Support AudioContextOptions latencyHint as double.

content/renderer/media/audio_renderer_mixer_manager.cc

 // Copyright (c) 2012 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/audio_renderer_mixer_manager.h"
 
 #include <algorithm>
 #include <string>
 
 #include "base/bind.h"
@@ skipping 45 matching lines @@
       output_buffer_size = media::AudioLatency::GetRtcBufferSize(
           output_sample_rate, valid_not_fake_hardware_params
                                   ? hardware_params.frames_per_buffer()
                                   : 0);
       break;
     case media::AudioLatency::LATENCY_PLAYBACK:
       output_buffer_size = media::AudioLatency::GetHighLatencyBufferSize(
           output_sample_rate, preferred_high_latency_output_buffer_size);
       break;
     case media::AudioLatency::LATENCY_EXACT_MS:
-    // TODO(olka): add support when WebAudio requires it.
+      output_buffer_size =
+          std::max(std::min(input_params.frames_per_buffer(),
+                            media::AudioLatency::GetHighLatencyBufferSize(

[o1ka, 2017/03/14 15:55:43]

HighLatencyBufferSize is 20 ms. Is it the the maximum value WebAudio want to
have?

rtoy@: latencies more than 10 ms will expose
https://bugs.chromium.org/p/chromium/issues/detail?id=701000 - something to be
aware of. I marked it as blocking
https://bugs.chromium.org/p/chromium/issues/detail?id=564276 (which does not
have an owner?)

[Raymond Toy, 2017/03/14 16:14:35]

Is 20ms the value for "playback"?

In any case, there are Android devices with huge HW buffer sizes, much greater
than 20 ms long. We need to support these.

IIRC Andrew said in some cases Android "interactive" was greater than
"balanced".  We'll need to fix these issus (in a different CL).

+                                output_sample_rate,
+                                preferred_high_latency_output_buffer_size)),
+                   hardware_params.frames_per_buffer());

[Raymond Toy, 2017/03/14 15:19:39]

Blink has a clampTo() function that's easier to read than this.  Don't know if
there's a Chromium equivalent.

Also, you do very similar clamping in several places.  Perhaps refactoring this
clamping into its own function would be good.

[Andrew MacPherson, 2017/03/15 15:08:17]

Makes sense, I've refactored the AudioContextTest in Blink to use clampTo() and
I've added a media::AudioLatency::GetExactBufferSize() as suggested by olka for
the Chromium side.

+      break;
     default:
       NOTREACHED();
   }
 
   DCHECK_NE(output_buffer_size, 0);
 
   // Force to 16-bit output for now since we know that works everywhere;
   // ChromeOS does not support other bit depths.
   media::AudioParameters params(input_params.format(),
                                 input_params.channel_layout(),
@@ skipping 170 matching lines @@
     const url::Origin& security_origin)
     : source_render_frame_id(source_render_frame_id),
       params(params),
       latency(latency),
       device_id(device_id),
       security_origin(security_origin) {}
 
 AudioRendererMixerManager::MixerKey::MixerKey(const MixerKey& other) = default;
 
 }  // namespace content