summaryrefslogtreecommitdiffstats
path: root/media
diff options
context:
space:
mode:
Diffstat (limited to 'media')
-rw-r--r--media/audio/simple_sources.cc18
-rw-r--r--media/audio/simple_sources.h1
-rw-r--r--media/audio/win/audio_output_win_unittest.cc59
3 files changed, 28 insertions, 50 deletions
diff --git a/media/audio/simple_sources.cc b/media/audio/simple_sources.cc
index 4cd0ba3..4078315 100644
--- a/media/audio/simple_sources.cc
+++ b/media/audio/simple_sources.cc
@@ -21,8 +21,7 @@ SineWaveAudioSource::SineWaveAudioSource(Format format, int channels,
: format_(format),
channels_(channels),
freq_(freq),
- sample_freq_(sample_freq),
- time_state_(0) {
+ sample_freq_(sample_freq) {
// TODO(cpu): support other formats.
DCHECK((format_ == FORMAT_16BIT_LINEAR_PCM) && (channels_ == 1));
}
@@ -32,19 +31,14 @@ SineWaveAudioSource::SineWaveAudioSource(Format format, int channels,
uint32 SineWaveAudioSource::OnMoreData(
AudioOutputStream* stream, uint8* dest, uint32 max_size,
AudioBuffersState audio_buffers) {
- const double kTwoPi = 2.0 * 3.141592653589793;
+ const double kTwoPi = 2.0 * 3.141592653589;
double f = freq_ / sample_freq_;
int16* sin_tbl = reinterpret_cast<int16*>(dest);
uint32 len = max_size / sizeof(int16);
-
- // The table is filled with s(t) = kint16max*sin(Theta*t),
- // where Theta = 2*PI*fs.
- // We store the discrete time value |t| in a member to ensure that the
- // next pass starts at a correct state.
- for (uint32 n = 0; n < len; ++n) {
- double theta = kTwoPi * f;
- sin_tbl[n] = static_cast<int16>(kint16max * sin(theta * time_state_));
- ++time_state_;
+ // The table is filled with s(t) = 32768*sin(2PI*f*t).
+ for (uint32 ix = 0; ix != len; ++ix) {
+ double th = kTwoPi * ix * f;
+ sin_tbl[ix] = static_cast<int16>((1 << 15) * sin(th));
}
return max_size;
}
diff --git a/media/audio/simple_sources.h b/media/audio/simple_sources.h
index a4373a0..dd2782c 100644
--- a/media/audio/simple_sources.h
+++ b/media/audio/simple_sources.h
@@ -36,7 +36,6 @@ class MEDIA_EXPORT SineWaveAudioSource
int channels_;
double freq_;
double sample_freq_;
- int time_state_;
};
// Defines an interface for pushing audio output. In contrast, the interfaces
diff --git a/media/audio/win/audio_output_win_unittest.cc b/media/audio/win/audio_output_win_unittest.cc
index d2447c5..4fcc5c8 100644
--- a/media/audio/win/audio_output_win_unittest.cc
+++ b/media/audio/win/audio_output_win_unittest.cc
@@ -10,11 +10,7 @@
#include "base/file_util.h"
#include "base/path_service.h"
#include "base/sync_socket.h"
-#include "base/win/scoped_com_initializer.h"
-#include "base/win/windows_version.h"
-#include "media/base/limits.h"
#include "media/audio/audio_io.h"
-#include "media/audio/audio_util.h"
#include "media/audio/audio_manager.h"
#include "media/audio/simple_sources.h"
#include "testing/gmock/include/gmock/gmock.h"
@@ -29,8 +25,6 @@ using ::testing::NiceMock;
using ::testing::NotNull;
using ::testing::Return;
-using base::win::ScopedCOMInitializer;
-
static const wchar_t kAudioFile1_16b_m_16K[]
= L"media\\test\\data\\sweep02_16b_mono_16KHz.raw";
@@ -237,7 +231,7 @@ TEST(WinAudioTest, PCMWaveStreamGetAndClose) {
oas->Close();
}
-// Test that can it be cannot be created with invalid parameters.
+// Test that can it be cannot be created with crazy parameters
TEST(WinAudioTest, SanityOnMakeParams) {
if (IsRunningHeadless())
return;
@@ -261,8 +255,7 @@ TEST(WinAudioTest, SanityOnMakeParams) {
EXPECT_TRUE(NULL == audio_man->MakeAudioOutputStream(
AudioParameters(fmt, CHANNEL_LAYOUT_MONO, 8000, 16, 0)));
EXPECT_TRUE(NULL == audio_man->MakeAudioOutputStream(
- AudioParameters(fmt, CHANNEL_LAYOUT_MONO, 8000, 16,
- media::Limits::kMaxSamplesPerPacket + 1)));
+ AudioParameters(fmt, CHANNEL_LAYOUT_MONO, 8000, 16, 100000)));
}
// Test that it can be opened and closed.
@@ -540,10 +533,11 @@ TEST(WinAudioTest, PCMWaveStreamPlayTwice200HzTone44Kss) {
oas->Close();
}
-// With the low latency mode, WASAPI is utilized by default for Vista and
-// higher and Wave is used for XP and lower. It is possible to utilize a
-// smaller buffer size for WASAPI than for Wave.
-TEST(WinAudioTest, PCMWaveStreamPlay200HzToneLowLatency) {
+// With the low latency mode, we have two buffers instead of 3 and we
+// should be able to handle 20ms buffers at 44KHz. See also the SyncSocketBasic
+// test below.
+// TODO(cpu): right now the best we can do is 50ms before it sounds choppy.
+TEST(WinAudioTest, PCMWaveStreamPlay200HzTone44KssLowLatency) {
if (IsRunningHeadless())
return;
AudioManager* audio_man = AudioManager::GetAudioManager();
@@ -551,23 +545,15 @@ TEST(WinAudioTest, PCMWaveStreamPlay200HzToneLowLatency) {
if (!audio_man->HasAudioOutputDevices())
return;
- // The WASAPI API requires a correct COM environment.
- ScopedCOMInitializer com_init(ScopedCOMInitializer::kMTA);
-
- // Use 10 ms buffer size for WASAPI and 50 ms buffer size for Wave.
- // Take the existing native sample rate into account.
- int sample_rate = static_cast<int>(media::GetAudioHardwareSampleRate());
- uint32 samples_10_ms = sample_rate / 100;
- int n = 1;
- (base::win::GetVersion() <= base::win::VERSION_XP) ? n = 5 : n = 1;
+ uint32 samples_50_ms = AudioParameters::kAudioCDSampleRate / 20;
AudioOutputStream* oas = audio_man->MakeAudioOutputStream(
AudioParameters(AudioParameters::AUDIO_PCM_LOW_LATENCY,
- CHANNEL_LAYOUT_MONO, sample_rate,
- 16, n * samples_10_ms));
+ CHANNEL_LAYOUT_MONO, AudioParameters::kAudioCDSampleRate,
+ 16, samples_50_ms));
ASSERT_TRUE(NULL != oas);
SineWaveAudioSource source(SineWaveAudioSource::FORMAT_16BIT_LINEAR_PCM, 1,
- 200.0, sample_rate);
+ 200.0, AudioParameters::kAudioCDSampleRate);
EXPECT_TRUE(oas->Open());
oas->SetVolume(1.0);
@@ -665,7 +651,7 @@ struct SyncThreadContext {
base::SyncSocket* socket;
int sample_rate;
double sine_freq;
- uint32 packet_size_bytes;
+ uint32 packet_size;
};
// This thread provides the data that the SyncSocketSource above needs
@@ -686,11 +672,11 @@ DWORD __stdcall SyncSocketThread(void* context) {
AudioBuffersState buffers_state;
int times = 0;
- for (int ix = 0; ix < kTwoSecBytes; ix += ctx.packet_size_bytes) {
+ for (int ix = 0; ix < kTwoSecBytes; ix += ctx.packet_size) {
if (ctx.socket->Receive(&buffers_state, sizeof(buffers_state)) == 0)
break;
if ((times > 0) && (buffers_state.pending_bytes < 1000)) __debugbreak();
- ctx.socket->Send(&buffer[ix], ctx.packet_size_bytes);
+ ctx.socket->Send(&buffer[ix], ctx.packet_size);
++times;
}
@@ -699,13 +685,11 @@ DWORD __stdcall SyncSocketThread(void* context) {
}
// Test the basic operation of AudioOutputStream used with a SyncSocket.
-// The emphasis is to verify that it is possible to feed data to the audio
-// layer using a source based on SyncSocket. In a real situation we would
-// go for the low-latency version in combination with SyncSocket, but to keep
-// the test more simple, AUDIO_PCM_LINEAR is utilized instead. The main
-// principle of the test still remains and we avoid the additional complexity
-// related to the two different audio-layers for AUDIO_PCM_LOW_LATENCY.
-// In this test you should hear a continuous 200Hz tone for 2 seconds.
+// The emphasis is to test low-latency with buffers less than 100ms. With
+// the waveout api it seems not possible to go below 50ms. In this test
+// you should hear a continous 200Hz tone.
+//
+// TODO(cpu): This actually sounds choppy most of the time. Fix it.
TEST(WinAudioTest, SyncSocketBasic) {
if (IsRunningHeadless())
return;
@@ -718,10 +702,11 @@ TEST(WinAudioTest, SyncSocketBasic) {
int sample_rate = AudioParameters::kAudioCDSampleRate;
const uint32 kSamples20ms = sample_rate / 50;
AudioOutputStream* oas = audio_man->MakeAudioOutputStream(
- AudioParameters(AudioParameters::AUDIO_PCM_LINEAR,
+ AudioParameters(AudioParameters::AUDIO_PCM_LOW_LATENCY,
CHANNEL_LAYOUT_MONO, sample_rate, 16, kSamples20ms));
ASSERT_TRUE(NULL != oas);
+ // compute buffer size for 20ms of audio, 882 samples (mono).
ASSERT_TRUE(oas->Open());
base::SyncSocket* sockets[2];
@@ -732,7 +717,7 @@ TEST(WinAudioTest, SyncSocketBasic) {
SyncThreadContext thread_context;
thread_context.sample_rate = sample_rate;
thread_context.sine_freq = 200.0;
- thread_context.packet_size_bytes = kSamples20ms * 2;
+ thread_context.packet_size = kSamples20ms;
thread_context.socket = sockets[1];
HANDLE thread = ::CreateThread(NULL, 0, SyncSocketThread,
generated by cgit v1.1 at 2024-09-23 00:15:25 +0000