Index: webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc |
diff --git a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc |
index 6df5430cba7c2f853c0b6e156a8ec62fafbf6bf6..43b097fa0eacd432727042209fcc5ab6af7bad1b 100644 |
--- a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc |
+++ b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc |
@@ -45,7 +45,7 @@ AudioEncoderG722::AudioEncoderG722(const Config& config) |
first_timestamp_in_buffer_(0), |
encoders_(new EncoderState[num_channels_]), |
interleave_buffer_(2 * num_channels_) { |
- CHECK(config.IsOk()); |
+ RTC_CHECK(config.IsOk()); |
const size_t samples_per_channel = |
kSampleRateHz / 100 * num_10ms_frames_per_packet_; |
for (int i = 0; i < num_channels_; ++i) { |
@@ -96,7 +96,7 @@ AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal( |
const int16_t* audio, |
size_t max_encoded_bytes, |
uint8_t* encoded) { |
- CHECK_GE(max_encoded_bytes, MaxEncodedBytes()); |
+ RTC_CHECK_GE(max_encoded_bytes, MaxEncodedBytes()); |
if (num_10ms_frames_buffered_ == 0) |
first_timestamp_in_buffer_ = rtp_timestamp; |
@@ -113,14 +113,14 @@ AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal( |
} |
// Encode each channel separately. |
- CHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_); |
+ RTC_CHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_); |
num_10ms_frames_buffered_ = 0; |
const size_t samples_per_channel = SamplesPerChannel(); |
for (int i = 0; i < num_channels_; ++i) { |
const size_t encoded = WebRtcG722_Encode( |
encoders_[i].encoder, encoders_[i].speech_buffer.get(), |
samples_per_channel, encoders_[i].encoded_buffer.data()); |
- CHECK_EQ(encoded, samples_per_channel / 2); |
+ RTC_CHECK_EQ(encoded, samples_per_channel / 2); |
} |
// Interleave the encoded bytes of the different channels. Each separate |
@@ -146,15 +146,15 @@ AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal( |
void AudioEncoderG722::Reset() { |
num_10ms_frames_buffered_ = 0; |
for (int i = 0; i < num_channels_; ++i) |
- CHECK_EQ(0, WebRtcG722_EncoderInit(encoders_[i].encoder)); |
+ RTC_CHECK_EQ(0, WebRtcG722_EncoderInit(encoders_[i].encoder)); |
} |
AudioEncoderG722::EncoderState::EncoderState() { |
- CHECK_EQ(0, WebRtcG722_CreateEncoder(&encoder)); |
+ RTC_CHECK_EQ(0, WebRtcG722_CreateEncoder(&encoder)); |
} |
AudioEncoderG722::EncoderState::~EncoderState() { |
- CHECK_EQ(0, WebRtcG722_FreeEncoder(encoder)); |
+ RTC_CHECK_EQ(0, WebRtcG722_FreeEncoder(encoder)); |
} |
size_t AudioEncoderG722::SamplesPerChannel() const { |