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1 // Copyright 2013 The Chromium Authors. All rights reserved. | 1 // Copyright 2013 The Chromium Authors. All rights reserved. |
2 // Use of this source code is governed by a BSD-style license that can be | 2 // Use of this source code is governed by a BSD-style license that can be |
3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
4 | 4 |
5 #include "content/renderer/media/gpu/rtc_video_encoder.h" | 5 #include "content/renderer/media/gpu/rtc_video_encoder.h" |
6 | 6 |
7 #include <string.h> | 7 #include <string.h> |
8 | 8 |
9 #include "base/bind.h" | 9 #include "base/bind.h" |
10 #include "base/location.h" | 10 #include "base/location.h" |
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25 #include "media/filters/h264_parser.h" | 25 #include "media/filters/h264_parser.h" |
26 #include "media/renderers/gpu_video_accelerator_factories.h" | 26 #include "media/renderers/gpu_video_accelerator_factories.h" |
27 #include "media/video/video_encode_accelerator.h" | 27 #include "media/video/video_encode_accelerator.h" |
28 #include "third_party/libyuv/include/libyuv.h" | 28 #include "third_party/libyuv/include/libyuv.h" |
29 #include "third_party/webrtc/base/timeutils.h" | 29 #include "third_party/webrtc/base/timeutils.h" |
30 | 30 |
31 namespace content { | 31 namespace content { |
32 | 32 |
33 namespace { | 33 namespace { |
34 | 34 |
| 35 // Used for timestamp conversions. |
| 36 static const int64_t kMsToRtpTimestamp = 90; |
| 37 |
35 // Translate from webrtc::VideoCodecType and webrtc::VideoCodec to | 38 // Translate from webrtc::VideoCodecType and webrtc::VideoCodec to |
36 // media::VideoCodecProfile. | 39 // media::VideoCodecProfile. |
37 media::VideoCodecProfile WebRTCVideoCodecToVideoCodecProfile( | 40 media::VideoCodecProfile WebRTCVideoCodecToVideoCodecProfile( |
38 webrtc::VideoCodecType type, | 41 webrtc::VideoCodecType type, |
39 const webrtc::VideoCodec* codec_settings) { | 42 const webrtc::VideoCodec* codec_settings) { |
40 DCHECK_EQ(type, codec_settings->codecType); | 43 DCHECK_EQ(type, codec_settings->codecType); |
41 switch (type) { | 44 switch (type) { |
42 case webrtc::kVideoCodecVP8: | 45 case webrtc::kVideoCodecVP8: |
43 return media::VP8PROFILE_ANY; | 46 return media::VP8PROFILE_ANY; |
44 case webrtc::kVideoCodecH264: { | 47 case webrtc::kVideoCodecH264: { |
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467 } | 470 } |
468 base::SharedMemory* output_buffer = output_buffers_[bitstream_buffer_id]; | 471 base::SharedMemory* output_buffer = output_buffers_[bitstream_buffer_id]; |
469 if (payload_size > output_buffer->mapped_size()) { | 472 if (payload_size > output_buffer->mapped_size()) { |
470 LogAndNotifyError(FROM_HERE, "invalid payload_size", | 473 LogAndNotifyError(FROM_HERE, "invalid payload_size", |
471 media::VideoEncodeAccelerator::kPlatformFailureError); | 474 media::VideoEncodeAccelerator::kPlatformFailureError); |
472 return; | 475 return; |
473 } | 476 } |
474 output_buffers_free_count_--; | 477 output_buffers_free_count_--; |
475 | 478 |
476 // Derive the capture time (in ms) and RTP timestamp (in 90KHz ticks). | 479 // Derive the capture time (in ms) and RTP timestamp (in 90KHz ticks). |
477 int64_t capture_time_us, capture_time_ms; | 480 int64_t rtp_timestamp, capture_time_ms; |
478 uint32_t rtp_timestamp; | |
479 | |
480 if (!timestamp.is_zero()) { | 481 if (!timestamp.is_zero()) { |
481 capture_time_us = timestamp.InMicroseconds();; | 482 // Get RTP timestamp value. |
482 capture_time_ms = timestamp.InMilliseconds(); | 483 rtp_timestamp = timestamp.ToInternalValue(); |
| 484 capture_time_ms = rtp_timestamp / kMsToRtpTimestamp; |
483 } else { | 485 } else { |
484 // Fallback to the current time if encoder does not provide timestamp. | 486 // Fallback to the current time if encoder does not provide timestamp. |
485 capture_time_us = rtc::TimeMicros(); | 487 rtp_timestamp = rtc::TimeMicros() * kMsToRtpTimestamp / |
486 capture_time_ms = capture_time_us / base::Time::kMicrosecondsPerMillisecond; | 488 base::Time::kMicrosecondsPerMillisecond; |
| 489 capture_time_ms = |
| 490 rtc::TimeMicros() / base::Time::kMicrosecondsPerMillisecond; |
487 } | 491 } |
488 // RTP timestamp can wrap around. Get the lower 32 bits. | |
489 rtp_timestamp = static_cast<uint32_t>( | |
490 capture_time_us * 90 / base::Time::kMicrosecondsPerMillisecond); | |
491 | 492 |
492 webrtc::EncodedImage image( | 493 webrtc::EncodedImage image( |
493 reinterpret_cast<uint8_t*>(output_buffer->memory()), payload_size, | 494 reinterpret_cast<uint8_t*>(output_buffer->memory()), payload_size, |
494 output_buffer->mapped_size()); | 495 output_buffer->mapped_size()); |
495 image._encodedWidth = input_visible_size_.width(); | 496 image._encodedWidth = input_visible_size_.width(); |
496 image._encodedHeight = input_visible_size_.height(); | 497 image._encodedHeight = input_visible_size_.height(); |
497 image._timeStamp = rtp_timestamp; | 498 image._timeStamp = static_cast<uint32_t>(rtp_timestamp); |
498 image.capture_time_ms_ = capture_time_ms; | 499 image.capture_time_ms_ = capture_time_ms; |
499 image._frameType = | 500 image._frameType = |
500 (key_frame ? webrtc::kVideoFrameKey : webrtc::kVideoFrameDelta); | 501 (key_frame ? webrtc::kVideoFrameKey : webrtc::kVideoFrameDelta); |
501 image._completeFrame = true; | 502 image._completeFrame = true; |
502 | 503 |
503 ReturnEncodedImage(image, bitstream_buffer_id, picture_id_); | 504 ReturnEncodedImage(image, bitstream_buffer_id, picture_id_); |
504 // Picture ID must wrap after reaching the maximum. | 505 // Picture ID must wrap after reaching the maximum. |
505 picture_id_ = (picture_id_ + 1) & 0x7FFF; | 506 picture_id_ = (picture_id_ + 1) & 0x7FFF; |
506 } | 507 } |
507 | 508 |
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564 scoped_refptr<media::VideoFrame> frame; | 565 scoped_refptr<media::VideoFrame> frame; |
565 if (next_frame->video_frame_buffer()->native_handle()) { | 566 if (next_frame->video_frame_buffer()->native_handle()) { |
566 frame = static_cast<media::VideoFrame*>( | 567 frame = static_cast<media::VideoFrame*>( |
567 next_frame->video_frame_buffer()->native_handle()); | 568 next_frame->video_frame_buffer()->native_handle()); |
568 requires_copy = RequiresSizeChange(frame); | 569 requires_copy = RequiresSizeChange(frame); |
569 } else { | 570 } else { |
570 requires_copy = true; | 571 requires_copy = true; |
571 } | 572 } |
572 | 573 |
573 if (requires_copy) { | 574 if (requires_copy) { |
574 const base::TimeDelta timestamp = | |
575 frame ? frame->timestamp() | |
576 : base::TimeDelta::FromMilliseconds(next_frame->ntp_time_ms()); | |
577 base::SharedMemory* input_buffer = input_buffers_[index]; | 575 base::SharedMemory* input_buffer = input_buffers_[index]; |
578 frame = media::VideoFrame::WrapExternalSharedMemory( | 576 frame = media::VideoFrame::WrapExternalSharedMemory( |
579 media::PIXEL_FORMAT_I420, input_frame_coded_size_, | 577 media::PIXEL_FORMAT_I420, input_frame_coded_size_, |
580 gfx::Rect(input_visible_size_), input_visible_size_, | 578 gfx::Rect(input_visible_size_), input_visible_size_, |
581 reinterpret_cast<uint8_t*>(input_buffer->memory()), | 579 reinterpret_cast<uint8_t*>(input_buffer->memory()), |
582 input_buffer->mapped_size(), input_buffer->handle(), 0, timestamp); | 580 input_buffer->mapped_size(), input_buffer->handle(), 0, |
| 581 base::TimeDelta()); |
583 if (!frame.get()) { | 582 if (!frame.get()) { |
584 LogAndNotifyError(FROM_HERE, "failed to create frame", | 583 LogAndNotifyError(FROM_HERE, "failed to create frame", |
585 media::VideoEncodeAccelerator::kPlatformFailureError); | 584 media::VideoEncodeAccelerator::kPlatformFailureError); |
586 return; | 585 return; |
587 } | 586 } |
588 | 587 |
589 // Do a strided copy and scale (if necessary) the input frame to match | 588 // Do a strided copy and scale (if necessary) the input frame to match |
590 // the input requirements for the encoder. | 589 // the input requirements for the encoder. |
591 // TODO(sheu): Support zero-copy from WebRTC. http://crbug.com/269312 | 590 // TODO(sheu): Support zero-copy from WebRTC. http://crbug.com/269312 |
592 // TODO(magjed): Downscale with kFilterBox in an image pyramid instead. | 591 // TODO(magjed): Downscale with kFilterBox in an image pyramid instead. |
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604 frame->visible_data(media::VideoFrame::kVPlane), | 603 frame->visible_data(media::VideoFrame::kVPlane), |
605 frame->stride(media::VideoFrame::kVPlane), | 604 frame->stride(media::VideoFrame::kVPlane), |
606 frame->visible_rect().width(), | 605 frame->visible_rect().width(), |
607 frame->visible_rect().height(), | 606 frame->visible_rect().height(), |
608 libyuv::kFilterBox)) { | 607 libyuv::kFilterBox)) { |
609 LogAndNotifyError(FROM_HERE, "Failed to copy buffer", | 608 LogAndNotifyError(FROM_HERE, "Failed to copy buffer", |
610 media::VideoEncodeAccelerator::kPlatformFailureError); | 609 media::VideoEncodeAccelerator::kPlatformFailureError); |
611 return; | 610 return; |
612 } | 611 } |
613 } | 612 } |
| 613 // Use the timestamp set from WebRTC and set it in 90 kHz. |
| 614 frame->set_timestamp( |
| 615 base::TimeDelta::FromInternalValue(next_frame->timestamp())); |
614 frame->AddDestructionObserver(media::BindToCurrentLoop( | 616 frame->AddDestructionObserver(media::BindToCurrentLoop( |
615 base::Bind(&RTCVideoEncoder::Impl::EncodeFrameFinished, this, index))); | 617 base::Bind(&RTCVideoEncoder::Impl::EncodeFrameFinished, this, index))); |
616 video_encoder_->Encode(frame, next_frame_keyframe); | 618 video_encoder_->Encode(frame, next_frame_keyframe); |
617 input_buffers_free_.pop_back(); | 619 input_buffers_free_.pop_back(); |
618 SignalAsyncWaiter(WEBRTC_VIDEO_CODEC_OK); | 620 SignalAsyncWaiter(WEBRTC_VIDEO_CODEC_OK); |
619 } | 621 } |
620 | 622 |
621 void RTCVideoEncoder::Impl::EncodeFrameFinished(int index) { | 623 void RTCVideoEncoder::Impl::EncodeFrameFinished(int index) { |
622 DVLOG(3) << "Impl::EncodeFrameFinished(): index=" << index; | 624 DVLOG(3) << "Impl::EncodeFrameFinished(): index=" << index; |
623 DCHECK(thread_checker_.CalledOnValidThread()); | 625 DCHECK(thread_checker_.CalledOnValidThread()); |
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879 UMA_HISTOGRAM_BOOLEAN("Media.RTCVideoEncoderInitEncodeSuccess", | 881 UMA_HISTOGRAM_BOOLEAN("Media.RTCVideoEncoderInitEncodeSuccess", |
880 init_retval == WEBRTC_VIDEO_CODEC_OK); | 882 init_retval == WEBRTC_VIDEO_CODEC_OK); |
881 if (init_retval == WEBRTC_VIDEO_CODEC_OK) { | 883 if (init_retval == WEBRTC_VIDEO_CODEC_OK) { |
882 UMA_HISTOGRAM_ENUMERATION("Media.RTCVideoEncoderProfile", | 884 UMA_HISTOGRAM_ENUMERATION("Media.RTCVideoEncoderProfile", |
883 profile, | 885 profile, |
884 media::VIDEO_CODEC_PROFILE_MAX + 1); | 886 media::VIDEO_CODEC_PROFILE_MAX + 1); |
885 } | 887 } |
886 } | 888 } |
887 | 889 |
888 } // namespace content | 890 } // namespace content |
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