summaryrefslogtreecommitdiffstats
path: root/media/audio/win
diff options
context:
space:
mode:
Diffstat (limited to 'media/audio/win')
-rw-r--r--media/audio/win/audio_low_latency_output_win.cc6
-rw-r--r--media/audio/win/audio_low_latency_output_win_unittest.cc27
-rw-r--r--media/audio/win/audio_output_win_unittest.cc44
-rw-r--r--media/audio/win/waveout_output_win.cc5
4 files changed, 30 insertions, 52 deletions
diff --git a/media/audio/win/audio_low_latency_output_win.cc b/media/audio/win/audio_low_latency_output_win.cc
index 15cabbaf..99576a6 100644
--- a/media/audio/win/audio_low_latency_output_win.cc
+++ b/media/audio/win/audio_low_latency_output_win.cc
@@ -498,13 +498,9 @@ bool WASAPIAudioOutputStream::RenderAudioFromSource(UINT64 device_frequency) {
// Read a data packet from the registered client source and
// deliver a delay estimate in the same callback to the client.
- // A time stamp is also stored in the AudioBuffersState. This
- // time stamp can be used at the client side to compensate for
- // the delay between the usage of the delay value and the time
- // of generation.
int frames_filled = source_->OnMoreData(
- audio_bus_.get(), AudioBuffersState(0, audio_delay_bytes));
+ audio_bus_.get(), audio_delay_bytes);
uint32 num_filled_bytes = frames_filled * format_.Format.nBlockAlign;
DCHECK_LE(num_filled_bytes, packet_size_bytes_);
diff --git a/media/audio/win/audio_low_latency_output_win_unittest.cc b/media/audio/win/audio_low_latency_output_win_unittest.cc
index ed03d2b..30fa9ca 100644
--- a/media/audio/win/audio_low_latency_output_win_unittest.cc
+++ b/media/audio/win/audio_low_latency_output_win_unittest.cc
@@ -52,7 +52,7 @@ MATCHER_P(HasValidDelay, value, "") {
// It is difficult to come up with a perfect test condition for the delay
// estimation. For now, verify that the produced output delay is always
// larger than the selected buffer size.
- return arg.hardware_delay_bytes >= value.hardware_delay_bytes;
+ return arg >= value;
}
// Used to terminate a loop from a different thread than the loop belongs to.
@@ -103,7 +103,7 @@ class ReadFromFileAudioSource : public AudioOutputStream::AudioSourceCallback {
// AudioOutputStream::AudioSourceCallback implementation.
virtual int OnMoreData(AudioBus* audio_bus,
- AudioBuffersState buffers_state) {
+ int total_bytes_delay) {
// Store time difference between two successive callbacks in an array.
// These values will be written to a file in the destructor.
const base::TimeTicks now_time = base::TimeTicks::Now();
@@ -396,14 +396,11 @@ TEST(WASAPIAudioOutputStreamTest, ValidPacketSize) {
EXPECT_TRUE(aos->Open());
// Derive the expected size in bytes of each packet.
- uint32 bytes_per_packet = aosw.channels() * aosw.samples_per_packet() *
- (aosw.bits_per_sample() / 8);
-
- // Set up expected minimum delay estimation.
- AudioBuffersState state(0, bytes_per_packet);
+ int bytes_per_packet = aosw.channels() * aosw.samples_per_packet() *
+ (aosw.bits_per_sample() / 8);
// Wait for the first callback and verify its parameters.
- EXPECT_CALL(source, OnMoreData(NotNull(), HasValidDelay(state)))
+ EXPECT_CALL(source, OnMoreData(NotNull(), HasValidDelay(bytes_per_packet)))
.WillOnce(DoAll(
QuitLoop(loop.message_loop_proxy()),
Return(aosw.samples_per_packet())));
@@ -600,14 +597,11 @@ TEST(WASAPIAudioOutputStreamTest, ExclusiveModeMinBufferSizeAt48kHz) {
EXPECT_TRUE(aos->Open());
// Derive the expected size in bytes of each packet.
- uint32 bytes_per_packet = aosw.channels() * aosw.samples_per_packet() *
+ int bytes_per_packet = aosw.channels() * aosw.samples_per_packet() *
(aosw.bits_per_sample() / 8);
- // Set up expected minimum delay estimation.
- AudioBuffersState state(0, bytes_per_packet);
-
// Wait for the first callback and verify its parameters.
- EXPECT_CALL(source, OnMoreData(NotNull(), HasValidDelay(state)))
+ EXPECT_CALL(source, OnMoreData(NotNull(), HasValidDelay(bytes_per_packet)))
.WillOnce(DoAll(
QuitLoop(loop.message_loop_proxy()),
Return(aosw.samples_per_packet())))
@@ -641,14 +635,11 @@ TEST(WASAPIAudioOutputStreamTest, ExclusiveModeMinBufferSizeAt44kHz) {
EXPECT_TRUE(aos->Open());
// Derive the expected size in bytes of each packet.
- uint32 bytes_per_packet = aosw.channels() * aosw.samples_per_packet() *
+ int bytes_per_packet = aosw.channels() * aosw.samples_per_packet() *
(aosw.bits_per_sample() / 8);
- // Set up expected minimum delay estimation.
- AudioBuffersState state(0, bytes_per_packet);
-
// Wait for the first callback and verify its parameters.
- EXPECT_CALL(source, OnMoreData(NotNull(), HasValidDelay(state)))
+ EXPECT_CALL(source, OnMoreData(NotNull(), HasValidDelay(bytes_per_packet)))
.WillOnce(DoAll(
QuitLoop(loop.message_loop_proxy()),
Return(aosw.samples_per_packet())))
diff --git a/media/audio/win/audio_output_win_unittest.cc b/media/audio/win/audio_output_win_unittest.cc
index 2902657..6257a60 100644
--- a/media/audio/win/audio_output_win_unittest.cc
+++ b/media/audio/win/audio_output_win_unittest.cc
@@ -37,7 +37,7 @@ namespace media {
static const wchar_t kAudioFile1_16b_m_16K[]
= L"media\\test\\data\\sweep02_16b_mono_16KHz.raw";
-static int ClearData(AudioBus* audio_bus, AudioBuffersState buffers_state) {
+static int ClearData(AudioBus* audio_bus, int total_bytes_delay) {
audio_bus->Zero();
return audio_bus->frames();
}
@@ -52,7 +52,7 @@ class TestSourceBasic : public AudioOutputStream::AudioSourceCallback {
}
// AudioSourceCallback::OnMoreData implementation:
virtual int OnMoreData(AudioBus* audio_bus,
- AudioBuffersState buffers_state) {
+ int total_bytes_delay) {
++callback_count_;
// Touch the channel memory value to make sure memory is good.
audio_bus->Zero();
@@ -89,9 +89,9 @@ class TestSourceLaggy : public TestSourceBasic {
: laggy_after_buffer_(laggy_after_buffer), lag_in_ms_(lag_in_ms) {
}
virtual int OnMoreData(AudioBus* audio_bus,
- AudioBuffersState buffers_state) {
+ int total_bytes_delay) {
// Call the base, which increments the callback_count_.
- TestSourceBasic::OnMoreData(audio_bus, buffers_state);
+ TestSourceBasic::OnMoreData(audio_bus, total_bytes_delay);
if (callback_count() > kMaxNumBuffers) {
::Sleep(lag_in_ms_);
}
@@ -520,32 +520,22 @@ TEST(WinAudioTest, PCMWaveStreamPendingBytes) {
// pending bytes will go down and eventually read zero.
InSequence s;
- EXPECT_CALL(source, OnMoreData(NotNull(),
- Field(&AudioBuffersState::pending_bytes, 0)))
+ EXPECT_CALL(source, OnMoreData(NotNull(), 0))
.WillOnce(Invoke(ClearData));
// Note: If AudioManagerWin::NumberOfWaveOutBuffers() ever changes, or if this
// test is run on Vista, these expectations will fail.
- EXPECT_CALL(source, OnMoreData(NotNull(),
- Field(&AudioBuffersState::pending_bytes,
- bytes_100_ms)))
+ EXPECT_CALL(source, OnMoreData(NotNull(), bytes_100_ms))
.WillOnce(Invoke(ClearData));
- EXPECT_CALL(source, OnMoreData(NotNull(),
- Field(&AudioBuffersState::pending_bytes,
- 2 * bytes_100_ms)))
+ EXPECT_CALL(source, OnMoreData(NotNull(), 2 * bytes_100_ms))
.WillOnce(Invoke(ClearData));
- EXPECT_CALL(source, OnMoreData(NotNull(),
- Field(&AudioBuffersState::pending_bytes,
- 2 * bytes_100_ms)))
+ EXPECT_CALL(source, OnMoreData(NotNull(), 2 * bytes_100_ms))
.Times(AnyNumber())
.WillRepeatedly(Return(0));
- EXPECT_CALL(source, OnMoreData(NotNull(),
- Field(&AudioBuffersState::pending_bytes,
- bytes_100_ms)))
+ EXPECT_CALL(source, OnMoreData(NotNull(), bytes_100_ms))
.Times(AnyNumber())
.WillRepeatedly(Return(0));
- EXPECT_CALL(source, OnMoreData(NotNull(),
- Field(&AudioBuffersState::pending_bytes, 0)))
+ EXPECT_CALL(source, OnMoreData(NotNull(), 0))
.Times(AnyNumber())
.WillRepeatedly(Return(0));
@@ -571,8 +561,8 @@ class SyncSocketSource : public AudioOutputStream::AudioSourceCallback {
// AudioSourceCallback::OnMoreData implementation:
virtual int OnMoreData(AudioBus* audio_bus,
- AudioBuffersState buffers_state) {
- socket_->Send(&buffers_state, sizeof(buffers_state));
+ int total_bytes_delay) {
+ socket_->Send(&total_bytes_delay, sizeof(total_bytes_delay));
uint32 size = socket_->Receive(data_.get(), data_size_);
DCHECK_EQ(static_cast<size_t>(size) % sizeof(*audio_bus_->channel(0)), 0U);
audio_bus_->CopyTo(audio_bus);
@@ -580,7 +570,7 @@ class SyncSocketSource : public AudioOutputStream::AudioSourceCallback {
}
virtual int OnMoreIOData(AudioBus* source,
AudioBus* dest,
- AudioBuffersState buffers_state) {
+ int total_bytes_delay) {
NOTREACHED();
return 0;
}
@@ -622,13 +612,13 @@ DWORD __stdcall SyncSocketThread(void* context) {
SineWaveAudioSource sine(1, ctx.sine_freq, ctx.sample_rate);
const int kTwoSecFrames = ctx.sample_rate * 2;
- AudioBuffersState buffers_state;
+ int total_bytes_delay = 0;
int times = 0;
for (int ix = 0; ix < kTwoSecFrames; ix += ctx.frames) {
- if (ctx.socket->Receive(&buffers_state, sizeof(buffers_state)) == 0)
+ if (ctx.socket->Receive(&total_bytes_delay, sizeof(total_bytes_delay)) == 0)
break;
- if ((times > 0) && (buffers_state.pending_bytes < 1000)) __debugbreak();
- sine.OnMoreData(audio_bus.get(), buffers_state);
+ if ((times > 0) && (total_bytes_delay < 1000)) __debugbreak();
+ sine.OnMoreData(audio_bus.get(), total_bytes_delay);
ctx.socket->Send(data.get(), ctx.packet_size_bytes);
++times;
}
diff --git a/media/audio/win/waveout_output_win.cc b/media/audio/win/waveout_output_win.cc
index 0f54817..4d11e08 100644
--- a/media/audio/win/waveout_output_win.cc
+++ b/media/audio/win/waveout_output_win.cc
@@ -325,9 +325,10 @@ void PCMWaveOutAudioOutputStream::QueueNextPacket(WAVEHDR *buffer) {
// return to us how many bytes were used.
// TODO(fbarchard): Handle used 0 by queueing more.
- // TODO(sergeyu): Specify correct hardware delay for AudioBuffersState.
+ // TODO(sergeyu): Specify correct hardware delay for |total_delay_bytes|.
+ int total_delay_bytes = pending_bytes_;
int frames_filled = callback_->OnMoreData(
- audio_bus_.get(), AudioBuffersState(pending_bytes_, 0));
+ audio_bus_.get(), total_delay_bytes);
uint32 used = frames_filled * audio_bus_->channels() *
format_.Format.wBitsPerSample / 8;
generated by cgit v1.1 at 2025-02-04 17:40:49 +0000