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andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +00001/*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 *
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.org281cff82013-05-17 13:44:48 +000011#include "webrtc/video_engine/vie_encoder.h"
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +000012
pbos@webrtc.org3f45c2e2013-08-05 16:22:53 +000013#include <assert.h>
14
stefan@webrtc.org695ff2a2013-06-04 09:36:56 +000015#include <algorithm>
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +000016
sprang@webrtc.org2e98d452013-11-26 11:41:59 +000017#include "webrtc/common_video/interface/video_image.h"
pbos@webrtc.org281cff82013-05-17 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.org2e98d452013-11-26 11:41:59 +000025#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
stefan@webrtc.orgc9995bc2014-07-04 09:20:42 +000026#include "webrtc/system_wrappers/interface/clock.h"
pbos@webrtc.org281cff82013-05-17 13:44:48 +000027#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
28#include "webrtc/system_wrappers/interface/logging.h"
29#include "webrtc/system_wrappers/interface/tick_util.h"
pbos@webrtc.org281cff82013-05-17 13:44:48 +000030#include "webrtc/system_wrappers/interface/trace_event.h"
31#include "webrtc/video_engine/include/vie_codec.h"
32#include "webrtc/video_engine/include/vie_image_process.h"
pbos@webrtc.org24e20892013-10-28 16:32:01 +000033#include "webrtc/frame_callback.h"
pbos@webrtc.org281cff82013-05-17 13:44:48 +000034#include "webrtc/video_engine/vie_defines.h"
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +000035
36namespace webrtc {
37
pwestin@webrtc.org36bdba42013-05-02 19:02:17 +000038// Margin on when we pause the encoder when the pacing buffer overflows relative
39// to the configured buffer delay.
40static const float kEncoderPausePacerMargin = 2.0f;
41
pwestin@webrtc.org3816c522013-04-25 22:20:08 +000042// Don't stop the encoder unless the delay is above this configured value.
43static const int kMinPacingDelayMs = 200;
44
stefan@webrtc.orgdca71b22013-03-27 16:36:01 +000045// Allow packets to be transmitted in up to 2 times max video bitrate if the
46// bandwidth estimate allows it.
47// TODO(holmer): Expose transmission start, min and max bitrates in the
48// VideoEngine API and remove the kTransmissionMaxBitrateMultiplier.
49static const int kTransmissionMaxBitrateMultiplier = 2;
pwestin@webrtc.org5e87b5f2012-11-13 21:12:39 +000050
stefan@webrtc.org5e74d962013-10-18 15:05:29 +000051static const float kStopPaddingThresholdMs = 2000;
52
stefan@webrtc.org7dc17902013-09-06 13:58:01 +000053std::vector<uint32_t> AllocateStreamBitrates(
54 uint32_t total_bitrate,
55 const SimulcastStream* stream_configs,
56 size_t number_of_streams) {
57 if (number_of_streams == 0) {
58 std::vector<uint32_t> stream_bitrates(1, 0);
59 stream_bitrates[0] = total_bitrate;
60 return stream_bitrates;
61 }
62 std::vector<uint32_t> stream_bitrates(number_of_streams, 0);
63 uint32_t bitrate_remainder = total_bitrate;
64 for (size_t i = 0; i < stream_bitrates.size() && bitrate_remainder > 0; ++i) {
65 if (stream_configs[i].maxBitrate * 1000 > bitrate_remainder) {
66 stream_bitrates[i] = bitrate_remainder;
67 } else {
68 stream_bitrates[i] = stream_configs[i].maxBitrate * 1000;
69 }
70 bitrate_remainder -= stream_bitrates[i];
71 }
72 return stream_bitrates;
73}
74
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +000075class QMVideoSettingsCallback : public VCMQMSettingsCallback {
76 public:
77 explicit QMVideoSettingsCallback(VideoProcessingModule* vpm);
mflodman@webrtc.orgbf76ae22013-07-23 11:35:00 +000078
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +000079 ~QMVideoSettingsCallback();
80
81 // Update VPM with QM (quality modes: frame size & frame rate) settings.
pbos@webrtc.org67879bc2013-04-09 13:41:51 +000082 int32_t SetVideoQMSettings(const uint32_t frame_rate,
83 const uint32_t width,
84 const uint32_t height);
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +000085
86 private:
87 VideoProcessingModule* vpm_;
88};
89
90class ViEBitrateObserver : public BitrateObserver {
91 public:
92 explicit ViEBitrateObserver(ViEEncoder* owner)
93 : owner_(owner) {
94 }
mflodman@webrtc.orgbf76ae22013-07-23 11:35:00 +000095 virtual ~ViEBitrateObserver() {}
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +000096 // Implements BitrateObserver.
97 virtual void OnNetworkChanged(const uint32_t bitrate_bps,
98 const uint8_t fraction_lost,
99 const uint32_t rtt) {
100 owner_->OnNetworkChanged(bitrate_bps, fraction_lost, rtt);
101 }
102 private:
103 ViEEncoder* owner_;
104};
105
pwestin@webrtc.org5e87b5f2012-11-13 21:12:39 +0000106class ViEPacedSenderCallback : public PacedSender::Callback {
107 public:
108 explicit ViEPacedSenderCallback(ViEEncoder* owner)
109 : owner_(owner) {
110 }
mflodman@webrtc.orgbf76ae22013-07-23 11:35:00 +0000111 virtual ~ViEPacedSenderCallback() {}
hclam@chromium.org0f6f7cb2013-06-20 20:18:31 +0000112 virtual bool TimeToSendPacket(uint32_t ssrc, uint16_t sequence_number,
stefan@webrtc.orgb748c9d2013-11-13 15:29:21 +0000113 int64_t capture_time_ms, bool retransmission) {
114 return owner_->TimeToSendPacket(ssrc, sequence_number, capture_time_ms,
115 retransmission);
pwestin@webrtc.org5e87b5f2012-11-13 21:12:39 +0000116 }
stefan@webrtc.org695ff2a2013-06-04 09:36:56 +0000117 virtual int TimeToSendPadding(int bytes) {
stefan@webrtc.org69f76052013-06-17 12:53:37 +0000118 return owner_->TimeToSendPadding(bytes);
pwestin@webrtc.org5e87b5f2012-11-13 21:12:39 +0000119 }
120 private:
121 ViEEncoder* owner_;
122};
123
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000124ViEEncoder::ViEEncoder(int32_t engine_id,
125 int32_t channel_id,
126 uint32_t number_of_cores,
andresp@webrtc.orgac6d9192013-05-13 10:50:50 +0000127 const Config& config,
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000128 ProcessThread& module_process_thread,
129 BitrateController* bitrate_controller)
130 : engine_id_(engine_id),
131 channel_id_(channel_id),
132 number_of_cores_(number_of_cores),
stefan@webrtc.org8edccce2014-04-11 14:08:35 +0000133 vcm_(*webrtc::VideoCodingModule::Create()),
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000134 vpm_(*webrtc::VideoProcessingModule::Create(ViEModuleId(engine_id,
135 channel_id))),
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000136 callback_cs_(CriticalSectionWrapper::CreateCriticalSection()),
137 data_cs_(CriticalSectionWrapper::CreateCriticalSection()),
138 bitrate_controller_(bitrate_controller),
stefan@webrtc.org5e74d962013-10-18 15:05:29 +0000139 time_of_last_incoming_frame_ms_(0),
stefan@webrtc.org69f76052013-06-17 12:53:37 +0000140 send_padding_(false),
pbos@webrtc.org3f83f9c2014-03-13 12:52:27 +0000141 min_transmit_bitrate_kbps_(0),
stefan@webrtc.orgdca71b22013-03-27 16:36:01 +0000142 target_delay_ms_(0),
143 network_is_transmitting_(true),
144 encoder_paused_(false),
pwestin@webrtc.org36bdba42013-05-02 19:02:17 +0000145 encoder_paused_and_dropped_frame_(false),
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000146 fec_enabled_(false),
147 nack_enabled_(false),
148 codec_observer_(NULL),
149 effect_filter_(NULL),
150 module_process_thread_(module_process_thread),
151 has_received_sli_(false),
152 picture_id_sli_(0),
153 has_received_rpsi_(false),
154 picture_id_rpsi_(0),
henrik.lundin@webrtc.org39079d12013-10-02 13:34:26 +0000155 qm_callback_(NULL),
henrik.lundin@webrtc.org45901772013-11-18 12:18:43 +0000156 video_suspended_(false),
pbos@webrtc.org63301bd2013-10-21 10:34:43 +0000157 pre_encode_callback_(NULL) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000158 RtpRtcp::Configuration configuration;
159 configuration.id = ViEModuleId(engine_id_, channel_id_);
160 configuration.audio = false; // Video.
161
162 default_rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(configuration));
163 bitrate_observer_.reset(new ViEBitrateObserver(this));
pwestin@webrtc.org5e87b5f2012-11-13 21:12:39 +0000164 pacing_callback_.reset(new ViEPacedSenderCallback(this));
pwestin@webrtc.org36bdba42013-05-02 19:02:17 +0000165 paced_sender_.reset(
stefan@webrtc.orgc9995bc2014-07-04 09:20:42 +0000166 new PacedSender(Clock::GetRealTimeClock(), pacing_callback_.get(),
167 PacedSender::kDefaultInitialPaceKbps, 0));
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000168}
169
170bool ViEEncoder::Init() {
171 if (vcm_.InitializeSender() != 0) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000172 return false;
173 }
174 vpm_.EnableTemporalDecimation(true);
175
176 // Enable/disable content analysis: off by default for now.
177 vpm_.EnableContentAnalysis(false);
178
pwestin@webrtc.org5e87b5f2012-11-13 21:12:39 +0000179 if (module_process_thread_.RegisterModule(&vcm_) != 0 ||
180 module_process_thread_.RegisterModule(default_rtp_rtcp_.get()) != 0 ||
181 module_process_thread_.RegisterModule(paced_sender_.get()) != 0) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000182 return false;
183 }
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000184 if (qm_callback_) {
185 delete qm_callback_;
186 }
187 qm_callback_ = new QMVideoSettingsCallback(&vpm_);
188
189#ifdef VIDEOCODEC_VP8
andresp@webrtc.orgc1696da2014-08-14 16:46:46 +0000190 VideoCodecType codec_type = webrtc::kVideoCodecVP8;
191#else
192 VideoCodecType codec_type = webrtc::kVideoCodecI420;
193#endif
194
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000195 VideoCodec video_codec;
andresp@webrtc.orgc1696da2014-08-14 16:46:46 +0000196 if (vcm_.Codec(codec_type, &video_codec) != VCM_OK) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000197 return false;
198 }
stefan@webrtc.org5fcd7df2014-02-13 13:48:38 +0000199 {
200 CriticalSectionScoped cs(data_cs_.get());
201 send_padding_ = video_codec.numberOfSimulcastStreams > 1;
202 }
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000203 if (vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
204 default_rtp_rtcp_->MaxDataPayloadLength()) != 0) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000205 return false;
206 }
207 if (default_rtp_rtcp_->RegisterSendPayload(video_codec) != 0) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000208 return false;
209 }
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000210 if (vcm_.RegisterTransportCallback(this) != 0) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000211 return false;
212 }
213 if (vcm_.RegisterSendStatisticsCallback(this) != 0) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000214 return false;
215 }
216 if (vcm_.RegisterVideoQMCallback(qm_callback_) != 0) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000217 return false;
218 }
219 return true;
220}
221
222ViEEncoder::~ViEEncoder() {
stefan@webrtc.orgc0539d92012-11-29 09:18:53 +0000223 if (bitrate_controller_) {
224 bitrate_controller_->RemoveBitrateObserver(bitrate_observer_.get());
225 }
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000226 module_process_thread_.DeRegisterModule(&vcm_);
227 module_process_thread_.DeRegisterModule(&vpm_);
228 module_process_thread_.DeRegisterModule(default_rtp_rtcp_.get());
pwestin@webrtc.org5e87b5f2012-11-13 21:12:39 +0000229 module_process_thread_.DeRegisterModule(paced_sender_.get());
mflodman@webrtc.orgc6242c92013-03-01 14:51:23 +0000230 VideoCodingModule::Destroy(&vcm_);
231 VideoProcessingModule::Destroy(&vpm_);
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000232 delete qm_callback_;
233}
234
235int ViEEncoder::Owner() const {
236 return channel_id_;
237}
238
stefan@webrtc.orgdca71b22013-03-27 16:36:01 +0000239void ViEEncoder::SetNetworkTransmissionState(bool is_transmitting) {
stefan@webrtc.orgdca71b22013-03-27 16:36:01 +0000240 {
241 CriticalSectionScoped cs(data_cs_.get());
242 network_is_transmitting_ = is_transmitting;
243 }
244 if (is_transmitting) {
245 paced_sender_->Resume();
246 } else {
247 paced_sender_->Pause();
248 }
249}
250
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000251void ViEEncoder::Pause() {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000252 CriticalSectionScoped cs(data_cs_.get());
stefan@webrtc.orgdca71b22013-03-27 16:36:01 +0000253 encoder_paused_ = true;
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000254}
255
256void ViEEncoder::Restart() {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000257 CriticalSectionScoped cs(data_cs_.get());
stefan@webrtc.orgdca71b22013-03-27 16:36:01 +0000258 encoder_paused_ = false;
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000259}
260
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000261uint8_t ViEEncoder::NumberOfCodecs() {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000262 return vcm_.NumberOfCodecs();
263}
264
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000265int32_t ViEEncoder::GetCodec(uint8_t list_index, VideoCodec* video_codec) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000266 if (vcm_.Codec(list_index, video_codec) != 0) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000267 return -1;
268 }
269 return 0;
270}
271
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000272int32_t ViEEncoder::RegisterExternalEncoder(webrtc::VideoEncoder* encoder,
273 uint8_t pl_type,
274 bool internal_source) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000275 if (encoder == NULL)
276 return -1;
277
stefan@webrtc.org71f3f682013-01-09 08:35:40 +0000278 if (vcm_.RegisterExternalEncoder(encoder, pl_type, internal_source) !=
mflodman@webrtc.org022615b2014-04-07 10:56:31 +0000279 VCM_OK) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000280 return -1;
281 }
282 return 0;
283}
284
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000285int32_t ViEEncoder::DeRegisterExternalEncoder(uint8_t pl_type) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000286 webrtc::VideoCodec current_send_codec;
287 if (vcm_.SendCodec(&current_send_codec) == VCM_OK) {
stefan@webrtc.org63136922013-03-19 10:04:57 +0000288 uint32_t current_bitrate_bps = 0;
289 if (vcm_.Bitrate(&current_bitrate_bps) != 0) {
mflodman@webrtc.org022615b2014-04-07 10:56:31 +0000290 LOG(LS_WARNING) << "Failed to get the current encoder target bitrate.";
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000291 }
stefan@webrtc.org63136922013-03-19 10:04:57 +0000292 current_send_codec.startBitrate = (current_bitrate_bps + 500) / 1000;
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000293 }
294
295 if (vcm_.RegisterExternalEncoder(NULL, pl_type) != VCM_OK) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000296 return -1;
297 }
298
stefan@webrtc.org69f76052013-06-17 12:53:37 +0000299 // If the external encoder is the current send codec, use vcm internal
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000300 // encoder.
301 if (current_send_codec.plType == pl_type) {
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000302 uint16_t max_data_payload_length =
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000303 default_rtp_rtcp_->MaxDataPayloadLength();
stefan@webrtc.org5fcd7df2014-02-13 13:48:38 +0000304 {
305 CriticalSectionScoped cs(data_cs_.get());
306 send_padding_ = current_send_codec.numberOfSimulcastStreams > 1;
307 }
fischman@webrtc.orgbbbe9b82014-03-07 18:00:05 +0000308 // TODO(mflodman): Unfortunately the VideoCodec that VCM has cached a
309 // raw pointer to an |extra_options| that's long gone. Clearing it here is
310 // a hack to prevent the following code from crashing. This should be fixed
311 // for realz. https://code.google.com/p/chromium/issues/detail?id=348222
312 current_send_codec.extra_options = NULL;
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000313 if (vcm_.RegisterSendCodec(&current_send_codec, number_of_cores_,
314 max_data_payload_length) != VCM_OK) {
stefan@webrtc.org6111d792014-07-16 11:20:40 +0000315 LOG(LS_INFO) << "De-registered the currently used external encoder ("
316 << static_cast<int>(pl_type) << ") and therefore tried to "
317 << "register the corresponding internal encoder, but none "
318 << "was supported.";
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000319 }
320 }
321 return 0;
322}
323
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000324int32_t ViEEncoder::SetEncoder(const webrtc::VideoCodec& video_codec) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000325 // Setting target width and height for VPM.
326 if (vpm_.SetTargetResolution(video_codec.width, video_codec.height,
327 video_codec.maxFramerate) != VPM_OK) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000328 return -1;
329 }
330
331 if (default_rtp_rtcp_->RegisterSendPayload(video_codec) != 0) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000332 return -1;
333 }
334 // Convert from kbps to bps.
stefan@webrtc.org7dc17902013-09-06 13:58:01 +0000335 std::vector<uint32_t> stream_bitrates = AllocateStreamBitrates(
336 video_codec.startBitrate * 1000,
337 video_codec.simulcastStream,
338 video_codec.numberOfSimulcastStreams);
339 default_rtp_rtcp_->SetTargetSendBitrate(stream_bitrates);
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000340
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000341 uint16_t max_data_payload_length =
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000342 default_rtp_rtcp_->MaxDataPayloadLength();
343
stefan@webrtc.org5a6e6912014-02-14 09:45:58 +0000344 {
345 CriticalSectionScoped cs(data_cs_.get());
346 send_padding_ = video_codec.numberOfSimulcastStreams > 1;
347 }
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000348 if (vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
349 max_data_payload_length) != VCM_OK) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000350 return -1;
351 }
352
353 // Set this module as sending right away, let the slave module in the channel
354 // start and stop sending.
andresp@webrtc.orgc1696da2014-08-14 16:46:46 +0000355 if (default_rtp_rtcp_->SetSendingStatus(true) != 0) {
356 return -1;
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000357 }
andresp@webrtc.orgc1696da2014-08-14 16:46:46 +0000358
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000359 bitrate_controller_->SetBitrateObserver(bitrate_observer_.get(),
360 video_codec.startBitrate * 1000,
361 video_codec.minBitrate * 1000,
stefan@webrtc.orgdca71b22013-03-27 16:36:01 +0000362 kTransmissionMaxBitrateMultiplier *
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000363 video_codec.maxBitrate * 1000);
364
pbos@webrtc.org3f83f9c2014-03-13 12:52:27 +0000365 CriticalSectionScoped crit(data_cs_.get());
366 int pad_up_to_bitrate_kbps = video_codec.startBitrate;
367 if (pad_up_to_bitrate_kbps < min_transmit_bitrate_kbps_)
368 pad_up_to_bitrate_kbps = min_transmit_bitrate_kbps_;
369
stefan@webrtc.orgc9995bc2014-07-04 09:20:42 +0000370 paced_sender_->UpdateBitrate(
371 PacedSender::kDefaultPaceMultiplier * video_codec.startBitrate,
372 pad_up_to_bitrate_kbps);
pbos@webrtc.org3f83f9c2014-03-13 12:52:27 +0000373
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000374 return 0;
375}
376
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000377int32_t ViEEncoder::GetEncoder(VideoCodec* video_codec) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000378 if (vcm_.SendCodec(video_codec) != 0) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000379 return -1;
380 }
381 return 0;
382}
383
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000384int32_t ViEEncoder::GetCodecConfigParameters(
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000385 unsigned char config_parameters[kConfigParameterSize],
386 unsigned char& config_parameters_size) {
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000387 int32_t num_parameters =
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000388 vcm_.CodecConfigParameters(config_parameters, kConfigParameterSize);
389 if (num_parameters <= 0) {
390 config_parameters_size = 0;
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000391 return -1;
392 }
393 config_parameters_size = static_cast<unsigned char>(num_parameters);
394 return 0;
395}
396
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000397int32_t ViEEncoder::ScaleInputImage(bool enable) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000398 VideoFrameResampling resampling_mode = kFastRescaling;
mflodman@webrtc.org022615b2014-04-07 10:56:31 +0000399 // TODO(mflodman) What?
400 if (enable) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000401 // kInterpolation is currently not supported.
mflodman@webrtc.org022615b2014-04-07 10:56:31 +0000402 LOG_F(LS_ERROR) << "Not supported.";
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000403 return -1;
404 }
405 vpm_.SetInputFrameResampleMode(resampling_mode);
406
407 return 0;
408}
409
stefan@webrtc.orgb748c9d2013-11-13 15:29:21 +0000410bool ViEEncoder::TimeToSendPacket(uint32_t ssrc,
411 uint16_t sequence_number,
412 int64_t capture_time_ms,
413 bool retransmission) {
hclam@chromium.org0f6f7cb2013-06-20 20:18:31 +0000414 return default_rtp_rtcp_->TimeToSendPacket(ssrc, sequence_number,
stefan@webrtc.orgb748c9d2013-11-13 15:29:21 +0000415 capture_time_ms, retransmission);
pwestin@webrtc.org5e87b5f2012-11-13 21:12:39 +0000416}
417
stefan@webrtc.org69f76052013-06-17 12:53:37 +0000418int ViEEncoder::TimeToSendPadding(int bytes) {
henrik.lundin@webrtc.orgb9f1eb82013-11-21 14:05:40 +0000419 bool send_padding;
420 {
421 CriticalSectionScoped cs(data_cs_.get());
pbos@webrtc.org3f83f9c2014-03-13 12:52:27 +0000422 send_padding =
423 send_padding_ || video_suspended_ || min_transmit_bitrate_kbps_ > 0;
henrik.lundin@webrtc.orgb9f1eb82013-11-21 14:05:40 +0000424 }
425 if (send_padding) {
stefan@webrtc.org69f76052013-06-17 12:53:37 +0000426 return default_rtp_rtcp_->TimeToSendPadding(bytes);
427 }
428 return 0;
429}
430
stefan@webrtc.orgdca71b22013-03-27 16:36:01 +0000431bool ViEEncoder::EncoderPaused() const {
pwestin@webrtc.org3816c522013-04-25 22:20:08 +0000432 // Pause video if paused by caller or as long as the network is down or the
433 // pacer queue has grown too large in buffered mode.
434 if (encoder_paused_) {
435 return true;
436 }
437 if (target_delay_ms_ > 0) {
438 // Buffered mode.
439 // TODO(pwestin): Workaround until nack is configured as a time and not
440 // number of packets.
441 return paced_sender_->QueueInMs() >=
pwestin@webrtc.org36bdba42013-05-02 19:02:17 +0000442 std::max(static_cast<int>(target_delay_ms_ * kEncoderPausePacerMargin),
443 kMinPacingDelayMs);
pwestin@webrtc.org3816c522013-04-25 22:20:08 +0000444 }
445 return !network_is_transmitting_;
stefan@webrtc.orgdca71b22013-03-27 16:36:01 +0000446}
447
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000448RtpRtcp* ViEEncoder::SendRtpRtcpModule() {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000449 return default_rtp_rtcp_.get();
450}
451
452void ViEEncoder::DeliverFrame(int id,
mikhal@webrtc.org3bbed742012-10-24 18:33:04 +0000453 I420VideoFrame* video_frame,
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000454 int num_csrcs,
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000455 const uint32_t CSRC[kRtpCsrcSize]) {
wuchengli@chromium.org76cd2f72014-03-19 03:44:20 +0000456 if (default_rtp_rtcp_->SendingMedia() == false) {
457 // We've paused or we have no channels attached, don't encode.
458 return;
459 }
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000460 {
461 CriticalSectionScoped cs(data_cs_.get());
stefan@webrtc.org5e74d962013-10-18 15:05:29 +0000462 time_of_last_incoming_frame_ms_ = TickTime::MillisecondTimestamp();
pwestin@webrtc.org36bdba42013-05-02 19:02:17 +0000463 if (EncoderPaused()) {
464 if (!encoder_paused_and_dropped_frame_) {
465 TRACE_EVENT_ASYNC_BEGIN0("webrtc", "EncoderPaused", this);
466 }
467 encoder_paused_and_dropped_frame_ = true;
468 return;
469 }
470 if (encoder_paused_and_dropped_frame_) {
471 TRACE_EVENT_ASYNC_END0("webrtc", "EncoderPaused", this);
472 }
473 encoder_paused_and_dropped_frame_ = false;
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000474 }
475
476 // Convert render time, in ms, to RTP timestamp.
477 const int kMsToRtpTimestamp = 90;
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000478 const uint32_t time_stamp =
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000479 kMsToRtpTimestamp *
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000480 static_cast<uint32_t>(video_frame->render_time_ms());
hclam@chromium.org74472fe2013-04-09 19:54:10 +0000481
hclam@chromium.org9c0f14d2013-07-08 21:31:18 +0000482 TRACE_EVENT_ASYNC_STEP0("webrtc", "Video", video_frame->render_time_ms(),
483 "Encode");
mikhal@webrtc.org3bbed742012-10-24 18:33:04 +0000484 video_frame->set_timestamp(time_stamp);
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000485
486 // Make sure the CSRC list is correct.
487 if (num_csrcs > 0) {
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000488 uint32_t tempCSRC[kRtpCsrcSize];
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000489 for (int i = 0; i < num_csrcs; i++) {
490 if (CSRC[i] == 1) {
491 tempCSRC[i] = default_rtp_rtcp_->SSRC();
492 } else {
493 tempCSRC[i] = CSRC[i];
494 }
495 }
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000496 default_rtp_rtcp_->SetCSRCs(tempCSRC, (uint8_t) num_csrcs);
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000497 }
wuchengli@chromium.org4ee63482014-06-20 12:04:05 +0000498
pwestin@webrtc.org28f76e52012-10-30 16:21:52 +0000499 I420VideoFrame* decimated_frame = NULL;
wuchengli@chromium.org4ee63482014-06-20 12:04:05 +0000500 // TODO(wuchengli): support texture frames.
501 if (video_frame->native_handle() == NULL) {
502 {
503 CriticalSectionScoped cs(callback_cs_.get());
504 if (effect_filter_) {
505 unsigned int length =
506 CalcBufferSize(kI420, video_frame->width(), video_frame->height());
507 scoped_ptr<uint8_t[]> video_buffer(new uint8_t[length]);
508 ExtractBuffer(*video_frame, length, video_buffer.get());
509 effect_filter_->Transform(length,
510 video_buffer.get(),
511 video_frame->ntp_time_ms(),
512 video_frame->timestamp(),
513 video_frame->width(),
514 video_frame->height());
515 }
516 }
517
518 // Pass frame via preprocessor.
519 const int ret = vpm_.PreprocessFrame(*video_frame, &decimated_frame);
520 if (ret == 1) {
521 // Drop this frame.
522 return;
523 }
524 if (ret != VPM_OK) {
525 return;
526 }
pwestin@webrtc.org28f76e52012-10-30 16:21:52 +0000527 }
wuchengli@chromium.org4ee63482014-06-20 12:04:05 +0000528 // If the frame was not resampled or scaled => use original.
pwestin@webrtc.org28f76e52012-10-30 16:21:52 +0000529 if (decimated_frame == NULL) {
530 decimated_frame = video_frame;
531 }
pbos@webrtc.org63301bd2013-10-21 10:34:43 +0000532
533 {
534 CriticalSectionScoped cs(callback_cs_.get());
535 if (pre_encode_callback_)
536 pre_encode_callback_->FrameCallback(decimated_frame);
537 }
538
wuchengli@chromium.org4ee63482014-06-20 12:04:05 +0000539 if (video_frame->native_handle() != NULL) {
540 // TODO(wuchengli): add texture support. http://crbug.com/362437
541 return;
542 }
543
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000544#ifdef VIDEOCODEC_VP8
545 if (vcm_.SendCodec() == webrtc::kVideoCodecVP8) {
546 webrtc::CodecSpecificInfo codec_specific_info;
547 codec_specific_info.codecType = webrtc::kVideoCodecVP8;
stefan@webrtc.orgfedbe8b2014-07-09 14:46:31 +0000548 {
549 CriticalSectionScoped cs(data_cs_.get());
550 codec_specific_info.codecSpecific.VP8.hasReceivedRPSI =
551 has_received_rpsi_;
552 codec_specific_info.codecSpecific.VP8.hasReceivedSLI =
553 has_received_sli_;
554 codec_specific_info.codecSpecific.VP8.pictureIdRPSI =
555 picture_id_rpsi_;
556 codec_specific_info.codecSpecific.VP8.pictureIdSLI =
557 picture_id_sli_;
558 has_received_sli_ = false;
559 has_received_rpsi_ = false;
560 }
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000561
mflodman@webrtc.org022615b2014-04-07 10:56:31 +0000562 vcm_.AddVideoFrame(*decimated_frame, vpm_.ContentMetrics(),
563 &codec_specific_info);
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000564 return;
565 }
566#endif
mflodman@webrtc.org022615b2014-04-07 10:56:31 +0000567 vcm_.AddVideoFrame(*decimated_frame);
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000568}
569
570void ViEEncoder::DelayChanged(int id, int frame_delay) {
stefan@webrtc.org73ebe672013-04-09 14:56:29 +0000571 default_rtp_rtcp_->SetCameraDelay(frame_delay);
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000572}
573
574int ViEEncoder::GetPreferedFrameSettings(int* width,
575 int* height,
576 int* frame_rate) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000577 webrtc::VideoCodec video_codec;
578 memset(&video_codec, 0, sizeof(video_codec));
579 if (vcm_.SendCodec(&video_codec) != VCM_OK) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000580 return -1;
581 }
582
583 *width = video_codec.width;
584 *height = video_codec.height;
585 *frame_rate = video_codec.maxFramerate;
586 return 0;
587}
588
589int ViEEncoder::SendKeyFrame() {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000590 return vcm_.IntraFrameRequest(0);
591}
592
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000593int32_t ViEEncoder::SendCodecStatistics(
594 uint32_t* num_key_frames, uint32_t* num_delta_frames) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000595 webrtc::VCMFrameCount sent_frames;
596 if (vcm_.SentFrameCount(sent_frames) != VCM_OK) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000597 return -1;
598 }
599 *num_key_frames = sent_frames.numKeyFrames;
600 *num_delta_frames = sent_frames.numDeltaFrames;
601 return 0;
602}
603
jiayl@webrtc.org55a2a272014-02-27 22:32:40 +0000604int32_t ViEEncoder::PacerQueuingDelayMs() const {
605 return paced_sender_->QueueInMs();
606}
607
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000608int ViEEncoder::CodecTargetBitrate(uint32_t* bitrate) const {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000609 if (vcm_.Bitrate(bitrate) != 0)
610 return -1;
611 return 0;
612}
613
wu@webrtc.org7fc75bb2013-08-15 23:38:54 +0000614int32_t ViEEncoder::UpdateProtectionMethod(bool enable_nack) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000615 bool fec_enabled = false;
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000616 uint8_t dummy_ptype_red = 0;
617 uint8_t dummy_ptypeFEC = 0;
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000618
619 // Updated protection method to VCM to get correct packetization sizes.
620 // FEC has larger overhead than NACK -> set FEC if used.
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000621 int32_t error = default_rtp_rtcp_->GenericFECStatus(fec_enabled,
622 dummy_ptype_red,
623 dummy_ptypeFEC);
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000624 if (error) {
625 return -1;
626 }
wu@webrtc.org7fc75bb2013-08-15 23:38:54 +0000627 if (fec_enabled_ == fec_enabled && nack_enabled_ == enable_nack) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000628 // No change needed, we're already in correct state.
629 return 0;
630 }
631 fec_enabled_ = fec_enabled;
wu@webrtc.org7fc75bb2013-08-15 23:38:54 +0000632 nack_enabled_ = enable_nack;
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000633
634 // Set Video Protection for VCM.
wu@webrtc.org7fc75bb2013-08-15 23:38:54 +0000635 if (fec_enabled && nack_enabled_) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000636 vcm_.SetVideoProtection(webrtc::kProtectionNackFEC, true);
637 } else {
638 vcm_.SetVideoProtection(webrtc::kProtectionFEC, fec_enabled_);
wu@webrtc.org7fc75bb2013-08-15 23:38:54 +0000639 vcm_.SetVideoProtection(webrtc::kProtectionNackSender, nack_enabled_);
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000640 vcm_.SetVideoProtection(webrtc::kProtectionNackFEC, false);
641 }
642
wu@webrtc.org7fc75bb2013-08-15 23:38:54 +0000643 if (fec_enabled_ || nack_enabled_) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000644 vcm_.RegisterProtectionCallback(this);
645 // The send codec must be registered to set correct MTU.
646 webrtc::VideoCodec codec;
647 if (vcm_.SendCodec(&codec) == 0) {
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000648 uint16_t max_pay_load = default_rtp_rtcp_->MaxDataPayloadLength();
stefan@webrtc.org63136922013-03-19 10:04:57 +0000649 uint32_t current_bitrate_bps = 0;
650 if (vcm_.Bitrate(&current_bitrate_bps) != 0) {
mflodman@webrtc.org022615b2014-04-07 10:56:31 +0000651 LOG_F(LS_WARNING) <<
652 "Failed to get the current encoder target bitrate.";
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000653 }
stefan@webrtc.org63136922013-03-19 10:04:57 +0000654 // Convert to start bitrate in kbps.
655 codec.startBitrate = (current_bitrate_bps + 500) / 1000;
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000656 if (vcm_.RegisterSendCodec(&codec, number_of_cores_, max_pay_load) != 0) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000657 return -1;
658 }
659 }
660 return 0;
661 } else {
662 // FEC and NACK are disabled.
663 vcm_.RegisterProtectionCallback(NULL);
664 }
665 return 0;
666}
667
mikhal@webrtc.org9d6fcb32013-02-15 23:22:18 +0000668void ViEEncoder::SetSenderBufferingMode(int target_delay_ms) {
stefan@webrtc.orgdca71b22013-03-27 16:36:01 +0000669 {
670 CriticalSectionScoped cs(data_cs_.get());
671 target_delay_ms_ = target_delay_ms;
672 }
mikhal@webrtc.org0c66de62013-02-10 18:42:55 +0000673 if (target_delay_ms > 0) {
stefan@webrtc.orgdca71b22013-03-27 16:36:01 +0000674 // Disable external frame-droppers.
675 vcm_.EnableFrameDropper(false);
676 vpm_.EnableTemporalDecimation(false);
stefan@webrtc.org04d65932013-11-27 14:16:20 +0000677 // We don't put any limits on the pacer queue when running in buffered mode
678 // since the encoder will be paused if the queue grow too large.
679 paced_sender_->set_max_queue_length_ms(-1);
mikhal@webrtc.org0c66de62013-02-10 18:42:55 +0000680 } else {
mikhal@webrtc.org9d6fcb32013-02-15 23:22:18 +0000681 // Real-time mode - enable frame droppers.
mikhal@webrtc.org0c66de62013-02-10 18:42:55 +0000682 vpm_.EnableTemporalDecimation(true);
683 vcm_.EnableFrameDropper(true);
stefan@webrtc.org04d65932013-11-27 14:16:20 +0000684 paced_sender_->set_max_queue_length_ms(
685 PacedSender::kDefaultMaxQueueLengthMs);
mikhal@webrtc.org0c66de62013-02-10 18:42:55 +0000686 }
687}
688
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000689int32_t ViEEncoder::SendData(
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000690 const FrameType frame_type,
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000691 const uint8_t payload_type,
692 const uint32_t time_stamp,
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000693 int64_t capture_time_ms,
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000694 const uint8_t* payload_data,
695 const uint32_t payload_size,
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000696 const webrtc::RTPFragmentationHeader& fragmentation_header,
697 const RTPVideoHeader* rtp_video_hdr) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000698 // New encoded data, hand over to the rtp module.
699 return default_rtp_rtcp_->SendOutgoingData(frame_type,
700 payload_type,
701 time_stamp,
702 capture_time_ms,
703 payload_data,
704 payload_size,
705 &fragmentation_header,
706 rtp_video_hdr);
707}
708
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000709int32_t ViEEncoder::ProtectionRequest(
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000710 const FecProtectionParams* delta_fec_params,
711 const FecProtectionParams* key_fec_params,
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000712 uint32_t* sent_video_rate_bps,
713 uint32_t* sent_nack_rate_bps,
714 uint32_t* sent_fec_rate_bps) {
mflodman@webrtc.org022615b2014-04-07 10:56:31 +0000715 default_rtp_rtcp_->SetFecParameters(delta_fec_params, key_fec_params);
716 default_rtp_rtcp_->BitrateSent(NULL, sent_video_rate_bps, sent_fec_rate_bps,
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000717 sent_nack_rate_bps);
718 return 0;
719}
720
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000721int32_t ViEEncoder::SendStatistics(const uint32_t bit_rate,
722 const uint32_t frame_rate) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000723 CriticalSectionScoped cs(callback_cs_.get());
724 if (codec_observer_) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000725 codec_observer_->OutgoingRate(channel_id_, frame_rate, bit_rate);
726 }
727 return 0;
728}
729
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000730int32_t ViEEncoder::RegisterCodecObserver(ViEEncoderObserver* observer) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000731 CriticalSectionScoped cs(callback_cs_.get());
mflodman@webrtc.org022615b2014-04-07 10:56:31 +0000732 if (observer && codec_observer_) {
733 LOG_F(LS_ERROR) << "Observer already set.";
734 return -1;
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000735 }
mflodman@webrtc.org022615b2014-04-07 10:56:31 +0000736 codec_observer_ = observer;
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000737 return 0;
738}
739
740void ViEEncoder::OnReceivedSLI(uint32_t /*ssrc*/,
741 uint8_t picture_id) {
stefan@webrtc.orgfedbe8b2014-07-09 14:46:31 +0000742 CriticalSectionScoped cs(data_cs_.get());
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000743 picture_id_sli_ = picture_id;
744 has_received_sli_ = true;
745}
746
747void ViEEncoder::OnReceivedRPSI(uint32_t /*ssrc*/,
748 uint64_t picture_id) {
stefan@webrtc.orgfedbe8b2014-07-09 14:46:31 +0000749 CriticalSectionScoped cs(data_cs_.get());
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000750 picture_id_rpsi_ = picture_id;
751 has_received_rpsi_ = true;
752}
753
mflodman@webrtc.orgb6d9cfc2012-10-25 11:30:29 +0000754void ViEEncoder::OnReceivedIntraFrameRequest(uint32_t ssrc) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000755 // Key frame request from remote side, signal to VCM.
justinlin@chromium.orgd474c132013-05-13 22:59:00 +0000756 TRACE_EVENT0("webrtc", "OnKeyFrameRequest");
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000757
mflodman@webrtc.orgb6d9cfc2012-10-25 11:30:29 +0000758 int idx = 0;
759 {
760 CriticalSectionScoped cs(data_cs_.get());
761 std::map<unsigned int, int>::iterator stream_it = ssrc_streams_.find(ssrc);
762 if (stream_it == ssrc_streams_.end()) {
mflodman@webrtc.orgac094232012-12-20 09:26:17 +0000763 LOG_F(LS_WARNING) << "ssrc not found: " << ssrc << ", map size "
764 << ssrc_streams_.size();
mflodman@webrtc.orgb6d9cfc2012-10-25 11:30:29 +0000765 return;
766 }
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000767 std::map<unsigned int, int64_t>::iterator time_it =
mflodman@webrtc.orgb6d9cfc2012-10-25 11:30:29 +0000768 time_last_intra_request_ms_.find(ssrc);
769 if (time_it == time_last_intra_request_ms_.end()) {
770 time_last_intra_request_ms_[ssrc] = 0;
771 }
772
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000773 int64_t now = TickTime::MillisecondTimestamp();
mflodman@webrtc.orgb6d9cfc2012-10-25 11:30:29 +0000774 if (time_last_intra_request_ms_[ssrc] + kViEMinKeyRequestIntervalMs > now) {
mflodman@webrtc.orgb6d9cfc2012-10-25 11:30:29 +0000775 return;
776 }
777 time_last_intra_request_ms_[ssrc] = now;
778 idx = stream_it->second;
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000779 }
mflodman@webrtc.orgb6d9cfc2012-10-25 11:30:29 +0000780 // Release the critsect before triggering key frame.
781 vcm_.IntraFrameRequest(idx);
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000782}
783
784void ViEEncoder::OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) {
mflodman@webrtc.orgb6d9cfc2012-10-25 11:30:29 +0000785 CriticalSectionScoped cs(data_cs_.get());
786 std::map<unsigned int, int>::iterator it = ssrc_streams_.find(old_ssrc);
787 if (it == ssrc_streams_.end()) {
788 return;
789 }
790
791 ssrc_streams_[new_ssrc] = it->second;
792 ssrc_streams_.erase(it);
793
794 std::map<unsigned int, int64_t>::iterator time_it =
795 time_last_intra_request_ms_.find(old_ssrc);
796 int64_t last_intra_request_ms = 0;
797 if (time_it != time_last_intra_request_ms_.end()) {
798 last_intra_request_ms = time_it->second;
799 time_last_intra_request_ms_.erase(time_it);
800 }
801 time_last_intra_request_ms_[new_ssrc] = last_intra_request_ms;
802}
803
804bool ViEEncoder::SetSsrcs(const std::list<unsigned int>& ssrcs) {
805 VideoCodec codec;
806 if (vcm_.SendCodec(&codec) != 0)
807 return false;
808
809 if (codec.numberOfSimulcastStreams > 0 &&
810 ssrcs.size() != codec.numberOfSimulcastStreams) {
811 return false;
812 }
813
814 CriticalSectionScoped cs(data_cs_.get());
815 ssrc_streams_.clear();
816 time_last_intra_request_ms_.clear();
817 int idx = 0;
818 for (std::list<unsigned int>::const_iterator it = ssrcs.begin();
819 it != ssrcs.end(); ++it, ++idx) {
820 unsigned int ssrc = *it;
821 ssrc_streams_[ssrc] = idx;
822 }
823 return true;
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000824}
825
pbos@webrtc.org3f83f9c2014-03-13 12:52:27 +0000826void ViEEncoder::SetMinTransmitBitrate(int min_transmit_bitrate_kbps) {
827 assert(min_transmit_bitrate_kbps >= 0);
828 CriticalSectionScoped crit(data_cs_.get());
829 min_transmit_bitrate_kbps_ = min_transmit_bitrate_kbps;
830}
831
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000832// Called from ViEBitrateObserver.
833void ViEEncoder::OnNetworkChanged(const uint32_t bitrate_bps,
834 const uint8_t fraction_lost,
835 const uint32_t round_trip_time_ms) {
mflodman@webrtc.org022615b2014-04-07 10:56:31 +0000836 LOG(LS_VERBOSE) << "OnNetworkChanged, bitrate" << bitrate_bps
837 << " packet loss " << fraction_lost
838 << " rtt " << round_trip_time_ms;
stefan@webrtc.org72e204a2013-03-18 17:00:51 +0000839 vcm_.SetChannelParameters(bitrate_bps, fraction_lost, round_trip_time_ms);
henrik.lundin@webrtc.org45901772013-11-18 12:18:43 +0000840 bool video_is_suspended = vcm_.VideoSuspended();
pwestin@webrtc.org5e87b5f2012-11-13 21:12:39 +0000841 int bitrate_kbps = bitrate_bps / 1000;
stefan@webrtc.org69f76052013-06-17 12:53:37 +0000842 VideoCodec send_codec;
843 if (vcm_.SendCodec(&send_codec) != 0) {
844 return;
845 }
stefan@webrtc.org7dc17902013-09-06 13:58:01 +0000846 SimulcastStream* stream_configs = send_codec.simulcastStream;
847 // Allocate the bandwidth between the streams.
848 std::vector<uint32_t> stream_bitrates = AllocateStreamBitrates(
849 bitrate_bps,
850 stream_configs,
851 send_codec.numberOfSimulcastStreams);
852 // Find the max amount of padding we can allow ourselves to send at this
853 // point, based on which streams are currently active and what our current
854 // available bandwidth is.
stefan@webrtc.org93cd3972013-10-16 13:03:10 +0000855 int pad_up_to_bitrate_kbps = 0;
856 if (send_codec.numberOfSimulcastStreams == 0) {
stefan@webrtc.org93cd3972013-10-16 13:03:10 +0000857 pad_up_to_bitrate_kbps = send_codec.minBitrate;
858 } else {
stefan@webrtc.org93cd3972013-10-16 13:03:10 +0000859 pad_up_to_bitrate_kbps =
860 stream_configs[send_codec.numberOfSimulcastStreams - 1].minBitrate;
861 for (int i = 0; i < send_codec.numberOfSimulcastStreams - 1; ++i) {
862 pad_up_to_bitrate_kbps += stream_configs[i].targetBitrate;
863 }
stefan@webrtc.org7dc17902013-09-06 13:58:01 +0000864 }
pbos@webrtc.org1d61e3a2014-03-19 10:59:52 +0000865
866 // Disable padding if only sending one stream and video isn't suspended and
867 // min-transmit bitrate isn't used (applied later).
868 if (!video_is_suspended && send_codec.numberOfSimulcastStreams <= 1)
stefan@webrtc.org93cd3972013-10-16 13:03:10 +0000869 pad_up_to_bitrate_kbps = 0;
stefan@webrtc.org5e74d962013-10-18 15:05:29 +0000870
871 {
stefan@webrtc.org5e74d962013-10-18 15:05:29 +0000872 CriticalSectionScoped cs(data_cs_.get());
pbos@webrtc.org1d61e3a2014-03-19 10:59:52 +0000873 // The amount of padding should decay to zero if no frames are being
874 // captured unless a min-transmit bitrate is used.
stefan@webrtc.org5e74d962013-10-18 15:05:29 +0000875 int64_t now_ms = TickTime::MillisecondTimestamp();
876 if (now_ms - time_of_last_incoming_frame_ms_ > kStopPaddingThresholdMs)
pbos@webrtc.org1d61e3a2014-03-19 10:59:52 +0000877 pad_up_to_bitrate_kbps = 0;
stefan@webrtc.org5e74d962013-10-18 15:05:29 +0000878
pbos@webrtc.org1d61e3a2014-03-19 10:59:52 +0000879 // Pad up to min bitrate.
pbos@webrtc.org3f83f9c2014-03-13 12:52:27 +0000880 if (pad_up_to_bitrate_kbps < min_transmit_bitrate_kbps_)
881 pad_up_to_bitrate_kbps = min_transmit_bitrate_kbps_;
pbos@webrtc.org1d61e3a2014-03-19 10:59:52 +0000882
883 // Padding may never exceed bitrate estimate.
884 if (pad_up_to_bitrate_kbps > bitrate_kbps)
885 pad_up_to_bitrate_kbps = bitrate_kbps;
886
stefan@webrtc.orgc9995bc2014-07-04 09:20:42 +0000887 paced_sender_->UpdateBitrate(
888 PacedSender::kDefaultPaceMultiplier * bitrate_kbps,
889 pad_up_to_bitrate_kbps);
pbos@webrtc.org3f83f9c2014-03-13 12:52:27 +0000890 default_rtp_rtcp_->SetTargetSendBitrate(stream_bitrates);
pbos@webrtc.org76738712013-11-21 18:44:23 +0000891 if (video_suspended_ == video_is_suspended)
892 return;
893 video_suspended_ = video_is_suspended;
894 }
pbos@webrtc.org1d61e3a2014-03-19 10:59:52 +0000895
896 // Video suspend-state changed, inform codec observer.
pbos@webrtc.org3f83f9c2014-03-13 12:52:27 +0000897 CriticalSectionScoped crit(callback_cs_.get());
pbos@webrtc.org76738712013-11-21 18:44:23 +0000898 if (codec_observer_) {
mflodman@webrtc.org022615b2014-04-07 10:56:31 +0000899 LOG(LS_INFO) << "Video suspended " << video_is_suspended
900 << " for channel " << channel_id_;
henrik.lundin@webrtc.org82b883c2013-11-21 23:00:40 +0000901 codec_observer_->SuspendChange(channel_id_, video_is_suspended);
henrik.lundin@webrtc.org39079d12013-10-02 13:34:26 +0000902 }
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000903}
904
pwestin@webrtc.org5e87b5f2012-11-13 21:12:39 +0000905PacedSender* ViEEncoder::GetPacedSender() {
906 return paced_sender_.get();
907}
908
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000909int32_t ViEEncoder::RegisterEffectFilter(ViEEffectFilter* effect_filter) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000910 CriticalSectionScoped cs(callback_cs_.get());
mflodman@webrtc.org022615b2014-04-07 10:56:31 +0000911 if (effect_filter != NULL && effect_filter_ != NULL) {
912 LOG_F(LS_ERROR) << "Filter already set.";
913 return -1;
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000914 }
915 effect_filter_ = effect_filter;
916 return 0;
917}
918
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000919int ViEEncoder::StartDebugRecording(const char* fileNameUTF8) {
920 return vcm_.StartDebugRecording(fileNameUTF8);
921}
922
923int ViEEncoder::StopDebugRecording() {
924 return vcm_.StopDebugRecording();
925}
926
henrik.lundin@webrtc.org45901772013-11-18 12:18:43 +0000927void ViEEncoder::SuspendBelowMinBitrate() {
928 vcm_.SuspendBelowMinBitrate();
henrik.lundin@webrtc.org4ce75902013-10-28 10:16:14 +0000929 bitrate_controller_->EnforceMinBitrate(false);
henrik.lundin@webrtc.org39079d12013-10-02 13:34:26 +0000930}
931
pbos@webrtc.org63301bd2013-10-21 10:34:43 +0000932void ViEEncoder::RegisterPreEncodeCallback(
933 I420FrameCallback* pre_encode_callback) {
934 CriticalSectionScoped cs(callback_cs_.get());
935 pre_encode_callback_ = pre_encode_callback;
936}
937
938void ViEEncoder::DeRegisterPreEncodeCallback() {
939 CriticalSectionScoped cs(callback_cs_.get());
940 pre_encode_callback_ = NULL;
941}
942
sprang@webrtc.org2e98d452013-11-26 11:41:59 +0000943void ViEEncoder::RegisterPostEncodeImageCallback(
944 EncodedImageCallback* post_encode_callback) {
945 vcm_.RegisterPostEncodeImageCallback(post_encode_callback);
946}
947
948void ViEEncoder::DeRegisterPostEncodeImageCallback() {
949 vcm_.RegisterPostEncodeImageCallback(NULL);
950}
951
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000952QMVideoSettingsCallback::QMVideoSettingsCallback(VideoProcessingModule* vpm)
953 : vpm_(vpm) {
954}
955
956QMVideoSettingsCallback::~QMVideoSettingsCallback() {
957}
958
pbos@webrtc.org67879bc2013-04-09 13:41:51 +0000959int32_t QMVideoSettingsCallback::SetVideoQMSettings(
960 const uint32_t frame_rate,
961 const uint32_t width,
962 const uint32_t height) {
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000963 return vpm_->SetTargetResolution(width, height, frame_rate);
964}
965
966} // namespace webrtc