MediaRecorder: support sampling rate adaption in AudioTrackRecorder

content/renderer/media/media_recorder_handler_unittest.cc

Index: content/renderer/media/media_recorder_handler_unittest.cc
diff --git a/content/renderer/media/media_recorder_handler_unittest.cc b/content/renderer/media/media_recorder_handler_unittest.cc
index 4a92b223b26d5d55755dd62eeadb3999f94916b6..0769a263b2ec746e2358e28fbd3a5c18682f5606 100644
--- a/content/renderer/media/media_recorder_handler_unittest.cc
+++ b/content/renderer/media/media_recorder_handler_unittest.cc
@@ -41,7 +41,11 @@ static const std::string kTestAudioTrackId = "audio_track_id";
 static const int kTestAudioChannels = 2;
 static const int kTestAudioBitsPerSample = 16;
 static const int kTestAudioSampleRate = 48000;
-static const int kTestAudioBufferDurationMS = 60;
+// MediaStream Audio Track always provides audio in chunks of 10 ms.

[miu, 2016/01/29 02:37:15]

This is not a global assumption.  The current Chrome implementation may do this,
but that's not something we want to hard-code everywhere.  It's fine if you want
to use 10 ms chunks for these tests, but please don't write comments or var
names to indicate that it is a global assumption.

FYI, this code change will break that assumption:
https://codereview.chromium.org/1647773002/

...and there's been a long-standing TODO to fix the code that forces WebAudio to
use 10 ms chunks.  Chris Jones, the original author and spec editor for WebAudio
was always asking Chrome Eng to use minimally-sized buffers (on the order of 2
ms or less).

[mcasas, 2016/01/29 20:37:34]

Removed the comment. For these tests is fine to use 10ms.

+static const int kMediaStreamAudioTrackBufferDurationMs = 10;
+// Opus works with 60ms buffers, so 6 MediaStreamAudioTrack Buffers are needed
+// to encode one output buffer.
+static const int kRatioOpusToMediaStreamTrackBuffers = 6;
 
 struct MediaRecorderTestParams {
   const bool has_video;
@@ -59,7 +63,7 @@ struct MediaRecorderTestParams {
 static const MediaRecorderTestParams kMediaRecorderTestParams[] = {
     {true, false, "video/webm", "vp8", 52, 32, 0, 0},
     {true, false, "video/webm", "vp9", 33, 18, 0, 0},
-    {false, true, "video/webm", "vp8", 0, 0, 990, 706}};
+    {false, true, "video/webm", "vp8", 0, 0, 996, 746}};

[miu, 2016/01/29 02:37:15]

It's a bit brittle to require the tests pass based on the number of output bytes
from the encoder.  For instance, if libvpx is updated with some kind of
improvement that changes the number of bytes, you really don't want these unit
tests to start failing, right?  Instead, I'd suggest just checking for an empty
or non-empty result to confirm "the plumbing" is good.

[mcasas, 2016/01/29 20:37:34]

My opinion is that a potential libvpx roller should 
have to deal (personally or not) with the  
consequences of a roll, including potentially 
the breaking these tests, but I understand this is
a judgement call so I'll follow your advice: removed
the whole parameter set in favour of a single
threshold.

 
 class MediaRecorderHandlerTest : public TestWithParam<MediaRecorderTestParams>,
                                  public blink::WebMediaRecorderHandlerClient {
@@ -113,7 +117,7 @@ class MediaRecorderHandlerTest : public TestWithParam<MediaRecorderTestParams>,
   scoped_ptr<media::AudioBus> NextAudioBus() {
     scoped_ptr<media::AudioBus> bus(media::AudioBus::Create(
         kTestAudioChannels,
-        kTestAudioSampleRate * kTestAudioBufferDurationMS / 1000));
+        kTestAudioSampleRate * kMediaStreamAudioTrackBufferDurationMs / 1000));
     audio_source_.OnMoreData(bus.get(), 0, 0);
     return bus;
   }
@@ -284,7 +288,7 @@ TEST_P(MediaRecorderHandlerTest, EncodeAudioFrames) {
   media::AudioParameters params(
       media::AudioParameters::AUDIO_PCM_LINEAR, media::CHANNEL_LAYOUT_STEREO,
       kTestAudioSampleRate, kTestAudioBitsPerSample,
-      kTestAudioSampleRate * kTestAudioBufferDurationMS / 1000);
+      kTestAudioSampleRate * kMediaStreamAudioTrackBufferDurationMs / 1000);
   SetAudioFormatForTesting(params);
 
   {
@@ -298,7 +302,8 @@ TEST_P(MediaRecorderHandlerTest, EncodeAudioFrames) {
         .Times(1)
         .WillOnce(RunClosure(quit_closure));
 
-    OnAudioBusForTesting(*audio_bus1);
+    for (int i = 0; i < kRatioOpusToMediaStreamTrackBuffers; ++i)
+      OnAudioBusForTesting(*audio_bus1);
     run_loop.Run();
   }
 
@@ -315,7 +320,8 @@ TEST_P(MediaRecorderHandlerTest, EncodeAudioFrames) {
         .Times(1)
         .WillOnce(RunClosure(quit_closure));
 
-    OnAudioBusForTesting(*audio_bus2);
+    for (int i = 0; i < kRatioOpusToMediaStreamTrackBuffers; ++i)
+      OnAudioBusForTesting(*audio_bus2);
     run_loop.Run();
   }