| 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
|
|
|