| Index: webrtc/common_audio/audio_converter.cc
|
| diff --git a/webrtc/common_audio/audio_converter.cc b/webrtc/common_audio/audio_converter.cc
|
| index 9bb5895b2540f3f3e9e0628d8130ef902b61658e..9ebfabc2860c6aeee019b9bc26615ae23d2d9070 100644
|
| --- a/webrtc/common_audio/audio_converter.cc
|
| +++ b/webrtc/common_audio/audio_converter.cc
|
| @@ -25,7 +25,7 @@ namespace webrtc {
|
|
|
| class CopyConverter : public AudioConverter {
|
| public:
|
| - CopyConverter(int src_channels, size_t src_frames, int dst_channels,
|
| + CopyConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
|
| size_t dst_frames)
|
| : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
|
| ~CopyConverter() override {};
|
| @@ -34,7 +34,7 @@ class CopyConverter : public AudioConverter {
|
| size_t dst_capacity) override {
|
| CheckSizes(src_size, dst_capacity);
|
| if (src != dst) {
|
| - for (int i = 0; i < src_channels(); ++i)
|
| + for (size_t i = 0; i < src_channels(); ++i)
|
| std::memcpy(dst[i], src[i], dst_frames() * sizeof(*dst[i]));
|
| }
|
| }
|
| @@ -42,7 +42,7 @@ class CopyConverter : public AudioConverter {
|
|
|
| class UpmixConverter : public AudioConverter {
|
| public:
|
| - UpmixConverter(int src_channels, size_t src_frames, int dst_channels,
|
| + UpmixConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
|
| size_t dst_frames)
|
| : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
|
| ~UpmixConverter() override {};
|
| @@ -52,7 +52,7 @@ class UpmixConverter : public AudioConverter {
|
| CheckSizes(src_size, dst_capacity);
|
| for (size_t i = 0; i < dst_frames(); ++i) {
|
| const float value = src[0][i];
|
| - for (int j = 0; j < dst_channels(); ++j)
|
| + for (size_t j = 0; j < dst_channels(); ++j)
|
| dst[j][i] = value;
|
| }
|
| }
|
| @@ -60,7 +60,7 @@ class UpmixConverter : public AudioConverter {
|
|
|
| class DownmixConverter : public AudioConverter {
|
| public:
|
| - DownmixConverter(int src_channels, size_t src_frames, int dst_channels,
|
| + DownmixConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
|
| size_t dst_frames)
|
| : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
|
| }
|
| @@ -72,7 +72,7 @@ class DownmixConverter : public AudioConverter {
|
| float* dst_mono = dst[0];
|
| for (size_t i = 0; i < src_frames(); ++i) {
|
| float sum = 0;
|
| - for (int j = 0; j < src_channels(); ++j)
|
| + for (size_t j = 0; j < src_channels(); ++j)
|
| sum += src[j][i];
|
| dst_mono[i] = sum / src_channels();
|
| }
|
| @@ -81,11 +81,11 @@ class DownmixConverter : public AudioConverter {
|
|
|
| class ResampleConverter : public AudioConverter {
|
| public:
|
| - ResampleConverter(int src_channels, size_t src_frames, int dst_channels,
|
| + ResampleConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
|
| size_t dst_frames)
|
| : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
|
| resamplers_.reserve(src_channels);
|
| - for (int i = 0; i < src_channels; ++i)
|
| + for (size_t i = 0; i < src_channels; ++i)
|
| resamplers_.push_back(new PushSincResampler(src_frames, dst_frames));
|
| }
|
| ~ResampleConverter() override {};
|
| @@ -136,9 +136,9 @@ class CompositionConverter : public AudioConverter {
|
| ScopedVector<ChannelBuffer<float>> buffers_;
|
| };
|
|
|
| -rtc::scoped_ptr<AudioConverter> AudioConverter::Create(int src_channels,
|
| +rtc::scoped_ptr<AudioConverter> AudioConverter::Create(size_t src_channels,
|
| size_t src_frames,
|
| - int dst_channels,
|
| + size_t dst_channels,
|
| size_t dst_frames) {
|
| rtc::scoped_ptr<AudioConverter> sp;
|
| if (src_channels > dst_channels) {
|
| @@ -183,8 +183,8 @@ AudioConverter::AudioConverter()
|
| dst_channels_(0),
|
| dst_frames_(0) {}
|
|
|
| -AudioConverter::AudioConverter(int src_channels, size_t src_frames,
|
| - int dst_channels, size_t dst_frames)
|
| +AudioConverter::AudioConverter(size_t src_channels, size_t src_frames,
|
| + size_t dst_channels, size_t dst_frames)
|
| : src_channels_(src_channels),
|
| src_frames_(src_frames),
|
| dst_channels_(dst_channels),
|
|
|