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Unify FrameType and VideoFrameType.

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Prevents some heap allocation and frame-type conversion since interfaces
mismatch. Also it's less confusing to have one type for this.

BUG=webrtc:5042
R=magjed@webrtc.org, mflodman@webrtc.org, henrik.lundin@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1371043003

Cr-Commit-Position: refs/heads/master@{#10320}
This commit is contained in:
pbos
2015-10-19 02:39:06 -07:00
committed by Commit bot
parent 4306fc70d7
commit 22993e1a0c
63 changed files with 179 additions and 237 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
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@ -46,7 +46,7 @@ class AcmReceiverTest : public AudioPacketizationCallback,
: timestamp_(0),
packet_sent_(false),
last_packet_send_timestamp_(timestamp_),
last_frame_type_(kFrameEmpty) {
last_frame_type_(kEmptyFrame) {
AudioCoding::Config config;
config.transport = this;
acm_.reset(new AudioCodingImpl(config));
@ -121,7 +121,7 @@ class AcmReceiverTest : public AudioPacketizationCallback,
const uint8_t* payload_data,
size_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) override {
if (frame_type == kFrameEmpty)
if (frame_type == kEmptyFrame)
return 0;
rtp_header_.header.payloadType = payload_type;

4
webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
View File

@ -46,7 +46,7 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
: timestamp_(0),
packet_sent_(false),
last_packet_send_timestamp_(timestamp_),
last_frame_type_(kFrameEmpty) {
last_frame_type_(kEmptyFrame) {
AudioCodingModule::Config config;
acm_.reset(new AudioCodingModuleImpl(config));
receiver_.reset(new AcmReceiver(config));
@ -120,7 +120,7 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
const uint8_t* payload_data,
size_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) override {
if (frame_type == kFrameEmpty)
if (frame_type == kEmptyFrame)
return 0;
rtp_header_.header.payloadType = payload_type;

2
webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
View File

@ -171,7 +171,7 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
ConvertEncodedInfoToFragmentationHeader(encoded_info, &my_fragmentation);
FrameType frame_type;
if (encode_buffer_.size() == 0 && encoded_info.send_even_if_empty) {
frame_type = kFrameEmpty;
frame_type = kEmptyFrame;
encoded_info.payload_type = previous_pltype;
} else {
RTC_DCHECK_GT(encode_buffer_.size(), 0u);

22
webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
View File

@ -92,7 +92,7 @@ class PacketizationCallbackStubOldApi : public AudioPacketizationCallback {
public:
PacketizationCallbackStubOldApi()
: num_calls_(0),
last_frame_type_(kFrameEmpty),
last_frame_type_(kEmptyFrame),
last_payload_type_(-1),
last_timestamp_(0),
crit_sect_(CriticalSectionWrapper::CreateCriticalSection()) {}
@ -416,18 +416,18 @@ class AudioCodingModuleTestWithComfortNoiseOldApi
int ix;
FrameType type;
} expectation[] = {{2, kAudioFrameCN},
{5, kFrameEmpty},
{8, kFrameEmpty},
{5, kEmptyFrame},
{8, kEmptyFrame},
{11, kAudioFrameCN},
{14, kFrameEmpty},
{17, kFrameEmpty},
{14, kEmptyFrame},
{17, kEmptyFrame},
{20, kAudioFrameCN},
{23, kFrameEmpty},
{26, kFrameEmpty},
{29, kFrameEmpty},
{23, kEmptyFrame},
{26, kEmptyFrame},
{29, kEmptyFrame},
{32, kAudioFrameCN},
{35, kFrameEmpty},
{38, kFrameEmpty}};
{35, kEmptyFrame},
{38, kEmptyFrame}};
for (int i = 0; i < kLoops; ++i) {
int num_calls_before = packet_cb_.num_calls();
EXPECT_EQ(i / blocks_per_packet, num_calls_before);
@ -447,7 +447,7 @@ class AudioCodingModuleTestWithComfortNoiseOldApi
// Checks that the transport callback is invoked once per frame period of the
// underlying speech encoder, even when comfort noise is produced.
// Also checks that the frame type is kAudioFrameCN or kFrameEmpty.
// Also checks that the frame type is kAudioFrameCN or kEmptyFrame.
// This test and the next check the same thing, but differ in the order of
// speech codec and CNG registration.
TEST_F(AudioCodingModuleTestWithComfortNoiseOldApi,

2
webrtc/modules/audio_coding/main/test/Channel.cc
View File

@ -42,7 +42,7 @@ int32_t Channel::SendData(FrameType frameType,
} else {
rtpInfo.type.Audio.isCNG = false;
}
if (frameType == kFrameEmpty) {
if (frameType == kEmptyFrame) {
// When frame is empty, we should not transmit it. The frame size of the
// next non-empty frame will be based on the previous frame size.
_useLastFrameSize = _lastFrameSizeSample > 0;

2
webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
View File

@ -74,7 +74,7 @@ int32_t TestPack::SendData(FrameType frame_type, uint8_t payload_type,
} else {
rtp_info.type.Audio.isCNG = false;
}
if (frame_type == kFrameEmpty) {
if (frame_type == kEmptyFrame) {
// Skip this frame.
return 0;
}

2
webrtc/modules/audio_coding/main/test/TestStereo.cc
View File

@ -58,7 +58,7 @@ int32_t TestPackStereo::SendData(const FrameType frame_type,
rtp_info.header.sequenceNumber = seq_no_++;
rtp_info.header.payloadType = payload_type;
rtp_info.header.timestamp = timestamp;
if (frame_type == kFrameEmpty) {
if (frame_type == kEmptyFrame) {
// Skip this frame
return 0;
}

8
webrtc/modules/audio_coding/main/test/TestVADDTX.cc
View File

@ -44,7 +44,7 @@ int32_t ActivityMonitor::InFrameType(FrameType frame_type) {
void ActivityMonitor::PrintStatistics() {
printf("\n");
printf("kFrameEmpty %u\n", counter_[kFrameEmpty]);
printf("kEmptyFrame %u\n", counter_[kEmptyFrame]);
printf("kAudioFrameSpeech %u\n", counter_[kAudioFrameSpeech]);
printf("kAudioFrameCN %u\n", counter_[kAudioFrameCN]);
printf("kVideoFrameKey %u\n", counter_[kVideoFrameKey]);
@ -248,7 +248,7 @@ void TestOpusDtx::Perform() {
32000, 1, out_filename, false, expects);
EXPECT_EQ(0, acm_send_->EnableOpusDtx());
expects[kFrameEmpty] = 1;
expects[kEmptyFrame] = 1;
Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
32000, 1, out_filename, true, expects);
@ -256,13 +256,13 @@ void TestOpusDtx::Perform() {
out_filename = webrtc::test::OutputPath() + "testOpusDtx_outFile_stereo.pcm";
RegisterCodec(kOpusStereo);
EXPECT_EQ(0, acm_send_->DisableOpusDtx());
expects[kFrameEmpty] = 0;
expects[kEmptyFrame] = 0;
Run(webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm"),
32000, 2, out_filename, false, expects);
EXPECT_EQ(0, acm_send_->EnableOpusDtx());
expects[kFrameEmpty] = 1;
expects[kEmptyFrame] = 1;
Run(webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm"),
32000, 2, out_filename, true, expects);
#endif

4
webrtc/modules/audio_coding/main/test/TestVADDTX.h
View File

@ -29,7 +29,7 @@ class ActivityMonitor : public ACMVADCallback {
void ResetStatistics();
void GetStatistics(uint32_t* stats);
private:
// 0 - kFrameEmpty
// 0 - kEmptyFrame
// 1 - kAudioFrameSpeech
// 2 - kAudioFrameCN
// 3 - kVideoFrameKey (not used by audio)
@ -60,7 +60,7 @@ class TestVadDtx : public ACMTest {
// 0 : there have been no packets of type |x|,
// 1 : there have been packets of type |x|,
// with |x| indicates the following packet types
// 0 - kFrameEmpty
// 0 - kEmptyFrame
// 1 - kAudioFrameSpeech
// 2 - kAudioFrameCN
// 3 - kVideoFrameKey (not used by audio)

2
webrtc/modules/audio_coding/main/test/utility.cc
View File

@ -288,7 +288,7 @@ VADCallback::VADCallback() {
}
void VADCallback::PrintFrameTypes() {
printf("kFrameEmpty......... %d\n", _numFrameTypes[kFrameEmpty]);
printf("kEmptyFrame......... %d\n", _numFrameTypes[kEmptyFrame]);
printf("kAudioFrameSpeech... %d\n", _numFrameTypes[kAudioFrameSpeech]);
printf("kAudioFrameCN....... %d\n", _numFrameTypes[kAudioFrameCN]);
printf("kVideoFrameKey...... %d\n", _numFrameTypes[kVideoFrameKey]);