webrtc_m130/webrtc/voice_engine/transmit_mixer_unittest.cc
Peter Kasting dce40cf804 Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.

This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.

This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002

The change is being landed as TBR to all the folks who reviewed the above.

BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher

Review URL: https://codereview.webrtc.org/1230503003 .

Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 21:52:45 +00:00

60 lines
2.5 KiB
C++

/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/voice_engine/transmit_mixer.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/voice_engine/include/voe_external_media.h"
namespace webrtc {
namespace voe {
namespace {
class MediaCallback : public VoEMediaProcess {
public:
virtual void Process(int channel, ProcessingTypes type,
int16_t audio[], size_t samples_per_channel,
int sample_rate_hz, bool is_stereo) {
}
};
// TODO(andrew): Mock VoEMediaProcess, and verify the behavior when calling
// PrepareDemux().
TEST(TransmitMixerTest, RegisterExternalMediaCallback) {
TransmitMixer* tm = NULL;
ASSERT_EQ(0, TransmitMixer::Create(tm, 0));
ASSERT_TRUE(tm != NULL);
MediaCallback callback;
EXPECT_EQ(-1, tm->RegisterExternalMediaProcessing(NULL,
kRecordingPreprocessing));
EXPECT_EQ(-1, tm->RegisterExternalMediaProcessing(&callback,
kPlaybackPerChannel));
EXPECT_EQ(-1, tm->RegisterExternalMediaProcessing(&callback,
kPlaybackAllChannelsMixed));
EXPECT_EQ(-1, tm->RegisterExternalMediaProcessing(&callback,
kRecordingPerChannel));
EXPECT_EQ(0, tm->RegisterExternalMediaProcessing(&callback,
kRecordingAllChannelsMixed));
EXPECT_EQ(0, tm->RegisterExternalMediaProcessing(&callback,
kRecordingPreprocessing));
EXPECT_EQ(-1, tm->DeRegisterExternalMediaProcessing(kPlaybackPerChannel));
EXPECT_EQ(-1, tm->DeRegisterExternalMediaProcessing(
kPlaybackAllChannelsMixed));
EXPECT_EQ(-1, tm->DeRegisterExternalMediaProcessing(kRecordingPerChannel));
EXPECT_EQ(0, tm->DeRegisterExternalMediaProcessing(
kRecordingAllChannelsMixed));
EXPECT_EQ(0, tm->DeRegisterExternalMediaProcessing(kRecordingPreprocessing));
TransmitMixer::Destroy(tm);
}
} // namespace
} // namespace voe
} // namespace webrtc