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Diffstat (limited to 'webrtc/modules/audio_coding/test/opus_test.cc')
-rw-r--r--webrtc/modules/audio_coding/test/opus_test.cc21
1 files changed, 10 insertions, 11 deletions
diff --git a/webrtc/modules/audio_coding/test/opus_test.cc b/webrtc/modules/audio_coding/test/opus_test.cc
index a68db910f5..466db9faa2 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