Refactor WebRtc_CreateBuffer to return the instance.
Review URL: https://webrtc-codereview.appspot.com/1140005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3574 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_processing/aec/aec_core.c b/webrtc/modules/audio_processing/aec/aec_core.c
index 1b747b8..d638b72 100644
--- a/webrtc/modules/audio_processing/aec/aec_core.c
+++ b/webrtc/modules/audio_processing/aec/aec_core.c
@@ -164,54 +164,57 @@
return -1;
}
- if (WebRtc_CreateBuffer(&aec->nearFrBuf,
- FRAME_LEN + PART_LEN,
- sizeof(int16_t)) == -1) {
+ aec->nearFrBuf = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN,
+ sizeof(int16_t));
+ if (!aec->nearFrBuf) {
WebRtcAec_FreeAec(aec);
aec = NULL;
return -1;
}
- if (WebRtc_CreateBuffer(&aec->outFrBuf,
- FRAME_LEN + PART_LEN,
- sizeof(int16_t)) == -1) {
+ aec->outFrBuf = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN,
+ sizeof(int16_t));
+ if (!aec->outFrBuf) {
WebRtcAec_FreeAec(aec);
aec = NULL;
return -1;
}
- if (WebRtc_CreateBuffer(&aec->nearFrBufH,
- FRAME_LEN + PART_LEN,
- sizeof(int16_t)) == -1) {
+ aec->nearFrBufH = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN,
+ sizeof(int16_t));
+ if (!aec->nearFrBufH) {
WebRtcAec_FreeAec(aec);
aec = NULL;
return -1;
}
- if (WebRtc_CreateBuffer(&aec->outFrBufH,
- FRAME_LEN + PART_LEN,
- sizeof(int16_t)) == -1) {
+ aec->outFrBufH = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN,
+ sizeof(int16_t));
+ if (!aec->outFrBufH) {
WebRtcAec_FreeAec(aec);
aec = NULL;
return -1;
}
// Create far-end buffers.
- if (WebRtc_CreateBuffer(&aec->far_buf, kBufSizePartitions,
- sizeof(float) * 2 * PART_LEN1) == -1) {
+ aec->far_buf = WebRtc_CreateBuffer(kBufSizePartitions,
+ sizeof(float) * 2 * PART_LEN1);
+ if (!aec->far_buf) {
WebRtcAec_FreeAec(aec);
aec = NULL;
return -1;
}
- if (WebRtc_CreateBuffer(&aec->far_buf_windowed, kBufSizePartitions,
- sizeof(float) * 2 * PART_LEN1) == -1) {
+ aec->far_buf_windowed = WebRtc_CreateBuffer(kBufSizePartitions,
+ sizeof(float) * 2 * PART_LEN1);
+ if (!aec->far_buf_windowed) {
WebRtcAec_FreeAec(aec);
aec = NULL;
return -1;
}
#ifdef WEBRTC_AEC_DEBUG_DUMP
- if (WebRtc_CreateBuffer(&aec->far_time_buf, kBufSizePartitions,
- sizeof(int16_t) * PART_LEN) == -1) {
+ aec->far_time_buf = WebRtc_CreateBuffer(kBufSizePartitions,
+ sizeof(int16_t) * PART_LEN);
+ if (!aec->far_time_buf) {
WebRtcAec_FreeAec(aec);
aec = NULL;
return -1;
diff --git a/webrtc/modules/audio_processing/aec/echo_cancellation.c b/webrtc/modules/audio_processing/aec/echo_cancellation.c
index 4066281..8fd9fe9 100644
--- a/webrtc/modules/audio_processing/aec/echo_cancellation.c
+++ b/webrtc/modules/audio_processing/aec/echo_cancellation.c
@@ -72,9 +72,9 @@
// Create far-end pre-buffer. The buffer size has to be large enough for
// largest possible drift compensation (kResamplerBufferSize) + "almost" an
// FFT buffer (PART_LEN2 - 1).
- if (WebRtc_CreateBuffer(&aecpc->far_pre_buf,
- PART_LEN2 + kResamplerBufferSize,
- sizeof(float)) == -1) {
+ aecpc->far_pre_buf = WebRtc_CreateBuffer(PART_LEN2 + kResamplerBufferSize,
+ sizeof(float));
+ if (!aecpc->far_pre_buf) {
WebRtcAec_Free(aecpc);
aecpc = NULL;
return -1;
@@ -84,9 +84,9 @@
aecpc->lastError = 0;
#ifdef WEBRTC_AEC_DEBUG_DUMP
- if (WebRtc_CreateBuffer(&aecpc->far_pre_buf_s16,
- PART_LEN2 + kResamplerBufferSize,
- sizeof(int16_t)) == -1) {
+ aecpc->far_pre_buf_s16 = WebRtc_CreateBuffer(
+ PART_LEN2 + kResamplerBufferSize, sizeof(float));
+ if (!aecpc->far_pre_buf_s16) {
WebRtcAec_Free(aecpc);
aecpc = NULL;
return -1;
diff --git a/webrtc/modules/audio_processing/aecm/aecm_core.c b/webrtc/modules/audio_processing/aecm/aecm_core.c
index a81912e..1368263 100644
--- a/webrtc/modules/audio_processing/aecm/aecm_core.c
+++ b/webrtc/modules/audio_processing/aecm/aecm_core.c
@@ -256,32 +256,36 @@
return -1;
}
- if (WebRtc_CreateBuffer(&aecm->farFrameBuf, FRAME_LEN + PART_LEN,
- sizeof(int16_t)) == -1)
+ aecm->farFrameBuf = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN,
+ sizeof(int16_t));
+ if (!aecm->farFrameBuf)
{
WebRtcAecm_FreeCore(aecm);
aecm = NULL;
return -1;
}
- if (WebRtc_CreateBuffer(&aecm->nearNoisyFrameBuf, FRAME_LEN + PART_LEN,
- sizeof(int16_t)) == -1)
+ aecm->nearNoisyFrameBuf = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN,
+ sizeof(int16_t));
+ if (!aecm->nearNoisyFrameBuf)
{
WebRtcAecm_FreeCore(aecm);
aecm = NULL;
return -1;
}
- if (WebRtc_CreateBuffer(&aecm->nearCleanFrameBuf, FRAME_LEN + PART_LEN,
- sizeof(int16_t)) == -1)
+ aecm->nearCleanFrameBuf = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN,
+ sizeof(int16_t));
+ if (!aecm->nearCleanFrameBuf)
{
WebRtcAecm_FreeCore(aecm);
aecm = NULL;
return -1;
}
- if (WebRtc_CreateBuffer(&aecm->outFrameBuf, FRAME_LEN + PART_LEN,
- sizeof(int16_t)) == -1)
+ aecm->outFrameBuf = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN,
+ sizeof(int16_t));
+ if (!aecm->outFrameBuf)
{
WebRtcAecm_FreeCore(aecm);
aecm = NULL;
diff --git a/webrtc/modules/audio_processing/aecm/echo_control_mobile.c b/webrtc/modules/audio_processing/aecm/echo_control_mobile.c
index 85bbcd4..881f7a5 100644
--- a/webrtc/modules/audio_processing/aecm/echo_control_mobile.c
+++ b/webrtc/modules/audio_processing/aecm/echo_control_mobile.c
@@ -102,8 +102,9 @@
return -1;
}
- if (WebRtc_CreateBuffer(&aecm->farendBuf, kBufSizeSamp,
- sizeof(int16_t)) == -1)
+ aecm->farendBuf = WebRtc_CreateBuffer(kBufSizeSamp,
+ sizeof(int16_t));
+ if (!aecm->farendBuf)
{
WebRtcAecm_Free(aecm);
aecm = NULL;
diff --git a/webrtc/modules/audio_processing/utility/ring_buffer.c b/webrtc/modules/audio_processing/utility/ring_buffer.c
index bc21f85..617c30f 100644
--- a/webrtc/modules/audio_processing/utility/ring_buffer.c
+++ b/webrtc/modules/audio_processing/utility/ring_buffer.c
@@ -65,34 +65,28 @@
return read_elements;
}
-int WebRtc_CreateBuffer(RingBuffer** handle,
- size_t element_count,
- size_t element_size) {
+RingBuffer* WebRtc_CreateBuffer(size_t element_count, size_t element_size) {
RingBuffer* self = NULL;
- if (!handle) {
- return -1;
- }
if (element_count == 0 || element_size == 0) {
- return -1;
+ return NULL;
}
self = malloc(sizeof(RingBuffer));
if (!self) {
- return -1;
+ return NULL;
}
- *handle = self;
self->data = malloc(element_count * element_size);
if (!self->data) {
free(self);
self = NULL;
- return -1;
+ return NULL;
}
self->element_count = element_count;
self->element_size = element_size;
- return 0;
+ return self;
}
int WebRtc_InitBuffer(RingBuffer* self) {
diff --git a/webrtc/modules/audio_processing/utility/ring_buffer.h b/webrtc/modules/audio_processing/utility/ring_buffer.h
index ee9a0c6..61d25ba 100644
--- a/webrtc/modules/audio_processing/utility/ring_buffer.h
+++ b/webrtc/modules/audio_processing/utility/ring_buffer.h
@@ -18,9 +18,8 @@
typedef struct RingBuffer RingBuffer;
-int WebRtc_CreateBuffer(RingBuffer** handle,
- size_t element_count,
- size_t element_size);
+// Returns NULL on failure.
+RingBuffer* WebRtc_CreateBuffer(size_t element_count, size_t element_size);
int WebRtc_InitBuffer(RingBuffer* handle);
void WebRtc_FreeBuffer(void* handle);
diff --git a/webrtc/modules/audio_processing/utility/ring_buffer_unittest.cc b/webrtc/modules/audio_processing/utility/ring_buffer_unittest.cc
index eef8b5f..365eb2d 100644
--- a/webrtc/modules/audio_processing/utility/ring_buffer_unittest.cc
+++ b/webrtc/modules/audio_processing/utility/ring_buffer_unittest.cc
@@ -58,9 +58,8 @@
const int buffer_size = std::max(rand() % kMaxBufferSize, 1);
scoped_array<int> write_data(new int[buffer_size]);
scoped_array<int> read_data(new int[buffer_size]);
- scoped_ring_buffer buffer;
- ASSERT_EQ(0, WebRtc_CreateBuffer(buffer.accept(), buffer_size,
- sizeof(int)));
+ scoped_ring_buffer buffer(WebRtc_CreateBuffer(buffer_size, sizeof(int)));
+ ASSERT_TRUE(buffer.get() != NULL);
ASSERT_EQ(0, WebRtc_InitBuffer(buffer.get()));
int buffer_consumed = 0;
int write_element = 0;
@@ -112,13 +111,13 @@
}
TEST(RingBufferTest, PassingNulltoReadBufferForcesMemcpy) {
- scoped_ring_buffer buffer;
const size_t kDataSize = 2;
int write_data[kDataSize];
int read_data[kDataSize];
int* data_ptr;
- ASSERT_EQ(0, WebRtc_CreateBuffer(buffer.accept(), kDataSize, sizeof(int)));
+ scoped_ring_buffer buffer(WebRtc_CreateBuffer(kDataSize, sizeof(int)));
+ ASSERT_TRUE(buffer.get() != NULL);
ASSERT_EQ(0, WebRtc_InitBuffer(buffer.get()));
SetIncrementingData(write_data, kDataSize, 0);
@@ -137,4 +136,12 @@
CheckIncrementingData(read_data, kDataSize, 0);
}
+TEST(RingBufferTest, CreateHandlesErrors) {
+ EXPECT_TRUE(WebRtc_CreateBuffer(0, 1) == NULL);
+ EXPECT_TRUE(WebRtc_CreateBuffer(1, 0) == NULL);
+ RingBuffer* buffer = WebRtc_CreateBuffer(1, 1);
+ EXPECT_TRUE(buffer != NULL);
+ WebRtc_FreeBuffer(buffer);
+}
+
} // namespace webrtc
