Make AudioFrameType an enum class, and move to audio_coding_module_typedefs.h
Bug: webrtc:5876
Change-Id: I0c92f9410fcf0832bfa321229b3437134255dba6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128085
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27190}diff --git a/modules/audio_coding/acm2/acm_receiver_unittest.cc b/modules/audio_coding/acm2/acm_receiver_unittest.cc
index 7667b71..747d4a3 100644
--- a/modules/audio_coding/acm2/acm_receiver_unittest.cc
+++ b/modules/audio_coding/acm2/acm_receiver_unittest.cc
@@ -36,7 +36,7 @@
: timestamp_(0),
packet_sent_(false),
last_packet_send_timestamp_(timestamp_),
- last_frame_type_(kEmptyFrame) {
+ last_frame_type_(AudioFrameType::kEmptyFrame) {
config_.decoder_factory = decoder_factory_;
}
@@ -109,7 +109,7 @@
const uint8_t* payload_data,
size_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) override {
- if (frame_type == kEmptyFrame)
+ if (frame_type == AudioFrameType::kEmptyFrame)
return 0;
rtp_header_.payloadType = payload_type;
@@ -336,7 +336,7 @@
SetEncoder(0, codecs.at(0), cng_payload_types)); // Enough to test
// with one codec.
ASSERT_TRUE(packet_sent_);
- EXPECT_EQ(kAudioFrameCN, last_frame_type_);
+ EXPECT_EQ(AudioFrameType::kAudioFrameCN, last_frame_type_);
// Has received, only, DTX. Last Audio codec is undefined.
EXPECT_EQ(absl::nullopt, receiver_->LastDecoder());
@@ -353,7 +353,7 @@
// Sanity check if Actually an audio payload received, and it should be
// of type "speech."
ASSERT_TRUE(packet_sent_);
- ASSERT_EQ(kAudioFrameSpeech, last_frame_type_);
+ ASSERT_EQ(AudioFrameType::kAudioFrameSpeech, last_frame_type_);
EXPECT_EQ(info_without_cng.sample_rate_hz,
receiver_->last_packet_sample_rate_hz());
@@ -361,7 +361,7 @@
// the expected codec. Encode repeatedly until a DTX is sent.
const AudioCodecInfo info_with_cng =
SetEncoder(payload_type, codecs.at(i), cng_payload_types);
- while (last_frame_type_ != kAudioFrameCN) {
+ while (last_frame_type_ != AudioFrameType::kAudioFrameCN) {
packet_sent_ = false;
InsertOnePacketOfSilence(info_with_cng);
ASSERT_TRUE(packet_sent_);
diff --git a/modules/audio_coding/acm2/acm_send_test.cc b/modules/audio_coding/acm2/acm_send_test.cc
index 4c34e41..c558f7b 100644
--- a/modules/audio_coding/acm2/acm_send_test.cc
+++ b/modules/audio_coding/acm2/acm_send_test.cc
@@ -44,7 +44,7 @@
static_cast<size_t>(source_rate_hz_ * kBlockSizeMs / 1000)),
codec_registered_(false),
test_duration_ms_(test_duration_ms),
- frame_type_(kAudioFrameSpeech),
+ frame_type_(AudioFrameType::kAudioFrameSpeech),
payload_type_(0),
timestamp_(0),
sequence_number_(0) {
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index a4b64b1..b5c5973 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -395,11 +395,12 @@
ConvertEncodedInfoToFragmentationHeader(encoded_info, &my_fragmentation);
AudioFrameType frame_type;
if (encode_buffer_.size() == 0 && encoded_info.send_even_if_empty) {
- frame_type = kEmptyFrame;
+ frame_type = AudioFrameType::kEmptyFrame;
encoded_info.payload_type = previous_pltype;
} else {
RTC_DCHECK_GT(encode_buffer_.size(), 0);
- frame_type = encoded_info.speech ? kAudioFrameSpeech : kAudioFrameCN;
+ frame_type = encoded_info.speech ? AudioFrameType::kAudioFrameSpeech
+ : AudioFrameType::kAudioFrameCN;
}
{
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index 797b9b1..e64077e 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -100,7 +100,7 @@
public:
PacketizationCallbackStubOldApi()
: num_calls_(0),
- last_frame_type_(kEmptyFrame),
+ last_frame_type_(AudioFrameType::kEmptyFrame),
last_payload_type_(-1),
last_timestamp_(0) {}
@@ -350,11 +350,12 @@
for (int i = 0; i < kLoops; ++i) {
EXPECT_EQ(i / k10MsBlocksPerPacket, packet_cb_.num_calls());
if (packet_cb_.num_calls() > 0)
- EXPECT_EQ(kAudioFrameSpeech, packet_cb_.last_frame_type());
+ EXPECT_EQ(AudioFrameType::kAudioFrameSpeech,
+ packet_cb_.last_frame_type());
InsertAudioAndVerifyEncoding();
}
EXPECT_EQ(kLoops / k10MsBlocksPerPacket, packet_cb_.num_calls());
- EXPECT_EQ(kAudioFrameSpeech, packet_cb_.last_frame_type());
+ EXPECT_EQ(AudioFrameType::kAudioFrameSpeech, packet_cb_.last_frame_type());
}
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
@@ -431,12 +432,19 @@
const struct {
int ix;
AudioFrameType type;
- } expectation[] = {
- {2, kAudioFrameCN}, {5, kEmptyFrame}, {8, kEmptyFrame},
- {11, kAudioFrameCN}, {14, kEmptyFrame}, {17, kEmptyFrame},
- {20, kAudioFrameCN}, {23, kEmptyFrame}, {26, kEmptyFrame},
- {29, kEmptyFrame}, {32, kAudioFrameCN}, {35, kEmptyFrame},
- {38, kEmptyFrame}};
+ } expectation[] = {{2, AudioFrameType::kAudioFrameCN},
+ {5, AudioFrameType::kEmptyFrame},
+ {8, AudioFrameType::kEmptyFrame},
+ {11, AudioFrameType::kAudioFrameCN},
+ {14, AudioFrameType::kEmptyFrame},
+ {17, AudioFrameType::kEmptyFrame},
+ {20, AudioFrameType::kAudioFrameCN},
+ {23, AudioFrameType::kEmptyFrame},
+ {26, AudioFrameType::kEmptyFrame},
+ {29, AudioFrameType::kEmptyFrame},
+ {32, AudioFrameType::kAudioFrameCN},
+ {35, AudioFrameType::kEmptyFrame},
+ {38, AudioFrameType::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);