Removed SSRC knowledge from ViEEncoder.

webrtc/video/video_send_stream.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/video/video_send_stream.h"
 
 #include <algorithm>
 #include <sstream>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/pacing/packet_router.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/utility/include/process_thread.h"
+#include "webrtc/modules/video_coding/utility/ivf_file_writer.h"
 #include "webrtc/video/call_stats.h"
 #include "webrtc/video/video_capture_input.h"
 #include "webrtc/video/vie_remb.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 
 class RtcpIntraFrameObserver;
 class TransportFeedbackObserver;
 
@@ skipping 329 matching lines @@
       encoder_thread_(EncoderThreadFunction, this, "EncoderThread"),
       encoder_wakeup_event_(false, false),
       stop_encoder_thread_(0),
       overuse_detector_(
           Clock::GetRealTimeClock(),
           GetCpuOveruseOptions(config.encoder_settings.full_overuse_time),
           this,
           config.post_encode_callback,
           &stats_proxy_),
       vie_encoder_(num_cpu_cores,
-                   config_.rtp.ssrcs,
                    module_process_thread_,
                    &stats_proxy_,
                    config.pre_encode_callback,
                    &overuse_detector_,
                    congestion_controller_->pacer()),
       video_sender_(vie_encoder_.video_sender()),
       bandwidth_observer_(congestion_controller_->GetBitrateController()
                               ->CreateRtcpBandwidthObserver()),
       rtp_rtcp_modules_(CreateRtpRtcpModules(
           config.send_transport,
@@ skipping 151 matching lines @@
       encoder_settings->video_codec.startBitrate =
           bitrate_allocator_->AddObserver(
               this, encoder_settings->video_codec.minBitrate * 1000,
               encoder_settings->video_codec.maxBitrate * 1000) /
           1000;
 
       payload_router_.SetSendStreams(encoder_settings->streams);
       vie_encoder_.SetEncoder(encoder_settings->video_codec,
                               encoder_settings->min_transmit_bitrate_bps,
                               payload_router_.MaxPayloadLength(), this);
+
+      // Clear stats for disabled layers.
+      for (size_t i = encoder_settings->streams.size();
+           i < config_.rtp.ssrcs.size(); ++i) {
+        stats_proxy_.OnInactiveSsrc(config_.rtp.ssrcs[i]);
+      }
+
       if (config_.suspend_below_min_bitrate) {
         video_sender_->SuspendBelowMinBitrate();
         bitrate_allocator_->EnforceMinBitrate(false);
       }
       // We might've gotten new settings while configuring the encoder settings,
       // restart from the top to see if that's the case before trying to encode
       // a frame (which might correspond to the last frame size).
       encoder_wakeup_event_.Set();
       continue;
     }
@@ skipping 40 matching lines @@
   if (config_.overuse_callback)
     config_.overuse_callback->OnLoadUpdate(LoadObserver::kUnderuse);
 }
 
 int32_t VideoSendStream::Encoded(const EncodedImage& encoded_image,
                                  const CodecSpecificInfo* codec_specific_info,
                                  const RTPFragmentationHeader* fragmentation) {
   // |encoded_frame_proxy_| forwards frames to |config_.post_encode_callback|;
   encoded_frame_proxy_.Encoded(encoded_image, codec_specific_info,
                                fragmentation);
-  return payload_router_.Encoded(encoded_image, codec_specific_info,
-                                 fragmentation);
+  bool return_value = payload_router_.Encoded(

[pbos-webrtc, 2016/05/02 01:06:46]

eh, no, this should be int32_t, ignore comment below

[perkj_webrtc, 2016/05/03 10:22:40]

Done.

+      encoded_image, codec_specific_info, fragmentation);

[pbos-webrtc, 2016/05/02 01:06:11]

== 0

+
+  if (kEnableFrameRecording) {
+    int layer = codec_specific_info->codecType == kVideoCodecVP8
+                    ? codec_specific_info->codecSpecific.VP8.simulcastIdx
+                    : 0;
+    IvfFileWriter* file_writer;
+    {
+      if (file_writers_[layer] == nullptr) {
+        std::ostringstream oss;
+        oss << "send_bitstream_ssrc";
+        for (uint32_t ssrc : config_.rtp.ssrcs)
+          oss << "_" << ssrc;
+        oss << "_layer" << layer << ".ivf";
+        file_writers_[layer] =
+            IvfFileWriter::Open(oss.str(), codec_specific_info->codecType);
+      }
+      file_writer = file_writers_[layer].get();
+    }
+    if (file_writer) {
+      bool ok = file_writer->WriteFrame(encoded_image);
+      RTC_DCHECK(ok);
+    }
+  }
+
+  return return_value;
 }
 
 void VideoSendStream::ConfigureProtection() {
   // Enable NACK, FEC or both.
   const bool enable_protection_nack = config_.rtp.nack.rtp_history_ms > 0;
   bool enable_protection_fec = config_.rtp.fec.red_payload_type != -1;
   // Payload types without picture ID cannot determine that a stream is complete
   // without retransmitting FEC, so using FEC + NACK for H.264 (for instance) is
   // a waste of bandwidth since FEC packets still have to be transmitted. Note
   // that this is not the case with FLEXFEC.
@@ skipping 147 matching lines @@
                           &module_nack_rate);
     *sent_video_rate_bps += module_video_rate;
     *sent_nack_rate_bps += module_nack_rate;
     *sent_fec_rate_bps += module_fec_rate;
   }
   return 0;
 }
 
 }  // namespace internal
 }  // namespace webrtc