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Unified Diff: content/renderer/media/webrtc_local_audio_track_unittest.cc

Issue 23691038: Switch LiveAudio to source provider solution. (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src
Patch Set: fixed the android bot Created 7 years, 3 months ago
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Index: content/renderer/media/webrtc_local_audio_track_unittest.cc
diff --git a/content/renderer/media/webrtc_local_audio_track_unittest.cc b/content/renderer/media/webrtc_local_audio_track_unittest.cc
index 7d125dcdf825e226b95ab1a842f5763ddce6b101..1014ebe33327e74900e1c1a2e8dfb710d9f886de 100644
--- a/content/renderer/media/webrtc_local_audio_track_unittest.cc
+++ b/content/renderer/media/webrtc_local_audio_track_unittest.cc
@@ -6,6 +6,7 @@
#include "base/test/test_timeouts.h"
#include "content/renderer/media/rtc_media_constraints.h"
#include "content/renderer/media/webrtc_audio_capturer.h"
+#include "content/renderer/media/webrtc_local_audio_source_provider.h"
#include "content/renderer/media/webrtc_local_audio_track.h"
#include "media/audio/audio_parameters.h"
#include "media/base/audio_bus.h"
@@ -133,13 +134,19 @@ class MockWebRtcAudioCapturerSink : public WebRtcAudioCapturerSink {
class WebRtcLocalAudioTrackTest : public ::testing::Test {
protected:
virtual void SetUp() OVERRIDE {
+ params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
+ media::CHANNEL_LAYOUT_STEREO, 2, 0, 48000, 16, 480);
capturer_ = WebRtcAudioCapturer::CreateCapturer();
+ WebRtcLocalAudioSourceProvider* source_provider =
+ static_cast<WebRtcLocalAudioSourceProvider*>(
+ capturer_->audio_source_provider());
+ source_provider->SetSinkParamsForTesting(params_);
capturer_source_ = new MockCapturerSource();
EXPECT_CALL(*capturer_source_.get(), Initialize(_, capturer_.get(), 0))
.WillOnce(Return());
capturer_->SetCapturerSource(capturer_source_,
- media::CHANNEL_LAYOUT_STEREO,
- 48000);
+ params_.channel_layout(),
+ params_.sample_rate());
EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(false))
.WillOnce(Return());
@@ -154,6 +161,7 @@ class WebRtcLocalAudioTrackTest : public ::testing::Test {
audio_thread_.reset();
}
+ media::AudioParameters params_;
scoped_refptr<MockCapturerSource> capturer_source_;
scoped_refptr<WebRtcAudioCapturer> capturer_;
scoped_ptr<FakeAudioThread> audio_thread_;
@@ -167,7 +175,7 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectAndDisconnectOneSink) {
EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(Return());
RTCMediaConstraints constraints;
scoped_refptr<WebRtcLocalAudioTrack> track =
- WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
+ WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
&constraints);
track->Start();
EXPECT_TRUE(track->enabled());
@@ -187,7 +195,7 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectAndDisconnectOneSink) {
CaptureData(kNumberOfNetworkChannels,
params.sample_rate(),
params.channels(),
- params.frames_per_buffer(),
+ params.sample_rate() / 100,
0,
0,
false,
@@ -213,7 +221,7 @@ TEST_F(WebRtcLocalAudioTrackTest, DISABLED_DisableEnableAudioTrack) {
EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(Return());
RTCMediaConstraints constraints;
scoped_refptr<WebRtcLocalAudioTrack> track =
- WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
+ WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
&constraints);
track->Start();
static_cast<webrtc::AudioTrackInterface*>(track.get())->
@@ -229,7 +237,7 @@ TEST_F(WebRtcLocalAudioTrackTest, DISABLED_DisableEnableAudioTrack) {
CaptureData(1,
params.sample_rate(),
params.channels(),
- params.frames_per_buffer(),
+ params.sample_rate() / 100,
0,
0,
false,
@@ -242,7 +250,7 @@ TEST_F(WebRtcLocalAudioTrackTest, DISABLED_DisableEnableAudioTrack) {
CaptureData(1,
params.sample_rate(),
params.channels(),
- params.frames_per_buffer(),
+ params.sample_rate() / 100,
0,
0,
false,
@@ -263,7 +271,7 @@ TEST_F(WebRtcLocalAudioTrackTest, MultipleAudioTracks) {
EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(Return());
RTCMediaConstraints constraints;
scoped_refptr<WebRtcLocalAudioTrack> track_1 =
- WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
+ WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
&constraints);
track_1->Start();
static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
@@ -278,7 +286,7 @@ TEST_F(WebRtcLocalAudioTrackTest, MultipleAudioTracks) {
CaptureData(1,
params.sample_rate(),
params.channels(),
- params.frames_per_buffer(),
+ params.sample_rate() / 100,
0,
0,
false,
@@ -288,7 +296,7 @@ TEST_F(WebRtcLocalAudioTrackTest, MultipleAudioTracks) {
EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
scoped_refptr<WebRtcLocalAudioTrack> track_2 =
- WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
+ WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
&constraints);
track_2->Start();
static_cast<webrtc::AudioTrackInterface*>(track_2.get())->
@@ -306,7 +314,7 @@ TEST_F(WebRtcLocalAudioTrackTest, MultipleAudioTracks) {
CaptureData(1,
params.sample_rate(),
params.channels(),
- params.frames_per_buffer(),
+ params.sample_rate() / 100,
0,
0,
false,
@@ -316,7 +324,7 @@ TEST_F(WebRtcLocalAudioTrackTest, MultipleAudioTracks) {
CaptureData(1,
params.sample_rate(),
params.channels(),
- params.frames_per_buffer(),
+ params.sample_rate() / 100,
0,
0,
false,
@@ -343,7 +351,7 @@ TEST_F(WebRtcLocalAudioTrackTest, StartOneAudioTrack) {
EXPECT_CALL(*capturer_source_.get(), Start()).Times(1);
RTCMediaConstraints constraints;
scoped_refptr<WebRtcLocalAudioTrack> track =
- WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
+ WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
&constraints);
track->Start();
@@ -362,7 +370,7 @@ TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) {
EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(SignalEvent(&event));
RTCMediaConstraints constraints;
scoped_refptr<WebRtcLocalAudioTrack> track_1 =
- WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
+ WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
&constraints);
static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
GetRenderer()->AddChannel(0);
@@ -382,7 +390,7 @@ TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) {
// since it has been started.
EXPECT_CALL(*capturer_source_.get(), Start()).Times(0);
scoped_refptr<WebRtcLocalAudioTrack> track_2 =
- WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
+ WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
&constraints);
track_2->Start();
static_cast<webrtc::AudioTrackInterface*>(track_2.get())->
@@ -415,7 +423,7 @@ TEST_F(WebRtcLocalAudioTrackTest, SetNewSourceForCapturerAfterStartTrack) {
EXPECT_CALL(*capturer_source_.get(), Start()).Times(1);
RTCMediaConstraints constraints;
scoped_refptr<WebRtcLocalAudioTrack> track =
- WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
+ WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
&constraints);
track->Start();
@@ -427,8 +435,8 @@ TEST_F(WebRtcLocalAudioTrackTest, SetNewSourceForCapturerAfterStartTrack) {
.WillOnce(Return());
EXPECT_CALL(*new_source.get(), Start()).WillOnce(Return());
capturer_->SetCapturerSource(new_source,
- media::CHANNEL_LAYOUT_STEREO,
- 48000);
+ params_.channel_layout(),
+ params_.sample_rate());
// Stop the track.
EXPECT_CALL(*new_source.get(), Stop());
@@ -442,7 +450,7 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectTracksToDifferentCapturers) {
EXPECT_CALL(*capturer_source_.get(), Start()).Times(1);
RTCMediaConstraints constraints;
scoped_refptr<WebRtcLocalAudioTrack> track_1 =
- WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
+ WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
&constraints);
track_1->Start();
@@ -466,6 +474,10 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectTracksToDifferentCapturers) {
// Create a new capturer with new source with different audio format.
scoped_refptr<WebRtcAudioCapturer> new_capturer(
WebRtcAudioCapturer::CreateCapturer());
+ WebRtcLocalAudioSourceProvider* source_provider =
+ static_cast<WebRtcLocalAudioSourceProvider*>(
+ new_capturer->audio_source_provider());
+ source_provider->SetSinkParamsForTesting(params_);
scoped_refptr<MockCapturerSource> new_source(new MockCapturerSource());
EXPECT_CALL(*new_source.get(), Initialize(_, new_capturer.get(), 0))
.WillOnce(Return());
@@ -482,7 +494,7 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectTracksToDifferentCapturers) {
// Setup the second audio track, connect it to the new capturer and start it.
EXPECT_CALL(*new_source.get(), Start()).Times(1);
scoped_refptr<WebRtcLocalAudioTrack> track_2 =
- WebRtcLocalAudioTrack::Create(std::string(), new_capturer, NULL,
+ WebRtcLocalAudioTrack::Create(std::string(), new_capturer, NULL, NULL,
&constraints);
track_2->Start();
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