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Unified Diff: trunk/src/media/audio/mac/audio_low_latency_input_mac_unittest.cc

Issue 335343004: Revert 277794 "Modifies AudioInputCallback::OnData and use media..." (Closed) Base URL: svn://svn.chromium.org/chrome/
Patch Set: Created 6 years, 6 months ago
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Index: trunk/src/media/audio/mac/audio_low_latency_input_mac_unittest.cc
===================================================================
--- trunk/src/media/audio/mac/audio_low_latency_input_mac_unittest.cc (revision 277811)
+++ trunk/src/media/audio/mac/audio_low_latency_input_mac_unittest.cc (working copy)
@@ -31,11 +31,9 @@
class MockAudioInputCallback : public AudioInputStream::AudioInputCallback {
public:
- MOCK_METHOD4(OnData,
- void(AudioInputStream* stream,
- const AudioBus* src,
- uint32 hardware_delay_bytes,
- double volume));
+ MOCK_METHOD5(OnData, void(AudioInputStream* stream,
+ const uint8* src, uint32 size,
+ uint32 hardware_delay_bytes, double volume));
MOCK_METHOD1(OnError, void(AudioInputStream* stream));
};
@@ -76,19 +74,12 @@
// AudioInputStream::AudioInputCallback implementation.
virtual void OnData(AudioInputStream* stream,
- const AudioBus* src,
- uint32 hardware_delay_bytes,
- double volume) OVERRIDE {
- const int num_samples = src->frames() * src->channels();
- scoped_ptr<int16> interleaved(new int16[num_samples]);
- const int bytes_per_sample = sizeof(*interleaved);
- src->ToInterleaved(src->frames(), bytes_per_sample, interleaved.get());
-
+ const uint8* src, uint32 size,
+ uint32 hardware_delay_bytes, double volume) OVERRIDE {
// Store data data in a temporary buffer to avoid making blocking
// fwrite() calls in the audio callback. The complete buffer will be
// written to file in the destructor.
- const int size = bytes_per_sample * num_samples;
- if (buffer_.Append((const uint8*)interleaved.get(), size)) {
+ if (buffer_.Append(src, size)) {
bytes_to_write_ += size;
}
}
@@ -233,13 +224,18 @@
AudioInputStream* ais = CreateAudioInputStream(CHANNEL_LAYOUT_MONO);
EXPECT_TRUE(ais->Open());
+ int fs = static_cast<int>(AUAudioInputStream::HardwareSampleRate());
+ int samples_per_packet = fs / 100;
+ int bits_per_sample = 16;
+ uint32 bytes_per_packet = samples_per_packet * (bits_per_sample / 8);
+
MockAudioInputCallback sink;
// We use 10ms packets and will run the test until ten packets are received.
// All should contain valid packets of the same size and a valid delay
// estimate.
base::RunLoop run_loop;
- EXPECT_CALL(sink, OnData(ais, NotNull(), _, _))
+ EXPECT_CALL(sink, OnData(ais, NotNull(), bytes_per_packet, _, _))
.Times(AtLeast(10))
.WillRepeatedly(CheckCountAndPostQuitTask(
&count, 10, &message_loop_, run_loop.QuitClosure()));
@@ -260,6 +256,11 @@
AudioInputStream* ais = CreateAudioInputStream(CHANNEL_LAYOUT_STEREO);
EXPECT_TRUE(ais->Open());
+ int fs = static_cast<int>(AUAudioInputStream::HardwareSampleRate());
+ int samples_per_packet = fs / 100;
+ int bits_per_sample = 16;
+ uint32 bytes_per_packet = 2 * samples_per_packet * (bits_per_sample / 8);
+
MockAudioInputCallback sink;
// We use 10ms packets and will run the test until ten packets are received.
@@ -273,7 +274,7 @@
// ensure that we can land the patch but will revisit this test again when
// more analysis of the delay estimates are done.
base::RunLoop run_loop;
- EXPECT_CALL(sink, OnData(ais, NotNull(), _, _))
+ EXPECT_CALL(sink, OnData(ais, NotNull(), bytes_per_packet, _, _))
.Times(AtLeast(10))
.WillRepeatedly(CheckCountAndPostQuitTask(
&count, 10, &message_loop_, run_loop.QuitClosure()));
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