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niklase@google.com470e71d2011-07-07 08:21:25 +00001/*
stefan@webrtc.org07b45a52012-02-02 08:37:48 +00002 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
niklase@google.com470e71d2011-07-07 08:21:25 +00003 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
pbos@webrtc.orgf5d4cb12013-05-17 13:44:48 +000011#include "webrtc/video_engine/vie_encoder.h"
mflodman@webrtc.org84d17832011-12-01 17:02:23 +000012
pbos@webrtc.org12dc1a32013-08-05 16:22:53 +000013#include <assert.h>
14
stefan@webrtc.orgc3cc3752013-06-04 09:36:56 +000015#include <algorithm>
niklase@google.com470e71d2011-07-07 08:21:25 +000016
sprang@webrtc.org40709352013-11-26 11:41:59 +000017#include "webrtc/common_video/interface/video_image.h"
pbos@webrtc.orgf5d4cb12013-05-17 13:44:48 +000018#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
19#include "webrtc/modules/pacing/include/paced_sender.h"
20#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
21#include "webrtc/modules/utility/interface/process_thread.h"
22#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
23#include "webrtc/modules/video_coding/main/interface/video_coding.h"
24#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
sprang@webrtc.org40709352013-11-26 11:41:59 +000025#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
stefan@webrtc.org88e0dda2014-07-04 09:20:42 +000026#include "webrtc/system_wrappers/interface/clock.h"
pbos@webrtc.orgf5d4cb12013-05-17 13:44:48 +000027#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
28#include "webrtc/system_wrappers/interface/logging.h"
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +000029#include "webrtc/system_wrappers/interface/metrics.h"
pbos@webrtc.orgf5d4cb12013-05-17 13:44:48 +000030#include "webrtc/system_wrappers/interface/tick_util.h"
pbos@webrtc.orgf5d4cb12013-05-17 13:44:48 +000031#include "webrtc/system_wrappers/interface/trace_event.h"
32#include "webrtc/video_engine/include/vie_codec.h"
33#include "webrtc/video_engine/include/vie_image_process.h"
pbos@webrtc.org16e03b72013-10-28 16:32:01 +000034#include "webrtc/frame_callback.h"
pbos@webrtc.orgf5d4cb12013-05-17 13:44:48 +000035#include "webrtc/video_engine/vie_defines.h"
niklase@google.com470e71d2011-07-07 08:21:25 +000036
niklase@google.com470e71d2011-07-07 08:21:25 +000037namespace webrtc {
38
pwestin@webrtc.org52b4e882013-05-02 19:02:17 +000039// Margin on when we pause the encoder when the pacing buffer overflows relative
40// to the configured buffer delay.
41static const float kEncoderPausePacerMargin = 2.0f;
42
pwestin@webrtc.org91563e42013-04-25 22:20:08 +000043// Don't stop the encoder unless the delay is above this configured value.
44static const int kMinPacingDelayMs = 200;
45
stefan@webrtc.orgbfacda62013-03-27 16:36:01 +000046// Allow packets to be transmitted in up to 2 times max video bitrate if the
47// bandwidth estimate allows it.
48// TODO(holmer): Expose transmission start, min and max bitrates in the
49// VideoEngine API and remove the kTransmissionMaxBitrateMultiplier.
50static const int kTransmissionMaxBitrateMultiplier = 2;
pwestin@webrtc.org571a1c02012-11-13 21:12:39 +000051
stefan@webrtc.org3e005052013-10-18 15:05:29 +000052static const float kStopPaddingThresholdMs = 2000;
53
stefan@webrtc.orgb2c8a952013-09-06 13:58:01 +000054std::vector<uint32_t> AllocateStreamBitrates(
55 uint32_t total_bitrate,
56 const SimulcastStream* stream_configs,
57 size_t number_of_streams) {
58 if (number_of_streams == 0) {
59 std::vector<uint32_t> stream_bitrates(1, 0);
60 stream_bitrates[0] = total_bitrate;
61 return stream_bitrates;
62 }
63 std::vector<uint32_t> stream_bitrates(number_of_streams, 0);
64 uint32_t bitrate_remainder = total_bitrate;
65 for (size_t i = 0; i < stream_bitrates.size() && bitrate_remainder > 0; ++i) {
66 if (stream_configs[i].maxBitrate * 1000 > bitrate_remainder) {
67 stream_bitrates[i] = bitrate_remainder;
68 } else {
69 stream_bitrates[i] = stream_configs[i].maxBitrate * 1000;
70 }
71 bitrate_remainder -= stream_bitrates[i];
72 }
73 return stream_bitrates;
74}
75
stefan@webrtc.org439be292012-02-16 14:45:37 +000076class QMVideoSettingsCallback : public VCMQMSettingsCallback {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +000077 public:
marpan@webrtc.orgefd01fd2012-04-18 15:56:34 +000078 explicit QMVideoSettingsCallback(VideoProcessingModule* vpm);
mflodman@webrtc.org6879c8a2013-07-23 11:35:00 +000079
stefan@webrtc.org439be292012-02-16 14:45:37 +000080 ~QMVideoSettingsCallback();
niklase@google.com470e71d2011-07-07 08:21:25 +000081
mflodman@webrtc.org84d17832011-12-01 17:02:23 +000082 // Update VPM with QM (quality modes: frame size & frame rate) settings.
pbos@webrtc.orgb238d122013-04-09 13:41:51 +000083 int32_t SetVideoQMSettings(const uint32_t frame_rate,
84 const uint32_t width,
85 const uint32_t height);
niklase@google.com470e71d2011-07-07 08:21:25 +000086
mflodman@webrtc.org84d17832011-12-01 17:02:23 +000087 private:
88 VideoProcessingModule* vpm_;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +000089};
niklase@google.com470e71d2011-07-07 08:21:25 +000090
pwestin@webrtc.org49888ce2012-04-27 05:25:53 +000091class ViEBitrateObserver : public BitrateObserver {
92 public:
mflodman@webrtc.orgf5e99db2012-06-27 09:49:37 +000093 explicit ViEBitrateObserver(ViEEncoder* owner)
pwestin@webrtc.org49888ce2012-04-27 05:25:53 +000094 : owner_(owner) {
95 }
mflodman@webrtc.org6879c8a2013-07-23 11:35:00 +000096 virtual ~ViEBitrateObserver() {}
pwestin@webrtc.org49888ce2012-04-27 05:25:53 +000097 // Implements BitrateObserver.
98 virtual void OnNetworkChanged(const uint32_t bitrate_bps,
99 const uint8_t fraction_lost,
100 const uint32_t rtt) {
101 owner_->OnNetworkChanged(bitrate_bps, fraction_lost, rtt);
102 }
103 private:
104 ViEEncoder* owner_;
105};
niklase@google.com470e71d2011-07-07 08:21:25 +0000106
pwestin@webrtc.org571a1c02012-11-13 21:12:39 +0000107class ViEPacedSenderCallback : public PacedSender::Callback {
108 public:
109 explicit ViEPacedSenderCallback(ViEEncoder* owner)
110 : owner_(owner) {
111 }
mflodman@webrtc.org6879c8a2013-07-23 11:35:00 +0000112 virtual ~ViEPacedSenderCallback() {}
stefan@webrtc.org82462aa2014-10-23 11:57:05 +0000113 virtual bool TimeToSendPacket(uint32_t ssrc,
114 uint16_t sequence_number,
115 int64_t capture_time_ms,
116 bool retransmission) {
stefan@webrtc.org9b82f5a2013-11-13 15:29:21 +0000117 return owner_->TimeToSendPacket(ssrc, sequence_number, capture_time_ms,
118 retransmission);
pwestin@webrtc.org571a1c02012-11-13 21:12:39 +0000119 }
pkasting@chromium.org4591fbd2014-11-20 22:28:14 +0000120 virtual size_t TimeToSendPadding(size_t bytes) {
stefan@webrtc.org508a84b2013-06-17 12:53:37 +0000121 return owner_->TimeToSendPadding(bytes);
pwestin@webrtc.org571a1c02012-11-13 21:12:39 +0000122 }
123 private:
124 ViEEncoder* owner_;
125};
126
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000127ViEEncoder::ViEEncoder(int32_t engine_id,
128 int32_t channel_id,
129 uint32_t number_of_cores,
andresp@webrtc.org7707d062013-05-13 10:50:50 +0000130 const Config& config,
pwestin@webrtc.org49888ce2012-04-27 05:25:53 +0000131 ProcessThread& module_process_thread,
132 BitrateController* bitrate_controller)
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000133 : engine_id_(engine_id),
134 channel_id_(channel_id),
135 number_of_cores_(number_of_cores),
stefan@webrtc.org34c5da62014-04-11 14:08:35 +0000136 vcm_(*webrtc::VideoCodingModule::Create()),
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000137 vpm_(*webrtc::VideoProcessingModule::Create(ViEModuleId(engine_id,
138 channel_id))),
mflodman@webrtc.orgd32c4472011-12-22 14:17:53 +0000139 callback_cs_(CriticalSectionWrapper::CreateCriticalSection()),
140 data_cs_(CriticalSectionWrapper::CreateCriticalSection()),
pwestin@webrtc.org49888ce2012-04-27 05:25:53 +0000141 bitrate_controller_(bitrate_controller),
stefan@webrtc.org3e005052013-10-18 15:05:29 +0000142 time_of_last_incoming_frame_ms_(0),
stefan@webrtc.org508a84b2013-06-17 12:53:37 +0000143 send_padding_(false),
pbos@webrtc.org3349ae02014-03-13 12:52:27 +0000144 min_transmit_bitrate_kbps_(0),
stefan@webrtc.orgbfacda62013-03-27 16:36:01 +0000145 target_delay_ms_(0),
146 network_is_transmitting_(true),
147 encoder_paused_(false),
pwestin@webrtc.org52b4e882013-05-02 19:02:17 +0000148 encoder_paused_and_dropped_frame_(false),
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000149 fec_enabled_(false),
150 nack_enabled_(false),
151 codec_observer_(NULL),
152 effect_filter_(NULL),
153 module_process_thread_(module_process_thread),
154 has_received_sli_(false),
155 picture_id_sli_(0),
156 has_received_rpsi_(false),
157 picture_id_rpsi_(0),
henrik.lundin@webrtc.org7ea4f242013-10-02 13:34:26 +0000158 qm_callback_(NULL),
henrik.lundin@webrtc.orgce8e0932013-11-18 12:18:43 +0000159 video_suspended_(false),
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000160 pre_encode_callback_(NULL),
161 start_ms_(Clock::GetRealTimeClock()->TimeInMilliseconds()) {
pwestin@webrtc.org2853dde2012-05-11 11:08:54 +0000162 RtpRtcp::Configuration configuration;
163 configuration.id = ViEModuleId(engine_id_, channel_id_);
164 configuration.audio = false; // Video.
165
166 default_rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(configuration));
pwestin@webrtc.org49888ce2012-04-27 05:25:53 +0000167 bitrate_observer_.reset(new ViEBitrateObserver(this));
pwestin@webrtc.org571a1c02012-11-13 21:12:39 +0000168 pacing_callback_.reset(new ViEPacedSenderCallback(this));
stefan@webrtc.org82462aa2014-10-23 11:57:05 +0000169 paced_sender_.reset(new PacedSender(
170 Clock::GetRealTimeClock(),
171 pacing_callback_.get(),
172 kDefaultStartBitrateKbps,
173 PacedSender::kDefaultPaceMultiplier * kDefaultStartBitrateKbps,
174 0));
wu@webrtc.org5d8c1022012-04-10 16:54:05 +0000175}
176
177bool ViEEncoder::Init() {
178 if (vcm_.InitializeSender() != 0) {
wu@webrtc.org5d8c1022012-04-10 16:54:05 +0000179 return false;
180 }
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000181 vpm_.EnableTemporalDecimation(true);
182
183 // Enable/disable content analysis: off by default for now.
184 vpm_.EnableContentAnalysis(false);
185
pwestin@webrtc.org571a1c02012-11-13 21:12:39 +0000186 if (module_process_thread_.RegisterModule(&vcm_) != 0 ||
187 module_process_thread_.RegisterModule(default_rtp_rtcp_.get()) != 0 ||
188 module_process_thread_.RegisterModule(paced_sender_.get()) != 0) {
wu@webrtc.org5d8c1022012-04-10 16:54:05 +0000189 return false;
190 }
stefan@webrtc.org97845122012-04-13 07:47:05 +0000191 if (qm_callback_) {
192 delete qm_callback_;
193 }
marpan@webrtc.orgefd01fd2012-04-18 15:56:34 +0000194 qm_callback_ = new QMVideoSettingsCallback(&vpm_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000195
196#ifdef VIDEOCODEC_VP8
andresp@webrtc.orga84b0a62014-08-14 16:46:46 +0000197 VideoCodecType codec_type = webrtc::kVideoCodecVP8;
198#else
199 VideoCodecType codec_type = webrtc::kVideoCodecI420;
200#endif
201
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000202 VideoCodec video_codec;
andresp@webrtc.orga84b0a62014-08-14 16:46:46 +0000203 if (vcm_.Codec(codec_type, &video_codec) != VCM_OK) {
wu@webrtc.org5d8c1022012-04-10 16:54:05 +0000204 return false;
205 }
stefan@webrtc.orgae2563a2014-02-13 13:48:38 +0000206 {
207 CriticalSectionScoped cs(data_cs_.get());
208 send_padding_ = video_codec.numberOfSimulcastStreams > 1;
209 }
wu@webrtc.org5d8c1022012-04-10 16:54:05 +0000210 if (vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
pwestin@webrtc.org2853dde2012-05-11 11:08:54 +0000211 default_rtp_rtcp_->MaxDataPayloadLength()) != 0) {
wu@webrtc.org5d8c1022012-04-10 16:54:05 +0000212 return false;
213 }
pwestin@webrtc.org2853dde2012-05-11 11:08:54 +0000214 if (default_rtp_rtcp_->RegisterSendPayload(video_codec) != 0) {
wu@webrtc.org5d8c1022012-04-10 16:54:05 +0000215 return false;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000216 }
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000217 if (vcm_.RegisterTransportCallback(this) != 0) {
wu@webrtc.org5d8c1022012-04-10 16:54:05 +0000218 return false;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000219 }
220 if (vcm_.RegisterSendStatisticsCallback(this) != 0) {
wu@webrtc.org5d8c1022012-04-10 16:54:05 +0000221 return false;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000222 }
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000223 if (vcm_.RegisterVideoQMCallback(qm_callback_) != 0) {
wu@webrtc.org5d8c1022012-04-10 16:54:05 +0000224 return false;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000225 }
wu@webrtc.org5d8c1022012-04-10 16:54:05 +0000226 return true;
niklase@google.com470e71d2011-07-07 08:21:25 +0000227}
228
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000229ViEEncoder::~ViEEncoder() {
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000230 UpdateHistograms();
stefan@webrtc.orgbf415082012-11-29 09:18:53 +0000231 if (bitrate_controller_) {
232 bitrate_controller_->RemoveBitrateObserver(bitrate_observer_.get());
233 }
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000234 module_process_thread_.DeRegisterModule(&vcm_);
235 module_process_thread_.DeRegisterModule(&vpm_);
pwestin@webrtc.org2853dde2012-05-11 11:08:54 +0000236 module_process_thread_.DeRegisterModule(default_rtp_rtcp_.get());
pwestin@webrtc.org571a1c02012-11-13 21:12:39 +0000237 module_process_thread_.DeRegisterModule(paced_sender_.get());
mflodman@webrtc.org66480932013-03-01 14:51:23 +0000238 VideoCodingModule::Destroy(&vcm_);
239 VideoProcessingModule::Destroy(&vpm_);
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000240 delete qm_callback_;
niklase@google.com470e71d2011-07-07 08:21:25 +0000241}
242
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000243void ViEEncoder::UpdateHistograms() {
244 const float kMinCallLengthInMinutes = 0.5f;
245 float elapsed_minutes =
246 (Clock::GetRealTimeClock()->TimeInMilliseconds() - start_ms_) / 60000.0f;
247 if (elapsed_minutes < kMinCallLengthInMinutes) {
248 return;
249 }
250 webrtc::VCMFrameCount frames;
251 if (vcm_.SentFrameCount(frames) != VCM_OK) {
252 return;
253 }
254 uint32_t total_frames = frames.numKeyFrames + frames.numDeltaFrames;
255 if (total_frames > 0) {
256 RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.KeyFramesSentInPermille",
257 static_cast<int>(
258 (frames.numKeyFrames * 1000.0f / total_frames) + 0.5f));
259 }
260}
261
mflodman@webrtc.org9ec883e2012-03-05 17:12:41 +0000262int ViEEncoder::Owner() const {
263 return channel_id_;
264}
265
stefan@webrtc.orgbfacda62013-03-27 16:36:01 +0000266void ViEEncoder::SetNetworkTransmissionState(bool is_transmitting) {
stefan@webrtc.orgbfacda62013-03-27 16:36:01 +0000267 {
268 CriticalSectionScoped cs(data_cs_.get());
269 network_is_transmitting_ = is_transmitting;
270 }
271 if (is_transmitting) {
272 paced_sender_->Resume();
273 } else {
274 paced_sender_->Pause();
275 }
276}
277
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000278void ViEEncoder::Pause() {
mflodman@webrtc.orgd32c4472011-12-22 14:17:53 +0000279 CriticalSectionScoped cs(data_cs_.get());
stefan@webrtc.orgbfacda62013-03-27 16:36:01 +0000280 encoder_paused_ = true;
niklase@google.com470e71d2011-07-07 08:21:25 +0000281}
282
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000283void ViEEncoder::Restart() {
mflodman@webrtc.orgd32c4472011-12-22 14:17:53 +0000284 CriticalSectionScoped cs(data_cs_.get());
stefan@webrtc.orgbfacda62013-03-27 16:36:01 +0000285 encoder_paused_ = false;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000286}
287
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000288uint8_t ViEEncoder::NumberOfCodecs() {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000289 return vcm_.NumberOfCodecs();
290}
291
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000292int32_t ViEEncoder::GetCodec(uint8_t list_index, VideoCodec* video_codec) {
mflodman@webrtc.orgf5e99db2012-06-27 09:49:37 +0000293 if (vcm_.Codec(list_index, video_codec) != 0) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000294 return -1;
295 }
296 return 0;
297}
298
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000299int32_t ViEEncoder::RegisterExternalEncoder(webrtc::VideoEncoder* encoder,
300 uint8_t pl_type,
301 bool internal_source) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000302 if (encoder == NULL)
303 return -1;
304
stefan@webrtc.orgfcd85852013-01-09 08:35:40 +0000305 if (vcm_.RegisterExternalEncoder(encoder, pl_type, internal_source) !=
mflodman@webrtc.org5574dac2014-04-07 10:56:31 +0000306 VCM_OK) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000307 return -1;
308 }
309 return 0;
310}
311
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000312int32_t ViEEncoder::DeRegisterExternalEncoder(uint8_t pl_type) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000313 webrtc::VideoCodec current_send_codec;
314 if (vcm_.SendCodec(&current_send_codec) == VCM_OK) {
stefan@webrtc.org3d0b0d62013-03-19 10:04:57 +0000315 uint32_t current_bitrate_bps = 0;
316 if (vcm_.Bitrate(&current_bitrate_bps) != 0) {
mflodman@webrtc.org5574dac2014-04-07 10:56:31 +0000317 LOG(LS_WARNING) << "Failed to get the current encoder target bitrate.";
stefan@webrtc.org439be292012-02-16 14:45:37 +0000318 }
stefan@webrtc.org3d0b0d62013-03-19 10:04:57 +0000319 current_send_codec.startBitrate = (current_bitrate_bps + 500) / 1000;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000320 }
321
322 if (vcm_.RegisterExternalEncoder(NULL, pl_type) != VCM_OK) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000323 return -1;
324 }
325
stefan@webrtc.org508a84b2013-06-17 12:53:37 +0000326 // If the external encoder is the current send codec, use vcm internal
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000327 // encoder.
328 if (current_send_codec.plType == pl_type) {
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000329 uint16_t max_data_payload_length =
pwestin@webrtc.org2853dde2012-05-11 11:08:54 +0000330 default_rtp_rtcp_->MaxDataPayloadLength();
stefan@webrtc.orgae2563a2014-02-13 13:48:38 +0000331 {
332 CriticalSectionScoped cs(data_cs_.get());
333 send_padding_ = current_send_codec.numberOfSimulcastStreams > 1;
334 }
fischman@webrtc.org64e04052014-03-07 18:00:05 +0000335 // TODO(mflodman): Unfortunately the VideoCodec that VCM has cached a
336 // raw pointer to an |extra_options| that's long gone. Clearing it here is
337 // a hack to prevent the following code from crashing. This should be fixed
338 // for realz. https://code.google.com/p/chromium/issues/detail?id=348222
339 current_send_codec.extra_options = NULL;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000340 if (vcm_.RegisterSendCodec(&current_send_codec, number_of_cores_,
341 max_data_payload_length) != VCM_OK) {
stefan@webrtc.org4070b1d2014-07-16 11:20:40 +0000342 LOG(LS_INFO) << "De-registered the currently used external encoder ("
343 << static_cast<int>(pl_type) << ") and therefore tried to "
344 << "register the corresponding internal encoder, but none "
345 << "was supported.";
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000346 }
347 }
348 return 0;
349}
350
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000351int32_t ViEEncoder::SetEncoder(const webrtc::VideoCodec& video_codec) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000352 // Setting target width and height for VPM.
353 if (vpm_.SetTargetResolution(video_codec.width, video_codec.height,
354 video_codec.maxFramerate) != VPM_OK) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000355 return -1;
356 }
357
pwestin@webrtc.org2853dde2012-05-11 11:08:54 +0000358 if (default_rtp_rtcp_->RegisterSendPayload(video_codec) != 0) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000359 return -1;
360 }
pwestin@webrtc.org49888ce2012-04-27 05:25:53 +0000361 // Convert from kbps to bps.
stefan@webrtc.orgb2c8a952013-09-06 13:58:01 +0000362 std::vector<uint32_t> stream_bitrates = AllocateStreamBitrates(
363 video_codec.startBitrate * 1000,
364 video_codec.simulcastStream,
365 video_codec.numberOfSimulcastStreams);
366 default_rtp_rtcp_->SetTargetSendBitrate(stream_bitrates);
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000367
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000368 uint16_t max_data_payload_length =
pwestin@webrtc.org2853dde2012-05-11 11:08:54 +0000369 default_rtp_rtcp_->MaxDataPayloadLength();
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000370
stefan@webrtc.org9075d512014-02-14 09:45:58 +0000371 {
372 CriticalSectionScoped cs(data_cs_.get());
373 send_padding_ = video_codec.numberOfSimulcastStreams > 1;
374 }
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000375 if (vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
376 max_data_payload_length) != VCM_OK) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000377 return -1;
378 }
379
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000380 // Set this module as sending right away, let the slave module in the channel
381 // start and stop sending.
andresp@webrtc.orga84b0a62014-08-14 16:46:46 +0000382 if (default_rtp_rtcp_->SetSendingStatus(true) != 0) {
383 return -1;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000384 }
andresp@webrtc.orga84b0a62014-08-14 16:46:46 +0000385
pwestin@webrtc.org49888ce2012-04-27 05:25:53 +0000386 bitrate_controller_->SetBitrateObserver(bitrate_observer_.get(),
387 video_codec.startBitrate * 1000,
388 video_codec.minBitrate * 1000,
stefan@webrtc.orgbfacda62013-03-27 16:36:01 +0000389 kTransmissionMaxBitrateMultiplier *
pwestin@webrtc.org49888ce2012-04-27 05:25:53 +0000390 video_codec.maxBitrate * 1000);
391
pbos@webrtc.org3349ae02014-03-13 12:52:27 +0000392 CriticalSectionScoped crit(data_cs_.get());
393 int pad_up_to_bitrate_kbps = video_codec.startBitrate;
394 if (pad_up_to_bitrate_kbps < min_transmit_bitrate_kbps_)
395 pad_up_to_bitrate_kbps = min_transmit_bitrate_kbps_;
396
stefan@webrtc.org88e0dda2014-07-04 09:20:42 +0000397 paced_sender_->UpdateBitrate(
stefan@webrtc.org82462aa2014-10-23 11:57:05 +0000398 video_codec.startBitrate,
stefan@webrtc.org88e0dda2014-07-04 09:20:42 +0000399 PacedSender::kDefaultPaceMultiplier * video_codec.startBitrate,
400 pad_up_to_bitrate_kbps);
pbos@webrtc.org3349ae02014-03-13 12:52:27 +0000401
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000402 return 0;
403}
404
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000405int32_t ViEEncoder::GetEncoder(VideoCodec* video_codec) {
mflodman@webrtc.orgf5e99db2012-06-27 09:49:37 +0000406 if (vcm_.SendCodec(video_codec) != 0) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000407 return -1;
408 }
409 return 0;
410}
niklase@google.com470e71d2011-07-07 08:21:25 +0000411
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000412int32_t ViEEncoder::GetCodecConfigParameters(
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000413 unsigned char config_parameters[kConfigParameterSize],
414 unsigned char& config_parameters_size) {
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000415 int32_t num_parameters =
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000416 vcm_.CodecConfigParameters(config_parameters, kConfigParameterSize);
417 if (num_parameters <= 0) {
418 config_parameters_size = 0;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000419 return -1;
420 }
421 config_parameters_size = static_cast<unsigned char>(num_parameters);
422 return 0;
niklase@google.com470e71d2011-07-07 08:21:25 +0000423}
424
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000425int32_t ViEEncoder::ScaleInputImage(bool enable) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000426 VideoFrameResampling resampling_mode = kFastRescaling;
mflodman@webrtc.org5574dac2014-04-07 10:56:31 +0000427 // TODO(mflodman) What?
428 if (enable) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000429 // kInterpolation is currently not supported.
mflodman@webrtc.org5574dac2014-04-07 10:56:31 +0000430 LOG_F(LS_ERROR) << "Not supported.";
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000431 return -1;
432 }
433 vpm_.SetInputFrameResampleMode(resampling_mode);
niklase@google.com470e71d2011-07-07 08:21:25 +0000434
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000435 return 0;
niklase@google.com470e71d2011-07-07 08:21:25 +0000436}
437
stefan@webrtc.org9b82f5a2013-11-13 15:29:21 +0000438bool ViEEncoder::TimeToSendPacket(uint32_t ssrc,
439 uint16_t sequence_number,
440 int64_t capture_time_ms,
441 bool retransmission) {
hclam@chromium.org2e402ce2013-06-20 20:18:31 +0000442 return default_rtp_rtcp_->TimeToSendPacket(ssrc, sequence_number,
stefan@webrtc.org9b82f5a2013-11-13 15:29:21 +0000443 capture_time_ms, retransmission);
pwestin@webrtc.org571a1c02012-11-13 21:12:39 +0000444}
445
pkasting@chromium.org4591fbd2014-11-20 22:28:14 +0000446size_t ViEEncoder::TimeToSendPadding(size_t bytes) {
henrik.lundin@webrtc.org331d4402013-11-21 14:05:40 +0000447 bool send_padding;
448 {
449 CriticalSectionScoped cs(data_cs_.get());
pbos@webrtc.org3349ae02014-03-13 12:52:27 +0000450 send_padding =
451 send_padding_ || video_suspended_ || min_transmit_bitrate_kbps_ > 0;
henrik.lundin@webrtc.org331d4402013-11-21 14:05:40 +0000452 }
453 if (send_padding) {
stefan@webrtc.org508a84b2013-06-17 12:53:37 +0000454 return default_rtp_rtcp_->TimeToSendPadding(bytes);
455 }
456 return 0;
457}
458
stefan@webrtc.orgbfacda62013-03-27 16:36:01 +0000459bool ViEEncoder::EncoderPaused() const {
pwestin@webrtc.org91563e42013-04-25 22:20:08 +0000460 // Pause video if paused by caller or as long as the network is down or the
461 // pacer queue has grown too large in buffered mode.
462 if (encoder_paused_) {
463 return true;
464 }
465 if (target_delay_ms_ > 0) {
466 // Buffered mode.
467 // TODO(pwestin): Workaround until nack is configured as a time and not
468 // number of packets.
469 return paced_sender_->QueueInMs() >=
pwestin@webrtc.org52b4e882013-05-02 19:02:17 +0000470 std::max(static_cast<int>(target_delay_ms_ * kEncoderPausePacerMargin),
471 kMinPacingDelayMs);
pwestin@webrtc.org91563e42013-04-25 22:20:08 +0000472 }
sprang@webrtc.orgdcebf2d2014-11-04 16:27:16 +0000473 if (paced_sender_->ExpectedQueueTimeMs() >
474 PacedSender::kDefaultMaxQueueLengthMs) {
475 // Too much data in pacer queue, drop frame.
476 return true;
477 }
pwestin@webrtc.org91563e42013-04-25 22:20:08 +0000478 return !network_is_transmitting_;
stefan@webrtc.orgbfacda62013-03-27 16:36:01 +0000479}
480
sprang@webrtc.orgdcebf2d2014-11-04 16:27:16 +0000481void ViEEncoder::TraceFrameDropStart() {
482 // Start trace event only on the first frame after encoder is paused.
483 if (!encoder_paused_and_dropped_frame_) {
484 TRACE_EVENT_ASYNC_BEGIN0("webrtc", "EncoderPaused", this);
485 }
486 encoder_paused_and_dropped_frame_ = true;
487 return;
488}
489
490void ViEEncoder::TraceFrameDropEnd() {
491 // End trace event on first frame after encoder resumes, if frame was dropped.
492 if (encoder_paused_and_dropped_frame_) {
493 TRACE_EVENT_ASYNC_END0("webrtc", "EncoderPaused", this);
494 }
495 encoder_paused_and_dropped_frame_ = false;
496}
497
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000498RtpRtcp* ViEEncoder::SendRtpRtcpModule() {
pwestin@webrtc.org2853dde2012-05-11 11:08:54 +0000499 return default_rtp_rtcp_.get();
niklase@google.com470e71d2011-07-07 08:21:25 +0000500}
501
mflodman@webrtc.org8baed512012-06-21 12:11:50 +0000502void ViEEncoder::DeliverFrame(int id,
mikhal@webrtc.org9fedff72012-10-24 18:33:04 +0000503 I420VideoFrame* video_frame,
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000504 int num_csrcs,
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000505 const uint32_t CSRC[kRtpCsrcSize]) {
wuchengli@chromium.orgac4b87c2014-03-19 03:44:20 +0000506 if (default_rtp_rtcp_->SendingMedia() == false) {
507 // We've paused or we have no channels attached, don't encode.
508 return;
509 }
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000510 {
mflodman@webrtc.orgd32c4472011-12-22 14:17:53 +0000511 CriticalSectionScoped cs(data_cs_.get());
stefan@webrtc.org3e005052013-10-18 15:05:29 +0000512 time_of_last_incoming_frame_ms_ = TickTime::MillisecondTimestamp();
pwestin@webrtc.org52b4e882013-05-02 19:02:17 +0000513 if (EncoderPaused()) {
sprang@webrtc.orgdcebf2d2014-11-04 16:27:16 +0000514 TraceFrameDropStart();
pwestin@webrtc.org52b4e882013-05-02 19:02:17 +0000515 return;
516 }
sprang@webrtc.orgdcebf2d2014-11-04 16:27:16 +0000517 TraceFrameDropEnd();
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000518 }
niklase@google.com470e71d2011-07-07 08:21:25 +0000519
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000520 // Convert render time, in ms, to RTP timestamp.
mflodman@webrtc.org8baed512012-06-21 12:11:50 +0000521 const int kMsToRtpTimestamp = 90;
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000522 const uint32_t time_stamp =
mflodman@webrtc.org8baed512012-06-21 12:11:50 +0000523 kMsToRtpTimestamp *
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000524 static_cast<uint32_t>(video_frame->render_time_ms());
hclam@chromium.org806dc3b2013-04-09 19:54:10 +0000525
hclam@chromium.org1a7b9b92013-07-08 21:31:18 +0000526 TRACE_EVENT_ASYNC_STEP0("webrtc", "Video", video_frame->render_time_ms(),
527 "Encode");
mikhal@webrtc.org9fedff72012-10-24 18:33:04 +0000528 video_frame->set_timestamp(time_stamp);
niklase@google.com470e71d2011-07-07 08:21:25 +0000529
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000530 // Make sure the CSRC list is correct.
531 if (num_csrcs > 0) {
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000532 uint32_t tempCSRC[kRtpCsrcSize];
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000533 for (int i = 0; i < num_csrcs; i++) {
534 if (CSRC[i] == 1) {
pwestin@webrtc.org2853dde2012-05-11 11:08:54 +0000535 tempCSRC[i] = default_rtp_rtcp_->SSRC();
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000536 } else {
537 tempCSRC[i] = CSRC[i];
538 }
niklase@google.com470e71d2011-07-07 08:21:25 +0000539 }
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000540 default_rtp_rtcp_->SetCSRCs(tempCSRC, (uint8_t) num_csrcs);
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000541 }
wuchengli@chromium.orgf425b552014-06-20 12:04:05 +0000542
pwestin@webrtc.org2f476ed2012-10-30 16:21:52 +0000543 I420VideoFrame* decimated_frame = NULL;
wuchengli@chromium.orgf425b552014-06-20 12:04:05 +0000544 // TODO(wuchengli): support texture frames.
545 if (video_frame->native_handle() == NULL) {
546 {
547 CriticalSectionScoped cs(callback_cs_.get());
548 if (effect_filter_) {
pkasting@chromium.org4591fbd2014-11-20 22:28:14 +0000549 size_t length =
wuchengli@chromium.orgf425b552014-06-20 12:04:05 +0000550 CalcBufferSize(kI420, video_frame->width(), video_frame->height());
551 scoped_ptr<uint8_t[]> video_buffer(new uint8_t[length]);
552 ExtractBuffer(*video_frame, length, video_buffer.get());
553 effect_filter_->Transform(length,
554 video_buffer.get(),
555 video_frame->ntp_time_ms(),
556 video_frame->timestamp(),
557 video_frame->width(),
558 video_frame->height());
559 }
560 }
561
562 // Pass frame via preprocessor.
563 const int ret = vpm_.PreprocessFrame(*video_frame, &decimated_frame);
564 if (ret == 1) {
565 // Drop this frame.
566 return;
567 }
568 if (ret != VPM_OK) {
569 return;
570 }
pwestin@webrtc.org2f476ed2012-10-30 16:21:52 +0000571 }
wuchengli@chromium.orgf425b552014-06-20 12:04:05 +0000572 // If the frame was not resampled or scaled => use original.
pwestin@webrtc.org2f476ed2012-10-30 16:21:52 +0000573 if (decimated_frame == NULL) {
574 decimated_frame = video_frame;
575 }
pbos@webrtc.orgfe1ef932013-10-21 10:34:43 +0000576
577 {
578 CriticalSectionScoped cs(callback_cs_.get());
579 if (pre_encode_callback_)
580 pre_encode_callback_->FrameCallback(decimated_frame);
581 }
582
wuchengli@chromium.orgf425b552014-06-20 12:04:05 +0000583 if (video_frame->native_handle() != NULL) {
584 // TODO(wuchengli): add texture support. http://crbug.com/362437
585 return;
586 }
587
niklase@google.com470e71d2011-07-07 08:21:25 +0000588#ifdef VIDEOCODEC_VP8
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000589 if (vcm_.SendCodec() == webrtc::kVideoCodecVP8) {
590 webrtc::CodecSpecificInfo codec_specific_info;
591 codec_specific_info.codecType = webrtc::kVideoCodecVP8;
stefan@webrtc.org7af12be2014-07-09 14:46:31 +0000592 {
593 CriticalSectionScoped cs(data_cs_.get());
594 codec_specific_info.codecSpecific.VP8.hasReceivedRPSI =
595 has_received_rpsi_;
596 codec_specific_info.codecSpecific.VP8.hasReceivedSLI =
597 has_received_sli_;
598 codec_specific_info.codecSpecific.VP8.pictureIdRPSI =
599 picture_id_rpsi_;
600 codec_specific_info.codecSpecific.VP8.pictureIdSLI =
601 picture_id_sli_;
602 has_received_sli_ = false;
603 has_received_rpsi_ = false;
604 }
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000605
mflodman@webrtc.org5574dac2014-04-07 10:56:31 +0000606 vcm_.AddVideoFrame(*decimated_frame, vpm_.ContentMetrics(),
607 &codec_specific_info);
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000608 return;
609 }
niklase@google.com470e71d2011-07-07 08:21:25 +0000610#endif
mflodman@webrtc.org5574dac2014-04-07 10:56:31 +0000611 vcm_.AddVideoFrame(*decimated_frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000612}
niklase@google.com470e71d2011-07-07 08:21:25 +0000613
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000614void ViEEncoder::DelayChanged(int id, int frame_delay) {
stefan@webrtc.org7da34592013-04-09 14:56:29 +0000615 default_rtp_rtcp_->SetCameraDelay(frame_delay);
niklase@google.com470e71d2011-07-07 08:21:25 +0000616}
niklase@google.com470e71d2011-07-07 08:21:25 +0000617
mflodman@webrtc.org8baed512012-06-21 12:11:50 +0000618int ViEEncoder::GetPreferedFrameSettings(int* width,
619 int* height,
620 int* frame_rate) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000621 webrtc::VideoCodec video_codec;
622 memset(&video_codec, 0, sizeof(video_codec));
623 if (vcm_.SendCodec(&video_codec) != VCM_OK) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000624 return -1;
625 }
626
mflodman@webrtc.org8baed512012-06-21 12:11:50 +0000627 *width = video_codec.width;
628 *height = video_codec.height;
629 *frame_rate = video_codec.maxFramerate;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000630 return 0;
631}
632
pwestin@webrtc.orgce330352012-04-12 06:59:14 +0000633int ViEEncoder::SendKeyFrame() {
stefan@webrtc.orgc5300432012-10-08 07:06:53 +0000634 return vcm_.IntraFrameRequest(0);
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000635}
636
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000637int32_t ViEEncoder::SendCodecStatistics(
638 uint32_t* num_key_frames, uint32_t* num_delta_frames) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000639 webrtc::VCMFrameCount sent_frames;
640 if (vcm_.SentFrameCount(sent_frames) != VCM_OK) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000641 return -1;
642 }
mflodman@webrtc.orgf5e99db2012-06-27 09:49:37 +0000643 *num_key_frames = sent_frames.numKeyFrames;
644 *num_delta_frames = sent_frames.numDeltaFrames;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000645 return 0;
646}
647
jiayl@webrtc.org9fd8d872014-02-27 22:32:40 +0000648int32_t ViEEncoder::PacerQueuingDelayMs() const {
649 return paced_sender_->QueueInMs();
650}
651
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000652int ViEEncoder::CodecTargetBitrate(uint32_t* bitrate) const {
stefan@webrtc.org439be292012-02-16 14:45:37 +0000653 if (vcm_.Bitrate(bitrate) != 0)
654 return -1;
655 return 0;
stefan@webrtc.org07b45a52012-02-02 08:37:48 +0000656}
657
wu@webrtc.org822fbd82013-08-15 23:38:54 +0000658int32_t ViEEncoder::UpdateProtectionMethod(bool enable_nack) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000659 bool fec_enabled = false;
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000660 uint8_t dummy_ptype_red = 0;
661 uint8_t dummy_ptypeFEC = 0;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000662
663 // Updated protection method to VCM to get correct packetization sizes.
664 // FEC has larger overhead than NACK -> set FEC if used.
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000665 int32_t error = default_rtp_rtcp_->GenericFECStatus(fec_enabled,
666 dummy_ptype_red,
667 dummy_ptypeFEC);
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000668 if (error) {
669 return -1;
670 }
wu@webrtc.org822fbd82013-08-15 23:38:54 +0000671 if (fec_enabled_ == fec_enabled && nack_enabled_ == enable_nack) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000672 // No change needed, we're already in correct state.
673 return 0;
674 }
675 fec_enabled_ = fec_enabled;
wu@webrtc.org822fbd82013-08-15 23:38:54 +0000676 nack_enabled_ = enable_nack;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000677
678 // Set Video Protection for VCM.
wu@webrtc.org822fbd82013-08-15 23:38:54 +0000679 if (fec_enabled && nack_enabled_) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000680 vcm_.SetVideoProtection(webrtc::kProtectionNackFEC, true);
681 } else {
682 vcm_.SetVideoProtection(webrtc::kProtectionFEC, fec_enabled_);
wu@webrtc.org822fbd82013-08-15 23:38:54 +0000683 vcm_.SetVideoProtection(webrtc::kProtectionNackSender, nack_enabled_);
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000684 vcm_.SetVideoProtection(webrtc::kProtectionNackFEC, false);
685 }
686
wu@webrtc.org822fbd82013-08-15 23:38:54 +0000687 if (fec_enabled_ || nack_enabled_) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000688 vcm_.RegisterProtectionCallback(this);
689 // The send codec must be registered to set correct MTU.
690 webrtc::VideoCodec codec;
691 if (vcm_.SendCodec(&codec) == 0) {
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000692 uint16_t max_pay_load = default_rtp_rtcp_->MaxDataPayloadLength();
stefan@webrtc.org3d0b0d62013-03-19 10:04:57 +0000693 uint32_t current_bitrate_bps = 0;
694 if (vcm_.Bitrate(&current_bitrate_bps) != 0) {
mflodman@webrtc.org5574dac2014-04-07 10:56:31 +0000695 LOG_F(LS_WARNING) <<
696 "Failed to get the current encoder target bitrate.";
stefan@webrtc.org439be292012-02-16 14:45:37 +0000697 }
stefan@webrtc.org3d0b0d62013-03-19 10:04:57 +0000698 // Convert to start bitrate in kbps.
699 codec.startBitrate = (current_bitrate_bps + 500) / 1000;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000700 if (vcm_.RegisterSendCodec(&codec, number_of_cores_, max_pay_load) != 0) {
niklase@google.com470e71d2011-07-07 08:21:25 +0000701 return -1;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000702 }
niklase@google.com470e71d2011-07-07 08:21:25 +0000703 }
704 return 0;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000705 } else {
706 // FEC and NACK are disabled.
707 vcm_.RegisterProtectionCallback(NULL);
708 }
709 return 0;
niklase@google.com470e71d2011-07-07 08:21:25 +0000710}
711
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000712void ViEEncoder::SetSenderBufferingMode(int target_delay_ms) {
stefan@webrtc.orgbfacda62013-03-27 16:36:01 +0000713 {
714 CriticalSectionScoped cs(data_cs_.get());
715 target_delay_ms_ = target_delay_ms;
716 }
mikhal@webrtc.org3d305c62013-02-10 18:42:55 +0000717 if (target_delay_ms > 0) {
stefan@webrtc.orgbfacda62013-03-27 16:36:01 +0000718 // Disable external frame-droppers.
719 vcm_.EnableFrameDropper(false);
720 vpm_.EnableTemporalDecimation(false);
mikhal@webrtc.org3d305c62013-02-10 18:42:55 +0000721 } else {
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000722 // Real-time mode - enable frame droppers.
mikhal@webrtc.org3d305c62013-02-10 18:42:55 +0000723 vpm_.EnableTemporalDecimation(true);
724 vcm_.EnableFrameDropper(true);
725 }
726}
727
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000728int32_t ViEEncoder::SendData(
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000729 const FrameType frame_type,
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000730 const uint8_t payload_type,
731 const uint32_t time_stamp,
stefan@webrtc.orgddfdfed2012-07-03 13:21:22 +0000732 int64_t capture_time_ms,
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000733 const uint8_t* payload_data,
pkasting@chromium.org4591fbd2014-11-20 22:28:14 +0000734 const size_t payload_size,
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000735 const webrtc::RTPFragmentationHeader& fragmentation_header,
736 const RTPVideoHeader* rtp_video_hdr) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000737 // New encoded data, hand over to the rtp module.
stefan@webrtc.orgddfdfed2012-07-03 13:21:22 +0000738 return default_rtp_rtcp_->SendOutgoingData(frame_type,
739 payload_type,
740 time_stamp,
741 capture_time_ms,
742 payload_data,
pwestin@webrtc.org2853dde2012-05-11 11:08:54 +0000743 payload_size,
744 &fragmentation_header,
745 rtp_video_hdr);
niklase@google.com470e71d2011-07-07 08:21:25 +0000746}
747
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000748int32_t ViEEncoder::ProtectionRequest(
stefan@webrtc.orge0d6fa42012-03-20 22:10:56 +0000749 const FecProtectionParams* delta_fec_params,
750 const FecProtectionParams* key_fec_params,
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000751 uint32_t* sent_video_rate_bps,
752 uint32_t* sent_nack_rate_bps,
753 uint32_t* sent_fec_rate_bps) {
mflodman@webrtc.org5574dac2014-04-07 10:56:31 +0000754 default_rtp_rtcp_->SetFecParameters(delta_fec_params, key_fec_params);
755 default_rtp_rtcp_->BitrateSent(NULL, sent_video_rate_bps, sent_fec_rate_bps,
stefan@webrtc.orgf4c82862011-12-13 15:38:14 +0000756 sent_nack_rate_bps);
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000757 return 0;
niklase@google.com470e71d2011-07-07 08:21:25 +0000758}
759
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000760int32_t ViEEncoder::SendStatistics(const uint32_t bit_rate,
761 const uint32_t frame_rate) {
mflodman@webrtc.orgd32c4472011-12-22 14:17:53 +0000762 CriticalSectionScoped cs(callback_cs_.get());
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000763 if (codec_observer_) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000764 codec_observer_->OutgoingRate(channel_id_, frame_rate, bit_rate);
765 }
766 return 0;
niklase@google.com470e71d2011-07-07 08:21:25 +0000767}
768
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000769int32_t ViEEncoder::RegisterCodecObserver(ViEEncoderObserver* observer) {
mflodman@webrtc.orgd32c4472011-12-22 14:17:53 +0000770 CriticalSectionScoped cs(callback_cs_.get());
mflodman@webrtc.org5574dac2014-04-07 10:56:31 +0000771 if (observer && codec_observer_) {
772 LOG_F(LS_ERROR) << "Observer already set.";
773 return -1;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000774 }
mflodman@webrtc.org5574dac2014-04-07 10:56:31 +0000775 codec_observer_ = observer;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000776 return 0;
niklase@google.com470e71d2011-07-07 08:21:25 +0000777}
778
andrew@webrtc.org96636862012-09-20 23:33:17 +0000779void ViEEncoder::OnReceivedSLI(uint32_t /*ssrc*/,
780 uint8_t picture_id) {
stefan@webrtc.org7af12be2014-07-09 14:46:31 +0000781 CriticalSectionScoped cs(data_cs_.get());
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000782 picture_id_sli_ = picture_id;
783 has_received_sli_ = true;
niklase@google.com470e71d2011-07-07 08:21:25 +0000784}
785
andrew@webrtc.org96636862012-09-20 23:33:17 +0000786void ViEEncoder::OnReceivedRPSI(uint32_t /*ssrc*/,
787 uint64_t picture_id) {
stefan@webrtc.org7af12be2014-07-09 14:46:31 +0000788 CriticalSectionScoped cs(data_cs_.get());
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000789 picture_id_rpsi_ = picture_id;
790 has_received_rpsi_ = true;
niklase@google.com470e71d2011-07-07 08:21:25 +0000791}
792
mflodman@webrtc.orgd6ec3862012-10-25 11:30:29 +0000793void ViEEncoder::OnReceivedIntraFrameRequest(uint32_t ssrc) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000794 // Key frame request from remote side, signal to VCM.
justinlin@chromium.org7bfb3a32013-05-13 22:59:00 +0000795 TRACE_EVENT0("webrtc", "OnKeyFrameRequest");
pwestin@webrtc.org1da1ce02011-10-13 15:19:55 +0000796
mflodman@webrtc.orgd6ec3862012-10-25 11:30:29 +0000797 int idx = 0;
798 {
799 CriticalSectionScoped cs(data_cs_.get());
800 std::map<unsigned int, int>::iterator stream_it = ssrc_streams_.find(ssrc);
801 if (stream_it == ssrc_streams_.end()) {
mflodman@webrtc.orgd73527c2012-12-20 09:26:17 +0000802 LOG_F(LS_WARNING) << "ssrc not found: " << ssrc << ", map size "
803 << ssrc_streams_.size();
mflodman@webrtc.orgd6ec3862012-10-25 11:30:29 +0000804 return;
805 }
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000806 std::map<unsigned int, int64_t>::iterator time_it =
mflodman@webrtc.orgd6ec3862012-10-25 11:30:29 +0000807 time_last_intra_request_ms_.find(ssrc);
808 if (time_it == time_last_intra_request_ms_.end()) {
809 time_last_intra_request_ms_[ssrc] = 0;
810 }
811
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000812 int64_t now = TickTime::MillisecondTimestamp();
mflodman@webrtc.orgd6ec3862012-10-25 11:30:29 +0000813 if (time_last_intra_request_ms_[ssrc] + kViEMinKeyRequestIntervalMs > now) {
mflodman@webrtc.orgd6ec3862012-10-25 11:30:29 +0000814 return;
815 }
816 time_last_intra_request_ms_[ssrc] = now;
817 idx = stream_it->second;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000818 }
mflodman@webrtc.orgd6ec3862012-10-25 11:30:29 +0000819 // Release the critsect before triggering key frame.
820 vcm_.IntraFrameRequest(idx);
niklase@google.com470e71d2011-07-07 08:21:25 +0000821}
822
mflodman@webrtc.orgaca26292012-10-05 16:17:41 +0000823void ViEEncoder::OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) {
mflodman@webrtc.orgd6ec3862012-10-25 11:30:29 +0000824 CriticalSectionScoped cs(data_cs_.get());
825 std::map<unsigned int, int>::iterator it = ssrc_streams_.find(old_ssrc);
826 if (it == ssrc_streams_.end()) {
827 return;
828 }
829
830 ssrc_streams_[new_ssrc] = it->second;
831 ssrc_streams_.erase(it);
832
833 std::map<unsigned int, int64_t>::iterator time_it =
834 time_last_intra_request_ms_.find(old_ssrc);
835 int64_t last_intra_request_ms = 0;
836 if (time_it != time_last_intra_request_ms_.end()) {
837 last_intra_request_ms = time_it->second;
838 time_last_intra_request_ms_.erase(time_it);
839 }
840 time_last_intra_request_ms_[new_ssrc] = last_intra_request_ms;
841}
842
843bool ViEEncoder::SetSsrcs(const std::list<unsigned int>& ssrcs) {
844 VideoCodec codec;
845 if (vcm_.SendCodec(&codec) != 0)
846 return false;
847
848 if (codec.numberOfSimulcastStreams > 0 &&
849 ssrcs.size() != codec.numberOfSimulcastStreams) {
850 return false;
851 }
852
853 CriticalSectionScoped cs(data_cs_.get());
854 ssrc_streams_.clear();
855 time_last_intra_request_ms_.clear();
856 int idx = 0;
857 for (std::list<unsigned int>::const_iterator it = ssrcs.begin();
858 it != ssrcs.end(); ++it, ++idx) {
859 unsigned int ssrc = *it;
860 ssrc_streams_[ssrc] = idx;
861 }
862 return true;
mflodman@webrtc.orgaca26292012-10-05 16:17:41 +0000863}
864
pbos@webrtc.org3349ae02014-03-13 12:52:27 +0000865void ViEEncoder::SetMinTransmitBitrate(int min_transmit_bitrate_kbps) {
866 assert(min_transmit_bitrate_kbps >= 0);
867 CriticalSectionScoped crit(data_cs_.get());
868 min_transmit_bitrate_kbps_ = min_transmit_bitrate_kbps;
869}
870
pwestin@webrtc.org49888ce2012-04-27 05:25:53 +0000871// Called from ViEBitrateObserver.
872void ViEEncoder::OnNetworkChanged(const uint32_t bitrate_bps,
873 const uint8_t fraction_lost,
874 const uint32_t round_trip_time_ms) {
mflodman@webrtc.org5574dac2014-04-07 10:56:31 +0000875 LOG(LS_VERBOSE) << "OnNetworkChanged, bitrate" << bitrate_bps
876 << " packet loss " << fraction_lost
877 << " rtt " << round_trip_time_ms;
stefan@webrtc.orgabc9d5b2013-03-18 17:00:51 +0000878 vcm_.SetChannelParameters(bitrate_bps, fraction_lost, round_trip_time_ms);
henrik.lundin@webrtc.orgce8e0932013-11-18 12:18:43 +0000879 bool video_is_suspended = vcm_.VideoSuspended();
pwestin@webrtc.org571a1c02012-11-13 21:12:39 +0000880 int bitrate_kbps = bitrate_bps / 1000;
stefan@webrtc.org508a84b2013-06-17 12:53:37 +0000881 VideoCodec send_codec;
882 if (vcm_.SendCodec(&send_codec) != 0) {
883 return;
884 }
stefan@webrtc.orgb2c8a952013-09-06 13:58:01 +0000885 SimulcastStream* stream_configs = send_codec.simulcastStream;
886 // Allocate the bandwidth between the streams.
887 std::vector<uint32_t> stream_bitrates = AllocateStreamBitrates(
888 bitrate_bps,
889 stream_configs,
890 send_codec.numberOfSimulcastStreams);
891 // Find the max amount of padding we can allow ourselves to send at this
892 // point, based on which streams are currently active and what our current
893 // available bandwidth is.
stefan@webrtc.orgb400aa72013-10-16 13:03:10 +0000894 int pad_up_to_bitrate_kbps = 0;
895 if (send_codec.numberOfSimulcastStreams == 0) {
stefan@webrtc.orgb400aa72013-10-16 13:03:10 +0000896 pad_up_to_bitrate_kbps = send_codec.minBitrate;
897 } else {
stefan@webrtc.orgb400aa72013-10-16 13:03:10 +0000898 pad_up_to_bitrate_kbps =
899 stream_configs[send_codec.numberOfSimulcastStreams - 1].minBitrate;
900 for (int i = 0; i < send_codec.numberOfSimulcastStreams - 1; ++i) {
901 pad_up_to_bitrate_kbps += stream_configs[i].targetBitrate;
902 }
stefan@webrtc.orgb2c8a952013-09-06 13:58:01 +0000903 }
pbos@webrtc.org709e2972014-03-19 10:59:52 +0000904
905 // Disable padding if only sending one stream and video isn't suspended and
906 // min-transmit bitrate isn't used (applied later).
907 if (!video_is_suspended && send_codec.numberOfSimulcastStreams <= 1)
stefan@webrtc.orgb400aa72013-10-16 13:03:10 +0000908 pad_up_to_bitrate_kbps = 0;
stefan@webrtc.org3e005052013-10-18 15:05:29 +0000909
910 {
stefan@webrtc.org3e005052013-10-18 15:05:29 +0000911 CriticalSectionScoped cs(data_cs_.get());
pbos@webrtc.org709e2972014-03-19 10:59:52 +0000912 // The amount of padding should decay to zero if no frames are being
913 // captured unless a min-transmit bitrate is used.
stefan@webrtc.org3e005052013-10-18 15:05:29 +0000914 int64_t now_ms = TickTime::MillisecondTimestamp();
915 if (now_ms - time_of_last_incoming_frame_ms_ > kStopPaddingThresholdMs)
pbos@webrtc.org709e2972014-03-19 10:59:52 +0000916 pad_up_to_bitrate_kbps = 0;
stefan@webrtc.org3e005052013-10-18 15:05:29 +0000917
pbos@webrtc.org709e2972014-03-19 10:59:52 +0000918 // Pad up to min bitrate.
pbos@webrtc.org3349ae02014-03-13 12:52:27 +0000919 if (pad_up_to_bitrate_kbps < min_transmit_bitrate_kbps_)
920 pad_up_to_bitrate_kbps = min_transmit_bitrate_kbps_;
pbos@webrtc.org709e2972014-03-19 10:59:52 +0000921
922 // Padding may never exceed bitrate estimate.
923 if (pad_up_to_bitrate_kbps > bitrate_kbps)
924 pad_up_to_bitrate_kbps = bitrate_kbps;
925
stefan@webrtc.org88e0dda2014-07-04 09:20:42 +0000926 paced_sender_->UpdateBitrate(
stefan@webrtc.org82462aa2014-10-23 11:57:05 +0000927 bitrate_kbps,
stefan@webrtc.org88e0dda2014-07-04 09:20:42 +0000928 PacedSender::kDefaultPaceMultiplier * bitrate_kbps,
929 pad_up_to_bitrate_kbps);
pbos@webrtc.org3349ae02014-03-13 12:52:27 +0000930 default_rtp_rtcp_->SetTargetSendBitrate(stream_bitrates);
pbos@webrtc.org484ee962013-11-21 18:44:23 +0000931 if (video_suspended_ == video_is_suspended)
932 return;
933 video_suspended_ = video_is_suspended;
934 }
pbos@webrtc.org709e2972014-03-19 10:59:52 +0000935
936 // Video suspend-state changed, inform codec observer.
pbos@webrtc.org3349ae02014-03-13 12:52:27 +0000937 CriticalSectionScoped crit(callback_cs_.get());
pbos@webrtc.org484ee962013-11-21 18:44:23 +0000938 if (codec_observer_) {
mflodman@webrtc.org5574dac2014-04-07 10:56:31 +0000939 LOG(LS_INFO) << "Video suspended " << video_is_suspended
940 << " for channel " << channel_id_;
henrik.lundin@webrtc.org9fe36032013-11-21 23:00:40 +0000941 codec_observer_->SuspendChange(channel_id_, video_is_suspended);
henrik.lundin@webrtc.org7ea4f242013-10-02 13:34:26 +0000942 }
niklase@google.com470e71d2011-07-07 08:21:25 +0000943}
944
pwestin@webrtc.org571a1c02012-11-13 21:12:39 +0000945PacedSender* ViEEncoder::GetPacedSender() {
946 return paced_sender_.get();
947}
948
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000949int32_t ViEEncoder::RegisterEffectFilter(ViEEffectFilter* effect_filter) {
mflodman@webrtc.orgd32c4472011-12-22 14:17:53 +0000950 CriticalSectionScoped cs(callback_cs_.get());
mflodman@webrtc.org5574dac2014-04-07 10:56:31 +0000951 if (effect_filter != NULL && effect_filter_ != NULL) {
952 LOG_F(LS_ERROR) << "Filter already set.";
953 return -1;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000954 }
955 effect_filter_ = effect_filter;
956 return 0;
niklase@google.com470e71d2011-07-07 08:21:25 +0000957}
958
mikhal@webrtc.orge41bbdf2012-08-28 16:15:16 +0000959int ViEEncoder::StartDebugRecording(const char* fileNameUTF8) {
960 return vcm_.StartDebugRecording(fileNameUTF8);
961}
962
963int ViEEncoder::StopDebugRecording() {
964 return vcm_.StopDebugRecording();
965}
966
henrik.lundin@webrtc.orgce8e0932013-11-18 12:18:43 +0000967void ViEEncoder::SuspendBelowMinBitrate() {
968 vcm_.SuspendBelowMinBitrate();
henrik.lundin@webrtc.org1a3a6e52013-10-28 10:16:14 +0000969 bitrate_controller_->EnforceMinBitrate(false);
henrik.lundin@webrtc.org7ea4f242013-10-02 13:34:26 +0000970}
971
pbos@webrtc.orgfe1ef932013-10-21 10:34:43 +0000972void ViEEncoder::RegisterPreEncodeCallback(
973 I420FrameCallback* pre_encode_callback) {
974 CriticalSectionScoped cs(callback_cs_.get());
975 pre_encode_callback_ = pre_encode_callback;
976}
977
978void ViEEncoder::DeRegisterPreEncodeCallback() {
979 CriticalSectionScoped cs(callback_cs_.get());
980 pre_encode_callback_ = NULL;
981}
982
sprang@webrtc.org40709352013-11-26 11:41:59 +0000983void ViEEncoder::RegisterPostEncodeImageCallback(
984 EncodedImageCallback* post_encode_callback) {
985 vcm_.RegisterPostEncodeImageCallback(post_encode_callback);
986}
987
988void ViEEncoder::DeRegisterPostEncodeImageCallback() {
989 vcm_.RegisterPostEncodeImageCallback(NULL);
990}
991
marpan@webrtc.orgefd01fd2012-04-18 15:56:34 +0000992QMVideoSettingsCallback::QMVideoSettingsCallback(VideoProcessingModule* vpm)
993 : vpm_(vpm) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000994}
niklase@google.com470e71d2011-07-07 08:21:25 +0000995
stefan@webrtc.org439be292012-02-16 14:45:37 +0000996QMVideoSettingsCallback::~QMVideoSettingsCallback() {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000997}
998
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000999int32_t QMVideoSettingsCallback::SetVideoQMSettings(
1000 const uint32_t frame_rate,
1001 const uint32_t width,
1002 const uint32_t height) {
marpan@webrtc.orgcf706c22012-03-27 21:04:13 +00001003 return vpm_->SetTargetResolution(width, height, frame_rate);
mflodman@webrtc.org84d17832011-12-01 17:02:23 +00001004}
1005
mflodman@webrtc.org84d17832011-12-01 17:02:23 +00001006} // namespace webrtc