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| 1 /* | 1 /* |
| 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
| 3 * | 3 * |
| 4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ | 9 */ |
| 10 | 10 |
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| 79 int32_t VideoStreamDecoder::FrameToRender(VideoFrame& video_frame) { // NOLINT | 79 int32_t VideoStreamDecoder::FrameToRender(VideoFrame& video_frame) { // NOLINT |
| 80 if (pre_render_callback_) { | 80 if (pre_render_callback_) { |
| 81 // Post processing is not supported if the frame is backed by a texture. | 81 // Post processing is not supported if the frame is backed by a texture. |
| 82 if (!video_frame.video_frame_buffer()->native_handle()) { | 82 if (!video_frame.video_frame_buffer()->native_handle()) { |
| 83 pre_render_callback_->FrameCallback(&video_frame); | 83 pre_render_callback_->FrameCallback(&video_frame); |
| 84 } | 84 } |
| 85 } | 85 } |
| 86 | 86 |
| 87 if (incoming_video_stream_) | 87 if (incoming_video_stream_) |
| 88 incoming_video_stream_->OnFrame(video_frame); | 88 incoming_video_stream_->OnFrame(video_frame); |
| 89 else | |
| 90 LOG(LS_INFO) << "Dropping frame because incoming_video_stream_ is NULL."; | |
|
stefan-webrtc
2016/09/26 10:44:09
AFAICT this can never happen as incoming_video_str
sakal
2016/09/26 11:23:59
It looks like this class is instantiated here: htt
stefan-webrtc
2016/09/26 11:39:22
Seems like it's always being set outside of Call t
magjed_webrtc
2016/09/27 15:06:11
There is probably no reason to not attach a render
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| 89 | 91 |
| 90 return 0; | 92 return 0; |
| 91 } | 93 } |
| 92 | 94 |
| 93 int32_t VideoStreamDecoder::ReceivedDecodedReferenceFrame( | 95 int32_t VideoStreamDecoder::ReceivedDecodedReferenceFrame( |
| 94 const uint64_t picture_id) { | 96 const uint64_t picture_id) { |
| 95 RTC_NOTREACHED(); | 97 RTC_NOTREACHED(); |
| 96 return 0; | 98 return 0; |
| 97 } | 99 } |
| 98 | 100 |
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| 136 jitter_buffer_ms, min_playout_delay_ms, render_delay_ms, last_rtt); | 138 jitter_buffer_ms, min_playout_delay_ms, render_delay_ms, last_rtt); |
| 137 } | 139 } |
| 138 | 140 |
| 139 void VideoStreamDecoder::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) { | 141 void VideoStreamDecoder::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) { |
| 140 video_receiver_->SetReceiveChannelParameters(max_rtt_ms); | 142 video_receiver_->SetReceiveChannelParameters(max_rtt_ms); |
| 141 | 143 |
| 142 rtc::CritScope lock(&crit_); | 144 rtc::CritScope lock(&crit_); |
| 143 last_rtt_ms_ = avg_rtt_ms; | 145 last_rtt_ms_ = avg_rtt_ms; |
| 144 } | 146 } |
| 145 } // namespace webrtc | 147 } // namespace webrtc |
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