Index: webrtc/modules/audio_coding/test/opus_test.cc |
diff --git a/webrtc/modules/audio_coding/test/opus_test.cc b/webrtc/modules/audio_coding/test/opus_test.cc |
index a68db910f5e8631fa16d8fd1d3b270bcc3e6244f..466db9faa2a6b17bbe1fd767236524087d14703d 100644 |
--- a/webrtc/modules/audio_coding/test/opus_test.cc |
+++ b/webrtc/modules/audio_coding/test/opus_test.cc |
@@ -206,16 +206,16 @@ void OpusTest::Perform() { |
} |
void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate, |
- int frame_length, int percent_loss) { |
+ size_t frame_length, int percent_loss) { |
AudioFrame audio_frame; |
int32_t out_freq_hz_b = out_file_.SamplingFrequency(); |
- const int kBufferSizeSamples = 480 * 12 * 2; // Can hold 120 ms stereo audio. |
+ const size_t kBufferSizeSamples = 480 * 12 * 2; // 120 ms stereo audio. |
int16_t audio[kBufferSizeSamples]; |
int16_t out_audio[kBufferSizeSamples]; |
int16_t audio_type; |
- int written_samples = 0; |
- int read_samples = 0; |
- int decoded_samples = 0; |
+ size_t written_samples = 0; |
+ size_t read_samples = 0; |
+ size_t decoded_samples = 0; |
bool first_packet = true; |
uint32_t start_time_stamp = 0; |
@@ -268,14 +268,14 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate, |
// Sometimes we need to loop over the audio vector to produce the right |
// number of packets. |
- int loop_encode = (written_samples - read_samples) / |
+ size_t loop_encode = (written_samples - read_samples) / |
(channels * frame_length); |
if (loop_encode > 0) { |
- const int kMaxBytes = 1000; // Maximum number of bytes for one packet. |
+ const size_t kMaxBytes = 1000; // Maximum number of bytes for one packet. |
size_t bitstream_len_byte; |
uint8_t bitstream[kMaxBytes]; |
- for (int i = 0; i < loop_encode; i++) { |
+ for (size_t i = 0; i < loop_encode; i++) { |
int bitstream_len_byte_int = WebRtcOpus_Encode( |
(channels == 1) ? opus_mono_encoder_ : opus_stereo_encoder_, |
&audio[read_samples], frame_length, kMaxBytes, bitstream); |
@@ -326,7 +326,7 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate, |
first_packet = false; |
start_time_stamp = rtp_timestamp_; |
} |
- rtp_timestamp_ += frame_length; |
+ rtp_timestamp_ += static_cast<uint32_t>(frame_length); |
read_samples += frame_length * channels; |
} |
if (read_samples == written_samples) { |
@@ -344,8 +344,7 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate, |
audio_frame.samples_per_channel_ * audio_frame.num_channels_); |
// Write stand-alone speech to file. |
- out_file_standalone_.Write10MsData( |
- out_audio, static_cast<size_t>(decoded_samples) * channels); |
+ out_file_standalone_.Write10MsData(out_audio, decoded_samples * channels); |
if (audio_frame.timestamp_ > start_time_stamp) { |
// Number of channels should be the same for both stand-alone and |