Unify FrameType and VideoFrameType.
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}
diff --git a/talk/app/webrtc/java/jni/androidmediaencoder_jni.cc b/talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
index 76a675d..d2ec552 100644
--- a/talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
+++ b/talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
@@ -86,10 +86,9 @@
int32_t InitEncode(const webrtc::VideoCodec* codec_settings,
int32_t /* number_of_cores */,
size_t /* max_payload_size */) override;
- int32_t Encode(
- const webrtc::VideoFrame& input_image,
- const webrtc::CodecSpecificInfo* /* codec_specific_info */,
- const std::vector<webrtc::VideoFrameType>* frame_types) override;
+ int32_t Encode(const webrtc::VideoFrame& input_image,
+ const webrtc::CodecSpecificInfo* /* codec_specific_info */,
+ const std::vector<webrtc::FrameType>* frame_types) override;
int32_t RegisterEncodeCompleteCallback(
webrtc::EncodedImageCallback* callback) override;
int32_t Release() override;
@@ -121,7 +120,7 @@
int32_t InitEncodeOnCodecThread(int width, int height, int kbps, int fps);
int32_t EncodeOnCodecThread(
const webrtc::VideoFrame& input_image,
- const std::vector<webrtc::VideoFrameType>* frame_types);
+ const std::vector<webrtc::FrameType>* frame_types);
int32_t RegisterEncodeCompleteCallbackOnCodecThread(
webrtc::EncodedImageCallback* callback);
int32_t ReleaseOnCodecThread();
@@ -338,7 +337,7 @@
int32_t MediaCodecVideoEncoder::Encode(
const webrtc::VideoFrame& frame,
const webrtc::CodecSpecificInfo* /* codec_specific_info */,
- const std::vector<webrtc::VideoFrameType>* frame_types) {
+ const std::vector<webrtc::FrameType>* frame_types) {
return codec_thread_->Invoke<int32_t>(Bind(
&MediaCodecVideoEncoder::EncodeOnCodecThread, this, frame, frame_types));
}
@@ -501,7 +500,7 @@
int32_t MediaCodecVideoEncoder::EncodeOnCodecThread(
const webrtc::VideoFrame& frame,
- const std::vector<webrtc::VideoFrameType>* frame_types) {
+ const std::vector<webrtc::FrameType>* frame_types) {
CheckOnCodecThread();
JNIEnv* jni = AttachCurrentThreadIfNeeded();
ScopedLocalRefFrame local_ref_frame(jni);
diff --git a/talk/media/webrtc/fakewebrtcvideoengine.h b/talk/media/webrtc/fakewebrtcvideoengine.h
index d205839..8e4c7c8 100644
--- a/talk/media/webrtc/fakewebrtcvideoengine.h
+++ b/talk/media/webrtc/fakewebrtcvideoengine.h
@@ -155,10 +155,9 @@
return codec_settings_;
}
- virtual int32_t Encode(
- const webrtc::VideoFrame& inputImage,
- const webrtc::CodecSpecificInfo* codecSpecificInfo,
- const std::vector<webrtc::VideoFrameType>* frame_types) {
+ virtual int32_t Encode(const webrtc::VideoFrame& inputImage,
+ const webrtc::CodecSpecificInfo* codecSpecificInfo,
+ const std::vector<webrtc::FrameType>* frame_types) {
rtc::CritScope lock(&crit_);
++num_frames_encoded_;
return WEBRTC_VIDEO_CODEC_OK;
diff --git a/webrtc/common_types.h b/webrtc/common_types.h
index c11c4d7..6b624bf 100644
--- a/webrtc/common_types.h
+++ b/webrtc/common_types.h
@@ -156,15 +156,22 @@
kRecordingPreprocessing
};
-enum FrameType
-{
- kFrameEmpty = 0,
- kAudioFrameSpeech = 1,
- kAudioFrameCN = 2,
- kVideoFrameKey = 3, // independent frame
- kVideoFrameDelta = 4, // depends on the previus frame
+enum FrameType {
+ kEmptyFrame = 0,
+ kAudioFrameSpeech = 1,
+ kAudioFrameCN = 2,
+ kVideoFrameKey = 3,
+ kVideoFrameDelta = 4,
+ // TODO(pbos): Remove below aliases (non-kVideo prefixed) as soon as no
+ // VideoEncoder implementation in Chromium uses them.
+ kKeyFrame = kVideoFrameKey,
+ kDeltaFrame = kVideoFrameDelta,
};
+// TODO(pbos): Remove VideoFrameType when VideoEncoder implementations no longer
+// depend on it.
+using VideoFrameType = FrameType;
+
// Statistics for an RTCP channel
struct RtcpStatistics {
RtcpStatistics()
diff --git a/webrtc/frame_callback.h b/webrtc/frame_callback.h
index 3098ba7..b7f2210 100644
--- a/webrtc/frame_callback.h
+++ b/webrtc/frame_callback.h
@@ -21,7 +21,7 @@
struct EncodedFrame {
public:
- EncodedFrame() : data_(NULL), length_(0), frame_type_(kFrameEmpty) {}
+ EncodedFrame() : data_(NULL), length_(0), frame_type_(kEmptyFrame) {}
EncodedFrame(const uint8_t* data, size_t length, FrameType frame_type)
: data_(data), length_(length), frame_type_(frame_type) {}
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
index 9b5c17b..c6cd6de 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
@@ -46,7 +46,7 @@
: 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 @@
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;
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
index 0142275..12ea300 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
@@ -46,7 +46,7 @@
: 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 @@
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;
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
index a652278..879af49 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -171,7 +171,7 @@
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);
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
index 01c8bb8..e36d4e6 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
@@ -92,7 +92,7 @@
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 @@
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 @@
// 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,
diff --git a/webrtc/modules/audio_coding/main/test/Channel.cc b/webrtc/modules/audio_coding/main/test/Channel.cc
index 779718d..1b0a610 100644
--- a/webrtc/modules/audio_coding/main/test/Channel.cc
+++ b/webrtc/modules/audio_coding/main/test/Channel.cc
@@ -42,7 +42,7 @@
} 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;
diff --git a/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc b/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
index 21d97f1..85c2579 100644
--- a/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
+++ b/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
@@ -74,7 +74,7 @@
} else {
rtp_info.type.Audio.isCNG = false;
}
- if (frame_type == kFrameEmpty) {
+ if (frame_type == kEmptyFrame) {
// Skip this frame.
return 0;
}
diff --git a/webrtc/modules/audio_coding/main/test/TestStereo.cc b/webrtc/modules/audio_coding/main/test/TestStereo.cc
index 32ecadf..b0786be 100644
--- a/webrtc/modules/audio_coding/main/test/TestStereo.cc
+++ b/webrtc/modules/audio_coding/main/test/TestStereo.cc
@@ -58,7 +58,7 @@
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;
}
diff --git a/webrtc/modules/audio_coding/main/test/TestVADDTX.cc b/webrtc/modules/audio_coding/main/test/TestVADDTX.cc
index 0e42b9f..bd0335a 100644
--- a/webrtc/modules/audio_coding/main/test/TestVADDTX.cc
+++ b/webrtc/modules/audio_coding/main/test/TestVADDTX.cc
@@ -44,7 +44,7 @@
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 @@
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 @@
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
diff --git a/webrtc/modules/audio_coding/main/test/TestVADDTX.h b/webrtc/modules/audio_coding/main/test/TestVADDTX.h
index 8ef4228..d34b99f 100644
--- a/webrtc/modules/audio_coding/main/test/TestVADDTX.h
+++ b/webrtc/modules/audio_coding/main/test/TestVADDTX.h
@@ -29,7 +29,7 @@
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 @@
// 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)
diff --git a/webrtc/modules/audio_coding/main/test/utility.cc b/webrtc/modules/audio_coding/main/test/utility.cc
index 949ca617b..86e49f1 100644
--- a/webrtc/modules/audio_coding/main/test/utility.cc
+++ b/webrtc/modules/audio_coding/main/test/utility.cc
@@ -288,7 +288,7 @@
}
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]);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
index 3ad5686..1a14b55 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
@@ -83,7 +83,7 @@
fragmentation.fragmentationOffset[0] = 0;
fragmentation.fragmentationLength[0] = frame_size;
rtc::scoped_ptr<RtpPacketizer> packetizer(RtpPacketizer::Create(
- kRtpVideoH264, max_payload_size, NULL, kFrameEmpty));
+ kRtpVideoH264, max_payload_size, NULL, kEmptyFrame));
packetizer->SetPayloadData(frame.get(), frame_size, &fragmentation);
rtc::scoped_ptr<uint8_t[]> packet(new uint8_t[max_payload_size]);
@@ -157,7 +157,7 @@
fragmentation.fragmentationOffset[0] = 0;
fragmentation.fragmentationLength[0] = sizeof(frame);
rtc::scoped_ptr<RtpPacketizer> packetizer(
- RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kFrameEmpty));
+ RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kEmptyFrame));
packetizer->SetPayloadData(frame, sizeof(frame), &fragmentation);
uint8_t packet[kMaxPayloadSize] = {0};
size_t length = 0;
@@ -186,7 +186,7 @@
frame[fragmentation.fragmentationOffset[1]] = 0x01;
rtc::scoped_ptr<RtpPacketizer> packetizer(
- RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kFrameEmpty));
+ RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kEmptyFrame));
packetizer->SetPayloadData(frame, kFrameSize, &fragmentation);
uint8_t packet[kMaxPayloadSize] = {0};
@@ -223,7 +223,7 @@
fragmentation.fragmentationLength[2] =
kNalHeaderSize + kFrameSize - kPayloadOffset;
rtc::scoped_ptr<RtpPacketizer> packetizer(
- RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kFrameEmpty));
+ RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kEmptyFrame));
packetizer->SetPayloadData(frame, kFrameSize, &fragmentation);
uint8_t packet[kMaxPayloadSize] = {0};
@@ -258,7 +258,7 @@
fragmentation.fragmentationLength[2] =
kNalHeaderSize + kFrameSize - kPayloadOffset;
rtc::scoped_ptr<RtpPacketizer> packetizer(
- RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kFrameEmpty));
+ RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kEmptyFrame));
packetizer->SetPayloadData(frame, kFrameSize, &fragmentation);
uint8_t packet[kMaxPayloadSize] = {0};
@@ -306,7 +306,7 @@
}
}
rtc::scoped_ptr<RtpPacketizer> packetizer(
- RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kFrameEmpty));
+ RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kEmptyFrame));
packetizer->SetPayloadData(frame, kFrameSize, &fragmentation);
// First expecting two FU-A packets.
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 0a26f9e..1a56c63 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -35,7 +35,8 @@
const char* FrameTypeToString(FrameType frame_type) {
switch (frame_type) {
- case kFrameEmpty: return "empty";
+ case kEmptyFrame:
+ return "empty";
case kAudioFrameSpeech: return "audio_speech";
case kAudioFrameCN: return "audio_cn";
case kVideoFrameKey: return "video_key";
@@ -509,7 +510,7 @@
TRACE_EVENT_ASYNC_STEP1("webrtc", "Audio", capture_timestamp,
"Send", "type", FrameTypeToString(frame_type));
assert(frame_type == kAudioFrameSpeech || frame_type == kAudioFrameCN ||
- frame_type == kFrameEmpty);
+ frame_type == kEmptyFrame);
ret_val = audio_->SendAudio(frame_type, payload_type, capture_timestamp,
payload_data, payload_size, fragmentation);
@@ -518,7 +519,7 @@
"Send", "type", FrameTypeToString(frame_type));
assert(frame_type != kAudioFrameSpeech && frame_type != kAudioFrameCN);
- if (frame_type == kFrameEmpty)
+ if (frame_type == kEmptyFrame)
return 0;
ret_val =
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
index 3f55db4..30842bb 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -208,8 +208,8 @@
// A source MAY send events and coded audio packets for the same time
// but we don't support it
if (_dtmfEventIsOn) {
- if (frameType == kFrameEmpty) {
- // kFrameEmpty is used to drive the DTMF when in CN mode
+ if (frameType == kEmptyFrame) {
+ // kEmptyFrame is used to drive the DTMF when in CN mode
// it can be triggered more frequently than we want to send the
// DTMF packets.
if (packet_size_samples > (captureTimeStamp - _dtmfTimestampLastSent)) {
@@ -259,7 +259,7 @@
return 0;
}
if (payloadSize == 0 || payloadData == NULL) {
- if (frameType == kFrameEmpty) {
+ if (frameType == kEmptyFrame) {
// we don't send empty audio RTP packets
// no error since we use it to drive DTMF when we use VAD
return 0;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 6d30263..e4ace67 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -1266,7 +1266,7 @@
// audio channel.
// This test checks the marker bit for the first packet and the consequent
// packets of the same telephone event. Since it is specifically for DTMF
-// events, ignoring audio packets and sending kFrameEmpty instead of those.
+// events, ignoring audio packets and sending kEmptyFrame instead of those.
TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
char payload_name[RTP_PAYLOAD_NAME_SIZE] = "telephone-event";
uint8_t payload_type = 126;
@@ -1284,13 +1284,13 @@
// During start, it takes the starting timestamp as last sent timestamp.
// The duration is calculated as the difference of current and last sent
// timestamp. So for first call it will skip since the duration is zero.
- ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kFrameEmpty, payload_type,
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
capture_time_ms, 0, nullptr, 0,
nullptr));
// DTMF Sample Length is (Frequency/1000) * Duration.
// So in this case, it is (8000/1000) * 500 = 4000.
// Sending it as two packets.
- ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kFrameEmpty, payload_type,
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
capture_time_ms + 2000, 0, nullptr,
0, nullptr));
rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_parser(
@@ -1303,7 +1303,7 @@
// Marker Bit should be set to 1 for first packet.
EXPECT_TRUE(rtp_header.markerBit);
- ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kFrameEmpty, payload_type,
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
capture_time_ms + 4000, 0, nullptr,
0, nullptr));
ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_,
diff --git a/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.cc b/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.cc
index 69e52a5..22ace50 100644
--- a/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.cc
+++ b/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.cc
@@ -242,7 +242,7 @@
int H264VideoToolboxEncoder::Encode(
const VideoFrame& input_image,
const CodecSpecificInfo* codec_specific_info,
- const std::vector<VideoFrameType>* frame_types) {
+ const std::vector<FrameType>* frame_types) {
if (input_image.IsZeroSize()) {
// It's possible to get zero sizes as a signal to produce keyframes (this
// happens for internal sources). But this shouldn't happen in
diff --git a/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.h b/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.h
index 28cd63e..f4fb86f 100644
--- a/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.h
+++ b/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.h
@@ -38,7 +38,7 @@
int Encode(const VideoFrame& input_image,
const CodecSpecificInfo* codec_specific_info,
- const std::vector<VideoFrameType>* frame_types) override;
+ const std::vector<FrameType>* frame_types) override;
int RegisterEncodeCompleteCallback(EncodedImageCallback* callback) override;
diff --git a/webrtc/modules/video_coding/codecs/i420/main/interface/i420.h b/webrtc/modules/video_coding/codecs/i420/main/interface/i420.h
index 35f7b3e..e54e78d 100644
--- a/webrtc/modules/video_coding/codecs/i420/main/interface/i420.h
+++ b/webrtc/modules/video_coding/codecs/i420/main/interface/i420.h
@@ -50,7 +50,7 @@
// <0 - Error
int Encode(const VideoFrame& inputImage,
const CodecSpecificInfo* /*codecSpecificInfo*/,
- const std::vector<VideoFrameType>* /*frame_types*/) override;
+ const std::vector<FrameType>* /*frame_types*/) override;
// Register an encode complete callback object.
//
diff --git a/webrtc/modules/video_coding/codecs/i420/main/source/i420.cc b/webrtc/modules/video_coding/codecs/i420/main/source/i420.cc
index 5ac785a..065a2d7 100644
--- a/webrtc/modules/video_coding/codecs/i420/main/source/i420.cc
+++ b/webrtc/modules/video_coding/codecs/i420/main/source/i420.cc
@@ -74,7 +74,7 @@
int I420Encoder::Encode(const VideoFrame& inputImage,
const CodecSpecificInfo* /*codecSpecificInfo*/,
- const std::vector<VideoFrameType>* /*frame_types*/) {
+ const std::vector<FrameType>* /*frame_types*/) {
if (!_inited) {
return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
}
diff --git a/webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h b/webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h
index 5710446..5243d9a 100644
--- a/webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h
+++ b/webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h
@@ -35,7 +35,7 @@
MOCK_METHOD3(Encode,
int32_t(const VideoFrame& inputImage,
const CodecSpecificInfo* codecSpecificInfo,
- const std::vector<VideoFrameType>* frame_types));
+ const std::vector<FrameType>* frame_types));
MOCK_METHOD1(RegisterEncodeCompleteCallback,
int32_t(EncodedImageCallback* callback));
MOCK_METHOD0(Release, int32_t());
diff --git a/webrtc/modules/video_coding/codecs/test/stats.h b/webrtc/modules/video_coding/codecs/test/stats.h
index 8dc8f15..83ba108b 100644
--- a/webrtc/modules/video_coding/codecs/test/stats.h
+++ b/webrtc/modules/video_coding/codecs/test/stats.h
@@ -39,7 +39,7 @@
// Copied from EncodedImage
size_t encoded_frame_length_in_bytes;
- webrtc::VideoFrameType frame_type;
+ webrtc::FrameType frame_type;
};
// Handles statistics from a single video processing run.
diff --git a/webrtc/modules/video_coding/codecs/test/videoprocessor.cc b/webrtc/modules/video_coding/codecs/test/videoprocessor.cc
index 888adb8..3bb6b7a 100644
--- a/webrtc/modules/video_coding/codecs/test/videoprocessor.cc
+++ b/webrtc/modules/video_coding/codecs/test/videoprocessor.cc
@@ -162,7 +162,7 @@
return encoded_frame_size_;
}
-VideoFrameType VideoProcessorImpl::EncodedFrameType() {
+FrameType VideoProcessorImpl::EncodedFrameType() {
return encoded_frame_type_;
}
@@ -199,7 +199,7 @@
source_frame_.set_timestamp(frame_number);
// Decide if we're going to force a keyframe:
- std::vector<VideoFrameType> frame_types(1, kDeltaFrame);
+ std::vector<FrameType> frame_types(1, kDeltaFrame);
if (config_.keyframe_interval > 0 &&
frame_number % config_.keyframe_interval == 0) {
frame_types[0] = kKeyFrame;
diff --git a/webrtc/modules/video_coding/codecs/test/videoprocessor.h b/webrtc/modules/video_coding/codecs/test/videoprocessor.h
index 8c9cb12..4b6aed1 100644
--- a/webrtc/modules/video_coding/codecs/test/videoprocessor.h
+++ b/webrtc/modules/video_coding/codecs/test/videoprocessor.h
@@ -147,7 +147,7 @@
virtual size_t EncodedFrameSize() = 0;
// Return the encoded frame type (key or delta).
- virtual VideoFrameType EncodedFrameType() = 0;
+ virtual FrameType EncodedFrameType() = 0;
// Return the number of dropped frames.
virtual int NumberDroppedFrames() = 0;
@@ -183,7 +183,7 @@
// Return the size of the encoded frame in bytes.
size_t EncodedFrameSize() override;
// Return the encoded frame type (key or delta).
- VideoFrameType EncodedFrameType() override;
+ FrameType EncodedFrameType() override;
// Return the number of dropped frames.
int NumberDroppedFrames() override;
// Return the number of spatial resizes.
@@ -212,7 +212,7 @@
// If Init() has executed successfully.
bool initialized_;
size_t encoded_frame_size_;
- VideoFrameType encoded_frame_type_;
+ FrameType encoded_frame_type_;
int prev_time_stamp_;
int num_dropped_frames_;
int num_spatial_resizes_;
diff --git a/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc b/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
index 9f8ff49..ec9d756 100644
--- a/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
+++ b/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
@@ -268,7 +268,7 @@
}
// For every encoded frame, update the rate control metrics.
- void UpdateRateControlMetrics(int frame_num, VideoFrameType frame_type) {
+ void UpdateRateControlMetrics(int frame_num, FrameType frame_type) {
float encoded_size_kbits = processor_->EncodedFrameSize() * 8.0f / 1000.0f;
// Update layer data.
// Update rate mismatch relative to per-frame bandwidth for delta frames.
@@ -450,7 +450,7 @@
ResetRateControlMetrics(
rate_profile.frame_index_rate_update[update_index + 1]);
int frame_number = 0;
- VideoFrameType frame_type = kDeltaFrame;
+ FrameType frame_type = kDeltaFrame;
while (processor_->ProcessFrame(frame_number) &&
frame_number < num_frames) {
// Get the layer index for the frame |frame_number|.
diff --git a/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.cc b/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.cc
index 9ac2efa..4439267 100644
--- a/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.cc
+++ b/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.cc
@@ -233,7 +233,7 @@
int SimulcastEncoderAdapter::Encode(
const VideoFrame& input_image,
const CodecSpecificInfo* codec_specific_info,
- const std::vector<VideoFrameType>* frame_types) {
+ const std::vector<FrameType>* frame_types) {
if (!Initialized()) {
return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
}
@@ -267,7 +267,7 @@
if (!streaminfos_[stream_idx].send_stream)
continue;
- std::vector<VideoFrameType> stream_frame_types;
+ std::vector<FrameType> stream_frame_types;
if (send_key_frame) {
stream_frame_types.push_back(kKeyFrame);
streaminfos_[stream_idx].key_frame_request = false;
diff --git a/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h b/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h
index c00d8fb..afec024 100644
--- a/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h
+++ b/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h
@@ -42,7 +42,7 @@
size_t max_payload_size) override;
int Encode(const VideoFrame& input_image,
const CodecSpecificInfo* codec_specific_info,
- const std::vector<VideoFrameType>* frame_types) override;
+ const std::vector<FrameType>* frame_types) override;
int RegisterEncodeCompleteCallback(EncodedImageCallback* callback) override;
int SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
int SetRates(uint32_t new_bitrate_kbit, uint32_t new_framerate) override;
diff --git a/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter_unittest.cc b/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter_unittest.cc
index ed6541c..218b5e2 100644
--- a/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter_unittest.cc
+++ b/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter_unittest.cc
@@ -117,7 +117,7 @@
int32_t Encode(const VideoFrame& inputImage,
const CodecSpecificInfo* codecSpecificInfo,
- const std::vector<VideoFrameType>* frame_types) override {
+ const std::vector<FrameType>* frame_types) override {
return 0;
}
diff --git a/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h b/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h
index 672fa3a..a3d9e5a 100644
--- a/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h
+++ b/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h
@@ -338,7 +338,7 @@
decoder_->Release();
}
- void ExpectStreams(VideoFrameType frame_type, int expected_video_streams) {
+ void ExpectStreams(FrameType frame_type, int expected_video_streams) {
ASSERT_GE(expected_video_streams, 0);
ASSERT_LE(expected_video_streams, kNumberOfSimulcastStreams);
if (expected_video_streams >= 1) {
@@ -389,8 +389,7 @@
// a key frame was only requested for some of them.
void TestKeyFrameRequestsOnAllStreams() {
encoder_->SetRates(kMaxBitrates[2], 30); // To get all three streams.
- std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
- kDeltaFrame);
+ std::vector<FrameType> frame_types(kNumberOfSimulcastStreams, kDeltaFrame);
ExpectStreams(kKeyFrame, kNumberOfSimulcastStreams);
EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
@@ -424,8 +423,7 @@
void TestPaddingAllStreams() {
// We should always encode the base layer.
encoder_->SetRates(kMinBitrates[0] - 1, 30);
- std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
- kDeltaFrame);
+ std::vector<FrameType> frame_types(kNumberOfSimulcastStreams, kDeltaFrame);
ExpectStreams(kKeyFrame, 1);
EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
@@ -437,8 +435,7 @@
void TestPaddingTwoStreams() {
// We have just enough to get only the first stream and padding for two.
encoder_->SetRates(kMinBitrates[0], 30);
- std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
- kDeltaFrame);
+ std::vector<FrameType> frame_types(kNumberOfSimulcastStreams, kDeltaFrame);
ExpectStreams(kKeyFrame, 1);
EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
@@ -451,8 +448,7 @@
// We are just below limit of sending second stream, so we should get
// the first stream maxed out (at |maxBitrate|), and padding for two.
encoder_->SetRates(kTargetBitrates[0] + kMinBitrates[1] - 1, 30);
- std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
- kDeltaFrame);
+ std::vector<FrameType> frame_types(kNumberOfSimulcastStreams, kDeltaFrame);
ExpectStreams(kKeyFrame, 1);
EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
@@ -464,8 +460,7 @@
void TestPaddingOneStream() {
// We have just enough to send two streams, so padding for one stream.
encoder_->SetRates(kTargetBitrates[0] + kMinBitrates[1], 30);
- std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
- kDeltaFrame);
+ std::vector<FrameType> frame_types(kNumberOfSimulcastStreams, kDeltaFrame);
ExpectStreams(kKeyFrame, 2);
EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
@@ -479,8 +474,7 @@
// first stream's rate maxed out at |targetBitrate|, second at |maxBitrate|.
encoder_->SetRates(kTargetBitrates[0] + kTargetBitrates[1] +
kMinBitrates[2] - 1, 30);
- std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
- kDeltaFrame);
+ std::vector<FrameType> frame_types(kNumberOfSimulcastStreams, kDeltaFrame);
ExpectStreams(kKeyFrame, 2);
EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
@@ -493,8 +487,7 @@
// We have just enough to send all streams.
encoder_->SetRates(kTargetBitrates[0] + kTargetBitrates[1] +
kMinBitrates[2], 30);
- std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
- kDeltaFrame);
+ std::vector<FrameType> frame_types(kNumberOfSimulcastStreams, kDeltaFrame);
ExpectStreams(kKeyFrame, 3);
EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
@@ -507,8 +500,7 @@
// We should get three media streams.
encoder_->SetRates(kMaxBitrates[0] + kMaxBitrates[1] +
kMaxBitrates[2], 30);
- std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
- kDeltaFrame);
+ std::vector<FrameType> frame_types(kNumberOfSimulcastStreams, kDeltaFrame);
ExpectStreams(kKeyFrame, 3);
EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
@@ -589,8 +581,7 @@
// Encode one frame and verify.
encoder_->SetRates(kMaxBitrates[0] + kMaxBitrates[1], 30);
- std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
- kDeltaFrame);
+ std::vector<FrameType> frame_types(kNumberOfSimulcastStreams, kDeltaFrame);
EXPECT_CALL(encoder_callback_, Encoded(
AllOf(Field(&EncodedImage::_frameType, kKeyFrame),
Field(&EncodedImage::_encodedWidth, width),
diff --git a/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc b/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
index 2885bb8..d6f36f2 100644
--- a/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
+++ b/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
@@ -706,7 +706,7 @@
int VP8EncoderImpl::Encode(const VideoFrame& frame,
const CodecSpecificInfo* codec_specific_info,
- const std::vector<VideoFrameType>* frame_types) {
+ const std::vector<FrameType>* frame_types) {
TRACE_EVENT1("webrtc", "VP8::Encode", "timestamp", frame.timestamp());
if (!inited_)
diff --git a/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h b/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h
index 5ff1485..ba14ed5 100644
--- a/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h
+++ b/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h
@@ -48,7 +48,7 @@
virtual int Encode(const VideoFrame& input_image,
const CodecSpecificInfo* codec_specific_info,
- const std::vector<VideoFrameType>* frame_types);
+ const std::vector<FrameType>* frame_types);
virtual int RegisterEncodeCompleteCallback(EncodedImageCallback* callback);
diff --git a/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc b/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
index deb3bca..1a21371 100644
--- a/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
+++ b/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
@@ -422,7 +422,7 @@
int VP9EncoderImpl::Encode(const VideoFrame& input_image,
const CodecSpecificInfo* codec_specific_info,
- const std::vector<VideoFrameType>* frame_types) {
+ const std::vector<FrameType>* frame_types) {
if (!inited_) {
return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
}
@@ -432,7 +432,7 @@
if (encoded_complete_callback_ == NULL) {
return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
}
- VideoFrameType frame_type = kDeltaFrame;
+ FrameType frame_type = kDeltaFrame;
// We only support one stream at the moment.
if (frame_types && frame_types->size() > 0) {
frame_type = (*frame_types)[0];
diff --git a/webrtc/modules/video_coding/codecs/vp9/vp9_impl.h b/webrtc/modules/video_coding/codecs/vp9/vp9_impl.h
index c164a63..f9c1230 100644
--- a/webrtc/modules/video_coding/codecs/vp9/vp9_impl.h
+++ b/webrtc/modules/video_coding/codecs/vp9/vp9_impl.h
@@ -35,7 +35,7 @@
int Encode(const VideoFrame& input_image,
const CodecSpecificInfo* codec_specific_info,
- const std::vector<VideoFrameType>* frame_types) override;
+ const std::vector<FrameType>* frame_types) override;
int RegisterEncodeCompleteCallback(EncodedImageCallback* callback) override;
diff --git a/webrtc/modules/video_coding/main/source/decoding_state_unittest.cc b/webrtc/modules/video_coding/main/source/decoding_state_unittest.cc
index 10f1d6e..feae701 100644
--- a/webrtc/modules/video_coding/main/source/decoding_state_unittest.cc
+++ b/webrtc/modules/video_coding/main/source/decoding_state_unittest.cc
@@ -181,7 +181,7 @@
// Now insert empty packet belonging to the same frame.
packet.timestamp = 1;
packet.seqNum = 2;
- packet.frameType = kFrameEmpty;
+ packet.frameType = kEmptyFrame;
packet.sizeBytes = 0;
dec_state.UpdateOldPacket(&packet);
EXPECT_EQ(dec_state.sequence_num(), 2);
@@ -196,7 +196,7 @@
// sequence number.
packet.timestamp = 0;
packet.seqNum = 4;
- packet.frameType = kFrameEmpty;
+ packet.frameType = kEmptyFrame;
packet.sizeBytes = 0;
dec_state.UpdateOldPacket(&packet);
EXPECT_EQ(dec_state.sequence_num(), 3);
diff --git a/webrtc/modules/video_coding/main/source/encoded_frame.cc b/webrtc/modules/video_coding/main/source/encoded_frame.cc
index 646dae3..92d2cd0 100644
--- a/webrtc/modules/video_coding/main/source/encoded_frame.cc
+++ b/webrtc/modules/video_coding/main/source/encoded_frame.cc
@@ -226,38 +226,4 @@
}
}
-webrtc::FrameType VCMEncodedFrame::ConvertFrameType(VideoFrameType frameType)
-{
- switch(frameType) {
- case kKeyFrame:
- return kVideoFrameKey;
- case kDeltaFrame:
- return kVideoFrameDelta;
- }
- // Bogus default return value.
- return kVideoFrameDelta;
-}
-
-VideoFrameType VCMEncodedFrame::ConvertFrameType(webrtc::FrameType frame_type) {
- switch (frame_type) {
- case kVideoFrameKey:
- return kKeyFrame;
- case kVideoFrameDelta:
- return kDeltaFrame;
- default:
- assert(false);
- return kDeltaFrame;
- }
-}
-
-void VCMEncodedFrame::ConvertFrameTypes(
- const std::vector<webrtc::FrameType>& frame_types,
- std::vector<VideoFrameType>* video_frame_types) {
- assert(video_frame_types);
- video_frame_types->reserve(frame_types.size());
- for (size_t i = 0; i < frame_types.size(); ++i) {
- (*video_frame_types)[i] = ConvertFrameType(frame_types[i]);
- }
-}
-
-}
+} // namespace webrtc
diff --git a/webrtc/modules/video_coding/main/source/encoded_frame.h b/webrtc/modules/video_coding/main/source/encoded_frame.h
index 6caaf42..608578c 100644
--- a/webrtc/modules/video_coding/main/source/encoded_frame.h
+++ b/webrtc/modules/video_coding/main/source/encoded_frame.h
@@ -68,7 +68,7 @@
/**
* Get frame type
*/
- webrtc::FrameType FrameType() const {return ConvertFrameType(_frameType);}
+ webrtc::FrameType FrameType() const { return _frameType; }
/**
* Get frame rotation
*/
@@ -95,12 +95,6 @@
const RTPFragmentationHeader* FragmentationHeader() const;
- static webrtc::FrameType ConvertFrameType(VideoFrameType frameType);
- static VideoFrameType ConvertFrameType(webrtc::FrameType frameType);
- static void ConvertFrameTypes(
- const std::vector<webrtc::FrameType>& frame_types,
- std::vector<VideoFrameType>* video_frame_types);
-
protected:
/**
* Verifies that current allocated buffer size is larger than or equal to the input size.
diff --git a/webrtc/modules/video_coding/main/source/frame_buffer.cc b/webrtc/modules/video_coding/main/source/frame_buffer.cc
index 82a755a..192febe 100644
--- a/webrtc/modules/video_coding/main/source/frame_buffer.cc
+++ b/webrtc/modules/video_coding/main/source/frame_buffer.cc
@@ -98,7 +98,7 @@
// We only take the ntp timestamp of the first packet of a frame.
ntp_time_ms_ = packet.ntp_time_ms_;
_codec = packet.codec;
- if (packet.frameType != kFrameEmpty) {
+ if (packet.frameType != kEmptyFrame) {
// first media packet
SetState(kStateIncomplete);
}
@@ -280,7 +280,7 @@
#endif
// Transfer frame information to EncodedFrame and create any codec
// specific information.
- _frameType = ConvertFrameType(_sessionInfo.FrameType());
+ _frameType = _sessionInfo.FrameType();
_completeFrame = _sessionInfo.complete();
_missingFrame = !continuous;
}
diff --git a/webrtc/modules/video_coding/main/source/generic_encoder.cc b/webrtc/modules/video_coding/main/source/generic_encoder.cc
index 31c3f17..fe3d5cb 100644
--- a/webrtc/modules/video_coding/main/source/generic_encoder.cc
+++ b/webrtc/modules/video_coding/main/source/generic_encoder.cc
@@ -140,9 +140,8 @@
int32_t VCMGenericEncoder::Encode(const VideoFrame& inputFrame,
const CodecSpecificInfo* codecSpecificInfo,
const std::vector<FrameType>& frameTypes) {
- std::vector<VideoFrameType> video_frame_types(frameTypes.size(),
- kDeltaFrame);
- VCMEncodedFrame::ConvertFrameTypes(frameTypes, &video_frame_types);
+ for (FrameType frame_type : frameTypes)
+ RTC_DCHECK(frame_type == kVideoFrameKey || frame_type == kVideoFrameDelta);
rotation_ = inputFrame.rotation();
@@ -153,12 +152,11 @@
vcm_encoded_frame_callback_->SetRotation(rotation_);
}
- int32_t result =
- encoder_->Encode(inputFrame, codecSpecificInfo, &video_frame_types);
+ int32_t result = encoder_->Encode(inputFrame, codecSpecificInfo, &frameTypes);
if (is_screenshare_ &&
result == WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT) {
// Target bitrate exceeded, encoder state has been reset - try again.
- return encoder_->Encode(inputFrame, codecSpecificInfo, &video_frame_types);
+ return encoder_->Encode(inputFrame, codecSpecificInfo, &frameTypes);
}
return result;
@@ -223,10 +221,7 @@
int32_t VCMGenericEncoder::RequestFrame(
const std::vector<FrameType>& frame_types) {
VideoFrame image;
- std::vector<VideoFrameType> video_frame_types(frame_types.size(),
- kDeltaFrame);
- VCMEncodedFrame::ConvertFrameTypes(frame_types, &video_frame_types);
- return encoder_->Encode(image, NULL, &video_frame_types);
+ return encoder_->Encode(image, NULL, &frame_types);
}
int32_t
@@ -294,6 +289,8 @@
const EncodedImage& encodedImage,
const CodecSpecificInfo* codecSpecificInfo,
const RTPFragmentationHeader* fragmentationHeader) {
+ RTC_DCHECK(encodedImage._frameType == kVideoFrameKey ||
+ encodedImage._frameType == kVideoFrameDelta);
post_encode_callback_->Encoded(encodedImage, NULL, NULL);
if (_sendCallback == NULL) {
diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer.cc b/webrtc/modules/video_coding/main/source/jitter_buffer.cc
index 49c2325..92ffbc9 100644
--- a/webrtc/modules/video_coding/main/source/jitter_buffer.cc
+++ b/webrtc/modules/video_coding/main/source/jitter_buffer.cc
@@ -643,7 +643,7 @@
// Empty packets may bias the jitter estimate (lacking size component),
// therefore don't let empty packet trigger the following updates:
- if (packet.frameType != kFrameEmpty) {
+ if (packet.frameType != kEmptyFrame) {
if (waiting_for_completion_.timestamp == packet.timestamp) {
// This can get bad if we have a lot of duplicate packets,
// we will then count some packet multiple times.
diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc b/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc
index 7ba4d68..ab4d8cd 100644
--- a/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc
+++ b/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc
@@ -168,10 +168,9 @@
}
VCMFrameBufferEnum InsertFrame(FrameType frame_type) {
- stream_generator_->GenerateFrame(frame_type,
- (frame_type != kFrameEmpty) ? 1 : 0,
- (frame_type == kFrameEmpty) ? 1 : 0,
- clock_->TimeInMilliseconds());
+ stream_generator_->GenerateFrame(
+ frame_type, (frame_type != kEmptyFrame) ? 1 : 0,
+ (frame_type == kEmptyFrame) ? 1 : 0, clock_->TimeInMilliseconds());
VCMFrameBufferEnum ret = InsertPacketAndPop(0);
clock_->AdvanceTimeMilliseconds(kDefaultFramePeriodMs);
return ret;
@@ -1050,7 +1049,7 @@
packet_->markerBit = false;
packet_->seqNum = seq_num_;
packet_->completeNALU = kNaluEnd;
- packet_->frameType = kFrameEmpty;
+ packet_->frameType = kEmptyFrame;
EXPECT_EQ(jitter_buffer_->InsertPacket(*packet_, &retransmitted),
kDecodableSession);
@@ -1524,7 +1523,7 @@
packet_->markerBit = false;
packet_->seqNum = seq_num_;
packet_->timestamp = timestamp_;
- packet_->frameType = kFrameEmpty;
+ packet_->frameType = kEmptyFrame;
EXPECT_EQ(kNoError, jitter_buffer_->InsertPacket(*packet_,
&retransmitted));
@@ -1895,7 +1894,7 @@
TEST_F(TestJitterBufferNack, EmptyPackets) {
// Make sure empty packets doesn't clog the jitter buffer.
jitter_buffer_->SetNackMode(kNack, media_optimization::kLowRttNackMs, -1);
- EXPECT_GE(InsertFrames(kMaxNumberOfFrames, kFrameEmpty), kNoError);
+ EXPECT_GE(InsertFrames(kMaxNumberOfFrames, kEmptyFrame), kNoError);
InsertFrame(kVideoFrameKey);
EXPECT_TRUE(DecodeCompleteFrame());
}
diff --git a/webrtc/modules/video_coding/main/source/packet.cc b/webrtc/modules/video_coding/main/source/packet.cc
index 88838f3..fd5a6ab 100644
--- a/webrtc/modules/video_coding/main/source/packet.cc
+++ b/webrtc/modules/video_coding/main/source/packet.cc
@@ -16,23 +16,21 @@
namespace webrtc {
VCMPacket::VCMPacket()
- :
- payloadType(0),
- timestamp(0),
- ntp_time_ms_(0),
- seqNum(0),
- dataPtr(NULL),
- sizeBytes(0),
- markerBit(false),
- frameType(kFrameEmpty),
- codec(kVideoCodecUnknown),
- isFirstPacket(false),
- completeNALU(kNaluUnset),
- insertStartCode(false),
- width(0),
- height(0),
- codecSpecificHeader() {
-}
+ : payloadType(0),
+ timestamp(0),
+ ntp_time_ms_(0),
+ seqNum(0),
+ dataPtr(NULL),
+ sizeBytes(0),
+ markerBit(false),
+ frameType(kEmptyFrame),
+ codec(kVideoCodecUnknown),
+ isFirstPacket(false),
+ completeNALU(kNaluUnset),
+ insertStartCode(false),
+ width(0),
+ height(0),
+ codecSpecificHeader() {}
VCMPacket::VCMPacket(const uint8_t* ptr,
const size_t size,
@@ -88,7 +86,7 @@
dataPtr = NULL;
sizeBytes = 0;
markerBit = false;
- frameType = kFrameEmpty;
+ frameType = kEmptyFrame;
codec = kVideoCodecUnknown;
isFirstPacket = false;
completeNALU = kNaluUnset;
diff --git a/webrtc/modules/video_coding/main/source/receiver_unittest.cc b/webrtc/modules/video_coding/main/source/receiver_unittest.cc
index eb5e471..707e1f1 100644
--- a/webrtc/modules/video_coding/main/source/receiver_unittest.cc
+++ b/webrtc/modules/video_coding/main/source/receiver_unittest.cc
@@ -63,10 +63,8 @@
int32_t InsertFrame(FrameType frame_type, bool complete) {
int num_of_packets = complete ? 1 : 2;
stream_generator_->GenerateFrame(
- frame_type,
- (frame_type != kFrameEmpty) ? num_of_packets : 0,
- (frame_type == kFrameEmpty) ? 1 : 0,
- clock_->TimeInMilliseconds());
+ frame_type, (frame_type != kEmptyFrame) ? num_of_packets : 0,
+ (frame_type == kEmptyFrame) ? 1 : 0, clock_->TimeInMilliseconds());
int32_t ret = InsertPacketAndPop(0);
if (!complete) {
// Drop the second packet.
diff --git a/webrtc/modules/video_coding/main/source/session_info.cc b/webrtc/modules/video_coding/main/source/session_info.cc
index bf6bcb3..7a32504 100644
--- a/webrtc/modules/video_coding/main/source/session_info.cc
+++ b/webrtc/modules/video_coding/main/source/session_info.cc
@@ -464,7 +464,7 @@
uint8_t* frame_buffer,
VCMDecodeErrorMode decode_error_mode,
const FrameData& frame_data) {
- if (packet.frameType == kFrameEmpty) {
+ if (packet.frameType == kEmptyFrame) {
// Update sequence number of an empty packet.
// Only media packets are inserted into the packet list.
InformOfEmptyPacket(packet.seqNum);
@@ -516,7 +516,7 @@
LOG(LS_WARNING) << "Received packet with a sequence number which is out "
"of frame boundaries";
return -3;
- } else if (frame_type_ == kFrameEmpty && packet.frameType != kFrameEmpty) {
+ } else if (frame_type_ == kEmptyFrame && packet.frameType != kEmptyFrame) {
// Update the frame type with the type of the first media packet.
// TODO(mikhal): Can this trigger?
frame_type_ = packet.frameType;
diff --git a/webrtc/modules/video_coding/main/source/session_info_unittest.cc b/webrtc/modules/video_coding/main/source/session_info_unittest.cc
index fae55f4..58c352d 100644
--- a/webrtc/modules/video_coding/main/source/session_info_unittest.cc
+++ b/webrtc/modules/video_coding/main/source/session_info_unittest.cc
@@ -175,7 +175,7 @@
packet_.markerBit = true;
packet_.seqNum = 2;
packet_.sizeBytes = 0;
- packet_.frameType = kFrameEmpty;
+ packet_.frameType = kEmptyFrame;
EXPECT_EQ(0,
session_.InsertPacket(packet_,
frame_buffer_,
@@ -888,7 +888,7 @@
TEST_F(TestNalUnits, OnlyReceivedEmptyPacket) {
packet_.isFirstPacket = false;
packet_.completeNALU = kNaluComplete;
- packet_.frameType = kFrameEmpty;
+ packet_.frameType = kEmptyFrame;
packet_.sizeBytes = 0;
packet_.seqNum = 0;
packet_.markerBit = false;
diff --git a/webrtc/modules/video_coding/main/source/test/stream_generator.cc b/webrtc/modules/video_coding/main/source/test/stream_generator.cc
index dcf9b68..664764a 100644
--- a/webrtc/modules/video_coding/main/source/test/stream_generator.cc
+++ b/webrtc/modules/video_coding/main/source/test/stream_generator.cc
@@ -46,8 +46,8 @@
++sequence_number_;
}
for (int i = 0; i < num_empty_packets; ++i) {
- packets_.push_back(GeneratePacket(
- sequence_number_, timestamp, 0, false, false, kFrameEmpty));
+ packets_.push_back(GeneratePacket(sequence_number_, timestamp, 0, false,
+ false, kEmptyFrame));
++sequence_number_;
}
}
diff --git a/webrtc/modules/video_coding/main/source/video_receiver_unittest.cc b/webrtc/modules/video_coding/main/source/video_receiver_unittest.cc
index 1f1996e..16eaed3 100644
--- a/webrtc/modules/video_coding/main/source/video_receiver_unittest.cc
+++ b/webrtc/modules/video_coding/main/source/video_receiver_unittest.cc
@@ -88,7 +88,7 @@
const uint8_t payload[kPaddingSize] = {0};
WebRtcRTPHeader header;
memset(&header, 0, sizeof(header));
- header.frameType = kFrameEmpty;
+ header.frameType = kEmptyFrame;
header.header.markerBit = false;
header.header.paddingLength = kPaddingSize;
header.header.payloadType = kUnusedPayloadType;
@@ -112,7 +112,7 @@
const uint8_t payload[kFrameSize] = {0};
WebRtcRTPHeader header;
memset(&header, 0, sizeof(header));
- header.frameType = kFrameEmpty;
+ header.frameType = kEmptyFrame;
header.header.markerBit = false;
header.header.paddingLength = kPaddingSize;
header.header.payloadType = kUnusedPayloadType;
@@ -127,7 +127,7 @@
clock_.AdvanceTimeMilliseconds(33);
header.header.timestamp += 3000;
- header.frameType = kFrameEmpty;
+ header.frameType = kEmptyFrame;
header.type.Video.isFirstPacket = false;
header.header.markerBit = false;
// Insert padding frames.
@@ -163,7 +163,7 @@
const uint8_t payload[kFrameSize] = {0};
WebRtcRTPHeader header;
memset(&header, 0, sizeof(header));
- header.frameType = kFrameEmpty;
+ header.frameType = kEmptyFrame;
header.type.Video.isFirstPacket = false;
header.header.markerBit = false;
header.header.paddingLength = kPaddingSize;
@@ -188,7 +188,7 @@
}
// Insert 2 padding only frames.
- header.frameType = kFrameEmpty;
+ header.frameType = kEmptyFrame;
header.type.Video.isFirstPacket = false;
header.header.markerBit = false;
for (int j = 0; j < 2; ++j) {
diff --git a/webrtc/modules/video_coding/main/source/video_sender.cc b/webrtc/modules/video_coding/main/source/video_sender.cc
index a5ef22a..8401cb2 100644
--- a/webrtc/modules/video_coding/main/source/video_sender.cc
+++ b/webrtc/modules/video_coding/main/source/video_sender.cc
@@ -313,7 +313,7 @@
}
// TODO(holmer): Add support for dropping frames per stream. Currently we
// only have one frame dropper for all streams.
- if (_nextFrameTypes[0] == kFrameEmpty) {
+ if (_nextFrameTypes[0] == kEmptyFrame) {
return VCM_OK;
}
if (_mediaOpt.DropFrame()) {
diff --git a/webrtc/modules/video_coding/main/source/video_sender_unittest.cc b/webrtc/modules/video_coding/main/source/video_sender_unittest.cc
index 31c7173..13e6b99 100644
--- a/webrtc/modules/video_coding/main/source/video_sender_unittest.cc
+++ b/webrtc/modules/video_coding/main/source/video_sender_unittest.cc
@@ -244,7 +244,7 @@
}
assert(stream >= 0);
assert(stream < kNumberOfStreams);
- std::vector<VideoFrameType> frame_types(kNumberOfStreams, kDeltaFrame);
+ std::vector<FrameType> frame_types(kNumberOfStreams, kDeltaFrame);
frame_types[stream] = kKeyFrame;
EXPECT_CALL(
encoder_,
diff --git a/webrtc/test/configurable_frame_size_encoder.cc b/webrtc/test/configurable_frame_size_encoder.cc
index ea42e96..2dbda38 100644
--- a/webrtc/test/configurable_frame_size_encoder.cc
+++ b/webrtc/test/configurable_frame_size_encoder.cc
@@ -41,7 +41,7 @@
int32_t ConfigurableFrameSizeEncoder::Encode(
const VideoFrame& inputImage,
const CodecSpecificInfo* codecSpecificInfo,
- const std::vector<VideoFrameType>* frame_types) {
+ const std::vector<FrameType>* frame_types) {
EncodedImage encodedImage(
buffer_.get(), current_frame_size_, max_frame_size_);
encodedImage._completeFrame = true;
diff --git a/webrtc/test/configurable_frame_size_encoder.h b/webrtc/test/configurable_frame_size_encoder.h
index ffc095b..79514ef 100644
--- a/webrtc/test/configurable_frame_size_encoder.h
+++ b/webrtc/test/configurable_frame_size_encoder.h
@@ -30,7 +30,7 @@
int32_t Encode(const VideoFrame& input_image,
const CodecSpecificInfo* codec_specific_info,
- const std::vector<VideoFrameType>* frame_types) override;
+ const std::vector<FrameType>* frame_types) override;
int32_t RegisterEncodeCompleteCallback(
EncodedImageCallback* callback) override;
diff --git a/webrtc/test/fake_encoder.cc b/webrtc/test/fake_encoder.cc
index f85fb9a..db7a587 100644
--- a/webrtc/test/fake_encoder.cc
+++ b/webrtc/test/fake_encoder.cc
@@ -47,7 +47,7 @@
int32_t FakeEncoder::Encode(const VideoFrame& input_image,
const CodecSpecificInfo* codec_specific_info,
- const std::vector<VideoFrameType>* frame_types) {
+ const std::vector<FrameType>* frame_types) {
assert(config_.maxFramerate > 0);
int64_t time_since_last_encode_ms = 1000 / config_.maxFramerate;
int64_t time_now_ms = clock_->TimeInMilliseconds();
@@ -189,7 +189,7 @@
int32_t DelayedEncoder::Encode(const VideoFrame& input_image,
const CodecSpecificInfo* codec_specific_info,
- const std::vector<VideoFrameType>* frame_types) {
+ const std::vector<FrameType>* frame_types) {
SleepMs(delay_ms_);
return FakeEncoder::Encode(input_image, codec_specific_info, frame_types);
}
diff --git a/webrtc/test/fake_encoder.h b/webrtc/test/fake_encoder.h
index 3dce336..dc3a1ae 100644
--- a/webrtc/test/fake_encoder.h
+++ b/webrtc/test/fake_encoder.h
@@ -33,7 +33,7 @@
size_t max_payload_size) override;
int32_t Encode(const VideoFrame& input_image,
const CodecSpecificInfo* codec_specific_info,
- const std::vector<VideoFrameType>* frame_types) override;
+ const std::vector<FrameType>* frame_types) override;
int32_t RegisterEncodeCompleteCallback(
EncodedImageCallback* callback) override;
int32_t Release() override;
@@ -74,7 +74,7 @@
int32_t Encode(const VideoFrame& input_image,
const CodecSpecificInfo* codec_specific_info,
- const std::vector<VideoFrameType>* frame_types) override;
+ const std::vector<FrameType>* frame_types) override;
private:
const int delay_ms_;
diff --git a/webrtc/video/encoded_frame_callback_adapter.cc b/webrtc/video/encoded_frame_callback_adapter.cc
index 6726a37..407801f 100644
--- a/webrtc/video/encoded_frame_callback_adapter.cc
+++ b/webrtc/video/encoded_frame_callback_adapter.cc
@@ -27,11 +27,8 @@
const CodecSpecificInfo* codecSpecificInfo,
const RTPFragmentationHeader* fragmentation) {
RTC_DCHECK(observer_ != nullptr);
- FrameType frame_type =
- VCMEncodedFrame::ConvertFrameType(encodedImage._frameType);
- const EncodedFrame frame(encodedImage._buffer,
- encodedImage._length,
- frame_type);
+ const EncodedFrame frame(encodedImage._buffer, encodedImage._length,
+ encodedImage._frameType);
observer_->EncodedFrameCallback(frame);
return 0;
}
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index 01f132f..d889290 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -1589,7 +1589,7 @@
public:
EncodedFrameTestObserver()
: length_(0),
- frame_type_(kFrameEmpty),
+ frame_type_(kEmptyFrame),
called_(EventWrapper::Create()) {}
virtual ~EncodedFrameTestObserver() {}
@@ -2957,7 +2957,7 @@
int32_t Encode(const VideoFrame& input_image,
const CodecSpecificInfo* codec_specific_info,
- const std::vector<VideoFrameType>* frame_types) override {
+ const std::vector<FrameType>* frame_types) override {
{
rtc::CritScope lock(&test_crit_);
if (sender_state_ == kNetworkDown) {
@@ -3080,7 +3080,7 @@
UnusedEncoder() : FakeEncoder(Clock::GetRealTimeClock()) {}
int32_t Encode(const VideoFrame& input_image,
const CodecSpecificInfo* codec_specific_info,
- const std::vector<VideoFrameType>* frame_types) override {
+ const std::vector<FrameType>* frame_types) override {
ADD_FAILURE() << "Unexpected frame encode.";
return test::FakeEncoder::Encode(
input_image, codec_specific_info, frame_types);
diff --git a/webrtc/video/video_encoder.cc b/webrtc/video/video_encoder.cc
index 305406b..3e6ce63 100644
--- a/webrtc/video/video_encoder.cc
+++ b/webrtc/video/video_encoder.cc
@@ -99,7 +99,7 @@
int32_t VideoEncoderSoftwareFallbackWrapper::Encode(
const VideoFrame& frame,
const CodecSpecificInfo* codec_specific_info,
- const std::vector<VideoFrameType>* frame_types) {
+ const std::vector<FrameType>* frame_types) {
if (fallback_encoder_)
return fallback_encoder_->Encode(frame, codec_specific_info, frame_types);
return encoder_->Encode(frame, codec_specific_info, frame_types);
diff --git a/webrtc/video/video_encoder_unittest.cc b/webrtc/video/video_encoder_unittest.cc
index be0170c..5f2c37a 100644
--- a/webrtc/video/video_encoder_unittest.cc
+++ b/webrtc/video/video_encoder_unittest.cc
@@ -32,7 +32,7 @@
}
int32_t Encode(const VideoFrame& frame,
const CodecSpecificInfo* codec_specific_info,
- const std::vector<VideoFrameType>* frame_types) override {
+ const std::vector<FrameType>* frame_types) override {
++encode_count_;
return WEBRTC_VIDEO_CODEC_OK;
}
@@ -120,7 +120,7 @@
memset(frame_.buffer(webrtc::kVPlane), 128,
frame_.allocated_size(webrtc::kVPlane));
- std::vector<VideoFrameType> types(1, kKeyFrame);
+ std::vector<FrameType> types(1, kKeyFrame);
EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
fallback_wrapper_.Encode(frame_, nullptr, &types));
EXPECT_EQ(0, fake_encoder_.encode_count_);
@@ -163,7 +163,7 @@
EXPECT_EQ(&callback2, fake_encoder_.encode_complete_callback_);
// Encoding a frame using the fallback should arrive at the new callback.
- std::vector<VideoFrameType> types(1, kKeyFrame);
+ std::vector<FrameType> types(1, kKeyFrame);
frame_.set_timestamp(frame_.timestamp() + 1000);
EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
fallback_wrapper_.Encode(frame_, nullptr, &types));
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index e16b670..e30c903 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -1252,7 +1252,7 @@
int32_t Encode(const VideoFrame& inputImage,
const CodecSpecificInfo* codecSpecificInfo,
- const std::vector<VideoFrameType>* frame_types) override {
+ const std::vector<FrameType>* frame_types) override {
EXPECT_TRUE(IsReadyForEncode());
observation_complete_->Set();
@@ -1451,7 +1451,7 @@
int32_t Encode(const VideoFrame& input_image,
const CodecSpecificInfo* codec_specific_info,
- const std::vector<VideoFrameType>* frame_types) override {
+ const std::vector<FrameType>* frame_types) override {
// Silently skip the encode, FakeEncoder::Encode doesn't produce VP8.
return 0;
}
@@ -1742,7 +1742,7 @@
private:
int32_t Encode(const VideoFrame& input_image,
const CodecSpecificInfo* codecSpecificInfo,
- const std::vector<VideoFrameType>* frame_types) override {
+ const std::vector<FrameType>* frame_types) override {
CodecSpecificInfo specifics;
memset(&specifics, 0, sizeof(specifics));
specifics.codecType = kVideoCodecGeneric;
diff --git a/webrtc/video_encoder.h b/webrtc/video_encoder.h
index a512c98..0858a71 100644
--- a/webrtc/video_encoder.h
+++ b/webrtc/video_encoder.h
@@ -98,7 +98,7 @@
// WEBRTC_VIDEO_CODEC_TIMEOUT
virtual int32_t Encode(const VideoFrame& frame,
const CodecSpecificInfo* codec_specific_info,
- const std::vector<VideoFrameType>* frame_types) = 0;
+ const std::vector<FrameType>* frame_types) = 0;
// Inform the encoder of the new packet loss rate and the round-trip time of
// the network.
@@ -147,7 +147,7 @@
int32_t Release() override;
int32_t Encode(const VideoFrame& frame,
const CodecSpecificInfo* codec_specific_info,
- const std::vector<VideoFrameType>* frame_types) override;
+ const std::vector<FrameType>* frame_types) override;
int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
int32_t SetRates(uint32_t bitrate, uint32_t framerate) override;
diff --git a/webrtc/video_engine/vie_channel.h b/webrtc/video_engine/vie_channel.h
index 488923d..cd0646b 100644
--- a/webrtc/video_engine/vie_channel.h
+++ b/webrtc/video_engine/vie_channel.h
@@ -251,10 +251,10 @@
int min_playout_delay_ms,
int render_delay_ms);
- // Implements VideoFrameTypeCallback.
+ // Implements FrameTypeCallback.
virtual int32_t RequestKeyFrame();
- // Implements VideoFrameTypeCallback.
+ // Implements FrameTypeCallback.
virtual int32_t SliceLossIndicationRequest(
const uint64_t picture_id);
diff --git a/webrtc/video_engine/vie_encoder.cc b/webrtc/video_engine/vie_encoder.cc
index 45e9612..4d41e0e 100644
--- a/webrtc/video_engine/vie_encoder.cc
+++ b/webrtc/video_engine/vie_encoder.cc
@@ -558,10 +558,11 @@
stats_proxy_->OnSendEncodedImage(encoded_image, rtp_video_hdr);
return send_payload_router_->RoutePayload(
- VCMEncodedFrame::ConvertFrameType(encoded_image._frameType), payload_type,
- encoded_image._timeStamp, encoded_image.capture_time_ms_,
- encoded_image._buffer, encoded_image._length, &fragmentation_header,
- rtp_video_hdr) ? 0 : -1;
+ encoded_image._frameType, payload_type, encoded_image._timeStamp,
+ encoded_image.capture_time_ms_, encoded_image._buffer,
+ encoded_image._length, &fragmentation_header, rtp_video_hdr)
+ ? 0
+ : -1;
}
int32_t ViEEncoder::SendStatistics(const uint32_t bit_rate,
diff --git a/webrtc/video_frame.h b/webrtc/video_frame.h
index 3c0ad0c..1968a69 100644
--- a/webrtc/video_frame.h
+++ b/webrtc/video_frame.h
@@ -12,6 +12,7 @@
#define WEBRTC_VIDEO_FRAME_H_
#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/common_types.h"
#include "webrtc/common_video/interface/video_frame_buffer.h"
#include "webrtc/common_video/rotation.h"
#include "webrtc/typedefs.h"
@@ -166,11 +167,6 @@
VideoRotation rotation_;
};
-enum VideoFrameType {
- kKeyFrame = 0,
- kDeltaFrame = 1,
-};
-
// TODO(pbos): Rename EncodedFrame and reformat this class' members.
class EncodedImage {
public:
@@ -192,8 +188,7 @@
// NTP time of the capture time in local timebase in milliseconds.
int64_t ntp_time_ms_ = 0;
int64_t capture_time_ms_ = 0;
- // TODO(pbos): Use webrtc::FrameType directly (and remove VideoFrameType).
- VideoFrameType _frameType = kDeltaFrame;
+ FrameType _frameType = kDeltaFrame;
uint8_t* _buffer;
size_t _length;
size_t _size;
