Update webrtc/libjingle to 6346.

content/renderer/media/webrtc_audio_device_impl.cc

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 
 #include "base/bind.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
 #include "base/win/windows_version.h"
@@ skipping 132 matching lines @@
   DCHECK_EQ(audio_bus->frames() % frames_per_10_ms, 0);
 
   // Get audio frames in blocks of 10 milliseconds from the registered
   // webrtc::AudioTransport source. Keep reading until our internal buffer
   // is full.
   uint32_t num_audio_frames = 0;
   int accumulated_audio_frames = 0;
   int16* audio_data = &render_buffer_[0];
   while (accumulated_audio_frames < audio_bus->frames()) {
     // Get 10ms and append output to temporary byte buffer.
-    uint32_t rtp_ts = 0;
-    int64_t ntp_ts = 0;
+    int64_t elapsed_time_ms = -1;
+    int64_t ntp_time_ms = -1;
     if (is_audio_track_processing_enabled_) {
       // When audio processing is enabled in the audio track, we use
       // PullRenderData() instead of NeedMorePlayData() to avoid passing the
       // render data to the APM in WebRTC as reference signal for echo
       // cancellation.
       static const int kBitsPerByte = 8;
       audio_transport_callback_->PullRenderData(bytes_per_sample * kBitsPerByte,
                                                 sample_rate,
                                                 audio_bus->channels(),
                                                 frames_per_10_ms,
                                                 audio_data,
-                                                &rtp_ts,
-                                                &ntp_ts);
+                                                &elapsed_time_ms,
+                                                &ntp_time_ms);
       accumulated_audio_frames += frames_per_10_ms;
     } else {
       // TODO(xians): Remove the following code after the APM in WebRTC is
       // deprecated.
       audio_transport_callback_->NeedMorePlayData(frames_per_10_ms,
                                                   bytes_per_sample,
                                                   audio_bus->channels(),
                                                   sample_rate,
                                                   audio_data,
                                                   num_audio_frames,
-                                                  &rtp_ts,
-                                                  &ntp_ts);
+                                                  &elapsed_time_ms,
+                                                  &ntp_time_ms);
       accumulated_audio_frames += num_audio_frames;
     }
 
     audio_data += bytes_per_10_ms;
   }
 
   // De-interleave each channel and convert to 32-bit floating-point
   // with nominal range -1.0 -> +1.0 to match the callback format.
   audio_bus->FromInterleaved(&render_buffer_[0],
                              audio_bus->frames(),
@@ skipping 369 matching lines @@
 
   // Start the Aec dump on the current default capturer.
   scoped_refptr<WebRtcAudioCapturer> default_capturer(GetDefaultCapturer());
   if (!default_capturer)
     return;
 
   default_capturer->StartAecDump(aec_dump_file_.Pass());
 }
 
 }  // namespace content