AudioConverter: Express delay in frames rather than msec.

media/base/audio_converter_unittest.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.
 
 // MSVC++ requires this to be set before any other includes to get M_PI.
 #define _USE_MATH_DEFINES
 
 #include "media/base/audio_converter.h"
 
 #include <stddef.h>
@@ skipping 198 matching lines @@
                                     CHANNEL_LAYOUT_DISCRETE, kSampleRate * 2,
                                     kBitsPerChannel, kHighLatencyBufferSize);
   output_parameters.set_channels_for_discrete(5);
 
   AudioConverter converter(input_parameters, output_parameters, false);
   FakeAudioRenderCallback callback(0.2);
   std::unique_ptr<AudioBus> audio_bus = AudioBus::Create(output_parameters);
   converter.AddInput(&callback);
   converter.Convert(audio_bus.get());
 
-  // Calculate the expected buffer delay for given AudioParameters.
-  double input_sample_rate = input_parameters.sample_rate();
-  int fill_count =
-      (output_parameters.frames_per_buffer() * input_sample_rate /
-       output_parameters.sample_rate()) / input_parameters.frames_per_buffer();
+  // double input_sample_rate = input_parameters.sample_rate();
+  // int fill_count =
+  //     (output_parameters.frames_per_buffer() * input_sample_rate /
+  //      output_parameters.sample_rate()) /
+  //      input_parameters.frames_per_buffer();
+  //
+  // This magic number is the accumulated MultiChannelResampler delay after
+  // |fill_count| (4) callbacks to provide input. The number of frames delayed
+  // is an implementation detail of the SincResampler chunk size (480 for the
+  // first two callbacks, 512 for the last two callbacks). See
+  // SincResampler.ChunkSize().
+  int kExpectedDelay = 992;
 
-  base::TimeDelta input_frame_duration = base::TimeDelta::FromMicroseconds(
-      base::Time::kMicrosecondsPerSecond / input_sample_rate);
-
-  int expected_last_delay_milliseconds =
-      fill_count * input_parameters.frames_per_buffer() *
-      input_frame_duration.InMillisecondsF();
-
-  EXPECT_EQ(expected_last_delay_milliseconds,

[chcunningham, 2016/05/24 01:34:38]

This old method for building an expectation is busted. It relied heavily on
external factors, the imprecision of using milliseconds, and rounding done in
the FakeAudioRenderCallback. I'm not sure its worth it to check the delay as I'm
doing now, but this at least serves to document the details of building an
expectation. 

-            callback.last_audio_delay_milliseconds());
+  EXPECT_EQ(kExpectedDelay, callback.last_frames_delayed());
   EXPECT_EQ(input_parameters.channels(), callback.last_channel_count());
 }
 
 TEST_P(AudioConverterTest, ArbitraryOutputRequestSize) {
   // Resize output bus to be half of |output_parameters_|'s frames_per_buffer().
   audio_bus_ = AudioBus::Create(output_parameters_.channels(),
                                 output_parameters_.frames_per_buffer() / 2);
   RunTest(1);
 }
 
@@ skipping 15 matching lines @@
         // No resampling. No channel mixing.
         std::tr1::make_tuple(44100, 44100, CHANNEL_LAYOUT_STEREO, 0.00000048),
 
         // Upsampling. Channel upmixing.
         std::tr1::make_tuple(44100, 48000, CHANNEL_LAYOUT_QUAD, 0.033),
 
         // Downsampling. Channel downmixing.
         std::tr1::make_tuple(48000, 41000, CHANNEL_LAYOUT_MONO, 0.042)));
 
 }  // namespace media