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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"
pbos@webrtc.org273a4142014-12-01 15:23:21 +000019#include "webrtc/frame_callback.h"
pbos@webrtc.orgf5d4cb12013-05-17 13:44:48 +000020#include "webrtc/modules/pacing/include/paced_sender.h"
21#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
22#include "webrtc/modules/utility/interface/process_thread.h"
23#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
24#include "webrtc/modules/video_coding/main/interface/video_coding.h"
25#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
sprang@webrtc.org40709352013-11-26 11:41:59 +000026#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
stefan@webrtc.org88e0dda2014-07-04 09:20:42 +000027#include "webrtc/system_wrappers/interface/clock.h"
pbos@webrtc.orgf5d4cb12013-05-17 13:44:48 +000028#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
29#include "webrtc/system_wrappers/interface/logging.h"
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +000030#include "webrtc/system_wrappers/interface/metrics.h"
pbos@webrtc.orgf5d4cb12013-05-17 13:44:48 +000031#include "webrtc/system_wrappers/interface/tick_util.h"
pbos@webrtc.orgf5d4cb12013-05-17 13:44:48 +000032#include "webrtc/system_wrappers/interface/trace_event.h"
pbos@webrtc.org273a4142014-12-01 15:23:21 +000033#include "webrtc/video/send_statistics_proxy.h"
pbos@webrtc.orgf5d4cb12013-05-17 13:44:48 +000034#include "webrtc/video_engine/include/vie_codec.h"
35#include "webrtc/video_engine/include/vie_image_process.h"
36#include "webrtc/video_engine/vie_defines.h"
niklase@google.com470e71d2011-07-07 08:21:25 +000037
niklase@google.com470e71d2011-07-07 08:21:25 +000038namespace webrtc {
39
pwestin@webrtc.org52b4e882013-05-02 19:02:17 +000040// Margin on when we pause the encoder when the pacing buffer overflows relative
41// to the configured buffer delay.
42static const float kEncoderPausePacerMargin = 2.0f;
43
pwestin@webrtc.org91563e42013-04-25 22:20:08 +000044// Don't stop the encoder unless the delay is above this configured value.
45static const int kMinPacingDelayMs = 200;
46
stefan@webrtc.orgbfacda62013-03-27 16:36:01 +000047// Allow packets to be transmitted in up to 2 times max video bitrate if the
48// bandwidth estimate allows it.
49// TODO(holmer): Expose transmission start, min and max bitrates in the
50// VideoEngine API and remove the kTransmissionMaxBitrateMultiplier.
51static const int kTransmissionMaxBitrateMultiplier = 2;
pwestin@webrtc.org571a1c02012-11-13 21:12:39 +000052
stefan@webrtc.org3e005052013-10-18 15:05:29 +000053static const float kStopPaddingThresholdMs = 2000;
54
stefan@webrtc.orgb2c8a952013-09-06 13:58:01 +000055std::vector<uint32_t> AllocateStreamBitrates(
56 uint32_t total_bitrate,
57 const SimulcastStream* stream_configs,
58 size_t number_of_streams) {
59 if (number_of_streams == 0) {
60 std::vector<uint32_t> stream_bitrates(1, 0);
61 stream_bitrates[0] = total_bitrate;
62 return stream_bitrates;
63 }
64 std::vector<uint32_t> stream_bitrates(number_of_streams, 0);
65 uint32_t bitrate_remainder = total_bitrate;
66 for (size_t i = 0; i < stream_bitrates.size() && bitrate_remainder > 0; ++i) {
67 if (stream_configs[i].maxBitrate * 1000 > bitrate_remainder) {
68 stream_bitrates[i] = bitrate_remainder;
69 } else {
70 stream_bitrates[i] = stream_configs[i].maxBitrate * 1000;
71 }
72 bitrate_remainder -= stream_bitrates[i];
73 }
74 return stream_bitrates;
75}
76
stefan@webrtc.org439be292012-02-16 14:45:37 +000077class QMVideoSettingsCallback : public VCMQMSettingsCallback {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +000078 public:
marpan@webrtc.orgefd01fd2012-04-18 15:56:34 +000079 explicit QMVideoSettingsCallback(VideoProcessingModule* vpm);
mflodman@webrtc.org6879c8a2013-07-23 11:35:00 +000080
stefan@webrtc.org439be292012-02-16 14:45:37 +000081 ~QMVideoSettingsCallback();
niklase@google.com470e71d2011-07-07 08:21:25 +000082
mflodman@webrtc.org84d17832011-12-01 17:02:23 +000083 // Update VPM with QM (quality modes: frame size & frame rate) settings.
pbos@webrtc.orgb238d122013-04-09 13:41:51 +000084 int32_t SetVideoQMSettings(const uint32_t frame_rate,
85 const uint32_t width,
86 const uint32_t height);
niklase@google.com470e71d2011-07-07 08:21:25 +000087
mflodman@webrtc.org84d17832011-12-01 17:02:23 +000088 private:
89 VideoProcessingModule* vpm_;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +000090};
niklase@google.com470e71d2011-07-07 08:21:25 +000091
pwestin@webrtc.org49888ce2012-04-27 05:25:53 +000092class ViEBitrateObserver : public BitrateObserver {
93 public:
mflodman@webrtc.orgf5e99db2012-06-27 09:49:37 +000094 explicit ViEBitrateObserver(ViEEncoder* owner)
pwestin@webrtc.org49888ce2012-04-27 05:25:53 +000095 : owner_(owner) {
96 }
mflodman@webrtc.org6879c8a2013-07-23 11:35:00 +000097 virtual ~ViEBitrateObserver() {}
pwestin@webrtc.org49888ce2012-04-27 05:25:53 +000098 // Implements BitrateObserver.
99 virtual void OnNetworkChanged(const uint32_t bitrate_bps,
100 const uint8_t fraction_lost,
101 const uint32_t rtt) {
102 owner_->OnNetworkChanged(bitrate_bps, fraction_lost, rtt);
103 }
104 private:
105 ViEEncoder* owner_;
106};
niklase@google.com470e71d2011-07-07 08:21:25 +0000107
pwestin@webrtc.org571a1c02012-11-13 21:12:39 +0000108class ViEPacedSenderCallback : public PacedSender::Callback {
109 public:
110 explicit ViEPacedSenderCallback(ViEEncoder* owner)
111 : owner_(owner) {
112 }
mflodman@webrtc.org6879c8a2013-07-23 11:35:00 +0000113 virtual ~ViEPacedSenderCallback() {}
stefan@webrtc.org82462aa2014-10-23 11:57:05 +0000114 virtual bool TimeToSendPacket(uint32_t ssrc,
115 uint16_t sequence_number,
116 int64_t capture_time_ms,
117 bool retransmission) {
stefan@webrtc.org9b82f5a2013-11-13 15:29:21 +0000118 return owner_->TimeToSendPacket(ssrc, sequence_number, capture_time_ms,
119 retransmission);
pwestin@webrtc.org571a1c02012-11-13 21:12:39 +0000120 }
pkasting@chromium.org4591fbd2014-11-20 22:28:14 +0000121 virtual size_t TimeToSendPadding(size_t bytes) {
stefan@webrtc.org508a84b2013-06-17 12:53:37 +0000122 return owner_->TimeToSendPadding(bytes);
pwestin@webrtc.org571a1c02012-11-13 21:12:39 +0000123 }
124 private:
125 ViEEncoder* owner_;
126};
127
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000128ViEEncoder::ViEEncoder(int32_t engine_id,
129 int32_t channel_id,
130 uint32_t number_of_cores,
andresp@webrtc.org7707d062013-05-13 10:50:50 +0000131 const Config& config,
pwestin@webrtc.org49888ce2012-04-27 05:25:53 +0000132 ProcessThread& module_process_thread,
133 BitrateController* bitrate_controller)
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000134 : engine_id_(engine_id),
135 channel_id_(channel_id),
136 number_of_cores_(number_of_cores),
stefan@webrtc.org34c5da62014-04-11 14:08:35 +0000137 vcm_(*webrtc::VideoCodingModule::Create()),
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000138 vpm_(*webrtc::VideoProcessingModule::Create(ViEModuleId(engine_id,
139 channel_id))),
mflodman@webrtc.orgd32c4472011-12-22 14:17:53 +0000140 callback_cs_(CriticalSectionWrapper::CreateCriticalSection()),
141 data_cs_(CriticalSectionWrapper::CreateCriticalSection()),
pwestin@webrtc.org49888ce2012-04-27 05:25:53 +0000142 bitrate_controller_(bitrate_controller),
stefan@webrtc.org3e005052013-10-18 15:05:29 +0000143 time_of_last_incoming_frame_ms_(0),
stefan@webrtc.org508a84b2013-06-17 12:53:37 +0000144 send_padding_(false),
pbos@webrtc.org3349ae02014-03-13 12:52:27 +0000145 min_transmit_bitrate_kbps_(0),
stefan@webrtc.orgbfacda62013-03-27 16:36:01 +0000146 target_delay_ms_(0),
147 network_is_transmitting_(true),
148 encoder_paused_(false),
pwestin@webrtc.org52b4e882013-05-02 19:02:17 +0000149 encoder_paused_and_dropped_frame_(false),
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000150 fec_enabled_(false),
151 nack_enabled_(false),
152 codec_observer_(NULL),
153 effect_filter_(NULL),
154 module_process_thread_(module_process_thread),
155 has_received_sli_(false),
156 picture_id_sli_(0),
157 has_received_rpsi_(false),
158 picture_id_rpsi_(0),
henrik.lundin@webrtc.org7ea4f242013-10-02 13:34:26 +0000159 qm_callback_(NULL),
henrik.lundin@webrtc.orgce8e0932013-11-18 12:18:43 +0000160 video_suspended_(false),
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000161 pre_encode_callback_(NULL),
pbos@webrtc.org273a4142014-12-01 15:23:21 +0000162 start_ms_(Clock::GetRealTimeClock()->TimeInMilliseconds()),
163 send_statistics_proxy_(NULL) {
pwestin@webrtc.org2853dde2012-05-11 11:08:54 +0000164 RtpRtcp::Configuration configuration;
165 configuration.id = ViEModuleId(engine_id_, channel_id_);
166 configuration.audio = false; // Video.
167
168 default_rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(configuration));
pwestin@webrtc.org49888ce2012-04-27 05:25:53 +0000169 bitrate_observer_.reset(new ViEBitrateObserver(this));
pwestin@webrtc.org571a1c02012-11-13 21:12:39 +0000170 pacing_callback_.reset(new ViEPacedSenderCallback(this));
stefan@webrtc.org82462aa2014-10-23 11:57:05 +0000171 paced_sender_.reset(new PacedSender(
172 Clock::GetRealTimeClock(),
173 pacing_callback_.get(),
174 kDefaultStartBitrateKbps,
175 PacedSender::kDefaultPaceMultiplier * kDefaultStartBitrateKbps,
176 0));
wu@webrtc.org5d8c1022012-04-10 16:54:05 +0000177}
178
179bool ViEEncoder::Init() {
180 if (vcm_.InitializeSender() != 0) {
wu@webrtc.org5d8c1022012-04-10 16:54:05 +0000181 return false;
182 }
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000183 vpm_.EnableTemporalDecimation(true);
184
185 // Enable/disable content analysis: off by default for now.
186 vpm_.EnableContentAnalysis(false);
187
pwestin@webrtc.org571a1c02012-11-13 21:12:39 +0000188 if (module_process_thread_.RegisterModule(&vcm_) != 0 ||
189 module_process_thread_.RegisterModule(default_rtp_rtcp_.get()) != 0 ||
190 module_process_thread_.RegisterModule(paced_sender_.get()) != 0) {
wu@webrtc.org5d8c1022012-04-10 16:54:05 +0000191 return false;
192 }
stefan@webrtc.org97845122012-04-13 07:47:05 +0000193 if (qm_callback_) {
194 delete qm_callback_;
195 }
marpan@webrtc.orgefd01fd2012-04-18 15:56:34 +0000196 qm_callback_ = new QMVideoSettingsCallback(&vpm_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000197
198#ifdef VIDEOCODEC_VP8
andresp@webrtc.orga84b0a62014-08-14 16:46:46 +0000199 VideoCodecType codec_type = webrtc::kVideoCodecVP8;
200#else
201 VideoCodecType codec_type = webrtc::kVideoCodecI420;
202#endif
203
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000204 VideoCodec video_codec;
andresp@webrtc.orga84b0a62014-08-14 16:46:46 +0000205 if (vcm_.Codec(codec_type, &video_codec) != VCM_OK) {
wu@webrtc.org5d8c1022012-04-10 16:54:05 +0000206 return false;
207 }
stefan@webrtc.orgae2563a2014-02-13 13:48:38 +0000208 {
209 CriticalSectionScoped cs(data_cs_.get());
210 send_padding_ = video_codec.numberOfSimulcastStreams > 1;
211 }
wu@webrtc.org5d8c1022012-04-10 16:54:05 +0000212 if (vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
pwestin@webrtc.org2853dde2012-05-11 11:08:54 +0000213 default_rtp_rtcp_->MaxDataPayloadLength()) != 0) {
wu@webrtc.org5d8c1022012-04-10 16:54:05 +0000214 return false;
215 }
pwestin@webrtc.org2853dde2012-05-11 11:08:54 +0000216 if (default_rtp_rtcp_->RegisterSendPayload(video_codec) != 0) {
wu@webrtc.org5d8c1022012-04-10 16:54:05 +0000217 return false;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000218 }
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000219 if (vcm_.RegisterTransportCallback(this) != 0) {
wu@webrtc.org5d8c1022012-04-10 16:54:05 +0000220 return false;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000221 }
222 if (vcm_.RegisterSendStatisticsCallback(this) != 0) {
wu@webrtc.org5d8c1022012-04-10 16:54:05 +0000223 return false;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000224 }
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000225 if (vcm_.RegisterVideoQMCallback(qm_callback_) != 0) {
wu@webrtc.org5d8c1022012-04-10 16:54:05 +0000226 return false;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000227 }
wu@webrtc.org5d8c1022012-04-10 16:54:05 +0000228 return true;
niklase@google.com470e71d2011-07-07 08:21:25 +0000229}
230
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000231ViEEncoder::~ViEEncoder() {
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000232 UpdateHistograms();
stefan@webrtc.orgbf415082012-11-29 09:18:53 +0000233 if (bitrate_controller_) {
234 bitrate_controller_->RemoveBitrateObserver(bitrate_observer_.get());
235 }
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000236 module_process_thread_.DeRegisterModule(&vcm_);
237 module_process_thread_.DeRegisterModule(&vpm_);
pwestin@webrtc.org2853dde2012-05-11 11:08:54 +0000238 module_process_thread_.DeRegisterModule(default_rtp_rtcp_.get());
pwestin@webrtc.org571a1c02012-11-13 21:12:39 +0000239 module_process_thread_.DeRegisterModule(paced_sender_.get());
mflodman@webrtc.org66480932013-03-01 14:51:23 +0000240 VideoCodingModule::Destroy(&vcm_);
241 VideoProcessingModule::Destroy(&vpm_);
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000242 delete qm_callback_;
niklase@google.com470e71d2011-07-07 08:21:25 +0000243}
244
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000245void ViEEncoder::UpdateHistograms() {
asapersson@webrtc.org83b52002014-11-28 10:17:13 +0000246 int64_t elapsed_sec =
247 (Clock::GetRealTimeClock()->TimeInMilliseconds() - start_ms_) / 1000;
248 if (elapsed_sec < metrics::kMinRunTimeInSeconds) {
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000249 return;
250 }
251 webrtc::VCMFrameCount frames;
252 if (vcm_.SentFrameCount(frames) != VCM_OK) {
253 return;
254 }
255 uint32_t total_frames = frames.numKeyFrames + frames.numDeltaFrames;
256 if (total_frames > 0) {
257 RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.KeyFramesSentInPermille",
258 static_cast<int>(
259 (frames.numKeyFrames * 1000.0f / total_frames) + 0.5f));
260 }
261}
262
mflodman@webrtc.org9ec883e2012-03-05 17:12:41 +0000263int ViEEncoder::Owner() const {
264 return channel_id_;
265}
266
stefan@webrtc.orgbfacda62013-03-27 16:36:01 +0000267void ViEEncoder::SetNetworkTransmissionState(bool is_transmitting) {
stefan@webrtc.orgbfacda62013-03-27 16:36:01 +0000268 {
269 CriticalSectionScoped cs(data_cs_.get());
270 network_is_transmitting_ = is_transmitting;
271 }
272 if (is_transmitting) {
273 paced_sender_->Resume();
274 } else {
275 paced_sender_->Pause();
276 }
277}
278
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000279void ViEEncoder::Pause() {
mflodman@webrtc.orgd32c4472011-12-22 14:17:53 +0000280 CriticalSectionScoped cs(data_cs_.get());
stefan@webrtc.orgbfacda62013-03-27 16:36:01 +0000281 encoder_paused_ = true;
niklase@google.com470e71d2011-07-07 08:21:25 +0000282}
283
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000284void ViEEncoder::Restart() {
mflodman@webrtc.orgd32c4472011-12-22 14:17:53 +0000285 CriticalSectionScoped cs(data_cs_.get());
stefan@webrtc.orgbfacda62013-03-27 16:36:01 +0000286 encoder_paused_ = false;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000287}
288
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000289uint8_t ViEEncoder::NumberOfCodecs() {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000290 return vcm_.NumberOfCodecs();
291}
292
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000293int32_t ViEEncoder::GetCodec(uint8_t list_index, VideoCodec* video_codec) {
mflodman@webrtc.orgf5e99db2012-06-27 09:49:37 +0000294 if (vcm_.Codec(list_index, video_codec) != 0) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000295 return -1;
296 }
297 return 0;
298}
299
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000300int32_t ViEEncoder::RegisterExternalEncoder(webrtc::VideoEncoder* encoder,
301 uint8_t pl_type,
302 bool internal_source) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000303 if (encoder == NULL)
304 return -1;
305
stefan@webrtc.orgfcd85852013-01-09 08:35:40 +0000306 if (vcm_.RegisterExternalEncoder(encoder, pl_type, internal_source) !=
mflodman@webrtc.org5574dac2014-04-07 10:56:31 +0000307 VCM_OK) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000308 return -1;
309 }
310 return 0;
311}
312
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000313int32_t ViEEncoder::DeRegisterExternalEncoder(uint8_t pl_type) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000314 webrtc::VideoCodec current_send_codec;
315 if (vcm_.SendCodec(&current_send_codec) == VCM_OK) {
stefan@webrtc.org3d0b0d62013-03-19 10:04:57 +0000316 uint32_t current_bitrate_bps = 0;
317 if (vcm_.Bitrate(&current_bitrate_bps) != 0) {
mflodman@webrtc.org5574dac2014-04-07 10:56:31 +0000318 LOG(LS_WARNING) << "Failed to get the current encoder target bitrate.";
stefan@webrtc.org439be292012-02-16 14:45:37 +0000319 }
stefan@webrtc.org3d0b0d62013-03-19 10:04:57 +0000320 current_send_codec.startBitrate = (current_bitrate_bps + 500) / 1000;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000321 }
322
323 if (vcm_.RegisterExternalEncoder(NULL, pl_type) != VCM_OK) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000324 return -1;
325 }
326
stefan@webrtc.org508a84b2013-06-17 12:53:37 +0000327 // If the external encoder is the current send codec, use vcm internal
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000328 // encoder.
329 if (current_send_codec.plType == pl_type) {
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000330 uint16_t max_data_payload_length =
pwestin@webrtc.org2853dde2012-05-11 11:08:54 +0000331 default_rtp_rtcp_->MaxDataPayloadLength();
stefan@webrtc.orgae2563a2014-02-13 13:48:38 +0000332 {
333 CriticalSectionScoped cs(data_cs_.get());
334 send_padding_ = current_send_codec.numberOfSimulcastStreams > 1;
335 }
fischman@webrtc.org64e04052014-03-07 18:00:05 +0000336 // TODO(mflodman): Unfortunately the VideoCodec that VCM has cached a
337 // raw pointer to an |extra_options| that's long gone. Clearing it here is
338 // a hack to prevent the following code from crashing. This should be fixed
339 // for realz. https://code.google.com/p/chromium/issues/detail?id=348222
340 current_send_codec.extra_options = NULL;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000341 if (vcm_.RegisterSendCodec(&current_send_codec, number_of_cores_,
342 max_data_payload_length) != VCM_OK) {
stefan@webrtc.org4070b1d2014-07-16 11:20:40 +0000343 LOG(LS_INFO) << "De-registered the currently used external encoder ("
344 << static_cast<int>(pl_type) << ") and therefore tried to "
345 << "register the corresponding internal encoder, but none "
346 << "was supported.";
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000347 }
348 }
349 return 0;
350}
351
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000352int32_t ViEEncoder::SetEncoder(const webrtc::VideoCodec& video_codec) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000353 // Setting target width and height for VPM.
354 if (vpm_.SetTargetResolution(video_codec.width, video_codec.height,
355 video_codec.maxFramerate) != VPM_OK) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000356 return -1;
357 }
358
pwestin@webrtc.org2853dde2012-05-11 11:08:54 +0000359 if (default_rtp_rtcp_->RegisterSendPayload(video_codec) != 0) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000360 return -1;
361 }
pwestin@webrtc.org49888ce2012-04-27 05:25:53 +0000362 // Convert from kbps to bps.
stefan@webrtc.orgb2c8a952013-09-06 13:58:01 +0000363 std::vector<uint32_t> stream_bitrates = AllocateStreamBitrates(
364 video_codec.startBitrate * 1000,
365 video_codec.simulcastStream,
366 video_codec.numberOfSimulcastStreams);
367 default_rtp_rtcp_->SetTargetSendBitrate(stream_bitrates);
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000368
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000369 uint16_t max_data_payload_length =
pwestin@webrtc.org2853dde2012-05-11 11:08:54 +0000370 default_rtp_rtcp_->MaxDataPayloadLength();
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000371
stefan@webrtc.org9075d512014-02-14 09:45:58 +0000372 {
373 CriticalSectionScoped cs(data_cs_.get());
374 send_padding_ = video_codec.numberOfSimulcastStreams > 1;
375 }
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000376 if (vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
377 max_data_payload_length) != VCM_OK) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000378 return -1;
379 }
380
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000381 // Set this module as sending right away, let the slave module in the channel
382 // start and stop sending.
andresp@webrtc.orga84b0a62014-08-14 16:46:46 +0000383 if (default_rtp_rtcp_->SetSendingStatus(true) != 0) {
384 return -1;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000385 }
andresp@webrtc.orga84b0a62014-08-14 16:46:46 +0000386
pwestin@webrtc.org49888ce2012-04-27 05:25:53 +0000387 bitrate_controller_->SetBitrateObserver(bitrate_observer_.get(),
388 video_codec.startBitrate * 1000,
389 video_codec.minBitrate * 1000,
stefan@webrtc.orgbfacda62013-03-27 16:36:01 +0000390 kTransmissionMaxBitrateMultiplier *
pwestin@webrtc.org49888ce2012-04-27 05:25:53 +0000391 video_codec.maxBitrate * 1000);
392
pbos@webrtc.org3349ae02014-03-13 12:52:27 +0000393 CriticalSectionScoped crit(data_cs_.get());
394 int pad_up_to_bitrate_kbps = video_codec.startBitrate;
395 if (pad_up_to_bitrate_kbps < min_transmit_bitrate_kbps_)
396 pad_up_to_bitrate_kbps = min_transmit_bitrate_kbps_;
397
stefan@webrtc.org88e0dda2014-07-04 09:20:42 +0000398 paced_sender_->UpdateBitrate(
stefan@webrtc.org82462aa2014-10-23 11:57:05 +0000399 video_codec.startBitrate,
stefan@webrtc.org88e0dda2014-07-04 09:20:42 +0000400 PacedSender::kDefaultPaceMultiplier * video_codec.startBitrate,
401 pad_up_to_bitrate_kbps);
pbos@webrtc.org3349ae02014-03-13 12:52:27 +0000402
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000403 return 0;
404}
405
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000406int32_t ViEEncoder::GetEncoder(VideoCodec* video_codec) {
mflodman@webrtc.orgf5e99db2012-06-27 09:49:37 +0000407 if (vcm_.SendCodec(video_codec) != 0) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000408 return -1;
409 }
410 return 0;
411}
niklase@google.com470e71d2011-07-07 08:21:25 +0000412
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000413int32_t ViEEncoder::GetCodecConfigParameters(
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000414 unsigned char config_parameters[kConfigParameterSize],
415 unsigned char& config_parameters_size) {
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000416 int32_t num_parameters =
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000417 vcm_.CodecConfigParameters(config_parameters, kConfigParameterSize);
418 if (num_parameters <= 0) {
419 config_parameters_size = 0;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000420 return -1;
421 }
422 config_parameters_size = static_cast<unsigned char>(num_parameters);
423 return 0;
niklase@google.com470e71d2011-07-07 08:21:25 +0000424}
425
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000426int32_t ViEEncoder::ScaleInputImage(bool enable) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000427 VideoFrameResampling resampling_mode = kFastRescaling;
mflodman@webrtc.org5574dac2014-04-07 10:56:31 +0000428 // TODO(mflodman) What?
429 if (enable) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000430 // kInterpolation is currently not supported.
mflodman@webrtc.org5574dac2014-04-07 10:56:31 +0000431 LOG_F(LS_ERROR) << "Not supported.";
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000432 return -1;
433 }
434 vpm_.SetInputFrameResampleMode(resampling_mode);
niklase@google.com470e71d2011-07-07 08:21:25 +0000435
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000436 return 0;
niklase@google.com470e71d2011-07-07 08:21:25 +0000437}
438
stefan@webrtc.org9b82f5a2013-11-13 15:29:21 +0000439bool ViEEncoder::TimeToSendPacket(uint32_t ssrc,
440 uint16_t sequence_number,
441 int64_t capture_time_ms,
442 bool retransmission) {
hclam@chromium.org2e402ce2013-06-20 20:18:31 +0000443 return default_rtp_rtcp_->TimeToSendPacket(ssrc, sequence_number,
stefan@webrtc.org9b82f5a2013-11-13 15:29:21 +0000444 capture_time_ms, retransmission);
pwestin@webrtc.org571a1c02012-11-13 21:12:39 +0000445}
446
pkasting@chromium.org4591fbd2014-11-20 22:28:14 +0000447size_t ViEEncoder::TimeToSendPadding(size_t bytes) {
henrik.lundin@webrtc.org331d4402013-11-21 14:05:40 +0000448 bool send_padding;
449 {
450 CriticalSectionScoped cs(data_cs_.get());
pbos@webrtc.org3349ae02014-03-13 12:52:27 +0000451 send_padding =
452 send_padding_ || video_suspended_ || min_transmit_bitrate_kbps_ > 0;
henrik.lundin@webrtc.org331d4402013-11-21 14:05:40 +0000453 }
454 if (send_padding) {
stefan@webrtc.org508a84b2013-06-17 12:53:37 +0000455 return default_rtp_rtcp_->TimeToSendPadding(bytes);
456 }
457 return 0;
458}
459
stefan@webrtc.orgbfacda62013-03-27 16:36:01 +0000460bool ViEEncoder::EncoderPaused() const {
pwestin@webrtc.org91563e42013-04-25 22:20:08 +0000461 // Pause video if paused by caller or as long as the network is down or the
462 // pacer queue has grown too large in buffered mode.
463 if (encoder_paused_) {
464 return true;
465 }
466 if (target_delay_ms_ > 0) {
467 // Buffered mode.
468 // TODO(pwestin): Workaround until nack is configured as a time and not
469 // number of packets.
470 return paced_sender_->QueueInMs() >=
pwestin@webrtc.org52b4e882013-05-02 19:02:17 +0000471 std::max(static_cast<int>(target_delay_ms_ * kEncoderPausePacerMargin),
472 kMinPacingDelayMs);
pwestin@webrtc.org91563e42013-04-25 22:20:08 +0000473 }
sprang@webrtc.orgdcebf2d2014-11-04 16:27:16 +0000474 if (paced_sender_->ExpectedQueueTimeMs() >
475 PacedSender::kDefaultMaxQueueLengthMs) {
476 // Too much data in pacer queue, drop frame.
477 return true;
478 }
pwestin@webrtc.org91563e42013-04-25 22:20:08 +0000479 return !network_is_transmitting_;
stefan@webrtc.orgbfacda62013-03-27 16:36:01 +0000480}
481
sprang@webrtc.orgdcebf2d2014-11-04 16:27:16 +0000482void ViEEncoder::TraceFrameDropStart() {
483 // Start trace event only on the first frame after encoder is paused.
484 if (!encoder_paused_and_dropped_frame_) {
485 TRACE_EVENT_ASYNC_BEGIN0("webrtc", "EncoderPaused", this);
486 }
487 encoder_paused_and_dropped_frame_ = true;
488 return;
489}
490
491void ViEEncoder::TraceFrameDropEnd() {
492 // End trace event on first frame after encoder resumes, if frame was dropped.
493 if (encoder_paused_and_dropped_frame_) {
494 TRACE_EVENT_ASYNC_END0("webrtc", "EncoderPaused", this);
495 }
496 encoder_paused_and_dropped_frame_ = false;
497}
498
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000499RtpRtcp* ViEEncoder::SendRtpRtcpModule() {
pwestin@webrtc.org2853dde2012-05-11 11:08:54 +0000500 return default_rtp_rtcp_.get();
niklase@google.com470e71d2011-07-07 08:21:25 +0000501}
502
mflodman@webrtc.org8baed512012-06-21 12:11:50 +0000503void ViEEncoder::DeliverFrame(int id,
mikhal@webrtc.org9fedff72012-10-24 18:33:04 +0000504 I420VideoFrame* video_frame,
pbos@webrtc.org9334ac22014-11-24 08:25:50 +0000505 const std::vector<uint32_t>& csrcs) {
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.
pbos@webrtc.org9334ac22014-11-24 08:25:50 +0000531 if (csrcs.size() > 0) {
532 std::vector<uint32_t> temp_csrcs(csrcs.size());
533 for (size_t i = 0; i < csrcs.size(); i++) {
534 if (csrcs[i] == 1) {
535 temp_csrcs[i] = default_rtp_rtcp_->SSRC();
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000536 } else {
pbos@webrtc.org9334ac22014-11-24 08:25:50 +0000537 temp_csrcs[i] = csrcs[i];
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000538 }
niklase@google.com470e71d2011-07-07 08:21:25 +0000539 }
pbos@webrtc.org9334ac22014-11-24 08:25:50 +0000540 default_rtp_rtcp_->SetCsrcs(temp_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(
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000729 const uint8_t payload_type,
pbos@webrtc.org273a4142014-12-01 15:23:21 +0000730 const EncodedImage& encoded_image,
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000731 const webrtc::RTPFragmentationHeader& fragmentation_header,
732 const RTPVideoHeader* rtp_video_hdr) {
pbos@webrtc.org273a4142014-12-01 15:23:21 +0000733 if (send_statistics_proxy_ != NULL) {
734 send_statistics_proxy_->OnSendEncodedImage(encoded_image, rtp_video_hdr);
735 }
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000736 // New encoded data, hand over to the rtp module.
pbos@webrtc.org273a4142014-12-01 15:23:21 +0000737 return default_rtp_rtcp_->SendOutgoingData(
738 VCMEncodedFrame::ConvertFrameType(encoded_image._frameType), payload_type,
739 encoded_image._timeStamp, encoded_image.capture_time_ms_,
740 encoded_image._buffer, encoded_image._length, &fragmentation_header,
741 rtp_video_hdr);
niklase@google.com470e71d2011-07-07 08:21:25 +0000742}
743
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000744int32_t ViEEncoder::ProtectionRequest(
stefan@webrtc.orge0d6fa42012-03-20 22:10:56 +0000745 const FecProtectionParams* delta_fec_params,
746 const FecProtectionParams* key_fec_params,
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000747 uint32_t* sent_video_rate_bps,
748 uint32_t* sent_nack_rate_bps,
749 uint32_t* sent_fec_rate_bps) {
mflodman@webrtc.org5574dac2014-04-07 10:56:31 +0000750 default_rtp_rtcp_->SetFecParameters(delta_fec_params, key_fec_params);
751 default_rtp_rtcp_->BitrateSent(NULL, sent_video_rate_bps, sent_fec_rate_bps,
stefan@webrtc.orgf4c82862011-12-13 15:38:14 +0000752 sent_nack_rate_bps);
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000753 return 0;
niklase@google.com470e71d2011-07-07 08:21:25 +0000754}
755
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000756int32_t ViEEncoder::SendStatistics(const uint32_t bit_rate,
757 const uint32_t frame_rate) {
mflodman@webrtc.orgd32c4472011-12-22 14:17:53 +0000758 CriticalSectionScoped cs(callback_cs_.get());
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000759 if (codec_observer_) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000760 codec_observer_->OutgoingRate(channel_id_, frame_rate, bit_rate);
761 }
762 return 0;
niklase@google.com470e71d2011-07-07 08:21:25 +0000763}
764
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000765int32_t ViEEncoder::RegisterCodecObserver(ViEEncoderObserver* observer) {
mflodman@webrtc.orgd32c4472011-12-22 14:17:53 +0000766 CriticalSectionScoped cs(callback_cs_.get());
mflodman@webrtc.org5574dac2014-04-07 10:56:31 +0000767 if (observer && codec_observer_) {
768 LOG_F(LS_ERROR) << "Observer already set.";
769 return -1;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000770 }
mflodman@webrtc.org5574dac2014-04-07 10:56:31 +0000771 codec_observer_ = observer;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000772 return 0;
niklase@google.com470e71d2011-07-07 08:21:25 +0000773}
774
andrew@webrtc.org96636862012-09-20 23:33:17 +0000775void ViEEncoder::OnReceivedSLI(uint32_t /*ssrc*/,
776 uint8_t picture_id) {
stefan@webrtc.org7af12be2014-07-09 14:46:31 +0000777 CriticalSectionScoped cs(data_cs_.get());
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000778 picture_id_sli_ = picture_id;
779 has_received_sli_ = true;
niklase@google.com470e71d2011-07-07 08:21:25 +0000780}
781
andrew@webrtc.org96636862012-09-20 23:33:17 +0000782void ViEEncoder::OnReceivedRPSI(uint32_t /*ssrc*/,
783 uint64_t picture_id) {
stefan@webrtc.org7af12be2014-07-09 14:46:31 +0000784 CriticalSectionScoped cs(data_cs_.get());
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000785 picture_id_rpsi_ = picture_id;
786 has_received_rpsi_ = true;
niklase@google.com470e71d2011-07-07 08:21:25 +0000787}
788
mflodman@webrtc.orgd6ec3862012-10-25 11:30:29 +0000789void ViEEncoder::OnReceivedIntraFrameRequest(uint32_t ssrc) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000790 // Key frame request from remote side, signal to VCM.
justinlin@chromium.org7bfb3a32013-05-13 22:59:00 +0000791 TRACE_EVENT0("webrtc", "OnKeyFrameRequest");
pwestin@webrtc.org1da1ce02011-10-13 15:19:55 +0000792
mflodman@webrtc.orgd6ec3862012-10-25 11:30:29 +0000793 int idx = 0;
794 {
795 CriticalSectionScoped cs(data_cs_.get());
796 std::map<unsigned int, int>::iterator stream_it = ssrc_streams_.find(ssrc);
797 if (stream_it == ssrc_streams_.end()) {
mflodman@webrtc.orgd73527c2012-12-20 09:26:17 +0000798 LOG_F(LS_WARNING) << "ssrc not found: " << ssrc << ", map size "
799 << ssrc_streams_.size();
mflodman@webrtc.orgd6ec3862012-10-25 11:30:29 +0000800 return;
801 }
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000802 std::map<unsigned int, int64_t>::iterator time_it =
mflodman@webrtc.orgd6ec3862012-10-25 11:30:29 +0000803 time_last_intra_request_ms_.find(ssrc);
804 if (time_it == time_last_intra_request_ms_.end()) {
805 time_last_intra_request_ms_[ssrc] = 0;
806 }
807
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000808 int64_t now = TickTime::MillisecondTimestamp();
mflodman@webrtc.orgd6ec3862012-10-25 11:30:29 +0000809 if (time_last_intra_request_ms_[ssrc] + kViEMinKeyRequestIntervalMs > now) {
mflodman@webrtc.orgd6ec3862012-10-25 11:30:29 +0000810 return;
811 }
812 time_last_intra_request_ms_[ssrc] = now;
813 idx = stream_it->second;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000814 }
mflodman@webrtc.orgd6ec3862012-10-25 11:30:29 +0000815 // Release the critsect before triggering key frame.
816 vcm_.IntraFrameRequest(idx);
niklase@google.com470e71d2011-07-07 08:21:25 +0000817}
818
mflodman@webrtc.orgaca26292012-10-05 16:17:41 +0000819void ViEEncoder::OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) {
mflodman@webrtc.orgd6ec3862012-10-25 11:30:29 +0000820 CriticalSectionScoped cs(data_cs_.get());
821 std::map<unsigned int, int>::iterator it = ssrc_streams_.find(old_ssrc);
822 if (it == ssrc_streams_.end()) {
823 return;
824 }
825
826 ssrc_streams_[new_ssrc] = it->second;
827 ssrc_streams_.erase(it);
828
829 std::map<unsigned int, int64_t>::iterator time_it =
830 time_last_intra_request_ms_.find(old_ssrc);
831 int64_t last_intra_request_ms = 0;
832 if (time_it != time_last_intra_request_ms_.end()) {
833 last_intra_request_ms = time_it->second;
834 time_last_intra_request_ms_.erase(time_it);
835 }
836 time_last_intra_request_ms_[new_ssrc] = last_intra_request_ms;
837}
838
839bool ViEEncoder::SetSsrcs(const std::list<unsigned int>& ssrcs) {
840 VideoCodec codec;
841 if (vcm_.SendCodec(&codec) != 0)
842 return false;
843
844 if (codec.numberOfSimulcastStreams > 0 &&
845 ssrcs.size() != codec.numberOfSimulcastStreams) {
846 return false;
847 }
848
849 CriticalSectionScoped cs(data_cs_.get());
850 ssrc_streams_.clear();
851 time_last_intra_request_ms_.clear();
852 int idx = 0;
853 for (std::list<unsigned int>::const_iterator it = ssrcs.begin();
854 it != ssrcs.end(); ++it, ++idx) {
855 unsigned int ssrc = *it;
856 ssrc_streams_[ssrc] = idx;
857 }
858 return true;
mflodman@webrtc.orgaca26292012-10-05 16:17:41 +0000859}
860
pbos@webrtc.org3349ae02014-03-13 12:52:27 +0000861void ViEEncoder::SetMinTransmitBitrate(int min_transmit_bitrate_kbps) {
862 assert(min_transmit_bitrate_kbps >= 0);
863 CriticalSectionScoped crit(data_cs_.get());
864 min_transmit_bitrate_kbps_ = min_transmit_bitrate_kbps;
865}
866
pwestin@webrtc.org49888ce2012-04-27 05:25:53 +0000867// Called from ViEBitrateObserver.
868void ViEEncoder::OnNetworkChanged(const uint32_t bitrate_bps,
869 const uint8_t fraction_lost,
870 const uint32_t round_trip_time_ms) {
mflodman@webrtc.org5574dac2014-04-07 10:56:31 +0000871 LOG(LS_VERBOSE) << "OnNetworkChanged, bitrate" << bitrate_bps
872 << " packet loss " << fraction_lost
873 << " rtt " << round_trip_time_ms;
stefan@webrtc.orgabc9d5b2013-03-18 17:00:51 +0000874 vcm_.SetChannelParameters(bitrate_bps, fraction_lost, round_trip_time_ms);
henrik.lundin@webrtc.orgce8e0932013-11-18 12:18:43 +0000875 bool video_is_suspended = vcm_.VideoSuspended();
pwestin@webrtc.org571a1c02012-11-13 21:12:39 +0000876 int bitrate_kbps = bitrate_bps / 1000;
stefan@webrtc.org508a84b2013-06-17 12:53:37 +0000877 VideoCodec send_codec;
878 if (vcm_.SendCodec(&send_codec) != 0) {
879 return;
880 }
stefan@webrtc.orgb2c8a952013-09-06 13:58:01 +0000881 SimulcastStream* stream_configs = send_codec.simulcastStream;
882 // Allocate the bandwidth between the streams.
883 std::vector<uint32_t> stream_bitrates = AllocateStreamBitrates(
884 bitrate_bps,
885 stream_configs,
886 send_codec.numberOfSimulcastStreams);
887 // Find the max amount of padding we can allow ourselves to send at this
888 // point, based on which streams are currently active and what our current
889 // available bandwidth is.
stefan@webrtc.orgb400aa72013-10-16 13:03:10 +0000890 int pad_up_to_bitrate_kbps = 0;
891 if (send_codec.numberOfSimulcastStreams == 0) {
stefan@webrtc.orgb400aa72013-10-16 13:03:10 +0000892 pad_up_to_bitrate_kbps = send_codec.minBitrate;
893 } else {
stefan@webrtc.orgb400aa72013-10-16 13:03:10 +0000894 pad_up_to_bitrate_kbps =
895 stream_configs[send_codec.numberOfSimulcastStreams - 1].minBitrate;
896 for (int i = 0; i < send_codec.numberOfSimulcastStreams - 1; ++i) {
897 pad_up_to_bitrate_kbps += stream_configs[i].targetBitrate;
898 }
stefan@webrtc.orgb2c8a952013-09-06 13:58:01 +0000899 }
pbos@webrtc.org709e2972014-03-19 10:59:52 +0000900
901 // Disable padding if only sending one stream and video isn't suspended and
902 // min-transmit bitrate isn't used (applied later).
903 if (!video_is_suspended && send_codec.numberOfSimulcastStreams <= 1)
stefan@webrtc.orgb400aa72013-10-16 13:03:10 +0000904 pad_up_to_bitrate_kbps = 0;
stefan@webrtc.org3e005052013-10-18 15:05:29 +0000905
906 {
stefan@webrtc.org3e005052013-10-18 15:05:29 +0000907 CriticalSectionScoped cs(data_cs_.get());
pbos@webrtc.org709e2972014-03-19 10:59:52 +0000908 // The amount of padding should decay to zero if no frames are being
909 // captured unless a min-transmit bitrate is used.
stefan@webrtc.org3e005052013-10-18 15:05:29 +0000910 int64_t now_ms = TickTime::MillisecondTimestamp();
911 if (now_ms - time_of_last_incoming_frame_ms_ > kStopPaddingThresholdMs)
pbos@webrtc.org709e2972014-03-19 10:59:52 +0000912 pad_up_to_bitrate_kbps = 0;
stefan@webrtc.org3e005052013-10-18 15:05:29 +0000913
pbos@webrtc.org709e2972014-03-19 10:59:52 +0000914 // Pad up to min bitrate.
pbos@webrtc.org3349ae02014-03-13 12:52:27 +0000915 if (pad_up_to_bitrate_kbps < min_transmit_bitrate_kbps_)
916 pad_up_to_bitrate_kbps = min_transmit_bitrate_kbps_;
pbos@webrtc.org709e2972014-03-19 10:59:52 +0000917
918 // Padding may never exceed bitrate estimate.
919 if (pad_up_to_bitrate_kbps > bitrate_kbps)
920 pad_up_to_bitrate_kbps = bitrate_kbps;
921
stefan@webrtc.org88e0dda2014-07-04 09:20:42 +0000922 paced_sender_->UpdateBitrate(
stefan@webrtc.org82462aa2014-10-23 11:57:05 +0000923 bitrate_kbps,
stefan@webrtc.org88e0dda2014-07-04 09:20:42 +0000924 PacedSender::kDefaultPaceMultiplier * bitrate_kbps,
925 pad_up_to_bitrate_kbps);
pbos@webrtc.org3349ae02014-03-13 12:52:27 +0000926 default_rtp_rtcp_->SetTargetSendBitrate(stream_bitrates);
pbos@webrtc.org484ee962013-11-21 18:44:23 +0000927 if (video_suspended_ == video_is_suspended)
928 return;
929 video_suspended_ = video_is_suspended;
930 }
pbos@webrtc.org709e2972014-03-19 10:59:52 +0000931
932 // Video suspend-state changed, inform codec observer.
pbos@webrtc.org3349ae02014-03-13 12:52:27 +0000933 CriticalSectionScoped crit(callback_cs_.get());
pbos@webrtc.org484ee962013-11-21 18:44:23 +0000934 if (codec_observer_) {
mflodman@webrtc.org5574dac2014-04-07 10:56:31 +0000935 LOG(LS_INFO) << "Video suspended " << video_is_suspended
936 << " for channel " << channel_id_;
henrik.lundin@webrtc.org9fe36032013-11-21 23:00:40 +0000937 codec_observer_->SuspendChange(channel_id_, video_is_suspended);
henrik.lundin@webrtc.org7ea4f242013-10-02 13:34:26 +0000938 }
niklase@google.com470e71d2011-07-07 08:21:25 +0000939}
940
pwestin@webrtc.org571a1c02012-11-13 21:12:39 +0000941PacedSender* ViEEncoder::GetPacedSender() {
942 return paced_sender_.get();
943}
944
pbos@webrtc.orgb238d122013-04-09 13:41:51 +0000945int32_t ViEEncoder::RegisterEffectFilter(ViEEffectFilter* effect_filter) {
mflodman@webrtc.orgd32c4472011-12-22 14:17:53 +0000946 CriticalSectionScoped cs(callback_cs_.get());
mflodman@webrtc.org5574dac2014-04-07 10:56:31 +0000947 if (effect_filter != NULL && effect_filter_ != NULL) {
948 LOG_F(LS_ERROR) << "Filter already set.";
949 return -1;
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000950 }
951 effect_filter_ = effect_filter;
952 return 0;
niklase@google.com470e71d2011-07-07 08:21:25 +0000953}
954
mikhal@webrtc.orge41bbdf2012-08-28 16:15:16 +0000955int ViEEncoder::StartDebugRecording(const char* fileNameUTF8) {
956 return vcm_.StartDebugRecording(fileNameUTF8);
957}
958
959int ViEEncoder::StopDebugRecording() {
960 return vcm_.StopDebugRecording();
961}
962
henrik.lundin@webrtc.orgce8e0932013-11-18 12:18:43 +0000963void ViEEncoder::SuspendBelowMinBitrate() {
964 vcm_.SuspendBelowMinBitrate();
henrik.lundin@webrtc.org1a3a6e52013-10-28 10:16:14 +0000965 bitrate_controller_->EnforceMinBitrate(false);
henrik.lundin@webrtc.org7ea4f242013-10-02 13:34:26 +0000966}
967
pbos@webrtc.orgfe1ef932013-10-21 10:34:43 +0000968void ViEEncoder::RegisterPreEncodeCallback(
969 I420FrameCallback* pre_encode_callback) {
970 CriticalSectionScoped cs(callback_cs_.get());
971 pre_encode_callback_ = pre_encode_callback;
972}
973
974void ViEEncoder::DeRegisterPreEncodeCallback() {
975 CriticalSectionScoped cs(callback_cs_.get());
976 pre_encode_callback_ = NULL;
977}
978
sprang@webrtc.org40709352013-11-26 11:41:59 +0000979void ViEEncoder::RegisterPostEncodeImageCallback(
980 EncodedImageCallback* post_encode_callback) {
981 vcm_.RegisterPostEncodeImageCallback(post_encode_callback);
982}
983
984void ViEEncoder::DeRegisterPostEncodeImageCallback() {
985 vcm_.RegisterPostEncodeImageCallback(NULL);
986}
987
pbos@webrtc.org273a4142014-12-01 15:23:21 +0000988void ViEEncoder::RegisterSendStatisticsProxy(
989 SendStatisticsProxy* send_statistics_proxy) {
990 send_statistics_proxy_ = send_statistics_proxy;
991}
992
marpan@webrtc.orgefd01fd2012-04-18 15:56:34 +0000993QMVideoSettingsCallback::QMVideoSettingsCallback(VideoProcessingModule* vpm)
994 : vpm_(vpm) {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000995}
niklase@google.com470e71d2011-07-07 08:21:25 +0000996
stefan@webrtc.org439be292012-02-16 14:45:37 +0000997QMVideoSettingsCallback::~QMVideoSettingsCallback() {
mflodman@webrtc.org84d17832011-12-01 17:02:23 +0000998}
999
pbos@webrtc.orgb238d122013-04-09 13:41:51 +00001000int32_t QMVideoSettingsCallback::SetVideoQMSettings(
1001 const uint32_t frame_rate,
1002 const uint32_t width,
1003 const uint32_t height) {
marpan@webrtc.orgcf706c22012-03-27 21:04:13 +00001004 return vpm_->SetTargetResolution(width, height, frame_rate);
mflodman@webrtc.org84d17832011-12-01 17:02:23 +00001005}
1006
mflodman@webrtc.org84d17832011-12-01 17:02:23 +00001007} // namespace webrtc