| Index: webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
|
| diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
|
| index f7812b34f7db9082c6f23449df3532a52d7bc15a..dde3cc6799871dc1175d215858b1ca1cb6ca6fb2 100644
|
| --- a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
|
| +++ b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
|
| @@ -24,7 +24,7 @@ int16_t NumSamplesPerFrame(int num_channels,
|
| int frame_size_ms,
|
| int sample_rate_hz) {
|
| int samples_per_frame = num_channels * frame_size_ms * sample_rate_hz / 1000;
|
| - CHECK_LE(samples_per_frame, std::numeric_limits<int16_t>::max())
|
| + RTC_CHECK_LE(samples_per_frame, std::numeric_limits<int16_t>::max())
|
| << "Frame size too large.";
|
| return static_cast<int16_t>(samples_per_frame);
|
| }
|
| @@ -54,8 +54,8 @@ AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz)
|
| config.frame_size_ms,
|
| sample_rate_hz_)),
|
| first_timestamp_in_buffer_(0) {
|
| - CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz";
|
| - CHECK_EQ(config.frame_size_ms % 10, 0)
|
| + RTC_CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz";
|
| + RTC_CHECK_EQ(config.frame_size_ms % 10, 0)
|
| << "Frame size must be an integer multiple of 10 ms.";
|
| speech_buffer_.reserve(full_frame_samples_);
|
| }
|
| @@ -101,8 +101,8 @@ AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeInternal(
|
| if (speech_buffer_.size() < full_frame_samples_) {
|
| return EncodedInfo();
|
| }
|
| - CHECK_EQ(speech_buffer_.size(), full_frame_samples_);
|
| - CHECK_GE(max_encoded_bytes, full_frame_samples_);
|
| + RTC_CHECK_EQ(speech_buffer_.size(), full_frame_samples_);
|
| + RTC_CHECK_GE(max_encoded_bytes, full_frame_samples_);
|
| EncodedInfo info;
|
| info.encoded_timestamp = first_timestamp_in_buffer_;
|
| info.payload_type = payload_type_;
|
|
|