webrtc_audio_debug_recordings_browsertest.cc

// Copyright 2015 The Chromium Authors
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.

#include <stdint.h>

#include "base/command_line.h"
#include "base/files/file_enumerator.h"
#include "base/files/file_path.h"
#include "base/files/file_util.h"
#include "base/process/process_handle.h"
#include "base/strings/string_number_conversions.h"
#include "build/build_config.h"
#include "content/browser/web_contents/web_contents_impl.h"
#include "content/browser/webrtc/webrtc_content_browsertest_base.h"
#include "content/browser/webrtc/webrtc_internals.h"
#include "content/public/test/browser_test.h"
#include "content/public/test/browser_test_utils.h"
#include "content/public/test/content_browser_test_utils.h"
#include "content/shell/browser/shell.h"
#include "media/base/media_switches.h"
#include "net/test/embedded_test_server/embedded_test_server.h"
#include "third_party/blink/public/common/features.h"
#include "ui/shell_dialogs/selected_file_info.h"

namespace {

constexpr int kExpectedConsumerId = …;
constexpr int kWaveHeaderSizeBytes = …;
constexpr char kBaseFilename[] = …;

// Get the expected AEC dump file name. The name will be
// <temporary path>.<render process id>.aec_dump.<consumer id>, for example
// "/tmp/.com.google.Chrome.Z6UC3P.12345.aec_dump.1".
base::FilePath GetExpectedAecDumpFileName(const base::FilePath& base_file_path,
                                          int renderer_pid) { … }

// Get the file names of the recordings. The name will be
// <temporary path>.<kind>.<running stream id>.wav, for example
// "/tmp/.com.google.Chrome.Z6UC3P.output.1.wav". |kind| is output or input.
std::vector<base::FilePath> GetRecordingFileNames(
    base::FilePath::StringPieceType kind,
    const base::FilePath& base_file_path) { … }

// Deletes the the file specified by |path|. If that fails, waits for 100 ms and
// tries again. Returns true if the delete was successful.
// This is to handle when not being able to delete the file due to race when the
// file is being closed. See comment for CallWithAudioDebugRecordings test case
// below.
bool DeleteFileWithRetryAfterPause(const base::FilePath& path) { … }

}  // namespace

namespace content {

class WebRtcAudioDebugRecordingsBrowserTest
    : public WebRtcContentBrowserTestBase { … };

#if BUILDFLAG(IS_ANDROID) || BUILDFLAG(IS_LINUX) || BUILDFLAG(IS_CHROMEOS) || \
    BUILDFLAG(IS_FUCHSIA)
// Renderer crashes under Android ASAN: https://crbug.com/408496.
// Renderer crashes under Android: https://crbug.com/820934.
// Failures on Android M. https://crbug.com/535728.
// Flaky on Linux: https://crbug.com/871182
// Failed on Fuchsia: https://crbug.com/1470981
#define MAYBE_CallWithAudioDebugRecordings …
#else
#define MAYBE_CallWithAudioDebugRecordings …
#endif

// This tests will make a complete PeerConnection-based call, verify that
// video is playing for the call, and verify that non-empty audio debug
// recording files exist. The recording is enabled through webrtc-internals. The
// HTML and Javascript is bypassed since it would trigger a file picker dialog.
// Instead, the dialog callback FileSelected() is invoked directly. In fact,
// there's never a webrtc-internals page opened at all since that's not needed.
// Note: Both stopping the streams (at hangup()) and disabling the recordings
// are asynchronous without response when finished. This means that closing the
// files is asynchronous and being able to delete the files in the test is
// therefore timing dependent and flaky prone. For this reason we use
// DeleteFileWithRetryAfterPause() as a simple mitigation.
IN_PROC_BROWSER_TEST_F(WebRtcAudioDebugRecordingsBrowserTest,
                       MAYBE_CallWithAudioDebugRecordings) { … }

// TODO(grunell): Add test for multiple dumps when re-use of
// MediaStreamAudioProcessor in AudioCapturer has been removed.

#if BUILDFLAG(IS_ANDROID)
// Renderer crashes under Android ASAN: https://crbug.com/408496.
// Renderer crashes under Android: https://crbug.com/820934.
#define MAYBE_CallWithAudioDebugRecordingsEnabledThenDisabled …
#else
#define MAYBE_CallWithAudioDebugRecordingsEnabledThenDisabled …
#endif

// As above, but enable and disable recordings before starting a call. No files
// should be created.
IN_PROC_BROWSER_TEST_F(WebRtcAudioDebugRecordingsBrowserTest,
                       MAYBE_CallWithAudioDebugRecordingsEnabledThenDisabled) { … }

// Same test as CallWithAudioDebugRecordings, but does two parallel calls.
// TODO(crbug.com/40589452): Fix an re-enable test.
// List of issues filed before this test was disabled for all platforms:
// Renderer crashes under Android ASAN: https://crbug.com/408496.
// Renderer crashes under Android: https://crbug.com/820934.
// Failures on Android M. https://crbug.com/535728.
// Flaky on Linux: https://crbug.com/871182
IN_PROC_BROWSER_TEST_F(WebRtcAudioDebugRecordingsBrowserTest,
                       DISABLED_TwoCallsWithAudioDebugRecordings) { … }

}  // namespace content
chromium/content/browser/webrtc/webrtc_audio_debug_recordings_browsertest.cc
