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-rw-r--r--media/audio/simple_sources.cc35
-rw-r--r--media/audio/simple_sources.h4
-rw-r--r--media/audio/simple_sources_unittest.cc106
-rw-r--r--media/audio/win/audio_output_win_unittest.cc43
-rw-r--r--media/media.gyp1
5 files changed, 133 insertions, 56 deletions
diff --git a/media/audio/simple_sources.cc b/media/audio/simple_sources.cc
index 6643fc1..f941782 100644
--- a/media/audio/simple_sources.cc
+++ b/media/audio/simple_sources.cc
@@ -50,7 +50,9 @@ void SineWaveAudioSource::OnError(AudioOutputStream* stream, int code) {
// PushSource implementation.
PushSource::PushSource(size_t packet_size)
- : packet_size_(packet_size), buffered_bytes_(0) {
+ : packet_size_(packet_size),
+ buffered_bytes_(0),
+ front_buffer_consumed_(0) {
}
PushSource::~PushSource() {
@@ -59,20 +61,29 @@ PushSource::~PushSource() {
size_t PushSource::OnMoreData(AudioOutputStream* stream,
void* dest, size_t max_size) {
- Packet packet;
- {
+ size_t copied = 0;
+ while (copied < max_size) {
AutoLock auto_lock(lock_);
+
// Under lock processing in this scope.
if (!packets_.size())
- return 0;
- packet = packets_.front();
- packets_.pop_front();
- buffered_bytes_ -= packet.size;
+ break;
+ Packet packet = packets_.front();
+ size_t size = std::min(max_size - copied,
+ packet.size - front_buffer_consumed_);
+ memcpy(static_cast<char*>(dest) + copied,
+ packet.buffer + front_buffer_consumed_,
+ size);
+ front_buffer_consumed_ += size;
+ buffered_bytes_ -= size;
+ copied += size;
+ if (front_buffer_consumed_ == packet.size) {
+ delete [] packet.buffer;
+ packets_.pop_front();
+ front_buffer_consumed_ = 0;
+ }
}
- size_t size = std::min(max_size, packet.size);
- memcpy(dest, packet.buffer, size);
- delete [] packet.buffer;
- return size;
+ return copied;
}
void PushSource::OnClose(AudioOutputStream* stream) {
@@ -85,7 +96,7 @@ void PushSource::OnError(AudioOutputStream* stream, int code) {
// TODO(cpu): Manage arbitrary large sizes.
bool PushSource::Write(const void *data, size_t len) {
- if ((len == 0) || (len > packet_size_)) {
+ if (len == 0) {
NOTREACHED();
return false;
}
diff --git a/media/audio/simple_sources.h b/media/audio/simple_sources.h
index 1da4d4f..4a3db8d 100644
--- a/media/audio/simple_sources.h
+++ b/media/audio/simple_sources.h
@@ -60,11 +60,12 @@ class PushSource : public AudioOutputStream::AudioSourceCallback,
public:
// Construct the audio source. Pass the same |packet_size| specified in the
// AudioOutputStream::Open() here.
+ // TODO(hclam): |packet_size| is not used anymore, remove it.
explicit PushSource(size_t packet_size);
virtual ~PushSource();
// Write one buffer. The ideal size is |packet_size| but smaller sizes are
- // accepted. Bigger sizes are an error. Returns false on error.
+ // accepted.
virtual bool Write(const void* data, size_t len);
// Return the total number of bytes not given to the audio device yet.
@@ -90,6 +91,7 @@ class PushSource : public AudioOutputStream::AudioSourceCallback,
typedef std::list<Packet> PacketList;
PacketList packets_;
size_t buffered_bytes_;
+ size_t front_buffer_consumed_;
// Serialize access to packets_ and buffered_bytes_ using this lock.
Lock lock_;
};
diff --git a/media/audio/simple_sources_unittest.cc b/media/audio/simple_sources_unittest.cc
new file mode 100644
index 0000000..0ebb335
--- /dev/null
+++ b/media/audio/simple_sources_unittest.cc
@@ -0,0 +1,106 @@
+// Copyright (c) 2009 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#include "base/logging.h"
+#include "base/basictypes.h"
+#include "base/scoped_ptr.h"
+#include "base/time.h"
+#include "media/audio/simple_sources.h"
+#include "testing/gtest/include/gtest/gtest.h"
+
+namespace {
+
+void GenerateRandomData(char* buffer, size_t len) {
+ static bool called = false;
+ if (!called) {
+ called = true;
+ int seed = static_cast<int>(base::Time::Now().ToInternalValue());
+ srand(seed);
+ LOG(INFO) << "Random seed: " << seed;
+ }
+
+ for (size_t i = 0; i < len; i++) {
+ buffer[i] = static_cast<char>(rand());
+ }
+}
+
+} // namespace
+
+// To test write size smaller than read size.
+TEST(SimpleSourcesTest, PushSourceSmallerWrite) {
+ const size_t kDataSize = 40960;
+ scoped_array<char> data(new char[kDataSize]);
+ GenerateRandomData(data.get(), kDataSize);
+
+ // Choose two prime numbers for read and write sizes.
+ const size_t kWriteSize = 283;
+ const size_t kReadSize = 293;
+ scoped_array<char> read_data(new char[kReadSize]);
+
+ // Create a PushSource that assumes the hardware audio buffer size is always
+ // bigger than the write size.
+ PushSource push_source(kReadSize);
+ EXPECT_EQ(0u, push_source.UnProcessedBytes());
+
+ // Write everything into this push source.
+ for (size_t i = 0; i < kDataSize; i += kWriteSize) {
+ size_t size = std::min(kDataSize - i, kWriteSize);
+ EXPECT_TRUE(push_source.Write(data.get() + i, size));
+ }
+ EXPECT_EQ(kDataSize, push_source.UnProcessedBytes());
+
+ // Read everything from the push source.
+ for (size_t i = 0; i < kDataSize; i += kReadSize) {
+ size_t size = std::min(kDataSize - i , kReadSize);
+ EXPECT_EQ(size, push_source.OnMoreData(NULL, read_data.get(), size));
+ EXPECT_EQ(0, memcmp(data.get() + i, read_data.get(), size));
+ }
+ EXPECT_EQ(0u, push_source.UnProcessedBytes());
+
+ push_source.OnClose(NULL);
+}
+
+
+// Validate that the SineWaveAudioSource writes the expected values for
+// the FORMAT_16BIT_MONO. The values are carefully selected so rounding issues
+// do not affect the result. We also test that AudioManager::GetLastMockBuffer
+// works.
+TEST(SimpleSourcesTest, SineWaveAudio16MonoTest) {
+ const size_t samples = 1024;
+ const size_t bytes_per_sample = 2;
+ const int freq = 200;
+
+ SineWaveAudioSource source(SineWaveAudioSource::FORMAT_16BIT_LINEAR_PCM, 1,
+ freq, AudioManager::kTelephoneSampleRate);
+
+ AudioManager* audio_man = AudioManager::GetAudioManager();
+ ASSERT_TRUE(NULL != audio_man);
+ AudioOutputStream* oas =
+ audio_man->MakeAudioStream(AudioManager::AUDIO_MOCK, 1,
+ AudioManager::kTelephoneSampleRate,
+ bytes_per_sample * 2);
+ ASSERT_TRUE(NULL != oas);
+ EXPECT_TRUE(oas->Open(samples * bytes_per_sample));
+
+ oas->Start(&source);
+ oas->Stop();
+ oas->Close();
+
+ const int16* last_buffer =
+ reinterpret_cast<const int16*>(audio_man->GetLastMockBuffer());
+ ASSERT_TRUE(NULL != last_buffer);
+
+ size_t half_period = AudioManager::kTelephoneSampleRate / (freq * 2);
+
+ // Spot test positive incursion of sine wave.
+ EXPECT_EQ(0, last_buffer[0]);
+ EXPECT_EQ(5126, last_buffer[1]);
+ EXPECT_TRUE(last_buffer[1] < last_buffer[2]);
+ EXPECT_TRUE(last_buffer[2] < last_buffer[3]);
+ // Spot test negative incursion of sine wave.
+ EXPECT_EQ(0, last_buffer[half_period]);
+ EXPECT_EQ(-5126, last_buffer[half_period + 1]);
+ EXPECT_TRUE(last_buffer[half_period + 1] > last_buffer[half_period + 2]);
+ EXPECT_TRUE(last_buffer[half_period + 2] > last_buffer[half_period + 3]);
+}
diff --git a/media/audio/win/audio_output_win_unittest.cc b/media/audio/win/audio_output_win_unittest.cc
index 5366754..00b9dd7 100644
--- a/media/audio/win/audio_output_win_unittest.cc
+++ b/media/audio/win/audio_output_win_unittest.cc
@@ -193,49 +193,6 @@ TEST(WinAudioTest, MockStreamBasicCallbacks) {
EXPECT_EQ(1, source.was_closed());
}
-// Validate that the SineWaveAudioSource writes the expected values for
-// the FORMAT_16BIT_MONO. The values are carefully selected so rounding issues
-// do not affect the result. We also test that AudioManager::GetLastMockBuffer
-// works.
-TEST(WinAudioTest, SineWaveAudio16MonoTest) {
- const size_t samples = 1024;
- const size_t bytes_per_sample = 2;
- const int freq = 200;
-
- SineWaveAudioSource source(SineWaveAudioSource::FORMAT_16BIT_LINEAR_PCM, 1,
- freq, AudioManager::kTelephoneSampleRate);
-
- AudioManager* audio_man = AudioManager::GetAudioManager();
- ASSERT_TRUE(NULL != audio_man);
- AudioOutputStream* oas =
- audio_man->MakeAudioStream(AudioManager::AUDIO_MOCK, 1,
- AudioManager::kTelephoneSampleRate,
- bytes_per_sample * 2);
- ASSERT_TRUE(NULL != oas);
- EXPECT_TRUE(oas->Open(samples * bytes_per_sample));
-
- oas->Start(&source);
- oas->Stop();
- oas->Close();
-
- const int16* last_buffer =
- reinterpret_cast<const int16*>(audio_man->GetLastMockBuffer());
- ASSERT_TRUE(NULL != last_buffer);
-
- size_t half_period = AudioManager::kTelephoneSampleRate / (freq * 2);
-
- // Spot test positive incursion of sine wave.
- EXPECT_EQ(0, last_buffer[0]);
- EXPECT_EQ(5126, last_buffer[1]);
- EXPECT_TRUE(last_buffer[1] < last_buffer[2]);
- EXPECT_TRUE(last_buffer[2] < last_buffer[3]);
- // Spot test negative incursion of sine wave.
- EXPECT_EQ(0, last_buffer[half_period]);
- EXPECT_EQ(-5126, last_buffer[half_period + 1]);
- EXPECT_TRUE(last_buffer[half_period + 1] > last_buffer[half_period + 2]);
- EXPECT_TRUE(last_buffer[half_period + 2] > last_buffer[half_period + 3]);
-}
-
// ===========================================================================
// Validation of AudioManager::AUDIO_PCM_LINEAR
//
diff --git a/media/media.gyp b/media/media.gyp
index db387c8..a62da2a 100644
--- a/media/media.gyp
+++ b/media/media.gyp
@@ -145,6 +145,7 @@
'sources': [
'audio/win/audio_output_win_unittest.cc',
'audio/mac/audio_output_mac_unittest.cc',
+ 'audio/simple_sources_unittest.cc',
'base/data_buffer_unittest.cc',
'base/pipeline_impl_unittest.cc',
'base/run_all_unittests.cc',
generated by cgit v1.1 at 2025-02-01 03:05:03 +0000