Support configuring the audio buffer duration in the Pepper MediaStream API.

ppapi/tests/test_media_stream_audio_track.cc

 // Copyright 2014 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.
 //
 // Tests PPB_MediaStreamAudioTrack interface.
 
 #include "ppapi/tests/test_media_stream_audio_track.h"
 
 #include "ppapi/c/private/ppb_testing_private.h"
 #include "ppapi/cpp/audio_buffer.h"
 #include "ppapi/cpp/completion_callback.h"
 #include "ppapi/cpp/instance.h"
 #include "ppapi/cpp/var.h"
 #include "ppapi/tests/test_utils.h"
 #include "ppapi/tests/testing_instance.h"
 
 REGISTER_TEST_CASE(MediaStreamAudioTrack);
 
 namespace {
 
 // Real max defined in

[dmichael (off chromium), 2014/07/23 20:08:42]

s/max/constants

[Anand Mistry (off Chromium), 2014/07/24 01:28:28]

Done.

 // content/renderer/pepper/pepper_media_stream_audio_track_host.cc.
 const int32_t kMaxNumberOfBuffers = 1000;
+const int32_t kMinDuration = 10;
+const int32_t kMaxDuration = 10000;
 const int32_t kTimes = 3;
 const char kJSCode[] =
     "function gotStream(stream) {"
     "  test_stream = stream;"
     "  var track = stream.getAudioTracks()[0];"
     "  var plugin = document.getElementById('plugin');"
     "  plugin.postMessage(track);"
     "}"
     "var constraints = {"
     "  audio: true,"
@@ skipping 160 matching lines @@
       ASSERT_GE(buffer.GetTimestamp(), timestamp);
       timestamp = buffer.GetTimestamp();
 
       ASSERT_GT(buffer.GetDataBufferSize(), 0U);
       ASSERT_TRUE(buffer.GetDataBuffer() != NULL);
 
       audio_track_.RecycleBuffer(buffer);
     }
   }
 
+  // Configure buffer duration.
+  struct {
+    int32_t duration;
+    int32_t expect_result;
+  } durations[] = {
+    { kMinDuration, PP_OK },
+    { kMaxDuration, PP_OK },  // This will take ~30 seconds.

[dmichael (off chromium), 2014/07/23 20:08:42]

We can't have the test take that long.

Maybe you can separate that out, and just make sure it can Configure
successfully with a buffer that big?

[Anand Mistry (off Chromium), 2014/07/24 01:28:28]

Done.

+    { 123, PP_OK },
+    { kMinDuration - 1, PP_ERROR_BADARGUMENT },
+    { kMaxDuration + 1, PP_ERROR_BADARGUMENT },
+  };
+  for (size_t i = 0; i < sizeof(durations) / sizeof(durations[0]); ++i) {

[dmichael (off chromium), 2014/07/23 20:08:42]

Instead of duplicating this part, perhaps you can make a function?

std::string CheckGetBuffer(int32_t attrib_list[]);
(or whatever name you think makes sense)

and you would use it like this (just after making attrib_list on 219):
ASSERT_SUBTEST_SUCCESS(CheckGetBuffer(attrib_list));

[Anand Mistry (off Chromium), 2014/07/24 01:28:28]

Done.

+    TestCompletionCallback cc_configure(instance_->pp_instance(), false);
+    int32_t attrib_list[] = {
+      PP_MEDIASTREAMAUDIOTRACK_ATTRIB_DURATION, durations[i].duration,
+      PP_MEDIASTREAMAUDIOTRACK_ATTRIB_NONE,
+    };
+    cc_configure.WaitForResult(
+        audio_track_.Configure(attrib_list, cc_configure.GetCallback()));
+    ASSERT_EQ(durations[i].expect_result, cc_configure.result());
+
+    // Get some buffers. This always works, but the buffer size will vary.
+    for (int j = 0; j < kTimes; ++j) {
+      TestCompletionCallbackWithOutput<pp::AudioBuffer> cc_get_buffer(
+          instance_->pp_instance(), false);
+      cc_get_buffer.WaitForResult(
+          audio_track_.GetBuffer(cc_get_buffer.GetCallback()));
+      ASSERT_EQ(PP_OK, cc_get_buffer.result());
+      pp::AudioBuffer buffer = cc_get_buffer.output();
+      ASSERT_FALSE(buffer.is_null());
+      ASSERT_TRUE(IsSampleRateValid(buffer.GetSampleRate()));
+      ASSERT_EQ(buffer.GetSampleSize(), PP_AUDIOBUFFER_SAMPLESIZE_16_BITS);
+
+      ASSERT_GE(buffer.GetTimestamp(), timestamp);
+      timestamp = buffer.GetTimestamp();
+
+      ASSERT_TRUE(buffer.GetDataBuffer() != NULL);
+      if (durations[i].expect_result == PP_OK) {
+        uint32_t buffer_size = buffer.GetDataBufferSize();
+        uint32_t channels = buffer.GetNumberOfChannels();
+        uint32_t sample_rate = buffer.GetSampleRate();
+        uint32_t bytes_per_frame = channels * 2;
+        int32_t duration = durations[i].duration;
+        ASSERT_EQ(buffer_size % bytes_per_frame, 0U);
+        ASSERT_EQ(buffer_size,
+                  (duration * sample_rate * bytes_per_frame) / 1000);
+      } else {
+        ASSERT_GT(buffer.GetDataBufferSize(), 0U);
+      }
+
+      audio_track_.RecycleBuffer(buffer);
+    }
+  }
+
   // Configure should fail while plugin holds buffers.
   {
     TestCompletionCallbackWithOutput<pp::AudioBuffer> cc_get_buffer(
         instance_->pp_instance(), false);
     cc_get_buffer.WaitForResult(
         audio_track_.GetBuffer(cc_get_buffer.GetCallback()));
     ASSERT_EQ(PP_OK, cc_get_buffer.result());
     pp::AudioBuffer buffer = cc_get_buffer.output();
     int32_t attrib_list[] = {
       PP_MEDIASTREAMAUDIOTRACK_ATTRIB_BUFFERS, 0,
       PP_MEDIASTREAMAUDIOTRACK_ATTRIB_NONE,
     };
     TestCompletionCallback cc_configure(instance_->pp_instance(), false);
     cc_configure.WaitForResult(
         audio_track_.Configure(attrib_list, cc_configure.GetCallback()));
     ASSERT_EQ(PP_ERROR_INPROGRESS, cc_configure.result());
     audio_track_.RecycleBuffer(buffer);
   }
 
   // Close the track.
   audio_track_.Close();
   ASSERT_TRUE(audio_track_.HasEnded());
   audio_track_ = pp::MediaStreamAudioTrack();
   PASS();
 }