WebRTC: Use the MediaStream Recording API for the audio_quality_browsertest.
This CL uses the MediaStream Recording API to record the audio received
by the right tag. It saves it as a webm file that is later converted to a
wav file using ffmpeg.
This might cause a performance regression because we're using webm+ffmpeg
to generate the wav file. I set a, hopefully high enough, bitrate to
mitigate this
BUG=chromium:681577
NOTRY=True
Review-Url: https://codereview.chromium.org/2753543010
Cr-Commit-Position: refs/heads/master@{#459050}
diff --git a/chrome/browser/media/webrtc/webrtc_audio_quality_browsertest.cc b/chrome/browser/media/webrtc/webrtc_audio_quality_browsertest.cc
index 8107494..c17f1c3 100644
--- a/chrome/browser/media/webrtc/webrtc_audio_quality_browsertest.cc
+++ b/chrome/browser/media/webrtc/webrtc_audio_quality_browsertest.cc
@@ -18,6 +18,7 @@
#include "base/strings/string_util.h"
#include "base/strings/stringprintf.h"
#include "base/strings/utf_string_conversions.h"
+#include "base/test/test_file_util.h"
#include "build/build_config.h"
#include "chrome/browser/media/webrtc/webrtc_browsertest_audio.h"
#include "chrome/browser/media/webrtc/webrtc_browsertest_base.h"
@@ -28,7 +29,9 @@
#include "chrome/browser/ui/tabs/tab_strip_model.h"
#include "chrome/common/chrome_paths.h"
#include "chrome/common/chrome_switches.h"
+#include "chrome/common/pref_names.h"
#include "chrome/test/base/ui_test_utils.h"
+#include "components/prefs/pref_service.h"
#include "content/public/common/content_switches.h"
#include "content/public/test/browser_test_utils.h"
#include "media/base/audio_parameters.h"
@@ -50,6 +53,15 @@
static const char kWebRtcAudioTestHtmlPage[] =
"/webrtc/webrtc_audio_quality_test.html";
+// How long to record the audio in the receiving peerConnection.
+static const int kCaptureDurationInSeconds = 25;
+
+// The name where the recorded WebM audio file will be saved.
+static const char kWebmRecordingFilename[] = "recording.webm";
+
+// How often to ask the test page whether the audio recording is completed.
+const int kPollingIntervalInMs = 1000;
+
// For the AGC test, there are 6 speech segments split on silence. If one
// segment is significantly different in length compared to the same segment in
// the reference file, there's something fishy going on.
@@ -76,23 +88,8 @@
// pesq binary for your platform (and sox.exe on windows). Read more on how
// resources are managed in chrome/test/data/webrtc/resources/README.
//
-// This test will only work on machines that have been configured to record
-// their own input.
-//
// On Linux:
-// 1. # sudo apt-get install pavucontrol sox
-// 2. For the user who will run the test: # pavucontrol
-// 3. In a separate terminal, # arecord dummy
-// 4. In pavucontrol, go to the recording tab.
-// 5. For the ALSA plugin [aplay]: ALSA Capture from, change from <x> to
-// <Monitor of x>, where x is whatever your primary sound device is called.
-// 6. Try launching chrome as the target user on the target machine, try
-// playing, say, a YouTube video, and record with # arecord -f dat tmp.dat.
-// Verify the recording with aplay (should have recorded what you played
-// from chrome).
-//
-// Note: the volume for ALL your input devices will be forced to 100% by
-// running this test on Linux.
+// 1. # sudo apt-get install sox
//
// On Mac:
// TODO(phoglund): download sox from gs instead.
@@ -100,35 +97,10 @@
// 2. Install it + reboot.
// 3. Install MacPorts (http://www.macports.org/).
// 4. Install sox: sudo port install sox.
-// 5. (For Chrome bots) Ensure sox and rec are reachable from the env the test
-// executes in (sox and rec tends to install in /opt/, which generally isn't
-// in the Chrome bots' env). For instance, run
-// sudo ln -s /opt/local/bin/rec /usr/local/bin/rec
+// 5. (For Chrome bots) Ensure sox is reachable from the env the test
+// executes in (sox tends to install in /opt/, which generally isn't in the
+// Chrome bots' env). For instance, run
// sudo ln -s /opt/local/bin/sox /usr/local/bin/sox
-// 6. In Sound Preferences, set both input and output to Soundflower (2ch).
-// Note: You will no longer hear audio on this machine, and it will no
-// longer use any built-in mics.
-// 7. Try launching chrome as the target user on the target machine, try
-// playing, say, a YouTube video, and record with 'rec test.wav trim 0 5'.
-// Stop the video in chrome and try playing back the file; you should hear
-// a recording of the video (note; if you play back on the target machine
-// you must revert the changes in step 3 first).
-//
-// On Windows 7:
-// 1. Control panel > Sound > Manage audio devices.
-// 2. In the recording tab, right-click in an empty space in the pane with the
-// devices. Tick 'show disabled devices'.
-// 3. You should see a 'stereo mix' device - this is what your speakers output.
-// If you don't have one, your driver doesn't support stereo mix devices.
-// Some drivers use different names for the mix device though (like "Wave").
-// Right click > Properties.
-// 4. Ensure "listen to this device" is unchecked, otherwise you get echo.
-// 5. Ensure the mix device is the default recording device.
-// 6. Launch chrome and try playing a video with sound. You should see
-// in the volume meter for the mix device. Configure the mix device to have
-// 50 / 100 in level. Also go into the playback tab, right-click Speakers,
-// and set that level to 50 / 100. Otherwise you will get distortion in
-// the recording.
class MAYBE_WebRtcAudioQualityBrowserTest : public WebRtcTestBase {
public:
MAYBE_WebRtcAudioQualityBrowserTest() {}
@@ -136,6 +108,18 @@
DetectErrorsInJavaScript(); // Look for errors in our rather complex js.
}
+ void SetUpOnMainThread() override {
+ base::FilePath tmp_dir;
+ EXPECT_TRUE(base::GetTempDir(&tmp_dir));
+ webm_recorded_output_filename_ =
+ tmp_dir.Append(FILE_PATH_LITERAL("recording.webm"));
+
+ browser()->profile()->GetPrefs()->SetFilePath(
+ prefs::kDownloadDefaultDirectory, tmp_dir);
+ browser()->profile()->GetPrefs()->SetBoolean(prefs::kPromptForDownload,
+ false);
+ }
+
void SetUpCommandLine(base::CommandLine* command_line) override {
EXPECT_FALSE(command_line->HasSwitch(
switches::kUseFakeUIForMediaStream));
@@ -191,142 +175,16 @@
const std::string& constraints,
const std::string& perf_modifier);
void SetupAndRecordAudioCall(const base::FilePath& reference_file,
- const base::FilePath& recording,
- const std::string& constraints,
- const base::TimeDelta recording_time);
+ const base::FilePath& recorded_output_path,
+ const std::string& constraints);
void TestWithFakeDeviceGetUserMedia(const std::string& constraints,
const std::string& perf_modifier);
+
+ base::FilePath webm_recorded_output_filename_;
};
namespace {
-class AudioRecorder {
- public:
- AudioRecorder() {}
- ~AudioRecorder() {}
-
- // Starts the recording program for the specified duration. Returns true
- // on success. We record in 16-bit 44.1 kHz Stereo (mostly because that's
- // what SoundRecorder.exe will give us and we can't change that).
- bool StartRecording(base::TimeDelta recording_time,
- const base::FilePath& output_file) {
- EXPECT_FALSE(recording_application_.IsValid())
- << "Tried to record, but is already recording.";
-
- int duration_sec = static_cast<int>(recording_time.InSeconds());
- base::CommandLine command_line(base::CommandLine::NO_PROGRAM);
-
-#if defined(OS_WIN)
- // This disable is required to run SoundRecorder.exe on 64-bit Windows
- // from a 32-bit binary. We need to load the wow64 disable function from
- // the DLL since it doesn't exist on Windows XP.
- base::ScopedNativeLibrary kernel32_lib(base::FilePath(L"kernel32"));
- if (kernel32_lib.is_valid()) {
- typedef BOOL (WINAPI* Wow64DisableWow64FSRedirection)(PVOID*);
- Wow64DisableWow64FSRedirection wow_64_disable_wow_64_fs_redirection;
- wow_64_disable_wow_64_fs_redirection =
- reinterpret_cast<Wow64DisableWow64FSRedirection>(
- kernel32_lib.GetFunctionPointer(
- "Wow64DisableWow64FsRedirection"));
- if (wow_64_disable_wow_64_fs_redirection != NULL) {
- PVOID* ignored = NULL;
- wow_64_disable_wow_64_fs_redirection(ignored);
- }
- }
-
- char duration_in_hms[128] = {0};
- struct tm duration_tm = {0};
- duration_tm.tm_sec = duration_sec;
- EXPECT_NE(0u, strftime(duration_in_hms, arraysize(duration_in_hms),
- "%H:%M:%S", &duration_tm));
-
- command_line.SetProgram(
- base::FilePath(FILE_PATH_LITERAL("SoundRecorder.exe")));
- command_line.AppendArg("/FILE");
- command_line.AppendArgPath(output_file);
- command_line.AppendArg("/DURATION");
- command_line.AppendArg(duration_in_hms);
-#elif defined(OS_MACOSX)
- command_line.SetProgram(base::FilePath("rec"));
- command_line.AppendArg("-b");
- command_line.AppendArg("16");
- command_line.AppendArg("-q");
- command_line.AppendArgPath(output_file);
- command_line.AppendArg("trim");
- command_line.AppendArg("0");
- command_line.AppendArg(base::IntToString(duration_sec));
-#else
- command_line.SetProgram(base::FilePath("arecord"));
- command_line.AppendArg("-d");
- command_line.AppendArg(base::IntToString(duration_sec));
- command_line.AppendArg("-f");
- command_line.AppendArg("cd");
- command_line.AppendArg("-c");
- command_line.AppendArg("2");
- command_line.AppendArgPath(output_file);
-#endif
-
- DVLOG(0) << "Running " << command_line.GetCommandLineString();
- recording_application_ =
- base::LaunchProcess(command_line, base::LaunchOptions());
- return recording_application_.IsValid();
- }
-
- // Joins the recording program. Returns true on success.
- bool WaitForRecordingToEnd() {
- int exit_code = -1;
- recording_application_.WaitForExit(&exit_code);
- return exit_code == 0;
- }
- private:
- base::Process recording_application_;
-};
-
-bool ForceMicrophoneVolumeTo100Percent() {
-#if defined(OS_WIN)
- // Note: the force binary isn't in tools since it's one of our own.
- base::CommandLine command_line(test::GetReferenceFilesDir().Append(
- FILE_PATH_LITERAL("force_mic_volume_max.exe")));
- DVLOG(0) << "Running " << command_line.GetCommandLineString();
- std::string result;
- if (!base::GetAppOutput(command_line, &result)) {
- LOG(ERROR) << "Failed to set source volume: output was " << result;
- return false;
- }
-#elif defined(OS_MACOSX)
- base::CommandLine command_line(
- base::FilePath(FILE_PATH_LITERAL("osascript")));
- command_line.AppendArg("-e");
- command_line.AppendArg("set volume input volume 100");
- command_line.AppendArg("-e");
- command_line.AppendArg("set volume output volume 85");
-
- std::string result;
- if (!base::GetAppOutput(command_line, &result)) {
- LOG(ERROR) << "Failed to set source volume: output was " << result;
- return false;
- }
-#else
- // Just force the volume of, say the first 5 devices. A machine will rarely
- // have more input sources than that. This is way easier than finding the
- // input device we happen to be using.
- for (int device_index = 0; device_index < 5; ++device_index) {
- std::string result;
- const std::string kHundredPercentVolume = "65536";
- base::CommandLine command_line(base::FilePath(FILE_PATH_LITERAL("pacmd")));
- command_line.AppendArg("set-source-volume");
- command_line.AppendArg(base::IntToString(device_index));
- command_line.AppendArg(kHundredPercentVolume);
- DVLOG(0) << "Running " << command_line.GetCommandLineString();
- if (!base::GetAppOutput(command_line, &result)) {
- LOG(ERROR) << "Failed to set source volume: output was " << result;
- return false;
- }
- }
-#endif
- return true;
-}
-
// Sox is the "Swiss army knife" of audio processing. We mainly use it for
// silence trimming. See http://sox.sourceforge.net.
base::CommandLine MakeSoxCommandLine() {
@@ -387,6 +245,34 @@
return ok;
}
+// Runs ffmpeg on the captured webm video and writes it to a .wav file.
+bool RunWebmToWavConverter(const base::FilePath& webm_recorded_output_path,
+ const base::FilePath& wav_recorded_output_path) {
+ const base::FilePath path_to_ffmpeg = test::GetToolForPlatform("ffmpeg");
+ if (!base::PathExists(path_to_ffmpeg)) {
+ LOG(ERROR) << "Missing ffmpeg: should be in " << path_to_ffmpeg.value();
+ return false;
+ }
+
+ // Set up ffmpeg to output at a certain bitrate (-ab). This is hopefully set
+ // high enough to avoid degrading audio quality too much.
+ base::CommandLine ffmpeg_command(path_to_ffmpeg);
+ ffmpeg_command.AppendArg("-i");
+ ffmpeg_command.AppendArgPath(webm_recorded_output_path);
+ ffmpeg_command.AppendArg("-ab");
+ ffmpeg_command.AppendArg("300k");
+ ffmpeg_command.AppendArg("-y");
+ ffmpeg_command.AppendArgPath(wav_recorded_output_path);
+
+ // We produce an output file that will later be used as an input to the
+ // barcode decoder and frame analyzer tools.
+ DVLOG(0) << "Running " << ffmpeg_command.GetCommandLineString();
+ std::string result;
+ bool ok = base::GetAppOutputAndError(ffmpeg_command, &result);
+ DVLOG(0) << "Output was:\n\n" << result;
+ return ok;
+}
+
// Looks for 0.2 second audio segments surrounded by silences under 0.3% audio
// power and splits the input file on those silences. Output files are written
// according to the output file template (e.g. /tmp/out.wav writes
@@ -587,13 +473,13 @@
}
void ComputeAndPrintPesqResults(const base::FilePath& reference_file,
- const base::FilePath& recording,
+ const base::FilePath& recorded_output_path,
const std::string& perf_modifier) {
base::FilePath trimmed_reference = CreateTemporaryWaveFile();
base::FilePath trimmed_recording = CreateTemporaryWaveFile();
ASSERT_TRUE(RemoveSilence(reference_file, trimmed_reference));
- ASSERT_TRUE(RemoveSilence(recording, trimmed_recording));
+ ASSERT_TRUE(RemoveSilence(recorded_output_path, trimmed_recording));
std::string raw_mos;
std::string mos_lqo;
@@ -613,8 +499,8 @@
} // namespace
-// Sets up a two-way WebRTC call and records its output to |recording|, using
-// getUserMedia.
+// Sets up a two-way WebRTC call and records its output to
+// |recorded_output_path|, using getUserMedia.
//
// |reference_file| should have at least five seconds of silence in the
// beginning: otherwise all the reference audio will not be picked up by the
@@ -626,12 +512,10 @@
// file, you should end up with two time-synchronized files.
void MAYBE_WebRtcAudioQualityBrowserTest::SetupAndRecordAudioCall(
const base::FilePath& reference_file,
- const base::FilePath& recording,
- const std::string& constraints,
- const base::TimeDelta recording_time) {
+ const base::FilePath& recorded_output_path,
+ const std::string& constraints) {
ASSERT_TRUE(embedded_test_server()->Start());
ASSERT_TRUE(test::HasReferenceFilesInCheckout());
- ASSERT_TRUE(ForceMicrophoneVolumeTo100Percent());
ConfigureFakeDeviceToPlayFile(reference_file);
@@ -647,15 +531,24 @@
OpenPageAndGetUserMediaInNewTabWithConstraints(test_page, constraints);
SetupPeerconnectionWithLocalStream(right_tab);
- AudioRecorder recorder;
- ASSERT_TRUE(recorder.StartRecording(recording_time, recording));
-
NegotiateCall(left_tab, right_tab);
- ASSERT_TRUE(recorder.WaitForRecordingToEnd());
- DVLOG(0) << "Done recording to " << recording.value() << std::endl;
+ EXPECT_EQ(
+ "ok-capturing",
+ ExecuteJavascript(
+ base::StringPrintf("startAudioCapture(%d, \"%s\");",
+ kCaptureDurationInSeconds, kWebmRecordingFilename),
+ right_tab));
+
+ EXPECT_TRUE(test::PollingWaitUntil("testIsDoneCapturing();", "true",
+ right_tab, kPollingIntervalInMs));
HangUp(left_tab);
+
+ RunWebmToWavConverter(webm_recorded_output_filename_, recorded_output_path);
+ EXPECT_TRUE(base::DieFileDie(webm_recorded_output_filename_, false));
+
+ DVLOG(0) << "Done recording to " << recorded_output_path.MaybeAsASCII();
}
void MAYBE_WebRtcAudioQualityBrowserTest::TestWithFakeDeviceGetUserMedia(
@@ -669,14 +562,14 @@
base::FilePath reference_file =
test::GetReferenceFilesDir().Append(kReferenceFile);
- base::FilePath recording = CreateTemporaryWaveFile();
+ base::FilePath recorded_output_path = CreateTemporaryWaveFile();
ASSERT_NO_FATAL_FAILURE(SetupAndRecordAudioCall(
- reference_file, recording, constraints,
- base::TimeDelta::FromSeconds(30)));
+ reference_file, recorded_output_path, constraints));
- ComputeAndPrintPesqResults(reference_file, recording, perf_modifier);
- DeleteFileUnlessTestFailed(recording, false);
+ ComputeAndPrintPesqResults(reference_file, recorded_output_path,
+ perf_modifier);
+ DeleteFileUnlessTestFailed(recorded_output_path, false);
}
IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcAudioQualityBrowserTest,
@@ -694,8 +587,6 @@
ASSERT_TRUE(test::HasReferenceFilesInCheckout());
ASSERT_TRUE(embedded_test_server()->Start());
- ASSERT_TRUE(ForceMicrophoneVolumeTo100Percent());
-
content::WebContents* left_tab =
OpenPageWithoutGetUserMedia(kWebRtcAudioTestHtmlPage);
content::WebContents* right_tab =
@@ -705,26 +596,32 @@
NegotiateCall(left_tab, right_tab);
- base::FilePath recording = CreateTemporaryWaveFile();
-
- // Note: the sound clip is 21.6 seconds: record for 25 seconds to get some
- // safety margins on each side.
- AudioRecorder recorder;
- ASSERT_TRUE(recorder.StartRecording(base::TimeDelta::FromSeconds(25),
- recording));
+ const base::FilePath recorded_output_path = CreateTemporaryWaveFile();
PlayAudioFileThroughWebAudio(left_tab);
- ASSERT_TRUE(recorder.WaitForRecordingToEnd());
- DVLOG(0) << "Done recording to " << recording.value() << std::endl;
+ EXPECT_EQ(
+ "ok-capturing",
+ ExecuteJavascript(
+ base::StringPrintf("startAudioCapture(%d, \"%s\");",
+ kCaptureDurationInSeconds, kWebmRecordingFilename),
+ right_tab));
+
+ EXPECT_TRUE(test::PollingWaitUntil("testIsDoneCapturing();", "true",
+ right_tab, kPollingIntervalInMs));
HangUp(left_tab);
+ RunWebmToWavConverter(webm_recorded_output_filename_, recorded_output_path);
+ EXPECT_TRUE(base::DieFileDie(webm_recorded_output_filename_, false));
+
+ DVLOG(0) << "Done recording to " << recorded_output_path.MaybeAsASCII();
+
// Compare with the reference file on disk (this is the same file we played
// through WebAudio earlier).
base::FilePath reference_file =
test::GetReferenceFilesDir().Append(kReferenceFile);
- ComputeAndPrintPesqResults(reference_file, recording, "_webaudio");
+ ComputeAndPrintPesqResults(reference_file, recorded_output_path, "_webaudio");
}
/**
@@ -768,11 +665,10 @@
}
base::FilePath reference_file =
test::GetReferenceFilesDir().Append(reference_filename);
- base::FilePath recording = CreateTemporaryWaveFile();
+ base::FilePath recorded_output_path = CreateTemporaryWaveFile();
ASSERT_NO_FATAL_FAILURE(SetupAndRecordAudioCall(
- reference_file, recording, constraints,
- base::TimeDelta::FromSeconds(30)));
+ reference_file, recorded_output_path, constraints));
base::ScopedTempDir split_ref_files;
ASSERT_TRUE(split_ref_files.CreateUniqueTempDir());
@@ -783,8 +679,8 @@
base::ScopedTempDir split_actual_files;
ASSERT_TRUE(split_actual_files.CreateUniqueTempDir());
- ASSERT_NO_FATAL_FAILURE(
- SplitFileOnSilenceIntoDir(recording, split_actual_files.GetPath()));
+ ASSERT_NO_FATAL_FAILURE(SplitFileOnSilenceIntoDir(
+ recorded_output_path, split_actual_files.GetPath()));
// Keep the recording and split files if the analysis fails.
base::FilePath actual_files_dir = split_actual_files.Take();
@@ -794,7 +690,7 @@
AnalyzeSegmentsAndPrintResult(
ref_segments, actual_segments, reference_file, perf_modifier);
- DeleteFileUnlessTestFailed(recording, false);
+ DeleteFileUnlessTestFailed(recorded_output_path, false);
DeleteFileUnlessTestFailed(actual_files_dir, true);
}
diff --git a/chrome/test/data/webrtc/audio_extraction.js b/chrome/test/data/webrtc/audio_extraction.js
new file mode 100644
index 0000000..db7536bb
--- /dev/null
+++ b/chrome/test/data/webrtc/audio_extraction.js
@@ -0,0 +1,62 @@
+/**
+ * Copyright 2017 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.
+ */
+
+/**
+ * Aggregate target bits per second for encoding of the Audio track.
+ * @private
+ */
+const AUDIO_BITS_PER_SECOND = 3000000;
+
+var doneCapturing = false;
+
+/**
+ * Starts the audio capturing.
+ *
+ * @param {Number} The duration of the capture in seconds.
+ */
+function startAudioCapture(capture_duration_in_seconds, output_file) {
+ console.log('Started capturing for ' + capture_duration_in_seconds + 's to '
+ + output_file);
+ var inputElement = document.getElementById('remote-view');
+
+ // |audioBitsPerSecond| must set to a large number to throw as little
+ // information away as possible.
+ var mediaRecorderOptions = {
+ audioBitsPerSecond: AUDIO_BITS_PER_SECOND,
+ mimeType: 'audio/webm',
+ };
+ var stream = inputElement.srcObject;
+ var mediaRecorder = new MediaRecorder(stream, mediaRecorderOptions);
+
+ var audio_chunks = [];
+
+ mediaRecorder.ondataavailable = function(recording) {
+ audio_chunks.push(recording.data);
+ }
+ mediaRecorder.onstop = function() {
+ var audioBlob = new Blob(audio_chunks, {type: 'audio/webm'});
+
+ var url = window.URL.createObjectURL(audioBlob);
+ var a = document.createElement('a');
+ document.body.appendChild(a);
+
+ a.href = url;
+ a.download = output_file;
+ a.click();
+
+ doneCapturing = true;
+ }
+
+ mediaRecorder.start();
+ setTimeout(function() { mediaRecorder.stop(); },
+ capture_duration_in_seconds * 1000);
+
+ returnToTest('ok-capturing');
+}
+
+function testIsDoneCapturing() {
+ returnToTest(doneCapturing.toString());
+}
diff --git a/chrome/test/data/webrtc/resources/force_mic_volume_max.exe.sha1 b/chrome/test/data/webrtc/resources/force_mic_volume_max.exe.sha1
deleted file mode 100644
index 037caaca..0000000
--- a/chrome/test/data/webrtc/resources/force_mic_volume_max.exe.sha1
+++ /dev/null
@@ -1 +0,0 @@
-d721e0b1a8eec10d5d1e60ac22d755c1ec7a1587
\ No newline at end of file
diff --git a/chrome/test/data/webrtc/webrtc_audio_quality_test.html b/chrome/test/data/webrtc/webrtc_audio_quality_test.html
index 3601a23c..4b17c80 100644
--- a/chrome/test/data/webrtc/webrtc_audio_quality_test.html
+++ b/chrome/test/data/webrtc/webrtc_audio_quality_test.html
@@ -3,6 +3,7 @@
<head>
<title>WebRTC Automated Test</title>
<script type="text/javascript" src="adapter.js"></script>
+ <script type="text/javascript" src="audio_extraction.js"></script>
<script type="text/javascript" src="test_functions.js"></script>
<script type="text/javascript" src="peerconnection.js"></script>
<script type="text/javascript" src="munge_sdp.js"></script>
