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Make AudioFrameType an enum class, and move to audio_coding_module_typedefs.h

Browse Source 
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}
This commit is contained in:
Niels Möller
2019-03-19 14:10:16 +01:00
committed by Commit Bot
parent 1d14af908e
commit c936cb6a86
21 changed files with 91 additions and 102 deletions
Download Patch File Download Diff File Expand all files Collapse all files

36
modules/audio_coding/BUILD.gn
View File

@ -15,11 +15,11 @@ if (rtc_enable_protobuf) {
visibility = [ ":*" ]
rtc_source_set("audio_coding_module_typedefs") {
visibility += [ "*" ]
sources = [
"include/audio_coding_module_typedefs.h",
]
deps = [
"../..:webrtc_common",
"../../rtc_base:deprecation",
]
}
@ -44,7 +44,6 @@ rtc_static_library("audio_coding") {
":neteq",
"..:module_api",
"..:module_api_public",
"../..:webrtc_common",
"../../api:array_view",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:audio_codecs_api",
@ -137,7 +136,6 @@ rtc_static_library("g711") {
deps = [
":legacy_encoded_audio_frame",
"../..:webrtc_common",
"../../api:array_view",
"../../api/audio_codecs:audio_codecs_api",
"../../rtc_base:checks",
@ -155,7 +153,6 @@ rtc_source_set("g711_c") {
"codecs/g711/g711_interface.h",
]
deps = [
"../..:webrtc_common",
"../third_party/g711:g711_3p",
]
}
@ -172,7 +169,6 @@ rtc_static_library("g722") {
deps = [
":legacy_encoded_audio_frame",
"../..:webrtc_common",
"../../api:array_view",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs/g722:audio_encoder_g722_config",
@ -191,7 +187,6 @@ rtc_source_set("g722_c") {
"codecs/g722/g722_interface.h",
]
deps = [
"../..:webrtc_common",
"../third_party/g722:g722_3p",
]
}
@ -208,7 +203,6 @@ rtc_static_library("ilbc") {
deps = [
":legacy_encoded_audio_frame",
"../..:webrtc_common",
"../../api:array_view",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs/ilbc:audio_encoder_ilbc_config",
@ -366,7 +360,6 @@ rtc_source_set("ilbc_c") {
]
deps = [
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
"../../common_audio",
"../../common_audio:common_audio_c",
@ -390,7 +383,6 @@ rtc_static_library("isac_common") {
]
deps = [
":isac_bwinfo",
"../..:webrtc_common",
"../../api:scoped_refptr",
"../../api/audio_codecs:audio_codecs_api",
"../../rtc_base:checks",
@ -500,7 +492,6 @@ rtc_static_library("isac_c") {
deps = [
":isac_bwinfo",
":isac_vad",
"../..:webrtc_common",
"../../common_audio",
"../../common_audio:common_audio_c",
"../../rtc_base:checks",
@ -615,7 +606,6 @@ rtc_source_set("isac_fix_c") {
deps = [
":isac_bwinfo",
":isac_common",
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
"../../common_audio",
"../../common_audio:common_audio_c",
@ -726,7 +716,6 @@ rtc_static_library("pcm16b") {
deps = [
":g711",
":legacy_encoded_audio_frame",
"../..:webrtc_common",
"../../api:array_view",
"../../api/audio_codecs:audio_codecs_api",
"../../rtc_base:checks",
@ -743,10 +732,6 @@ rtc_source_set("pcm16b_c") {
"codecs/pcm16b/pcm16b.c",
"codecs/pcm16b/pcm16b.h",
]
deps = [
"../..:webrtc_common",
]
}
rtc_static_library("webrtc_opus") {
@ -761,7 +746,6 @@ rtc_static_library("webrtc_opus") {
deps = [
":audio_network_adaptor",
"../..:webrtc_common",
"../../api:array_view",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs/opus:audio_encoder_opus_config",
@ -808,7 +792,6 @@ rtc_source_set("webrtc_opus_c") {
}
deps = [
"../..:webrtc_common",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
]
@ -877,7 +860,6 @@ rtc_static_library("audio_network_adaptor") {
]
deps = [
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
"../../common_audio",
"../../logging:rtc_event_audio",
@ -977,7 +959,6 @@ rtc_static_library("neteq") {
":webrtc_cng",
"..:module_api",
"..:module_api_public",
"../..:webrtc_common",
"../../api:array_view",
"../../api:libjingle_peerconnection_api",
"../../api:scoped_refptr",
@ -1023,7 +1004,6 @@ rtc_source_set("neteq_tools_minimal") {
deps = [
":neteq",
"../..:webrtc_common",
"../../api:libjingle_peerconnection_api",
"../../api:neteq_simulator_api",
"../../api/audio:audio_frame_api",
@ -1058,7 +1038,6 @@ rtc_source_set("neteq_test_tools") {
deps = [
":pcm16b",
"../..:webrtc_common",
"../../api:array_view",
"../../api:libjingle_peerconnection_api",
"../../common_audio",
@ -1103,7 +1082,6 @@ rtc_source_set("neteq_tools") {
deps = [
"..:module_api",
"../..:webrtc_common",
"../../api:array_view",
"../../api/audio_codecs:audio_codecs_api",
"../../rtc_base:checks",
@ -1130,7 +1108,6 @@ rtc_source_set("neteq_input_audio_tools") {
]
deps = [
"../..:webrtc_common",
"../../common_audio",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
@ -1166,7 +1143,6 @@ if (rtc_include_tests) {
audio_coding_deps = [
"../../common_audio",
"../../system_wrappers",
"../..:webrtc_common",
":audio_encoder_cng",
":g711",
":g722",
@ -1274,7 +1250,6 @@ if (rtc_include_tests) {
":red",
":webrtc_opus_c",
"..:module_api",
"../..:webrtc_common",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
@ -1316,7 +1291,6 @@ if (rtc_include_tests) {
deps = [
":neteq_test_support",
":neteq_test_tools",
"../..:webrtc_common",
"../../api/audio_codecs/opus:audio_encoder_opus",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers",
@ -1446,7 +1420,6 @@ if (rtc_include_tests) {
deps += [
":neteq",
":neteq_test_tools",
"../..:webrtc_common",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../rtc_base:rtc_base_approved",
"../../test:audio_codec_mocks",
@ -1518,7 +1491,6 @@ if (rtc_include_tests) {
deps += [
":isac_fix",
":webrtc_opus",
"../..:webrtc_common",
"../../api:libjingle_peerconnection_api",
"../../rtc_base:rtc_base_approved",
"../../test:test_main",
@ -1539,7 +1511,6 @@ if (rtc_include_tests) {
":neteq",
":neteq_test_tools",
":pcm16b",
"../..:webrtc_common",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
@ -1562,7 +1533,6 @@ if (rtc_include_tests) {
deps = [
":neteq",
":neteq_test_tools",
"../..:webrtc_common",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
@ -1672,7 +1642,6 @@ if (rtc_include_tests) {
deps = [
":neteq",
":neteq_test_support",
"../..:webrtc_common",
"../../rtc_base:rtc_base_approved",
"../../test:test_support",
]
@ -1690,7 +1659,6 @@ if (rtc_include_tests) {
":neteq",
":neteq_quality_test_support",
":neteq_tools",
"../..:webrtc_common",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../test:fileutils",
@ -1815,7 +1783,6 @@ if (rtc_include_tests) {
deps = [
":g722",
"../..:webrtc_common",
]
}
@ -1982,7 +1949,6 @@ if (rtc_include_tests) {
":webrtc_opus",
"..:module_api",
"..:module_api_public",
"../..:webrtc_common",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",

10
modules/audio_coding/acm2/acm_receiver_unittest.cc
View File

@ -36,7 +36,7 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
: 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 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
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 @@ TEST_F(AcmReceiverTestOldApi, MAYBE_LastAudioCodec) {
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 @@ TEST_F(AcmReceiverTestOldApi, MAYBE_LastAudioCodec) {
// 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 @@ TEST_F(AcmReceiverTestOldApi, MAYBE_LastAudioCodec) {
// 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_);

2
modules/audio_coding/acm2/acm_send_test.cc
View File

@ -44,7 +44,7 @@ AcmSendTestOldApi::AcmSendTestOldApi(InputAudioFile* audio_source,
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) {

5
modules/audio_coding/acm2/audio_coding_module.cc
View File

@ -395,11 +395,12 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
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;
}
{

26
modules/audio_coding/acm2/audio_coding_module_unittest.cc
View File

@ -100,7 +100,7 @@ class PacketizationCallbackStubOldApi : public AudioPacketizationCallback {
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 @@ TEST_F(AudioCodingModuleTestOldApi, TransportCallbackIsInvokedForEachPacket) {
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 @@ class AudioCodingModuleTestWithComfortNoiseOldApi
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);

6
modules/audio_coding/include/audio_coding_module_typedefs.h
View File

@ -33,6 +33,12 @@ enum ACMVADMode {
VADVeryAggr = 3
};
enum class AudioFrameType {
kEmptyFrame = 0,
kAudioFrameSpeech = 1,
kAudioFrameCN = 2,
};
///////////////////////////////////////////////////////////////////////////
//
// Enumeration of Opus mode for intended application.

1
modules/audio_coding/neteq/neteq_network_stats_unittest.cc
View File

@ -13,7 +13,6 @@
#include "absl/memory/memory.h"
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/audio_decoder.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/audio_coding/neteq/tools/rtp_generator.h"

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

@ -39,7 +39,7 @@ int32_t Channel::SendData(AudioFrameType frameType,
? timeStamp
: static_cast<uint32_t>(external_send_timestamp_);
if (frameType == kEmptyFrame) {
if (frameType == AudioFrameType::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
modules/audio_coding/test/TestAllCodecs.cc
View File

@ -75,7 +75,7 @@ int32_t TestPack::SendData(AudioFrameType frame_type,
rtp_header.payloadType = payload_type;
rtp_header.timestamp = timestamp;
if (frame_type == kEmptyFrame) {
if (frame_type == AudioFrameType::kEmptyFrame) {
// Skip this frame.
return 0;
}

4
modules/audio_coding/test/TestStereo.cc
View File

@ -54,7 +54,7 @@ int32_t TestPackStereo::SendData(const AudioFrameType frame_type,
rtp_header.sequenceNumber = seq_no_++;
rtp_header.payloadType = payload_type;
rtp_header.timestamp = timestamp;
if (frame_type == kEmptyFrame) {
if (frame_type == AudioFrameType::kEmptyFrame) {
// Skip this frame
return 0;
}
@ -63,7 +63,7 @@ int32_t TestPackStereo::SendData(const AudioFrameType frame_type,
status =
receiver_acm_->IncomingPacket(payload_data, payload_size, rtp_header);
if (frame_type != kAudioFrameCN) {
if (frame_type != AudioFrameType::kAudioFrameCN) {
payload_size_ = static_cast<int>(payload_size);
} else {
payload_size_ = -1;

15
modules/audio_coding/test/TestVADDTX.cc
View File

@ -34,17 +34,18 @@ ActivityMonitor::ActivityMonitor() {
}
int32_t ActivityMonitor::InFrameType(AudioFrameType frame_type) {
counter_[frame_type]++;
counter_[static_cast<int>(frame_type)]++;
return 0;
}
void ActivityMonitor::PrintStatistics() {
printf("\n");
printf("kEmptyFrame %u\n", counter_[kEmptyFrame]);
printf("kAudioFrameSpeech %u\n", counter_[kAudioFrameSpeech]);
printf("kAudioFrameCN %u\n", counter_[kAudioFrameCN]);
printf("kVideoFrameKey %u\n", counter_[kVideoFrameKey]);
printf("kVideoFrameDelta %u\n", counter_[kVideoFrameDelta]);
printf("kEmptyFrame %u\n",
counter_[static_cast<int>(AudioFrameType::kEmptyFrame)]);
printf("kAudioFrameSpeech %u\n",
counter_[static_cast<int>(AudioFrameType::kAudioFrameSpeech)]);
printf("kAudioFrameCN %u\n",
counter_[static_cast<int>(AudioFrameType::kAudioFrameCN)]);
printf("\n\n");
}
@ -146,7 +147,7 @@ void TestVadDtx::Run(std::string in_filename,
monitor_->PrintStatistics();
#endif
uint32_t stats[5];
uint32_t stats[3];
monitor_->GetStatistics(stats);
monitor_->ResetStatistics();

6
modules/audio_coding/test/TestVADDTX.h
View File

@ -34,9 +34,7 @@ class ActivityMonitor : public ACMVADCallback {
// 0 - kEmptyFrame
// 1 - kAudioFrameSpeech
// 2 - kAudioFrameCN
// 3 - kVideoFrameKey (not used by audio)
// 4 - kVideoFrameDelta (not used by audio)
uint32_t counter_[5];
uint32_t counter_[3];
};
// TestVadDtx is to verify that VAD/DTX perform as they should. It runs through
@ -64,8 +62,6 @@ class TestVadDtx {
// 0 - kEmptyFrame
// 1 - kAudioFrameSpeech
// 2 - kAudioFrameCN
// 3 - kVideoFrameKey (not used by audio)
// 4 - kVideoFrameDelta (not used by audio)
void Run(std::string in_filename,
int frequency,
int channels,

4
modules/audio_coding/test/opus_test.cc
View File

@ -315,8 +315,8 @@ void OpusTest::Run(TestPackStereo* channel,
}
// Send data to the channel. "channel" will handle the loss simulation.
channel->SendData(kAudioFrameSpeech, payload_type_, rtp_timestamp_,
bitstream, bitstream_len_byte, NULL);
channel->SendData(AudioFrameType::kAudioFrameSpeech, payload_type_,
rtp_timestamp_, bitstream, bitstream_len_byte, NULL);
if (first_packet) {
first_packet = false;
start_time_stamp = rtp_timestamp_;