Adds trunk/talk folder of revision 359 from libjingles google code to
trunk/talk
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc b/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc
new file mode 100644
index 0000000..5738955
--- /dev/null
+++ b/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc
@@ -0,0 +1,212 @@
+/*
+ * libjingle
+ * Copyright 2012, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "talk/app/webrtc/test/fakeaudiocapturemodule.h"
+
+#include <algorithm>
+
+#include "talk/base/gunit.h"
+#include "talk/base/scoped_ref_ptr.h"
+#include "talk/base/thread.h"
+
+using std::min;
+
+class FakeAdmTest : public testing::Test,
+ public webrtc::AudioTransport {
+ protected:
+ static const int kMsInSecond = 1000;
+
+ FakeAdmTest()
+ : push_iterations_(0),
+ pull_iterations_(0),
+ rec_buffer_bytes_(0) {
+ memset(rec_buffer_, 0, sizeof(rec_buffer_));
+ }
+
+ virtual void SetUp() {
+ fake_audio_capture_module_ = FakeAudioCaptureModule::Create(
+ talk_base::Thread::Current());
+ EXPECT_TRUE(fake_audio_capture_module_.get() != NULL);
+ }
+
+ // Callbacks inherited from webrtc::AudioTransport.
+ // ADM is pushing data.
+ virtual int32_t RecordedDataIsAvailable(const void* audioSamples,
+ const uint32_t nSamples,
+ const uint8_t nBytesPerSample,
+ const uint8_t nChannels,
+ const uint32_t samplesPerSec,
+ const uint32_t totalDelayMS,
+ const int32_t clockDrift,
+ const uint32_t currentMicLevel,
+ const bool keyPressed,
+ uint32_t& newMicLevel) {
+ rec_buffer_bytes_ = nSamples * nBytesPerSample;
+ if ((rec_buffer_bytes_ <= 0) ||
+ (rec_buffer_bytes_ > FakeAudioCaptureModule::kNumberSamples *
+ FakeAudioCaptureModule::kNumberBytesPerSample)) {
+ ADD_FAILURE();
+ return -1;
+ }
+ memcpy(rec_buffer_, audioSamples, rec_buffer_bytes_);
+ ++push_iterations_;
+ newMicLevel = currentMicLevel;
+ return 0;
+ }
+
+ // ADM is pulling data.
+ virtual int32_t NeedMorePlayData(const uint32_t nSamples,
+ const uint8_t nBytesPerSample,
+ const uint8_t nChannels,
+ const uint32_t samplesPerSec,
+ void* audioSamples,
+ uint32_t& nSamplesOut) {
+ ++pull_iterations_;
+ const uint32_t audio_buffer_size = nSamples * nBytesPerSample;
+ const uint32_t bytes_out = RecordedDataReceived() ?
+ CopyFromRecBuffer(audioSamples, audio_buffer_size):
+ GenerateZeroBuffer(audioSamples, audio_buffer_size);
+ nSamplesOut = bytes_out / nBytesPerSample;
+ return 0;
+ }
+
+ int push_iterations() const { return push_iterations_; }
+ int pull_iterations() const { return pull_iterations_; }
+
+ talk_base::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
+
+ private:
+ bool RecordedDataReceived() const {
+ return rec_buffer_bytes_ != 0;
+ }
+ int32_t GenerateZeroBuffer(void* audio_buffer, uint32_t audio_buffer_size) {
+ memset(audio_buffer, 0, audio_buffer_size);
+ return audio_buffer_size;
+ }
+ int32_t CopyFromRecBuffer(void* audio_buffer, uint32_t audio_buffer_size) {
+ EXPECT_EQ(audio_buffer_size, rec_buffer_bytes_);
+ const uint32_t min_buffer_size = min(audio_buffer_size, rec_buffer_bytes_);
+ memcpy(audio_buffer, rec_buffer_, min_buffer_size);
+ return min_buffer_size;
+ }
+
+ int push_iterations_;
+ int pull_iterations_;
+
+ char rec_buffer_[FakeAudioCaptureModule::kNumberSamples *
+ FakeAudioCaptureModule::kNumberBytesPerSample];
+ uint32_t rec_buffer_bytes_;
+};
+
+TEST_F(FakeAdmTest, TestProccess) {
+ // Next process call must be some time in the future (or now).
+ EXPECT_LE(0, fake_audio_capture_module_->TimeUntilNextProcess());
+ // Process call updates TimeUntilNextProcess() but there are no guarantees on
+ // timing so just check that Process can ba called successfully.
+ EXPECT_LE(0, fake_audio_capture_module_->Process());
+}
+
+TEST_F(FakeAdmTest, PlayoutTest) {
+ EXPECT_EQ(0, fake_audio_capture_module_->RegisterAudioCallback(this));
+
+ bool speaker_available = false;
+ EXPECT_EQ(0, fake_audio_capture_module_->SpeakerIsAvailable(
+ &speaker_available));
+ EXPECT_TRUE(speaker_available);
+
+ bool stereo_available = false;
+ EXPECT_EQ(0,
+ fake_audio_capture_module_->StereoPlayoutIsAvailable(
+ &stereo_available));
+ EXPECT_TRUE(stereo_available);
+
+ EXPECT_NE(0, fake_audio_capture_module_->StartPlayout());
+ EXPECT_FALSE(fake_audio_capture_module_->PlayoutIsInitialized());
+ EXPECT_FALSE(fake_audio_capture_module_->Playing());
+ EXPECT_EQ(0, fake_audio_capture_module_->StopPlayout());
+
+ EXPECT_EQ(0, fake_audio_capture_module_->InitPlayout());
+ EXPECT_TRUE(fake_audio_capture_module_->PlayoutIsInitialized());
+ EXPECT_FALSE(fake_audio_capture_module_->Playing());
+
+ EXPECT_EQ(0, fake_audio_capture_module_->StartPlayout());
+ EXPECT_TRUE(fake_audio_capture_module_->Playing());
+
+ uint16_t delay_ms = 10;
+ EXPECT_EQ(0, fake_audio_capture_module_->PlayoutDelay(&delay_ms));
+ EXPECT_EQ(0, delay_ms);
+
+ EXPECT_TRUE_WAIT(pull_iterations() > 0, kMsInSecond);
+ EXPECT_GE(0, push_iterations());
+
+ EXPECT_EQ(0, fake_audio_capture_module_->StopPlayout());
+ EXPECT_FALSE(fake_audio_capture_module_->Playing());
+}
+
+TEST_F(FakeAdmTest, RecordTest) {
+ EXPECT_EQ(0, fake_audio_capture_module_->RegisterAudioCallback(this));
+
+ bool microphone_available = false;
+ EXPECT_EQ(0, fake_audio_capture_module_->MicrophoneIsAvailable(
+ µphone_available));
+ EXPECT_TRUE(microphone_available);
+
+ bool stereo_available = false;
+ EXPECT_EQ(0, fake_audio_capture_module_->StereoRecordingIsAvailable(
+ &stereo_available));
+ EXPECT_FALSE(stereo_available);
+
+ EXPECT_NE(0, fake_audio_capture_module_->StartRecording());
+ EXPECT_FALSE(fake_audio_capture_module_->Recording());
+ EXPECT_EQ(0, fake_audio_capture_module_->StopRecording());
+
+ EXPECT_EQ(0, fake_audio_capture_module_->InitRecording());
+ EXPECT_EQ(0, fake_audio_capture_module_->StartRecording());
+ EXPECT_TRUE(fake_audio_capture_module_->Recording());
+
+ EXPECT_TRUE_WAIT(push_iterations() > 0, kMsInSecond);
+ EXPECT_GE(0, pull_iterations());
+
+ EXPECT_EQ(0, fake_audio_capture_module_->StopRecording());
+ EXPECT_FALSE(fake_audio_capture_module_->Recording());
+}
+
+TEST_F(FakeAdmTest, DuplexTest) {
+ EXPECT_EQ(0, fake_audio_capture_module_->RegisterAudioCallback(this));
+
+ EXPECT_EQ(0, fake_audio_capture_module_->InitPlayout());
+ EXPECT_EQ(0, fake_audio_capture_module_->StartPlayout());
+
+ EXPECT_EQ(0, fake_audio_capture_module_->InitRecording());
+ EXPECT_EQ(0, fake_audio_capture_module_->StartRecording());
+
+ EXPECT_TRUE_WAIT(push_iterations() > 0, kMsInSecond);
+ EXPECT_TRUE_WAIT(pull_iterations() > 0, kMsInSecond);
+
+ EXPECT_EQ(0, fake_audio_capture_module_->StopPlayout());
+ EXPECT_EQ(0, fake_audio_capture_module_->StopRecording());
+}
