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authorrlarocque@chromium.org <rlarocque@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98>2013-12-05 23:10:19 +0000
committerrlarocque@chromium.org <rlarocque@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98>2013-12-05 23:10:19 +0000
commitc243e3bfac1f725ba2a8c29ce8924c09ff9c684d (patch)
tree111567576ac601403612f31f9f46abe658b140a0
parent2e3ad4f042dbdb2bdf784d328ea53b190d408bb6 (diff)
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Revert 238953 "Adding an audio-level WebRTC test to verify mutin..."
> Adding an audio-level WebRTC test to verify muting and playback. > > This test uses the stats API to measure the audioOutputLevel. Since the > content browsertests always uses fake devices we can adapt the test to > look for that particular signal. > > This test is primarily intended to run on Android but will work on > desktop as well. Tested by running manually on a Nexus 7 device > at my workstation. > > BUG=325164 > > Review URL: https://codereview.chromium.org/101773002 TBR=phoglund@chromium.org BUG=326338 Review URL: https://codereview.chromium.org/107443002 git-svn-id: svn://svn.chromium.org/chrome/trunk/src@239065 0039d316-1c4b-4281-b951-d872f2087c98
Diffstat
-rw-r--r--content/browser/media/webrtc_browsertest.cc41
-rw-r--r--content/test/data/media/peerconnection-call.html60
-rw-r--r--content/test/data/media/webrtc_test_audio.js74
3 files changed, 2 insertions, 173 deletions
diff --git a/content/browser/media/webrtc_browsertest.cc b/content/browser/media/webrtc_browsertest.cc
index f76a215..e63207d 100644
--- a/content/browser/media/webrtc_browsertest.cc
+++ b/content/browser/media/webrtc_browsertest.cc
@@ -21,14 +21,6 @@
#include "base/win/windows_version.h"
#endif
-const char kForceIsac16K[] =
-#ifdef OS_ANDROID
- // The default audio codec, Opus, doesn't work on Android.
- "true";
-#else
- "false";
-#endif
-
namespace {
static const char kGetUserMediaAndStop[] = "getUserMediaAndStop";
@@ -566,37 +558,4 @@ IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, AddTwoMediaStreamsToOnePC) {
ExpectTitle("OK");
}
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest,
- EstablishAudioVideoCallAndMeasureOutputLevel) {
- ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
- ASSERT_TRUE(CommandLine::ForCurrentProcess()->HasSwitch(
- switches::kUseFakeDeviceForMediaStream))
- << "Must run with fake devices since the test will explicitly look "
- << "for the fake device signal.";
-
- GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
- NavigateToURL(shell(), url);
-
- EXPECT_TRUE(ExecuteJavascript(
- base::StringPrintf("callAndEnsureAudioIsPlaying(%s);", kForceIsac16K)));
- ExpectTitle("OK");
-}
-
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest,
- EstablishAudioVideoCallAndVerifyMutingWorks) {
- ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
- ASSERT_TRUE(CommandLine::ForCurrentProcess()->HasSwitch(
- switches::kUseFakeDeviceForMediaStream))
- << "Must run with fake devices since the test will explicitly look "
- << "for the fake device signal.";
-
- GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
- NavigateToURL(shell(), url);
-
- EXPECT_TRUE(ExecuteJavascript(
- base::StringPrintf("callAndEnsureAudioMutingWorks(%s);",
- kForceIsac16K)));
- ExpectTitle("OK");
-}
-
} // namespace content
diff --git a/content/test/data/media/peerconnection-call.html b/content/test/data/media/peerconnection-call.html
index 3b9b053..3793dc4 100644
--- a/content/test/data/media/peerconnection-call.html
+++ b/content/test/data/media/peerconnection-call.html
@@ -1,7 +1,6 @@
<html>
<head>
<script type="text/javascript" src="webrtc_test_utilities.js"></script>
- <script type="text/javascript" src="webrtc_test_audio.js"></script>
<script type="text/javascript">
$ = function(id) {
return document.getElementById(id);
@@ -248,56 +247,6 @@
detectVideoPlaying('remote-view-2', onCallEstablished);
}
- //TODO(phoglund): do this for all tests on android if this works on bots.
- /** @private */
- function forceIsac16k_(sdp) {
- // Remove all other codecs (not the video codecs though).
- sdp = sdp.replace(/m=audio (\d+) RTP\/SAVPF.*\r\n/g,
- 'm=audio $1 RTP/SAVPF 103\r\n');
- sdp = sdp.replace('a=fmtp:111 minptime=10', 'a=fmtp:103 minptime=10');
- sdp = sdp.replace(/a=rtpmap:(?!103)\d{1,3} (?!VP8|red|ulpfec).*\r\n/g, '');
- return sdp;
- }
-
- function callAndEnsureAudioIsPlaying(force_isac_16k) {
- if (force_isac_16k)
- transformSdp = forceIsac16k_;
- createConnections(null);
- navigator.webkitGetUserMedia({audio: true, video: true},
- addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
-
- // Wait until we have gathered samples and can conclude if audio is playing.
- addExpectedEvent();
- var onCallEstablished = function() {
- gatherAudioLevelSamples(gSecondConnection, 300, 100,
- function(samples) {
- verifyAudioIsPlaying(samples);
- eventOccured();
- });
- };
-
- detectVideoPlaying('remote-view-2', onCallEstablished);
- }
-
- function callAndEnsureAudioMutingWorks() {
- callAndEnsureAudioIsPlaying();
- setAllEventsOccuredHandler(function() {
- var audioTrack =
- gSecondConnection.getRemoteStreams()[0].getAudioTracks()[0];
-
- // Call is up, now mute the track and check everything goes silent (give
- // it a small delay though, we don't expect it to happen instantly).
- audioTrack.enabled = false;
-
- setTimeout(function() {
- gatherAudioLevelSamples(gSecondConnection, 200, 100, function(samples) {
- verifyIsSilent(samples);
- document.title = 'OK';
- });
- }, 500);
- });
- }
-
// Test call with a new Video MediaStream that has been created based on a
// stream generated by getUserMedia.
function callWithNewVideoMediaStream() {
@@ -501,15 +450,10 @@
gLocalStream = localStream;
}
-
+
// Called if getUserMedia fails.
function printGetUserMediaError(error) {
- document.title = 'getUserMedia request failed:';
- if (error.constraintName)
- document.title += ' could not satisfy constraint ' + error.constraintName;
- else
- document.title += ' devices not working/user denied access.';
- console.log(document.title);
+ document.title = 'getUserMedia request failed with code ' + error.code;
}
// Called if getUserMedia succeeds and we want to send from both connections.
diff --git a/content/test/data/media/webrtc_test_audio.js b/content/test/data/media/webrtc_test_audio.js
deleted file mode 100644
index d0a8ed4..0000000
--- a/content/test/data/media/webrtc_test_audio.js
+++ /dev/null
@@ -1,74 +0,0 @@
-// 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.
-
-// Audio test utilities.
-
-// Gathers |numSamples| samples at |frequency| number of times per second and
-// calls back |callback| with an array with numbers in the [0, 32768] range.
-function gatherAudioLevelSamples(peerConnection, numSamples, frequency,
- callback) {
- var audioLevelSamples = []
- var gatherSamples = setInterval(function() {
- peerConnection.getStats(function(response) {
- audioLevelSamples.push(getAudioLevelFromStats_(response));
- if (audioLevelSamples.length == numSamples) {
- clearInterval(gatherSamples);
- callback(audioLevelSamples);
- }
- });
- }, 1000 / frequency);
-}
-
-// Tries to identify the beep-every-second signal generated by the fake audio
-// media/audio/fake_audio_input_stream.cc. Fails the test if we can't see a
-// signal.
-function verifyAudioIsPlaying(samples) {
- var average = 0;
- for (var i = 0; i < samples.length; ++i)
- average += samples[i] / samples.length;
-
- var largest = 0;
- for (var i = 0; i < samples.length; ++i)
- largest = Math.max(largest, samples[i]);
-
- console.log('Average audio level: ' + average + ', largest: ' + largest);
-
- // TODO(phoglund): Make a more sophisticated curve-fitting algorithm. We want
- // to see a number of peaks with relative silence between them. The following
- // seems to work fine on a nexus 7.
- if (average < 3000 || average > 8000)
- throw 'Unexpected avg audio level: got ' + average + ', expected it ' +
- 'to be 4000 < avg < 8000.'
- if (largest < 30000)
- throw 'Too low max audio level: got ' + largest + ', expected > 30000.';
-};
-
-// If silent (like when muted), we should get very near zero audio level.
-function verifyIsSilent(samples) {
- var average = 0;
- for (var i = 0; i < samples.length; ++i)
- average += samples[i] / samples.length;
-
- console.log('Average audio level: ' + average);
- if (average > 10)
- throw 'Expected silence, but avg audio level was ' + average;
-}
-
-/**
- * @private
- */
-function getAudioLevelFromStats_(response) {
- var reports = response.result();
- var audioOutputLevels = [];
- for (var i = 0; i < reports.length; ++i) {
- var report = reports[i];
- if (report.names().indexOf('audioOutputLevel') != -1) {
- audioOutputLevels.push(report.stat('audioOutputLevel'));
- }
- }
- // Should only be one audio level reported, otherwise we get confused.
- expectEquals(1, audioOutputLevels.length);
-
- return audioOutputLevels[0];
-} \ No newline at end of file
generated by cgit v1.1 at 2024-12-24 20:47:47 +0000