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 fa476e8b77e833402429e0e48adb8ecf820e5ffd..d7203b9da3ea8b5adb9f1998d285cd0320fbce4d 100644 |
--- a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc |
+++ b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc |
@@ -48,7 +48,7 @@ AudioEncoderG722::AudioEncoderG722(const Config& config) |
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) { |
+ for (size_t i = 0; i < num_channels_; ++i) { |
encoders_[i].speech_buffer.reset(new int16_t[samples_per_channel]); |
encoders_[i].encoded_buffer.SetSize(samples_per_channel / 2); |
} |
@@ -68,7 +68,7 @@ int AudioEncoderG722::SampleRateHz() const { |
return kSampleRateHz; |
} |
-int AudioEncoderG722::NumChannels() const { |
+size_t AudioEncoderG722::NumChannels() const { |
return num_channels_; |
} |
@@ -88,7 +88,7 @@ size_t AudioEncoderG722::Max10MsFramesInAPacket() const { |
int AudioEncoderG722::GetTargetBitrate() const { |
// 4 bits/sample, 16000 samples/s/channel. |
- return 64000 * NumChannels(); |
+ return static_cast<int>(64000 * NumChannels()); |
} |
AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal( |
@@ -104,7 +104,7 @@ AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal( |
// Deinterleave samples and save them in each channel's buffer. |
const size_t start = kSampleRateHz / 100 * num_10ms_frames_buffered_; |
for (size_t i = 0; i < kSampleRateHz / 100; ++i) |
- for (int j = 0; j < num_channels_; ++j) |
+ for (size_t j = 0; j < num_channels_; ++j) |
encoders_[j].speech_buffer[start + i] = audio[i * num_channels_ + j]; |
// If we don't yet have enough samples for a packet, we're done for now. |
@@ -116,7 +116,7 @@ AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal( |
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) { |
+ for (size_t 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()); |
@@ -127,12 +127,12 @@ AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal( |
// channel and the interleaved stream encodes two samples per byte, most |
// significant half first. |
for (size_t i = 0; i < samples_per_channel / 2; ++i) { |
- for (int j = 0; j < num_channels_; ++j) { |
+ for (size_t j = 0; j < num_channels_; ++j) { |
uint8_t two_samples = encoders_[j].encoded_buffer.data()[i]; |
interleave_buffer_.data()[j] = two_samples >> 4; |
interleave_buffer_.data()[num_channels_ + j] = two_samples & 0xf; |
} |
- for (int j = 0; j < num_channels_; ++j) |
+ for (size_t j = 0; j < num_channels_; ++j) |
encoded[i * num_channels_ + j] = interleave_buffer_.data()[2 * j] << 4 | |
interleave_buffer_.data()[2 * j + 1]; |
} |
@@ -145,7 +145,7 @@ AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal( |
void AudioEncoderG722::Reset() { |
num_10ms_frames_buffered_ = 0; |
- for (int i = 0; i < num_channels_; ++i) |
+ for (size_t i = 0; i < num_channels_; ++i) |
RTC_CHECK_EQ(0, WebRtcG722_EncoderInit(encoders_[i].encoder)); |
} |