Reformat the WebRTC code base
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index fc6d544..05d3b72 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -613,20 +613,17 @@
const size_t max_encoded_bytes = SufficientOutputBufferSize();
EncodedInfo info;
- info.encoded_bytes =
- encoded->AppendData(
- max_encoded_bytes, [&] (rtc::ArrayView<uint8_t> encoded) {
- int status = WebRtcOpus_Encode(
- inst_, &input_buffer_[0],
- rtc::CheckedDivExact(input_buffer_.size(),
- config_.num_channels),
- rtc::saturated_cast<int16_t>(max_encoded_bytes),
- encoded.data());
+ info.encoded_bytes = encoded->AppendData(
+ max_encoded_bytes, [&](rtc::ArrayView<uint8_t> encoded) {
+ int status = WebRtcOpus_Encode(
+ inst_, &input_buffer_[0],
+ rtc::CheckedDivExact(input_buffer_.size(), config_.num_channels),
+ rtc::saturated_cast<int16_t>(max_encoded_bytes), encoded.data());
- RTC_CHECK_GE(status, 0); // Fails only if fed invalid data.
+ RTC_CHECK_GE(status, 0); // Fails only if fed invalid data.
- return static_cast<size_t>(status);
- });
+ return static_cast<size_t>(status);
+ });
input_buffer_.clear();
bool dtx_frame = (info.encoded_bytes <= 2);
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
index c4d37da..dde2090 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -753,8 +753,8 @@
EXPECT_EQ(8000, config.max_playback_rate_hz);
EXPECT_EQ(12000, config.bitrate_bps);
- config = CreateConfigWithParameters({{"maxplaybackrate", "8000"},
- {"stereo", "1"}});
+ config = CreateConfigWithParameters(
+ {{"maxplaybackrate", "8000"}, {"stereo", "1"}});
EXPECT_EQ(8000, config.max_playback_rate_hz);
EXPECT_EQ(24000, config.bitrate_bps);
}
@@ -765,8 +765,8 @@
EXPECT_EQ(8001, config.max_playback_rate_hz);
EXPECT_EQ(20000, config.bitrate_bps);
- config = CreateConfigWithParameters({{"maxplaybackrate", "8001"},
- {"stereo", "1"}});
+ config = CreateConfigWithParameters(
+ {{"maxplaybackrate", "8001"}, {"stereo", "1"}});
EXPECT_EQ(8001, config.max_playback_rate_hz);
EXPECT_EQ(40000, config.bitrate_bps);
}
@@ -777,8 +777,8 @@
EXPECT_EQ(12001, config.max_playback_rate_hz);
EXPECT_EQ(20000, config.bitrate_bps);
- config = CreateConfigWithParameters({{"maxplaybackrate", "12001"},
- {"stereo", "1"}});
+ config = CreateConfigWithParameters(
+ {{"maxplaybackrate", "12001"}, {"stereo", "1"}});
EXPECT_EQ(12001, config.max_playback_rate_hz);
EXPECT_EQ(40000, config.bitrate_bps);
}
@@ -789,8 +789,8 @@
EXPECT_EQ(16001, config.max_playback_rate_hz);
EXPECT_EQ(32000, config.bitrate_bps);
- config = CreateConfigWithParameters({{"maxplaybackrate", "16001"},
- {"stereo", "1"}});
+ config = CreateConfigWithParameters(
+ {{"maxplaybackrate", "16001"}, {"stereo", "1"}});
EXPECT_EQ(16001, config.max_playback_rate_hz);
EXPECT_EQ(64000, config.bitrate_bps);
}
@@ -801,8 +801,8 @@
EXPECT_EQ(24001, config.max_playback_rate_hz);
EXPECT_EQ(32000, config.bitrate_bps);
- config = CreateConfigWithParameters({{"maxplaybackrate", "24001"},
- {"stereo", "1"}});
+ config = CreateConfigWithParameters(
+ {{"maxplaybackrate", "24001"}, {"stereo", "1"}});
EXPECT_EQ(24001, config.max_playback_rate_hz);
EXPECT_EQ(64000, config.bitrate_bps);
}
diff --git a/modules/audio_coding/codecs/opus/opus_fec_test.cc b/modules/audio_coding/codecs/opus/opus_fec_test.cc
index 4e0a17e..f1983ae 100644
--- a/modules/audio_coding/codecs/opus/opus_fec_test.cc
+++ b/modules/audio_coding/codecs/opus/opus_fec_test.cc
@@ -83,8 +83,8 @@
rewind(fp);
// Allocate memory to contain the whole file.
- in_data_.reset(new int16_t[loop_length_samples_ +
- block_length_sample_ * channels_]);
+ in_data_.reset(
+ new int16_t[loop_length_samples_ + block_length_sample_ * channels_]);
// Copy the file into the buffer.
ASSERT_EQ(fread(&in_data_[0], sizeof(int16_t), loop_length_samples_, fp),
@@ -130,14 +130,12 @@
max_bytes_(0),
encoded_bytes_(0),
opus_encoder_(NULL),
- opus_decoder_(NULL) {
-}
+ opus_decoder_(NULL) {}
void OpusFecTest::EncodeABlock() {
- int value = WebRtcOpus_Encode(opus_encoder_,
- &in_data_[data_pointer_],
- block_length_sample_,
- max_bytes_, &bit_stream_[0]);
+ int value =
+ WebRtcOpus_Encode(opus_encoder_, &in_data_[data_pointer_],
+ block_length_sample_, max_bytes_, &bit_stream_[0]);
EXPECT_GT(value, 0);
encoded_bytes_ = static_cast<size_t>(value);
@@ -151,9 +149,9 @@
// Decode previous frame.
if (!lost_current &&
WebRtcOpus_PacketHasFec(&bit_stream_[0], encoded_bytes_) == 1) {
- value_1 = WebRtcOpus_DecodeFec(opus_decoder_, &bit_stream_[0],
- encoded_bytes_, &out_data_[0],
- &audio_type);
+ value_1 =
+ WebRtcOpus_DecodeFec(opus_decoder_, &bit_stream_[0], encoded_bytes_,
+ &out_data_[0], &audio_type);
} else {
value_1 = WebRtcOpus_DecodePlc(opus_decoder_, &out_data_[0], 1);
}
@@ -173,16 +171,14 @@
int time_now_ms, fec_frames;
int actual_packet_loss_rate;
bool lost_current, lost_previous;
- mode mode_set[3] = {{true, 0},
- {false, 0},
- {true, 50}};
+ mode mode_set[3] = {{true, 0}, {false, 0}, {true, 50}};
lost_current = false;
for (int i = 0; i < 3; i++) {
if (mode_set[i].fec) {
EXPECT_EQ(0, WebRtcOpus_EnableFec(opus_encoder_));
- EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(opus_encoder_,
- mode_set[i].target_packet_loss_rate));
+ EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(
+ opus_encoder_, mode_set[i].target_packet_loss_rate));
printf("FEC is ON, target at packet loss rate %d percent.\n",
mode_set[i].target_packet_loss_rate);
} else {
@@ -218,7 +214,7 @@
// |data_pointer_| is incremented and wrapped across
// |loop_length_samples_|.
data_pointer_ = (data_pointer_ + block_length_sample_ * channels_) %
- loop_length_samples_;
+ loop_length_samples_;
}
if (mode_set[i].fec) {
printf("%.2f percent frames has FEC.\n",
@@ -242,7 +238,6 @@
string("pcm"))};
// 64 kbps, stereo
-INSTANTIATE_TEST_CASE_P(AllTest, OpusFecTest,
- ::testing::ValuesIn(param_set));
+INSTANTIATE_TEST_CASE_P(AllTest, OpusFecTest, ::testing::ValuesIn(param_set));
} // namespace webrtc
diff --git a/modules/audio_coding/codecs/opus/opus_inst.h b/modules/audio_coding/codecs/opus/opus_inst.h
index 066fa22..2473a5c 100644
--- a/modules/audio_coding/codecs/opus/opus_inst.h
+++ b/modules/audio_coding/codecs/opus/opus_inst.h
@@ -32,5 +32,4 @@
int in_dtx_mode;
};
-
#endif // MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
diff --git a/modules/audio_coding/codecs/opus/opus_interface.h b/modules/audio_coding/codecs/opus/opus_interface.h
index 4b8e892..0b1c64d 100644
--- a/modules/audio_coding/codecs/opus/opus_interface.h
+++ b/modules/audio_coding/codecs/opus/opus_interface.h
@@ -318,8 +318,10 @@
* Return value : >0 - Samples per channel in decoded vector
* -1 - Error
*/
-int WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
- size_t encoded_bytes, int16_t* decoded,
+int WebRtcOpus_Decode(OpusDecInst* inst,
+ const uint8_t* encoded,
+ size_t encoded_bytes,
+ int16_t* decoded,
int16_t* audio_type);
/****************************************************************************
@@ -336,7 +338,8 @@
* Return value : >0 - number of samples in decoded PLC vector
* -1 - Error
*/
-int WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
+int WebRtcOpus_DecodePlc(OpusDecInst* inst,
+ int16_t* decoded,
int number_of_lost_frames);
/****************************************************************************
@@ -357,8 +360,10 @@
* 0 - No FEC data in the packet
* -1 - Error
*/
-int WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
- size_t encoded_bytes, int16_t* decoded,
+int WebRtcOpus_DecodeFec(OpusDecInst* inst,
+ const uint8_t* encoded,
+ size_t encoded_bytes,
+ int16_t* decoded,
int16_t* audio_type);
/****************************************************************************
diff --git a/modules/audio_coding/codecs/opus/opus_speed_test.cc b/modules/audio_coding/codecs/opus/opus_speed_test.cc
index ca46aa1..03b59ed 100644
--- a/modules/audio_coding/codecs/opus/opus_speed_test.cc
+++ b/modules/audio_coding/codecs/opus/opus_speed_test.cc
@@ -23,9 +23,12 @@
OpusSpeedTest();
void SetUp() override;
void TearDown() override;
- float EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
- size_t max_bytes, size_t* encoded_bytes) override;
- float DecodeABlock(const uint8_t* bit_stream, size_t encoded_bytes,
+ float EncodeABlock(int16_t* in_data,
+ uint8_t* bit_stream,
+ size_t max_bytes,
+ size_t* encoded_bytes) override;
+ float DecodeABlock(const uint8_t* bit_stream,
+ size_t encoded_bytes,
int16_t* out_data) override;
WebRtcOpusEncInst* opus_encoder_;
WebRtcOpusDecInst* opus_decoder_;
@@ -36,8 +39,7 @@
kOpusSamplingKhz,
kOpusSamplingKhz),
opus_encoder_(NULL),
- opus_decoder_(NULL) {
-}
+ opus_decoder_(NULL) {}
void OpusSpeedTest::SetUp() {
AudioCodecSpeedTest::SetUp();
@@ -57,12 +59,13 @@
EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
}
-float OpusSpeedTest::EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
- size_t max_bytes, size_t* encoded_bytes) {
+float OpusSpeedTest::EncodeABlock(int16_t* in_data,
+ uint8_t* bit_stream,
+ size_t max_bytes,
+ size_t* encoded_bytes) {
clock_t clocks = clock();
- int value = WebRtcOpus_Encode(opus_encoder_, in_data,
- input_length_sample_, max_bytes,
- bit_stream);
+ int value = WebRtcOpus_Encode(opus_encoder_, in_data, input_length_sample_,
+ max_bytes, bit_stream);
clocks = clock() - clocks;
EXPECT_GT(value, 0);
*encoded_bytes = static_cast<size_t>(value);
@@ -70,7 +73,8 @@
}
float OpusSpeedTest::DecodeABlock(const uint8_t* bit_stream,
- size_t encoded_bytes, int16_t* out_data) {
+ size_t encoded_bytes,
+ int16_t* out_data) {
int value;
int16_t audio_type;
clock_t clocks = clock();
@@ -84,13 +88,13 @@
/* Test audio length in second. */
constexpr size_t kDurationSec = 400;
-#define ADD_TEST(complexity) \
-TEST_P(OpusSpeedTest, OpusSetComplexityTest##complexity) { \
- /* Set complexity. */ \
- printf("Setting complexity to %d ...\n", complexity); \
- EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, complexity)); \
- EncodeDecode(kDurationSec); \
-}
+#define ADD_TEST(complexity) \
+ TEST_P(OpusSpeedTest, OpusSetComplexityTest##complexity) { \
+ /* Set complexity. */ \
+ printf("Setting complexity to %d ...\n", complexity); \
+ EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, complexity)); \
+ EncodeDecode(kDurationSec); \
+ }
ADD_TEST(10);
ADD_TEST(9);
@@ -136,7 +140,6 @@
string("pcm"),
true)};
-INSTANTIATE_TEST_CASE_P(AllTest, OpusSpeedTest,
- ::testing::ValuesIn(param_set));
+INSTANTIATE_TEST_CASE_P(AllTest, OpusSpeedTest, ::testing::ValuesIn(param_set));
} // namespace webrtc
diff --git a/modules/audio_coding/codecs/opus/opus_unittest.cc b/modules/audio_coding/codecs/opus/opus_unittest.cc
index 12a1585..034f8cd 100644
--- a/modules/audio_coding/codecs/opus/opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/opus_unittest.cc
@@ -58,9 +58,12 @@
int16_t* audio_type);
void SetMaxPlaybackRate(WebRtcOpusEncInst* encoder,
- opus_int32 expect, int32_t set);
+ opus_int32 expect,
+ int32_t set);
- void CheckAudioBounded(const int16_t* audio, size_t samples, size_t channels,
+ void CheckAudioBounded(const int16_t* audio,
+ size_t samples,
+ size_t channels,
uint16_t bound) const;
WebRtcOpusEncInst* opus_encoder_;
@@ -78,15 +81,15 @@
opus_decoder_(NULL),
encoded_bytes_(0),
channels_(static_cast<size_t>(::testing::get<0>(GetParam()))),
- application_(::testing::get<1>(GetParam())) {
-}
+ application_(::testing::get<1>(GetParam())) {}
-void OpusTest::PrepareSpeechData(size_t channel, int block_length_ms,
+void OpusTest::PrepareSpeechData(size_t channel,
+ int block_length_ms,
int loop_length_ms) {
- const std::string file_name =
- webrtc::test::ResourcePath((channel == 1) ?
- "audio_coding/testfile32kHz" :
- "audio_coding/teststereo32kHz", "pcm");
+ const std::string file_name = webrtc::test::ResourcePath(
+ (channel == 1) ? "audio_coding/testfile32kHz"
+ : "audio_coding/teststereo32kHz",
+ "pcm");
if (loop_length_ms < block_length_ms) {
loop_length_ms = block_length_ms;
}
@@ -100,13 +103,14 @@
int32_t set) {
opus_int32 bandwidth;
EXPECT_EQ(0, WebRtcOpus_SetMaxPlaybackRate(opus_encoder_, set));
- opus_encoder_ctl(opus_encoder_->encoder,
- OPUS_GET_MAX_BANDWIDTH(&bandwidth));
+ opus_encoder_ctl(opus_encoder_->encoder, OPUS_GET_MAX_BANDWIDTH(&bandwidth));
EXPECT_EQ(expect, bandwidth);
}
-void OpusTest::CheckAudioBounded(const int16_t* audio, size_t samples,
- size_t channels, uint16_t bound) const {
+void OpusTest::CheckAudioBounded(const int16_t* audio,
+ size_t samples,
+ size_t channels,
+ uint16_t bound) const {
for (size_t i = 0; i < samples; ++i) {
for (size_t c = 0; c < channels; ++c) {
ASSERT_GE(audio[i * channels + c], -bound);
@@ -120,16 +124,15 @@
WebRtcOpusDecInst* decoder,
int16_t* output_audio,
int16_t* audio_type) {
- int encoded_bytes_int = WebRtcOpus_Encode(
- encoder, input_audio.data(),
- rtc::CheckedDivExact(input_audio.size(), channels_),
- kMaxBytes, bitstream_);
+ int encoded_bytes_int =
+ WebRtcOpus_Encode(encoder, input_audio.data(),
+ rtc::CheckedDivExact(input_audio.size(), channels_),
+ kMaxBytes, bitstream_);
EXPECT_GE(encoded_bytes_int, 0);
encoded_bytes_ = static_cast<size_t>(encoded_bytes_int);
int est_len = WebRtcOpus_DurationEst(decoder, bitstream_, encoded_bytes_);
- int act_len = WebRtcOpus_Decode(decoder, bitstream_,
- encoded_bytes_, output_audio,
- audio_type);
+ int act_len = WebRtcOpus_Decode(decoder, bitstream_, encoded_bytes_,
+ output_audio, audio_type);
EXPECT_EQ(est_len, act_len);
return act_len;
}
@@ -141,30 +144,28 @@
const size_t samples = kOpusRateKhz * block_length_ms;
// Create encoder memory.
- EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
- channels_,
- application_));
+ EXPECT_EQ(0,
+ WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_));
EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_));
// Set bitrate.
- EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_,
- channels_ == 1 ? 32000 : 64000));
+ EXPECT_EQ(
+ 0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
// Set input audio as silence.
std::vector<int16_t> silence(samples * channels_, 0);
// Setting DTX.
- EXPECT_EQ(0, dtx ? WebRtcOpus_EnableDtx(opus_encoder_) :
- WebRtcOpus_DisableDtx(opus_encoder_));
+ EXPECT_EQ(0, dtx ? WebRtcOpus_EnableDtx(opus_encoder_)
+ : WebRtcOpus_DisableDtx(opus_encoder_));
int16_t audio_type;
int16_t* output_data_decode = new int16_t[samples * channels_];
for (int i = 0; i < 100; ++i) {
- EXPECT_EQ(samples,
- static_cast<size_t>(EncodeDecode(
- opus_encoder_, speech_data_.GetNextBlock(), opus_decoder_,
- output_data_decode, &audio_type)));
+ EXPECT_EQ(samples, static_cast<size_t>(EncodeDecode(
+ opus_encoder_, speech_data_.GetNextBlock(),
+ opus_decoder_, output_data_decode, &audio_type)));
// If not DTX, it should never enter DTX mode. If DTX, we do not care since
// whether it enters DTX depends on the signal type.
if (!dtx) {
@@ -178,10 +179,9 @@
// We input some silent segments. In DTX mode, the encoder will stop sending.
// However, DTX may happen after a while.
for (int i = 0; i < 30; ++i) {
- EXPECT_EQ(samples,
- static_cast<size_t>(EncodeDecode(
- opus_encoder_, silence, opus_decoder_, output_data_decode,
- &audio_type)));
+ EXPECT_EQ(samples, static_cast<size_t>(
+ EncodeDecode(opus_encoder_, silence, opus_decoder_,
+ output_data_decode, &audio_type)));
if (!dtx) {
EXPECT_GT(encoded_bytes_, 1U);
EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
@@ -227,10 +227,9 @@
int i = 0;
for (; i < max_dtx_frames; ++i) {
time += block_length_ms;
- EXPECT_EQ(samples,
- static_cast<size_t>(EncodeDecode(
- opus_encoder_, silence, opus_decoder_, output_data_decode,
- &audio_type)));
+ EXPECT_EQ(samples, static_cast<size_t>(
+ EncodeDecode(opus_encoder_, silence, opus_decoder_,
+ output_data_decode, &audio_type)));
if (dtx) {
if (encoded_bytes_ > 1)
break;
@@ -263,10 +262,9 @@
// Enters DTX again immediately.
time += block_length_ms;
- EXPECT_EQ(samples,
- static_cast<size_t>(EncodeDecode(
- opus_encoder_, silence, opus_decoder_, output_data_decode,
- &audio_type)));
+ EXPECT_EQ(samples, static_cast<size_t>(
+ EncodeDecode(opus_encoder_, silence, opus_decoder_,
+ output_data_decode, &audio_type)));
if (dtx) {
EXPECT_EQ(1U, encoded_bytes_); // Send 1 byte.
EXPECT_EQ(1, opus_encoder_->in_dtx_mode);
@@ -287,10 +285,9 @@
silence[0] = 10000;
if (dtx) {
// Verify that encoder/decoder can jump out from DTX mode.
- EXPECT_EQ(samples,
- static_cast<size_t>(EncodeDecode(
- opus_encoder_, silence, opus_decoder_, output_data_decode,
- &audio_type)));
+ EXPECT_EQ(samples, static_cast<size_t>(
+ EncodeDecode(opus_encoder_, silence, opus_decoder_,
+ output_data_decode, &audio_type)));
EXPECT_GT(encoded_bytes_, 1U);
EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
@@ -375,9 +372,8 @@
// Test normal Create and Free.
TEST_P(OpusTest, OpusCreateFree) {
- EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
- channels_,
- application_));
+ EXPECT_EQ(0,
+ WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_));
EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_));
EXPECT_TRUE(opus_encoder_ != NULL);
EXPECT_TRUE(opus_decoder_ != NULL);
@@ -390,23 +386,20 @@
PrepareSpeechData(channels_, 20, 20);
// Create encoder memory.
- EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
- channels_,
- application_));
- EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_,
- channels_));
+ EXPECT_EQ(0,
+ WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_));
+ EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_));
// Set bitrate.
- EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_,
- channels_ == 1 ? 32000 : 64000));
+ EXPECT_EQ(
+ 0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
// Check number of channels for decoder.
EXPECT_EQ(channels_, WebRtcOpus_DecoderChannels(opus_decoder_));
// Check application mode.
opus_int32 app;
- opus_encoder_ctl(opus_encoder_->encoder,
- OPUS_GET_APPLICATION(&app));
+ opus_encoder_ctl(opus_encoder_->encoder, OPUS_GET_APPLICATION(&app));
EXPECT_EQ(application_ == 0 ? OPUS_APPLICATION_VOIP : OPUS_APPLICATION_AUDIO,
app);
@@ -429,9 +422,8 @@
EXPECT_EQ(-1, WebRtcOpus_SetBitRate(opus_encoder_, 60000));
// Create encoder memory, try with different bitrates.
- EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
- channels_,
- application_));
+ EXPECT_EQ(0,
+ WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_));
EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, 30000));
EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, 60000));
EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, 300000));
@@ -446,9 +438,8 @@
EXPECT_EQ(-1, WebRtcOpus_SetComplexity(opus_encoder_, 9));
// Create encoder memory, try with different complexities.
- EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
- channels_,
- application_));
+ EXPECT_EQ(0,
+ WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_));
EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, 0));
EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, 10));
@@ -524,9 +515,8 @@
PrepareSpeechData(channels_, 20, 20);
// Create encoder memory.
- EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
- channels_,
- application_));
+ EXPECT_EQ(0,
+ WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_));
EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_));
// Encode & decode.
@@ -540,9 +530,9 @@
WebRtcOpus_DecoderInit(opus_decoder_);
EXPECT_EQ(kOpus20msFrameSamples,
- static_cast<size_t>(WebRtcOpus_Decode(
- opus_decoder_, bitstream_, encoded_bytes_, output_data_decode,
- &audio_type)));
+ static_cast<size_t>(
+ WebRtcOpus_Decode(opus_decoder_, bitstream_, encoded_bytes_,
+ output_data_decode, &audio_type)));
// Free memory.
delete[] output_data_decode;
@@ -556,9 +546,8 @@
EXPECT_EQ(-1, WebRtcOpus_DisableFec(opus_encoder_));
// Create encoder memory.
- EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
- channels_,
- application_));
+ EXPECT_EQ(0,
+ WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_));
EXPECT_EQ(0, WebRtcOpus_EnableFec(opus_encoder_));
EXPECT_EQ(0, WebRtcOpus_DisableFec(opus_encoder_));
@@ -573,30 +562,25 @@
EXPECT_EQ(-1, WebRtcOpus_DisableDtx(opus_encoder_));
// Create encoder memory.
- EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
- channels_,
- application_));
+ EXPECT_EQ(0,
+ WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_));
opus_int32 dtx;
// DTX is off by default.
- opus_encoder_ctl(opus_encoder_->encoder,
- OPUS_GET_DTX(&dtx));
+ opus_encoder_ctl(opus_encoder_->encoder, OPUS_GET_DTX(&dtx));
EXPECT_EQ(0, dtx);
// Test to enable DTX.
EXPECT_EQ(0, WebRtcOpus_EnableDtx(opus_encoder_));
- opus_encoder_ctl(opus_encoder_->encoder,
- OPUS_GET_DTX(&dtx));
+ opus_encoder_ctl(opus_encoder_->encoder, OPUS_GET_DTX(&dtx));
EXPECT_EQ(1, dtx);
// Test to disable DTX.
EXPECT_EQ(0, WebRtcOpus_DisableDtx(opus_encoder_));
- opus_encoder_ctl(opus_encoder_->encoder,
- OPUS_GET_DTX(&dtx));
+ opus_encoder_ctl(opus_encoder_->encoder, OPUS_GET_DTX(&dtx));
EXPECT_EQ(0, dtx);
-
// Free memory.
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
}
@@ -630,9 +614,8 @@
EXPECT_EQ(-1, WebRtcOpus_SetPacketLossRate(opus_encoder_, 50));
// Create encoder memory.
- EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
- channels_,
- application_));
+ EXPECT_EQ(0,
+ WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_));
EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(opus_encoder_, 50));
EXPECT_EQ(-1, WebRtcOpus_SetPacketLossRate(opus_encoder_, -1));
@@ -647,9 +630,8 @@
EXPECT_EQ(-1, WebRtcOpus_SetMaxPlaybackRate(opus_encoder_, 20000));
// Create encoder memory.
- EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
- channels_,
- application_));
+ EXPECT_EQ(0,
+ WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_));
SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_FULLBAND, 48000);
SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_FULLBAND, 24001);
@@ -671,14 +653,13 @@
PrepareSpeechData(channels_, 20, 20);
// Create encoder memory.
- EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
- channels_,
- application_));
+ EXPECT_EQ(0,
+ WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_));
EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_));
// Set bitrate.
- EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_,
- channels_== 1 ? 32000 : 64000));
+ EXPECT_EQ(
+ 0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
// Check number of channels for decoder.
EXPECT_EQ(channels_, WebRtcOpus_DecoderChannels(opus_decoder_));
@@ -693,9 +674,8 @@
// Call decoder PLC.
int16_t* plc_buffer = new int16_t[kOpus20msFrameSamples * channels_];
- EXPECT_EQ(kOpus20msFrameSamples,
- static_cast<size_t>(WebRtcOpus_DecodePlc(
- opus_decoder_, plc_buffer, 1)));
+ EXPECT_EQ(kOpus20msFrameSamples, static_cast<size_t>(WebRtcOpus_DecodePlc(
+ opus_decoder_, plc_buffer, 1)));
// Free memory.
delete[] plc_buffer;
@@ -709,34 +689,33 @@
PrepareSpeechData(channels_, 20, 20);
// Create.
- EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
- channels_,
- application_));
+ EXPECT_EQ(0,
+ WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_));
EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_));
// 10 ms. We use only first 10 ms of a 20 ms block.
auto speech_block = speech_data_.GetNextBlock();
int encoded_bytes_int = WebRtcOpus_Encode(
opus_encoder_, speech_block.data(),
- rtc::CheckedDivExact(speech_block.size(), 2 * channels_),
- kMaxBytes, bitstream_);
+ rtc::CheckedDivExact(speech_block.size(), 2 * channels_), kMaxBytes,
+ bitstream_);
EXPECT_GE(encoded_bytes_int, 0);
- EXPECT_EQ(kOpus10msFrameSamples,
- static_cast<size_t>(WebRtcOpus_DurationEst(
- opus_decoder_, bitstream_,
- static_cast<size_t>(encoded_bytes_int))));
+ EXPECT_EQ(
+ kOpus10msFrameSamples,
+ static_cast<size_t>(WebRtcOpus_DurationEst(
+ opus_decoder_, bitstream_, static_cast<size_t>(encoded_bytes_int))));
// 20 ms
speech_block = speech_data_.GetNextBlock();
- encoded_bytes_int = WebRtcOpus_Encode(
- opus_encoder_, speech_block.data(),
- rtc::CheckedDivExact(speech_block.size(), channels_),
- kMaxBytes, bitstream_);
+ encoded_bytes_int =
+ WebRtcOpus_Encode(opus_encoder_, speech_block.data(),
+ rtc::CheckedDivExact(speech_block.size(), channels_),
+ kMaxBytes, bitstream_);
EXPECT_GE(encoded_bytes_int, 0);
- EXPECT_EQ(kOpus20msFrameSamples,
- static_cast<size_t>(WebRtcOpus_DurationEst(
- opus_decoder_, bitstream_,
- static_cast<size_t>(encoded_bytes_int))));
+ EXPECT_EQ(
+ kOpus20msFrameSamples,
+ static_cast<size_t>(WebRtcOpus_DurationEst(
+ opus_decoder_, bitstream_, static_cast<size_t>(encoded_bytes_int))));
// Free memory.
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
@@ -749,15 +728,13 @@
PrepareSpeechData(channels_, 20, 20 * kPackets);
// Create encoder memory.
- ASSERT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
- channels_,
- application_));
- ASSERT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_,
- channels_));
+ ASSERT_EQ(0,
+ WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_));
+ ASSERT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_));
// Set bitrate.
- EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_,
- channels_ == 1 ? 32000 : 64000));
+ EXPECT_EQ(
+ 0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
// Check number of channels for decoder.
EXPECT_EQ(channels_, WebRtcOpus_DecoderChannels(opus_decoder_));
@@ -776,9 +753,9 @@
WebRtcOpus_Encode(opus_encoder_, speech_block.data(),
rtc::CheckedDivExact(speech_block.size(), channels_),
kMaxBytes, bitstream_);
- if (opus_repacketizer_cat(
- rp, bitstream_,
- rtc::checked_cast<opus_int32>(encoded_bytes_)) == OPUS_OK) {
+ if (opus_repacketizer_cat(rp, bitstream_,
+ rtc::checked_cast<opus_int32>(encoded_bytes_)) ==
+ OPUS_OK) {
++num_packets;
if (num_packets == kPackets) {
break;
@@ -798,9 +775,9 @@
opus_decoder_, bitstream_, encoded_bytes_)));
EXPECT_EQ(kOpus20msFrameSamples * kPackets,
- static_cast<size_t>(WebRtcOpus_Decode(
- opus_decoder_, bitstream_, encoded_bytes_,
- output_data_decode.get(), &audio_type)));
+ static_cast<size_t>(
+ WebRtcOpus_Decode(opus_decoder_, bitstream_, encoded_bytes_,
+ output_data_decode.get(), &audio_type)));
// Free memory.
opus_repacketizer_destroy(rp);
@@ -812,5 +789,4 @@
OpusTest,
Combine(Values(1, 2), Values(0, 1)));
-
} // namespace webrtc