingModule::Destroy(&vpm_);
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delete qm_callback_;
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}
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int ViEEncoder::Owner() const {
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return channel_id_;
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}
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void ViEEncoder::SetNetworkTransmissionState(bool is_transmitting) {
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{
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CriticalSectionScoped cs(data_cs_.get());
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network_is_transmitting_ = is_transmitting;
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}
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if (is_transmitting) {
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paced_sender_->Resume();
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} else {
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paced_sender_->Pause();
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}
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}
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void ViEEncoder::Pause() {
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CriticalSectionScoped cs(data_cs_.get());
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encoder_paused_ = true;
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}
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void ViEEncoder::Restart() {
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CriticalSectionScoped cs(data_cs_.get());
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encoder_paused_ = false;
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}
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uint8_t ViEEncoder::NumberOfCodecs() {
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return vcm_.NumberOfCodecs();
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}
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int32_t ViEEncoder::GetCodec(uint8_t list_index, VideoCodec* video_codec) {
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if (vcm_.Codec(list_index, video_codec) != 0) {
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return -1;
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}
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return 0;
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}
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int32_t ViEEncoder::RegisterExternalEncoder(webrtc::VideoEncoder* encoder,
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uint8_t pl_type,
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bool internal_source) {
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if (encoder == NULL)
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return -1;
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if (vcm_.RegisterExternalEncoder(encoder, pl_type, internal_source) !=
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VCM_OK) {
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return -1;
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}
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return 0;
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}
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int32_t ViEEncoder::DeRegisterExternalEncoder(uint8_t pl_type) {
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webrtc::VideoCodec current_send_codec;
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if (vcm_.SendCodec(¤t_send_codec) == VCM_OK) {
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uint32_t current_bitrate_bps = 0;
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if (vcm_.Bitrate(¤t_bitrate_bps) != 0) {
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LOG(LS_WARNING) << "Failed to get the current encoder target bitrate.";
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}
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current_send_codec.startBitrate = (current_bitrate_bps + 500) / 1000;
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}
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if (vcm_.RegisterExternalEncoder(NULL, pl_type) != VCM_OK) {
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return -1;
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}
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// If the external encoder is the current send codec, use vcm internal
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// encoder.
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if (current_send_codec.plType == pl_type) {
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uint16_t max_data_payload_length =
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default_rtp_rtcp_->MaxDataPayloadLength();
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{
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CriticalSectionScoped cs(data_cs_.get());
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send_padding_ = current_send_codec.numberOfSimulcastStreams > 1;
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}
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// TODO(mflodman): Unfortunately the VideoCodec that VCM has cached a
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// raw pointer to an |extra_options| that's long gone. Clearing it here is
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// a hack to prevent the following code from crashing. This should be fixed
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// for realz. https://code.google.com/p/chromium/issues/detail?id=348222
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current_send_codec.extra_options = NULL;
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if (vcm_.RegisterSendCodec(¤t_send_codec, number_of_cores_,
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max_data_payload_length) != VCM_OK) {
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return -1;
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}
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}
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return 0;
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}
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int32_t ViEEncoder::SetEncoder(const webrtc::VideoCodec& video_codec) {
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// Setting target width and height for VPM.
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if (vpm_.SetTargetResolution(video_codec.width, video_codec.height,
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video_codec.maxFramerate) != VPM_OK) {
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return -1;
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}
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if (default_rtp_rtcp_->RegisterSendPayload(video_codec) != 0) {
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return -1;
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}
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// Convert from kbps to bps.
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std::vector<uint32_t> stream_bitrates = AllocateStreamBitrates(
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video_codec.startBitrate * 1000,
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video_codec.simulcastStream,
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video_codec.numberOfSimulcastStreams);
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default_rtp_rtcp_->SetTargetSendBitrate(stream_bitrates);
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uint16_t max_data_payload_length =
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default_rtp_rtcp_->MaxDataPayloadLength();
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{
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CriticalSectionScoped cs(data_cs_.get());
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send_padding_ = video_codec.numberOfSimulcastStreams > 1;
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}
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if (vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
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max_data_payload_length) != VCM_OK) {
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return -1;
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}
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// Set this module as sending right away, let the slave module in the channel
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// start and stop sending.
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if (default_rtp_rtcp_->Sending() == false) {
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if (default_rtp_rtcp_->SetSendingStatus(true) != 0) {
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return -1;
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}
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}
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bitrate_controller_->SetBitrateObserver(bitrate_observer_.get(),
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video_codec.startBitrate * 1000,
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video_codec.minBitrate * 1000,
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kTransmissionMaxBitrateMultiplier *
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video_codec.maxBitrate * 1000);
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CriticalSectionScoped crit(data_cs_.get());
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int pad_up_to_bitrate_kbps = video_codec.startBitrate;
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if (pad_up_to_bitrate_kbps < min_transmit_bitrate_kbps_)
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pad_up_to_bitrate_kbps = min_transmit_bitrate_kbps_;
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paced_sender_->UpdateBitrate(kPaceMultiplier * video_codec.startBitrate,
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pad_up_to_bitrate_kbps);
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return 0;
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}
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int32_t ViEEncoder::GetEncoder(VideoCodec* video_codec) {
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if (vcm_.SendCodec(video_codec) != 0) {
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return -1;
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}
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return 0;
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}
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int32_t ViEEncoder::GetCodecConfigParameters(
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unsigned char config_parameters[kConfigParameterSize],
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unsigned char& config_parameters_size) {
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int32_t num_parameters =
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vcm_.CodecConfigParameters(config_parameters, kConfigParameterSize);
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if (num_parameters <= 0) {
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config_parameters_size = 0;
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return -1;
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}
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config_parameters_size = static_cast<unsigned char>(num_parameters);
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return 0;
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}
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int32_t ViEEncoder::ScaleInputImage(bool enable) {
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VideoFrameResampling resampling_mode = kFastRescaling;
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// TODO(mflodman) What?
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if (enable) {
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// kInterpolation is currently not supported.
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LOG_F(LS_ERROR) << "Not supported.";
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return -1;
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}
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vpm_.SetInputFrameResampleMode(resampling_mode);
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return 0;
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}
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bool ViEEncoder::TimeToSendPacket(uint32_t ssrc,
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uint16_t sequence_number,
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int64_t capture_time_ms,
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bool retransmission) {
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return default_rtp_rtcp_->TimeToSendPacket(ssrc, sequence_number,
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capture_time_ms, retransmission);
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}
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int ViEEncoder::TimeToSendPadding(int bytes) {
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bool send_padding;
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{
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CriticalSectionScoped cs(data_cs_.get());
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send_padding =
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send_padding_ || video_suspended_ || min_transmit_bitrate_kbps_ > 0;
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}
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if (send_padding) {
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return default_rtp_rtcp_->TimeToSendPadding(bytes);
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}
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return 0;
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}
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bool ViEEncoder::EncoderPaused() const {
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// Pause video if paused by caller or as long as the network is down or the
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// pacer queue has grown too large in buffered mode.
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if (encoder_paused_) {
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return true;
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}
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if (target_delay_ms_ > 0) {
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// Buffered mode.
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// TODO(pwestin): Workaround until nack is configured as a time and not
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// number of packets.
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return paced_sender_->QueueInMs() >=
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std::max(static_cast<int>(target_delay_ms_ * kEncoderPausePacerMargin),
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kMinPacingDelayMs);
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}
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return !network_is_transmitting_;
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}
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RtpRtcp* ViEEncoder::SendRtpRtcpModule() {
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return default_rtp_rtcp_.get();
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}
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void ViEEncoder::DeliverFrame(int id,
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I420VideoFrame* video_frame,
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int num_csrcs,
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const uint32_t CSRC[kRtpCsrcSize]) {
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if (default_rtp_rtcp_->SendingMedia() == false) {
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// We've paused or we have no channels attached, don't encode.
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return;
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|
}
|
|
{
|
|
CriticalSectionScoped cs(data_cs_.get());
|
|
time_of_last_incoming_frame_ms_ = TickTime::MillisecondTimestamp();
|
|
if (EncoderPaused()) {
|
|
if (!encoder_paused_and_dropped_frame_) {
|
|
TRACE_EVENT_ASYNC_BEGIN0("webrtc", "EncoderPaused", this);
|
|
}
|
|
encoder_paused_and_dropped_frame_ = true;
|
|
return;
|
|
}
|
|
if (encoder_paused_and_dropped_frame_) {
|
|
TRACE_EVENT_ASYNC_END0("webrtc", "EncoderPaused", this);
|
|
}
|
|
encoder_paused_and_dropped_frame_ = false;
|
|
}
|
|
|
|
// Convert render time, in ms, to RTP timestamp.
|
|
const int kMsToRtpTimestamp = 90;
|
|
const uint32_t time_stamp =
|
|
kMsToRtpTimestamp *
|
|
static_cast<uint32_t>(video_frame->render_time_ms());
|
|
|
|
TRACE_EVENT_ASYNC_STEP0("webrtc", "Video", video_frame->render_time_ms(),
|
|
"Encode");
|
|
video_frame->set_timestamp(time_stamp);
|
|
{
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
if (effect_filter_) {
|
|
unsigned int length = CalcBufferSize(kI420,
|
|
video_frame->width(),
|
|
video_frame->height());
|
|
scoped_ptr<uint8_t[]> video_buffer(new uint8_t[length]);
|
|
ExtractBuffer(*video_frame, length, video_buffer.get());
|
|
effect_filter_->Transform(length,
|
|
video_buffer.get(),
|
|
video_frame->ntp_time_ms(),
|
|
video_frame->timestamp(),
|
|
video_frame->width(),
|
|
video_frame->height());
|
|
}
|
|
}
|
|
|
|
// Make sure the CSRC list is correct.
|
|
if (num_csrcs > 0) {
|
|
uint32_t tempCSRC[kRtpCsrcSize];
|
|
for (int i = 0; i < num_csrcs; i++) {
|
|
if (CSRC[i] == 1) {
|
|
tempCSRC[i] = default_rtp_rtcp_->SSRC();
|
|
} else {
|
|
tempCSRC[i] = CSRC[i];
|
|
}
|
|
}
|
|
default_rtp_rtcp_->SetCSRCs(tempCSRC, (uint8_t) num_csrcs);
|
|
}
|
|
// Pass frame via preprocessor.
|
|
I420VideoFrame* decimated_frame = NULL;
|
|
const int ret = vpm_.PreprocessFrame(*video_frame, &decimated_frame);
|
|
if (ret == 1) {
|
|
// Drop this frame.
|
|
return;
|
|
}
|
|
if (ret != VPM_OK) {
|
|
return;
|
|
}
|
|
// Frame was not sampled => use original.
|
|
if (decimated_frame == NULL) {
|
|
decimated_frame = video_frame;
|
|
}
|
|
|
|
{
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
if (pre_encode_callback_)
|
|
pre_encode_callback_->FrameCallback(decimated_frame);
|
|
}
|
|
|
|
#ifdef VIDEOCODEC_VP8
|
|
if (vcm_.SendCodec() == webrtc::kVideoCodecVP8) {
|
|
webrtc::CodecSpecificInfo codec_specific_info;
|
|
codec_specific_info.codecType = webrtc::kVideoCodecVP8;
|
|
codec_specific_info.codecSpecific.VP8.hasReceivedRPSI =
|
|
has_received_rpsi_;
|
|
codec_specific_info.codecSpecific.VP8.hasReceivedSLI =
|
|
has_received_sli_;
|
|
codec_specific_info.codecSpecific.VP8.pictureIdRPSI =
|
|
picture_id_rpsi_;
|
|
codec_specific_info.codecSpecific.VP8.pictureIdSLI =
|
|
picture_id_sli_;
|
|
has_received_sli_ = false;
|
|
has_received_rpsi_ = false;
|
|
|
|
vcm_.AddVideoFrame(*decimated_frame, vpm_.ContentMetrics(),
|
|
&codec_specific_info);
|
|
return;
|
|
}
|
|
#endif
|
|
vcm_.AddVideoFrame(*decimated_frame);
|
|
}
|
|
|
|
void ViEEncoder::DelayChanged(int id, int frame_delay) {
|
|
default_rtp_rtcp_->SetCameraDelay(frame_delay);
|
|
}
|
|
|
|
int ViEEncoder::GetPreferedFrameSettings(int* width,
|
|
int* height,
|
|
int* frame_rate) {
|
|
webrtc::VideoCodec video_codec;
|
|
memset(&video_codec, 0, sizeof(video_codec));
|
|
if (vcm_.SendCodec(&video_codec) != VCM_OK) {
|
|
return -1;
|
|
}
|
|
|
|
*width = video_codec.width;
|
|
*height = video_codec.height;
|
|
*frame_rate = video_codec.maxFramerate;
|
|
return 0;
|
|
}
|
|
|
|
int ViEEncoder::SendKeyFrame() {
|
|
return vcm_.IntraFrameRequest(0);
|
|
}
|
|
|
|
int32_t ViEEncoder::SendCodecStatistics(
|
|
uint32_t* num_key_frames, uint32_t* num_delta_frames) {
|
|
webrtc::VCMFrameCount sent_frames;
|
|
if (vcm_.SentFrameCount(sent_frames) != VCM_OK) {
|
|
return -1;
|
|
}
|
|
*num_key_frames = sent_frames.numKeyFrames;
|
|
*num_delta_frames = sent_frames.numDeltaFrames;
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEEncoder::PacerQueuingDelayMs() const {
|
|
return paced_sender_->QueueInMs();
|
|
}
|
|
|
|
int32_t ViEEncoder::EstimatedSendBandwidth(
|
|
uint32_t* available_bandwidth) const {
|
|
if (!bitrate_controller_->AvailableBandwidth(available_bandwidth)) {
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int ViEEncoder::CodecTargetBitrate(uint32_t* bitrate) const {
|
|
if (vcm_.Bitrate(bitrate) != 0)
|
|
return -1;
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEEncoder::UpdateProtectionMethod(bool enable_nack) {
|
|
bool fec_enabled = false;
|
|
uint8_t dummy_ptype_red = 0;
|
|
uint8_t dummy_ptypeFEC = 0;
|
|
|
|
// Updated protection method to VCM to get correct packetization sizes.
|
|
// FEC has larger overhead than NACK -> set FEC if used.
|
|
int32_t error = default_rtp_rtcp_->GenericFECStatus(fec_enabled,
|
|
dummy_ptype_red,
|
|
dummy_ptypeFEC);
|
|
if (error) {
|
|
return -1;
|
|
}
|
|
if (fec_enabled_ == fec_enabled && nack_enabled_ == enable_nack) {
|
|
// No change needed, we're already in correct state.
|
|
return 0;
|
|
}
|
|
fec_enabled_ = fec_enabled;
|
|
nack_enabled_ = enable_nack;
|
|
|
|
// Set Video Protection for VCM.
|
|
if (fec_enabled && nack_enabled_) {
|
|
vcm_.SetVideoProtection(webrtc::kProtectionNackFEC, true);
|
|
} else {
|
|
vcm_.SetVideoProtection(webrtc::kProtectionFEC, fec_enabled_);
|
|
vcm_.SetVideoProtection(webrtc::kProtectionNackSender, nack_enabled_);
|
|
vcm_.SetVideoProtection(webrtc::kProtectionNackFEC, false);
|
|
}
|
|
|
|
if (fec_enabled_ || nack_enabled_) {
|
|
vcm_.RegisterProtectionCallback(this);
|
|
// The send codec must be registered to set correct MTU.
|
|
webrtc::VideoCodec codec;
|
|
if (vcm_.SendCodec(&codec) == 0) {
|
|
uint16_t max_pay_load = default_rtp_rtcp_->MaxDataPayloadLength();
|
|
uint32_t current_bitrate_bps = 0;
|
|
if (vcm_.Bitrate(¤t_bitrate_bps) != 0) {
|
|
LOG_F(LS_WARNING) <<
|
|
"Failed to get the current encoder target bitrate.";
|
|
}
|
|
// Convert to start bitrate in kbps.
|
|
codec.startBitrate = (current_bitrate_bps + 500) / 1000;
|
|
if (vcm_.RegisterSendCodec(&codec, number_of_cores_, max_pay_load) != 0) {
|
|
return -1;
|
|
}
|
|
}
|
|
return 0;
|
|
} else {
|
|
// FEC and NACK are disabled.
|
|
vcm_.RegisterProtectionCallback(NULL);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
void ViEEncoder::SetSenderBufferingMode(int target_delay_ms) {
|
|
{
|
|
CriticalSectionScoped cs(data_cs_.get());
|
|
target_delay_ms_ = target_delay_ms;
|
|
}
|
|
if (target_delay_ms > 0) {
|
|
// Disable external frame-droppers.
|
|
vcm_.EnableFrameDropper(false);
|
|
vpm_.EnableTemporalDecimation(false);
|
|
// We don't put any limits on the pacer queue when running in buffered mode
|
|
// since the encoder will be paused if the queue grow too large.
|
|
paced_sender_->set_max_queue_length_ms(-1);
|
|
} else {
|
|
// Real-time mode - enable frame droppers.
|
|
vpm_.EnableTemporalDecimation(true);
|
|
vcm_.EnableFrameDropper(true);
|
|
paced_sender_->set_max_queue_length_ms(
|
|
PacedSender::kDefaultMaxQueueLengthMs);
|
|
}
|
|
}
|
|
|
|
int32_t ViEEncoder::SendData(
|
|
const FrameType frame_type,
|
|
const uint8_t payload_type,
|
|
const uint32_t time_stamp,
|
|
int64_t capture_time_ms,
|
|
const uint8_t* payload_data,
|
|
const uint32_t payload_size,
|
|
const webrtc::RTPFragmentationHeader& fragmentation_header,
|
|
const RTPVideoHeader* rtp_video_hdr) {
|
|
// New encoded data, hand over to the rtp module.
|
|
return default_rtp_rtcp_->SendOutgoingData(frame_type,
|
|
payload_type,
|
|
time_stamp,
|
|
capture_time_ms,
|
|
payload_data,
|
|
payload_size,
|
|
&fragmentation_header,
|
|
rtp_video_hdr);
|
|
}
|
|
|
|
int32_t ViEEncoder::ProtectionRequest(
|
|
const FecProtectionParams* delta_fec_params,
|
|
const FecProtectionParams* key_fec_params,
|
|
uint32_t* sent_video_rate_bps,
|
|
uint32_t* sent_nack_rate_bps,
|
|
uint32_t* sent_fec_rate_bps) {
|
|
default_rtp_rtcp_->SetFecParameters(delta_fec_params, key_fec_params);
|
|
default_rtp_rtcp_->BitrateSent(NULL, sent_video_rate_bps, sent_fec_rate_bps,
|
|
sent_nack_rate_bps);
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEEncoder::SendStatistics(const uint32_t bit_rate,
|
|
const uint32_t frame_rate) {
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
if (codec_observer_) {
|
|
codec_observer_->OutgoingRate(channel_id_, frame_rate, bit_rate);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEEncoder::RegisterCodecObserver(ViEEncoderObserver* observer) {
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
if (observer && codec_observer_) {
|
|
LOG_F(LS_ERROR) << "Observer already set.";
|
|
return -1;
|
|
}
|
|
codec_observer_ = observer;
|
|
return 0;
|
|
}
|
|
|
|
void ViEEncoder::OnReceivedSLI(uint32_t /*ssrc*/,
|
|
uint8_t picture_id) {
|
|
picture_id_sli_ = picture_id;
|
|
has_received_sli_ = true;
|
|
}
|
|
|
|
void ViEEncoder::OnReceivedRPSI(uint32_t /*ssrc*/,
|
|
uint64_t picture_id) {
|
|
picture_id_rpsi_ = picture_id;
|
|
has_received_rpsi_ = true;
|
|
}
|
|
|
|
void ViEEncoder::OnReceivedIntraFrameRequest(uint32_t ssrc) {
|
|
// Key frame request from remote side, signal to VCM.
|
|
TRACE_EVENT0("webrtc", "OnKeyFrameRequest");
|
|
|
|
int idx = 0;
|
|
{
|
|
CriticalSectionScoped cs(data_cs_.get());
|
|
std::map<unsigned int, int>::iterator stream_it = ssrc_streams_.find(ssrc);
|
|
if (stream_it == ssrc_streams_.end()) {
|
|
LOG_F(LS_WARNING) << "ssrc not found: " << ssrc << ", map size "
|
|
<< ssrc_streams_.size();
|
|
return;
|
|
}
|
|
std::map<unsigned int, int64_t>::iterator time_it =
|
|
time_last_intra_request_ms_.find(ssrc);
|
|
if (time_it == time_last_intra_request_ms_.end()) {
|
|
time_last_intra_request_ms_[ssrc] = 0;
|
|
}
|
|
|
|
int64_t now = TickTime::MillisecondTimestamp();
|
|
if (time_last_intra_request_ms_[ssrc] + kViEMinKeyRequestIntervalMs > now) {
|
|
return;
|
|
}
|
|
time_last_intra_request_ms_[ssrc] = now;
|
|
idx = stream_it->second;
|
|
}
|
|
// Release the critsect before triggering key frame.
|
|
vcm_.IntraFrameRequest(idx);
|
|
}
|
|
|
|
void ViEEncoder::OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) {
|
|
CriticalSectionScoped cs(data_cs_.get());
|
|
std::map<unsigned int, int>::iterator it = ssrc_streams_.find(old_ssrc);
|
|
if (it == ssrc_streams_.end()) {
|
|
return;
|
|
}
|
|
|
|
ssrc_streams_[new_ssrc] = it->second;
|
|
ssrc_streams_.erase(it);
|
|
|
|
std::map<unsigned int, int64_t>::iterator time_it =
|
|
time_last_intra_request_ms_.find(old_ssrc);
|
|
int64_t last_intra_request_ms = 0;
|
|
if (time_it != time_last_intra_request_ms_.end()) {
|
|
last_intra_request_ms = time_it->second;
|
|
time_last_intra_request_ms_.erase(time_it);
|
|
}
|
|
time_last_intra_request_ms_[new_ssrc] = last_intra_request_ms;
|
|
}
|
|
|
|
bool ViEEncoder::SetSsrcs(const std::list<unsigned int>& ssrcs) {
|
|
VideoCodec codec;
|
|
if (vcm_.SendCodec(&codec) != 0)
|
|
return false;
|
|
|
|
if (codec.numberOfSimulcastStreams > 0 &&
|
|
ssrcs.size() != codec.numberOfSimulcastStreams) {
|
|
return false;
|
|
}
|
|
|
|
CriticalSectionScoped cs(data_cs_.get());
|
|
ssrc_streams_.clear();
|
|
time_last_intra_request_ms_.clear();
|
|
int idx = 0;
|
|
for (std::list<unsigned int>::const_iterator it = ssrcs.begin();
|
|
it != ssrcs.end(); ++it, ++idx) {
|
|
unsigned int ssrc = *it;
|
|
ssrc_streams_[ssrc] = idx;
|
|
}
|
|
return true;
|
|
}
|
|
|
|
void ViEEncoder::SetMinTransmitBitrate(int min_transmit_bitrate_kbps) {
|
|
assert(min_transmit_bitrate_kbps >= 0);
|
|
CriticalSectionScoped crit(data_cs_.get());
|
|
min_transmit_bitrate_kbps_ = min_transmit_bitrate_kbps;
|
|
}
|
|
|
|
// Called from ViEBitrateObserver.
|
|
void ViEEncoder::OnNetworkChanged(const uint32_t bitrate_bps,
|
|
const uint8_t fraction_lost,
|
|
const uint32_t round_trip_time_ms) {
|
|
LOG(LS_VERBOSE) << "OnNetworkChanged, bitrate" << bitrate_bps
|
|
<< " packet loss " << fraction_lost
|
|
<< " rtt " << round_trip_time_ms;
|
|
vcm_.SetChannelParameters(bitrate_bps, fraction_lost, round_trip_time_ms);
|
|
bool video_is_suspended = vcm_.VideoSuspended();
|
|
int bitrate_kbps = bitrate_bps / 1000;
|
|
VideoCodec send_codec;
|
|
if (vcm_.SendCodec(&send_codec) != 0) {
|
|
return;
|
|
}
|
|
SimulcastStream* stream_configs = send_codec.simulcastStream;
|
|
// Allocate the bandwidth between the streams.
|
|
std::vector<uint32_t> stream_bitrates = AllocateStreamBitrates(
|
|
bitrate_bps,
|
|
stream_configs,
|
|
send_codec.numberOfSimulcastStreams);
|
|
// Find the max amount of padding we can allow ourselves to send at this
|
|
// point, based on which streams are currently active and what our current
|
|
// available bandwidth is.
|
|
int pad_up_to_bitrate_kbps = 0;
|
|
if (send_codec.numberOfSimulcastStreams == 0) {
|
|
pad_up_to_bitrate_kbps = send_codec.minBitrate;
|
|
} else {
|
|
pad_up_to_bitrate_kbps =
|
|
stream_configs[send_codec.numberOfSimulcastStreams - 1].minBitrate;
|
|
for (int i = 0; i < send_codec.numberOfSimulcastStreams - 1; ++i) {
|
|
pad_up_to_bitrate_kbps += stream_configs[i].targetBitrate;
|
|
}
|
|
}
|
|
|
|
// Disable padding if only sending one stream and video isn't suspended and
|
|
// min-transmit bitrate isn't used (applied later).
|
|
if (!video_is_suspended && send_codec.numberOfSimulcastStreams <= 1)
|
|
pad_up_to_bitrate_kbps = 0;
|
|
|
|
{
|
|
CriticalSectionScoped cs(data_cs_.get());
|
|
// The amount of padding should decay to zero if no frames are being
|
|
// captured unless a min-transmit bitrate is used.
|
|
int64_t now_ms = TickTime::MillisecondTimestamp();
|
|
if (now_ms - time_of_last_incoming_frame_ms_ > kStopPaddingThresholdMs)
|
|
pad_up_to_bitrate_kbps = 0;
|
|
|
|
// Pad up to min bitrate.
|
|
if (pad_up_to_bitrate_kbps < min_transmit_bitrate_kbps_)
|
|
pad_up_to_bitrate_kbps = min_transmit_bitrate_kbps_;
|
|
|
|
// Padding may never exceed bitrate estimate.
|
|
if (pad_up_to_bitrate_kbps > bitrate_kbps)
|
|
pad_up_to_bitrate_kbps = bitrate_kbps;
|
|
|
|
paced_sender_->UpdateBitrate(kPaceMultiplier * bitrate_kbps,
|
|
pad_up_to_bitrate_kbps);
|
|
default_rtp_rtcp_->SetTargetSendBitrate(stream_bitrates);
|
|
if (video_suspended_ == video_is_suspended)
|
|
return;
|
|
video_suspended_ = video_is_suspended;
|
|
}
|
|
|
|
// Video suspend-state changed, inform codec observer.
|
|
CriticalSectionScoped crit(callback_cs_.get());
|
|
if (codec_observer_) {
|
|
LOG(LS_INFO) << "Video suspended " << video_is_suspended
|
|
<< " for channel " << channel_id_;
|
|
codec_observer_->SuspendChange(channel_id_, video_is_suspended);
|
|
}
|
|
}
|
|
|
|
PacedSender* ViEEncoder::GetPacedSender() {
|
|
return paced_sender_.get();
|
|
}
|
|
|
|
int32_t ViEEncoder::RegisterEffectFilter(ViEEffectFilter* effect_filter) {
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
if (effect_filter != NULL && effect_filter_ != NULL) {
|
|
LOG_F(LS_ERROR) << "Filter already set.";
|
|
return -1;
|
|
}
|
|
effect_filter_ = effect_filter;
|
|
return 0;
|
|
}
|
|
|
|
int ViEEncoder::StartDebugRecording(const char* fileNameUTF8) {
|
|
return vcm_.StartDebugRecording(fileNameUTF8);
|
|
}
|
|
|
|
int ViEEncoder::StopDebugRecording() {
|
|
return vcm_.StopDebugRecording();
|
|
}
|
|
|
|
void ViEEncoder::SuspendBelowMinBitrate() {
|
|
vcm_.SuspendBelowMinBitrate();
|
|
bitrate_controller_->EnforceMinBitrate(false);
|
|
}
|
|
|
|
void ViEEncoder::RegisterPreEncodeCallback(
|
|
I420FrameCallback* pre_encode_callback) {
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
pre_encode_callback_ = pre_encode_callback;
|
|
}
|
|
|
|
void ViEEncoder::DeRegisterPreEncodeCallback() {
|
|
CriticalSectionScoped cs(callback_cs_.get());
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|
pre_encode_callback_ = NULL;
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|
}
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|
|
|
void ViEEncoder::RegisterPostEncodeImageCallback(
|
|
EncodedImageCallback* post_encode_callback) {
|
|
vcm_.RegisterPostEncodeImageCallback(post_encode_callback);
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|
}
|
|
|
|
void ViEEncoder::DeRegisterPostEncodeImageCallback() {
|
|
vcm_.RegisterPostEncodeImageCallback(NULL);
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|
}
|
|
|
|
QMVideoSettingsCallback::QMVideoSettingsCallback(VideoProcessingModule* vpm)
|
|
: vpm_(vpm) {
|
|
}
|
|
|
|
QMVideoSettingsCallback::~QMVideoSettingsCallback() {
|
|
}
|
|
|
|
int32_t QMVideoSettingsCallback::SetVideoQMSettings(
|
|
const uint32_t frame_rate,
|
|
const uint32_t width,
|
|
const uint32_t height) {
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|
return vpm_->SetTargetResolution(width, height, frame_rate);
|
|
}
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|
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} // namespace webrtc
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