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Remove old methods in AudioTransport, make it pass a gn build

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when building with default warnings.

This is in preparation for making a gn target for audio_device_tests.

BUG=webrtc:6170, webrtc:163
NOTRY=True

Review-Url: https://codereview.webrtc.org/2219653004
Cr-Commit-Position: refs/heads/master@{#13759}
This commit is contained in:
maxmorin
2016-08-15 11:46:19 -07:00
committed by Commit bot
parent c8c71f484e
commit 1aee0b5bd9
10 changed files with 108 additions and 112 deletions
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15
webrtc/api/test/fakeaudiocapturemodule_unittest.cc
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@ -62,6 +62,21 @@ class FakeAdmTest : public testing::Test,
return 0;
}
void PushCaptureData(int voe_channel,
const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames) override {}
void PullRenderData(int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
void* audio_data,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) override {}
// ADM is pulling data.
int32_t NeedMorePlayData(const size_t nSamples,
const size_t nBytesPerSample,

9
webrtc/media/engine/webrtcvoiceengine.cc
View File

@ -1218,12 +1218,9 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
RTC_DCHECK(!worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(audio_capture_thread_checker_.CalledOnValidThread());
RTC_DCHECK(voe_audio_transport_);
voe_audio_transport_->OnData(config_.voe_channel_id,
audio_data,
bits_per_sample,
sample_rate,
number_of_channels,
number_of_frames);
voe_audio_transport_->PushCaptureData(config_.voe_channel_id, audio_data,
bits_per_sample, sample_rate,
number_of_channels, number_of_frames);
}
// Callback from the |source_| when it is going away. In case Start() has

18
webrtc/modules/audio_device/android/audio_device_unittest.cc
View File

@ -390,6 +390,7 @@ class MockAudioTransport : public AudioTransport {
const uint32_t currentMicLevel,
const bool keyPressed,
uint32_t& newMicLevel));
MOCK_METHOD8(NeedMorePlayData,
int32_t(const size_t nSamples,
const size_t nBytesPerSample,
@ -400,6 +401,23 @@ class MockAudioTransport : public AudioTransport {
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms));
MOCK_METHOD6(PushCaptureData,
void(int voe_channel,
const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames));
MOCK_METHOD7(PullRenderData,
void(int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
void* audio_data,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms));
// Set default actions of the mock object. We are delegating to fake
// implementations (of AudioStreamInterface) here.
void HandleCallbacks(EventWrapper* test_is_done,

46
webrtc/modules/audio_device/include/audio_device_defines.h
View File

@ -66,58 +66,16 @@ class AudioTransport {
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) = 0;
// Method to pass captured data directly and unmixed to network channels.
// |channel_ids| contains a list of VoE channels which are the
// sinks to the capture data. |audio_delay_milliseconds| is the sum of
// recording delay and playout delay of the hardware. |current_volume| is
// in the range of [0, 255], representing the current microphone analog
// volume. |key_pressed| is used by the typing detection.
// |need_audio_processing| specify if the data needs to be processed by APM.
// Currently WebRtc supports only one APM, and Chrome will make sure only
// one stream goes through APM. When |need_audio_processing| is false, the
// values of |audio_delay_milliseconds|, |current_volume| and |key_pressed|
// will be ignored.
// The return value is the new microphone volume, in the range of |0, 255].
// When the volume does not need to be updated, it returns 0.
// TODO(xians): Remove this interface after Chrome and Libjingle switches
// to OnData().
virtual int OnDataAvailable(const int voe_channels[],
size_t number_of_voe_channels,
const int16_t* audio_data,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
int audio_delay_milliseconds,
int current_volume,
bool key_pressed,
bool need_audio_processing) {
return 0;
}
// Method to pass the captured audio data to the specific VoE channel.
// |voe_channel| is the id of the VoE channel which is the sink to the
// capture data.
// TODO(xians): Remove this interface after Libjingle switches to
// PushCaptureData().
virtual void OnData(int voe_channel,
const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames) {}
// Method to push the captured audio data to the specific VoE channel.
// The data will not undergo audio processing.
// |voe_channel| is the id of the VoE channel which is the sink to the
// capture data.
// TODO(xians): Make the interface pure virtual after Libjingle
// has its implementation.
virtual void PushCaptureData(int voe_channel,
const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames) {}
size_t number_of_frames) = 0;
// Method to pull mixed render audio data from all active VoE channels.
// The data will not be passed as reference for audio processing internally.
@ -129,7 +87,7 @@ class AudioTransport {
size_t number_of_frames,
void* audio_data,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) {}
int64_t* ntp_time_ms) = 0;
protected:
virtual ~AudioTransport() {}

18
webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc
View File

@ -380,6 +380,7 @@ class MockAudioTransport : public AudioTransport {
const uint32_t currentMicLevel,
const bool keyPressed,
uint32_t& newMicLevel));
MOCK_METHOD8(NeedMorePlayData,
int32_t(const size_t nSamples,
const size_t nBytesPerSample,
@ -390,6 +391,23 @@ class MockAudioTransport : public AudioTransport {
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms));
MOCK_METHOD6(PushCaptureData,
void(int voe_channel,
const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames));
MOCK_METHOD7(PullRenderData,
void(int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
void* audio_data,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms));
// Set default actions of the mock object. We are delegating to fake
// implementations (of AudioStreamInterface) here.
void HandleCallbacks(EventWrapper* test_is_done,

31
webrtc/modules/audio_device/test/audio_device_test_api.cc
View File

@ -54,14 +54,14 @@ class AudioEventObserverAPI: public AudioDeviceObserver {
warning_(kRecordingWarning),
audio_device_(audioDevice) {}
~AudioEventObserverAPI() {}
~AudioEventObserverAPI() override {}
virtual void OnErrorIsReported(const ErrorCode error) {
void OnErrorIsReported(const ErrorCode error) override {
TEST_LOG("\n[*** ERROR ***] => OnErrorIsReported(%d)\n\n", error);
error_ = error;
}
virtual void OnWarningIsReported(const WarningCode warning) {
void OnWarningIsReported(const WarningCode warning) override {
TEST_LOG("\n[*** WARNING ***] => OnWarningIsReported(%d)\n\n", warning);
warning_ = warning;
EXPECT_EQ(0, audio_device_->StopRecording());
@ -82,7 +82,7 @@ class AudioTransportAPI: public AudioTransport {
play_count_(0) {
}
~AudioTransportAPI() {}
~AudioTransportAPI() override {}
int32_t RecordedDataIsAvailable(const void* audioSamples,
const size_t nSamples,
@ -130,6 +130,21 @@ class AudioTransportAPI: public AudioTransport {
return 0;
}
void PushCaptureData(int voe_channel,
const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames) override {}
void PullRenderData(int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
void* audio_data,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) override {}
private:
uint32_t rec_count_;
uint32_t play_count_;
@ -139,7 +154,7 @@ class AudioDeviceAPITest: public testing::Test {
protected:
AudioDeviceAPITest() {}
virtual ~AudioDeviceAPITest() {}
~AudioDeviceAPITest() override {}
static void SetUpTestCase() {
process_thread_ = ProcessThread::Create("ProcessThread");
@ -258,7 +273,7 @@ class AudioDeviceAPITest: public testing::Test {
PRINT_TEST_RESULTS;
}
void SetUp() {
void SetUp() override {
if (linux_alsa_) {
FAIL() << "API Test is not available on ALSA on Linux!";
}
@ -266,9 +281,7 @@ class AudioDeviceAPITest: public testing::Test {
EXPECT_TRUE(audio_device_->Initialized());
}
void TearDown() {
EXPECT_EQ(0, audio_device_->Terminate());
}
void TearDown() override { EXPECT_EQ(0, audio_device_->Terminate()); }
void CheckVolume(uint32_t expected, uint32_t actual) {
// Mac and Windows have lower resolution on the volume settings.

15
webrtc/modules/audio_device/test/func_test_manager.cc
View File

@ -563,6 +563,21 @@ int32_t AudioTransportImpl::NeedMorePlayData(
return 0;
}
void AudioTransportImpl::PushCaptureData(int voe_channel,
const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames) {}
void AudioTransportImpl::PullRenderData(int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
void* audio_data,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) {}
FuncTestManager::FuncTestManager() :
_audioDevice(NULL),
_audioEventObserver(NULL),

15
webrtc/modules/audio_device/test/func_test_manager.h
View File

@ -106,6 +106,21 @@ public:
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) override;
void PushCaptureData(int voe_channel,
const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames) override;
void PullRenderData(int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
void* audio_data,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) override;
AudioTransportImpl(AudioDeviceModule* audioDevice);
~AudioTransportImpl();

37
webrtc/voice_engine/voe_base_impl.cc
View File

@ -107,43 +107,6 @@ int32_t VoEBaseImpl::NeedMorePlayData(const size_t nSamples,
return 0;
}
int VoEBaseImpl::OnDataAvailable(const int voe_channels[],
size_t number_of_voe_channels,
const int16_t* audio_data, int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
int audio_delay_milliseconds, int volume,
bool key_pressed, bool need_audio_processing) {
if (number_of_voe_channels == 0) return 0;
if (need_audio_processing) {
return ProcessRecordedDataWithAPM(
voe_channels, number_of_voe_channels, audio_data, sample_rate,
number_of_channels, number_of_frames, audio_delay_milliseconds, 0,
volume, key_pressed);
}
// No need to go through the APM, demultiplex the data to each VoE channel,
// encode and send to the network.
for (size_t i = 0; i < number_of_voe_channels; ++i) {
// TODO(ajm): In the case where multiple channels are using the same codec
// rate, this path needlessly does extra conversions. We should convert once
// and share between channels.
PushCaptureData(voe_channels[i], audio_data, 16, sample_rate,
number_of_channels, number_of_frames);
}
// Return 0 to indicate no need to change the volume.
return 0;
}
void VoEBaseImpl::OnData(int voe_channel, const void* audio_data,
int bits_per_sample, int sample_rate,
size_t number_of_channels, size_t number_of_frames) {
PushCaptureData(voe_channel, audio_data, bits_per_sample, sample_rate,
number_of_channels, number_of_frames);
}
void VoEBaseImpl::PushCaptureData(int voe_channel, const void* audio_data,
int bits_per_sample, int sample_rate,
size_t number_of_channels,

16
webrtc/voice_engine/voe_base_impl.h
View File

@ -78,22 +78,6 @@ class VoEBaseImpl : public VoEBase,
size_t& nSamplesOut,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) override;
int OnDataAvailable(const int voe_channels[],
size_t number_of_voe_channels,
const int16_t* audio_data,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
int audio_delay_milliseconds,
int current_volume,
bool key_pressed,
bool need_audio_processing) override;
void OnData(int voe_channel,
const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames) override;
void PushCaptureData(int voe_channel,
const void* audio_data,
int bits_per_sample,