Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1316523002 .
Cr-Commit-Position: refs/heads/master@{#11229}
diff --git a/talk/app/webrtc/mediastreaminterface.h b/talk/app/webrtc/mediastreaminterface.h
index 89a4abe..9b137d9 100644
--- a/talk/app/webrtc/mediastreaminterface.h
+++ b/talk/app/webrtc/mediastreaminterface.h
@@ -151,7 +151,7 @@
virtual void OnData(const void* audio_data,
int bits_per_sample,
int sample_rate,
- int number_of_channels,
+ size_t number_of_channels,
size_t number_of_frames) = 0;
protected:
diff --git a/talk/app/webrtc/rtpsender.cc b/talk/app/webrtc/rtpsender.cc
index c0d23a0..91e484b 100644
--- a/talk/app/webrtc/rtpsender.cc
+++ b/talk/app/webrtc/rtpsender.cc
@@ -44,7 +44,7 @@
void LocalAudioSinkAdapter::OnData(const void* audio_data,
int bits_per_sample,
int sample_rate,
- int number_of_channels,
+ size_t number_of_channels,
size_t number_of_frames) {
rtc::CritScope lock(&lock_);
if (sink_) {
diff --git a/talk/app/webrtc/rtpsender.h b/talk/app/webrtc/rtpsender.h
index d5f88a9..dd846b5 100644
--- a/talk/app/webrtc/rtpsender.h
+++ b/talk/app/webrtc/rtpsender.h
@@ -57,7 +57,7 @@
void OnData(const void* audio_data,
int bits_per_sample,
int sample_rate,
- int number_of_channels,
+ size_t number_of_channels,
size_t number_of_frames) override;
// cricket::AudioRenderer implementation.
diff --git a/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc b/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc
index e2dc123..6b675a9 100644
--- a/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc
+++ b/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc
@@ -58,7 +58,7 @@
int32_t RecordedDataIsAvailable(const void* audioSamples,
const size_t nSamples,
const size_t nBytesPerSample,
- const uint8_t nChannels,
+ const size_t nChannels,
const uint32_t samplesPerSec,
const uint32_t totalDelayMS,
const int32_t clockDrift,
@@ -82,7 +82,7 @@
// ADM is pulling data.
int32_t NeedMorePlayData(const size_t nSamples,
const size_t nBytesPerSample,
- const uint8_t nChannels,
+ const size_t nChannels,
const uint32_t samplesPerSec,
void* audioSamples,
size_t& nSamplesOut,
diff --git a/talk/app/webrtc/webrtcsdp.cc b/talk/app/webrtc/webrtcsdp.cc
index 07a4eb9..e287e90 100644
--- a/talk/app/webrtc/webrtcsdp.cc
+++ b/talk/app/webrtc/webrtcsdp.cc
@@ -2064,7 +2064,7 @@
struct StaticPayloadAudioCodec {
const char* name;
int clockrate;
- int channels;
+ size_t channels;
};
static const StaticPayloadAudioCodec kStaticPayloadAudioCodecs[] = {
{ "PCMU", 8000, 1 },
@@ -2103,7 +2103,7 @@
payload_type < arraysize(kStaticPayloadAudioCodecs)) {
std::string encoding_name = kStaticPayloadAudioCodecs[payload_type].name;
int clock_rate = kStaticPayloadAudioCodecs[payload_type].clockrate;
- int channels = kStaticPayloadAudioCodecs[payload_type].channels;
+ size_t channels = kStaticPayloadAudioCodecs[payload_type].channels;
media_desc->AddCodec(cricket::AudioCodec(payload_type, encoding_name,
clock_rate, 0, channels,
preference));
@@ -2838,7 +2838,7 @@
// Updates or creates a new codec entry in the audio description with according
// to |name|, |clockrate|, |bitrate|, |channels| and |preference|.
void UpdateCodec(int payload_type, const std::string& name, int clockrate,
- int bitrate, int channels, int preference,
+ int bitrate, size_t channels, int preference,
AudioContentDescription* audio_desc) {
// Codec may already be populated with (only) optional parameters
// (from an fmtp).
@@ -2937,7 +2937,7 @@
// of audio channels. This parameter is OPTIONAL and may be
// omitted if the number of channels is one, provided that no
// additional parameters are needed.
- int channels = 1;
+ size_t channels = 1;
if (codec_params.size() == 3) {
if (!GetValueFromString(line, codec_params[2], &channels, error)) {
return false;
diff --git a/talk/media/base/audiorenderer.h b/talk/media/base/audiorenderer.h
index 229c36e..a42cd7d 100644
--- a/talk/media/base/audiorenderer.h
+++ b/talk/media/base/audiorenderer.h
@@ -41,7 +41,7 @@
virtual void OnData(const void* audio_data,
int bits_per_sample,
int sample_rate,
- int number_of_channels,
+ size_t number_of_channels,
size_t number_of_frames) = 0;
// Called when the AudioRenderer is going away.
diff --git a/talk/media/base/codec.cc b/talk/media/base/codec.cc
index 101d2be..59708b3 100644
--- a/talk/media/base/codec.cc
+++ b/talk/media/base/codec.cc
@@ -167,7 +167,7 @@
const std::string& name,
int clockrate,
int bitrate,
- int channels,
+ size_t channels,
int preference)
: Codec(id, name, clockrate, preference),
bitrate(bitrate),
diff --git a/talk/media/base/codec.h b/talk/media/base/codec.h
index fcdb6ea..da78e1c 100644
--- a/talk/media/base/codec.h
+++ b/talk/media/base/codec.h
@@ -128,14 +128,14 @@
struct AudioCodec : public Codec {
int bitrate;
- int channels;
+ size_t channels;
// Creates a codec with the given parameters.
AudioCodec(int id,
const std::string& name,
int clockrate,
int bitrate,
- int channels,
+ size_t channels,
int preference);
// Creates an empty codec.
AudioCodec();
diff --git a/talk/media/base/fakemediaengine.h b/talk/media/base/fakemediaengine.h
index f5b2174..149704f 100644
--- a/talk/media/base/fakemediaengine.h
+++ b/talk/media/base/fakemediaengine.h
@@ -368,7 +368,7 @@
void OnData(const void* audio_data,
int bits_per_sample,
int sample_rate,
- int number_of_channels,
+ size_t number_of_channels,
size_t number_of_frames) override {}
void OnClose() override { renderer_ = NULL; }
AudioRenderer* renderer() const { return renderer_; }
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h
index eb3b4d3..65ba927 100644
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h
@@ -77,10 +77,10 @@
WEBRTC_STUB_CONST(input_sample_rate_hz, ());
WEBRTC_STUB_CONST(proc_sample_rate_hz, ());
WEBRTC_STUB_CONST(proc_split_sample_rate_hz, ());
- WEBRTC_STUB_CONST(num_input_channels, ());
- WEBRTC_STUB_CONST(num_proc_channels, ());
- WEBRTC_STUB_CONST(num_output_channels, ());
- WEBRTC_STUB_CONST(num_reverse_channels, ());
+ size_t num_input_channels() const override { return 0; }
+ size_t num_proc_channels() const override { return 0; }
+ size_t num_output_channels() const override { return 0; }
+ size_t num_reverse_channels() const override { return 0; }
WEBRTC_VOID_STUB(set_output_will_be_muted, (bool muted));
WEBRTC_STUB(ProcessStream, (webrtc::AudioFrame* frame));
WEBRTC_STUB(ProcessStream, (
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 9eee2af..9742564 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -410,7 +410,7 @@
struct CodecPref {
const char* name;
int clockrate;
- int channels;
+ size_t channels;
int payload_type;
bool is_multi_rate;
int packet_sizes_ms[kMaxNumPacketSize];
@@ -1155,7 +1155,7 @@
void OnData(const void* audio_data,
int bits_per_sample,
int sample_rate,
- int number_of_channels,
+ size_t number_of_channels,
size_t number_of_frames) override {
RTC_DCHECK(!worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(audio_capture_thread_checker_.CalledOnValidThread());
diff --git a/webrtc/audio/audio_sink.h b/webrtc/audio/audio_sink.h
index d022b32..999644f 100644
--- a/webrtc/audio/audio_sink.h
+++ b/webrtc/audio/audio_sink.h
@@ -30,7 +30,7 @@
Data(int16_t* data,
size_t samples_per_channel,
int sample_rate,
- int channels,
+ size_t channels,
uint32_t timestamp)
: data(data),
samples_per_channel(samples_per_channel),
@@ -41,7 +41,7 @@
int16_t* data; // The actual 16bit audio data.
size_t samples_per_channel; // Number of frames in the buffer.
int sample_rate; // Sample rate in Hz.
- int channels; // Number of channels in the audio data.
+ size_t channels; // Number of channels in the audio data.
uint32_t timestamp; // The RTP timestamp of the first sample.
};
diff --git a/webrtc/common_audio/audio_converter.cc b/webrtc/common_audio/audio_converter.cc
index 9bb5895..9ebfabc 100644
--- a/webrtc/common_audio/audio_converter.cc
+++ b/webrtc/common_audio/audio_converter.cc
@@ -25,7 +25,7 @@
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 @@
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 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 @@
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 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 @@
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 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 @@
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 @@
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),
diff --git a/webrtc/common_audio/audio_converter.h b/webrtc/common_audio/audio_converter.h
index 7d1513b..c5f08c1 100644
--- a/webrtc/common_audio/audio_converter.h
+++ b/webrtc/common_audio/audio_converter.h
@@ -26,9 +26,9 @@
public:
// Returns a new AudioConverter, which will use the supplied format for its
// lifetime. Caller is responsible for the memory.
- static rtc::scoped_ptr<AudioConverter> Create(int src_channels,
+ static rtc::scoped_ptr<AudioConverter> Create(size_t src_channels,
size_t src_frames,
- int dst_channels,
+ size_t dst_channels,
size_t dst_frames);
virtual ~AudioConverter() {};
@@ -39,23 +39,23 @@
virtual void Convert(const float* const* src, size_t src_size,
float* const* dst, size_t dst_capacity) = 0;
- int src_channels() const { return src_channels_; }
+ size_t src_channels() const { return src_channels_; }
size_t src_frames() const { return src_frames_; }
- int dst_channels() const { return dst_channels_; }
+ size_t dst_channels() const { return dst_channels_; }
size_t dst_frames() const { return dst_frames_; }
protected:
AudioConverter();
- AudioConverter(int src_channels, size_t src_frames, int dst_channels,
+ AudioConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
size_t dst_frames);
// Helper to RTC_CHECK that inputs are correctly sized.
void CheckSizes(size_t src_size, size_t dst_capacity) const;
private:
- const int src_channels_;
+ const size_t src_channels_;
const size_t src_frames_;
- const int dst_channels_;
+ const size_t dst_channels_;
const size_t dst_frames_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioConverter);
diff --git a/webrtc/common_audio/audio_converter_unittest.cc b/webrtc/common_audio/audio_converter_unittest.cc
index e373d78..dace0bd 100644
--- a/webrtc/common_audio/audio_converter_unittest.cc
+++ b/webrtc/common_audio/audio_converter_unittest.cc
@@ -26,9 +26,9 @@
// Sets the signal value to increase by |data| with every sample.
ScopedBuffer CreateBuffer(const std::vector<float>& data, size_t frames) {
- const int num_channels = static_cast<int>(data.size());
+ const size_t num_channels = data.size();
ScopedBuffer sb(new ChannelBuffer<float>(frames, num_channels));
- for (int i = 0; i < num_channels; ++i)
+ for (size_t i = 0; i < num_channels; ++i)
for (size_t j = 0; j < frames; ++j)
sb->channels()[i][j] = data[i] * j;
return sb;
@@ -57,7 +57,7 @@
float mse = 0;
float variance = 0;
float mean = 0;
- for (int i = 0; i < ref.num_channels(); ++i) {
+ for (size_t i = 0; i < ref.num_channels(); ++i) {
for (size_t j = 0; j < ref.num_frames() - delay; ++j) {
float error = ref.channels()[i][j] - test.channels()[i][j + delay];
mse += error * error;
@@ -86,9 +86,9 @@
// Sets the source to a linearly increasing signal for which we can easily
// generate a reference. Runs the AudioConverter and ensures the output has
// sufficiently high SNR relative to the reference.
-void RunAudioConverterTest(int src_channels,
+void RunAudioConverterTest(size_t src_channels,
int src_sample_rate_hz,
- int dst_channels,
+ size_t dst_channels,
int dst_sample_rate_hz) {
const float kSrcLeft = 0.0002f;
const float kSrcRight = 0.0001f;
@@ -128,8 +128,9 @@
static_cast<size_t>(
PushSincResampler::AlgorithmicDelaySeconds(src_sample_rate_hz) *
dst_sample_rate_hz);
- printf("(%d, %d Hz) -> (%d, %d Hz) ", // SNR reported on the same line later.
- src_channels, src_sample_rate_hz, dst_channels, dst_sample_rate_hz);
+ // SNR reported on the same line later.
+ printf("(%" PRIuS ", %d Hz) -> (%" PRIuS ", %d Hz) ",
+ src_channels, src_sample_rate_hz, dst_channels, dst_sample_rate_hz);
rtc::scoped_ptr<AudioConverter> converter = AudioConverter::Create(
src_channels, src_frames, dst_channels, dst_frames);
@@ -142,7 +143,7 @@
TEST(AudioConverterTest, ConversionsPassSNRThreshold) {
const int kSampleRates[] = {8000, 16000, 32000, 44100, 48000};
- const int kChannels[] = {1, 2};
+ const size_t kChannels[] = {1, 2};
for (size_t src_rate = 0; src_rate < arraysize(kSampleRates); ++src_rate) {
for (size_t dst_rate = 0; dst_rate < arraysize(kSampleRates); ++dst_rate) {
for (size_t src_channel = 0; src_channel < arraysize(kChannels);
diff --git a/webrtc/common_audio/blocker.cc b/webrtc/common_audio/blocker.cc
index 0133550..13432f2 100644
--- a/webrtc/common_audio/blocker.cc
+++ b/webrtc/common_audio/blocker.cc
@@ -22,10 +22,10 @@
const float* const* b,
int b_start_index,
size_t num_frames,
- int num_channels,
+ size_t num_channels,
float* const* result,
size_t result_start_index) {
- for (int i = 0; i < num_channels; ++i) {
+ for (size_t i = 0; i < num_channels; ++i) {
for (size_t j = 0; j < num_frames; ++j) {
result[i][j + result_start_index] =
a[i][j + a_start_index] + b[i][j + b_start_index];
@@ -37,10 +37,10 @@
void CopyFrames(const float* const* src,
size_t src_start_index,
size_t num_frames,
- int num_channels,
+ size_t num_channels,
float* const* dst,
size_t dst_start_index) {
- for (int i = 0; i < num_channels; ++i) {
+ for (size_t i = 0; i < num_channels; ++i) {
memcpy(&dst[i][dst_start_index],
&src[i][src_start_index],
num_frames * sizeof(dst[i][dst_start_index]));
@@ -51,10 +51,10 @@
void MoveFrames(const float* const* src,
size_t src_start_index,
size_t num_frames,
- int num_channels,
+ size_t num_channels,
float* const* dst,
size_t dst_start_index) {
- for (int i = 0; i < num_channels; ++i) {
+ for (size_t i = 0; i < num_channels; ++i) {
memmove(&dst[i][dst_start_index],
&src[i][src_start_index],
num_frames * sizeof(dst[i][dst_start_index]));
@@ -64,8 +64,8 @@
void ZeroOut(float* const* buffer,
size_t starting_idx,
size_t num_frames,
- int num_channels) {
- for (int i = 0; i < num_channels; ++i) {
+ size_t num_channels) {
+ for (size_t i = 0; i < num_channels; ++i) {
memset(&buffer[i][starting_idx], 0,
num_frames * sizeof(buffer[i][starting_idx]));
}
@@ -75,9 +75,9 @@
// stored in |frames|.
void ApplyWindow(const float* window,
size_t num_frames,
- int num_channels,
+ size_t num_channels,
float* const* frames) {
- for (int i = 0; i < num_channels; ++i) {
+ for (size_t i = 0; i < num_channels; ++i) {
for (size_t j = 0; j < num_frames; ++j) {
frames[i][j] = frames[i][j] * window[j];
}
@@ -100,8 +100,8 @@
Blocker::Blocker(size_t chunk_size,
size_t block_size,
- int num_input_channels,
- int num_output_channels,
+ size_t num_input_channels,
+ size_t num_output_channels,
const float* window,
size_t shift_amount,
BlockerCallback* callback)
@@ -166,8 +166,8 @@
// TODO(claguna): Look at using ring buffers to eliminate some copies.
void Blocker::ProcessChunk(const float* const* input,
size_t chunk_size,
- int num_input_channels,
- int num_output_channels,
+ size_t num_input_channels,
+ size_t num_output_channels,
float* const* output) {
RTC_CHECK_EQ(chunk_size, chunk_size_);
RTC_CHECK_EQ(num_input_channels, num_input_channels_);
diff --git a/webrtc/common_audio/blocker.h b/webrtc/common_audio/blocker.h
index 025638a..3a67c13 100644
--- a/webrtc/common_audio/blocker.h
+++ b/webrtc/common_audio/blocker.h
@@ -26,8 +26,8 @@
virtual void ProcessBlock(const float* const* input,
size_t num_frames,
- int num_input_channels,
- int num_output_channels,
+ size_t num_input_channels,
+ size_t num_output_channels,
float* const* output) = 0;
};
@@ -65,23 +65,23 @@
public:
Blocker(size_t chunk_size,
size_t block_size,
- int num_input_channels,
- int num_output_channels,
+ size_t num_input_channels,
+ size_t num_output_channels,
const float* window,
size_t shift_amount,
BlockerCallback* callback);
void ProcessChunk(const float* const* input,
size_t chunk_size,
- int num_input_channels,
- int num_output_channels,
+ size_t num_input_channels,
+ size_t num_output_channels,
float* const* output);
private:
const size_t chunk_size_;
const size_t block_size_;
- const int num_input_channels_;
- const int num_output_channels_;
+ const size_t num_input_channels_;
+ const size_t num_output_channels_;
// The number of frames of delay to add at the beginning of the first chunk.
const size_t initial_delay_;
diff --git a/webrtc/common_audio/blocker_unittest.cc b/webrtc/common_audio/blocker_unittest.cc
index 065c09e..a5a7b56 100644
--- a/webrtc/common_audio/blocker_unittest.cc
+++ b/webrtc/common_audio/blocker_unittest.cc
@@ -20,10 +20,10 @@
public:
void ProcessBlock(const float* const* input,
size_t num_frames,
- int num_input_channels,
- int num_output_channels,
+ size_t num_input_channels,
+ size_t num_output_channels,
float* const* output) override {
- for (int i = 0; i < num_output_channels; ++i) {
+ for (size_t i = 0; i < num_output_channels; ++i) {
for (size_t j = 0; j < num_frames; ++j) {
output[i][j] = input[i][j] + 3;
}
@@ -36,10 +36,10 @@
public:
void ProcessBlock(const float* const* input,
size_t num_frames,
- int num_input_channels,
- int num_output_channels,
+ size_t num_input_channels,
+ size_t num_output_channels,
float* const* output) override {
- for (int i = 0; i < num_output_channels; ++i) {
+ for (size_t i = 0; i < num_output_channels; ++i) {
for (size_t j = 0; j < num_frames; ++j) {
output[i][j] = input[i][j];
}
@@ -63,8 +63,8 @@
float* const* input_chunk,
float* const* output,
float* const* output_chunk,
- int num_input_channels,
- int num_output_channels) {
+ size_t num_input_channels,
+ size_t num_output_channels) {
size_t start = 0;
size_t end = chunk_size - 1;
while (end < num_frames) {
@@ -83,9 +83,9 @@
void ValidateSignalEquality(const float* const* expected,
const float* const* actual,
- int num_channels,
+ size_t num_channels,
size_t num_frames) {
- for (int i = 0; i < num_channels; ++i) {
+ for (size_t i = 0; i < num_channels; ++i) {
for (size_t j = 0; j < num_frames; ++j) {
EXPECT_FLOAT_EQ(expected[i][j], actual[i][j]);
}
@@ -93,10 +93,10 @@
}
void ValidateInitialDelay(const float* const* output,
- int num_channels,
+ size_t num_channels,
size_t num_frames,
size_t initial_delay) {
- for (int i = 0; i < num_channels; ++i) {
+ for (size_t i = 0; i < num_channels; ++i) {
for (size_t j = 0; j < num_frames; ++j) {
if (j < initial_delay) {
EXPECT_FLOAT_EQ(output[i][j], 0.f);
@@ -110,10 +110,10 @@
static void CopyTo(float* const* dst,
size_t start_index_dst,
size_t start_index_src,
- int num_channels,
+ size_t num_channels,
size_t num_frames,
const float* const* src) {
- for (int i = 0; i < num_channels; ++i) {
+ for (size_t i = 0; i < num_channels; ++i) {
memcpy(&dst[i][start_index_dst],
&src[i][start_index_src],
num_frames * sizeof(float));
@@ -122,8 +122,8 @@
};
TEST_F(BlockerTest, TestBlockerMutuallyPrimeChunkandBlockSize) {
- const int kNumInputChannels = 3;
- const int kNumOutputChannels = 2;
+ const size_t kNumInputChannels = 3;
+ const size_t kNumOutputChannels = 2;
const size_t kNumFrames = 10;
const size_t kBlockSize = 4;
const size_t kChunkSize = 5;
@@ -175,8 +175,8 @@
}
TEST_F(BlockerTest, TestBlockerMutuallyPrimeShiftAndBlockSize) {
- const int kNumInputChannels = 3;
- const int kNumOutputChannels = 2;
+ const size_t kNumInputChannels = 3;
+ const size_t kNumOutputChannels = 2;
const size_t kNumFrames = 12;
const size_t kBlockSize = 4;
const size_t kChunkSize = 6;
@@ -228,8 +228,8 @@
}
TEST_F(BlockerTest, TestBlockerNoOverlap) {
- const int kNumInputChannels = 3;
- const int kNumOutputChannels = 2;
+ const size_t kNumInputChannels = 3;
+ const size_t kNumOutputChannels = 2;
const size_t kNumFrames = 12;
const size_t kBlockSize = 4;
const size_t kChunkSize = 4;
@@ -281,8 +281,8 @@
}
TEST_F(BlockerTest, InitialDelaysAreMinimum) {
- const int kNumInputChannels = 3;
- const int kNumOutputChannels = 2;
+ const size_t kNumInputChannels = 3;
+ const size_t kNumOutputChannels = 2;
const size_t kNumFrames = 1280;
const size_t kChunkSize[] =
{80, 80, 80, 80, 80, 80, 160, 160, 160, 160, 160, 160};
@@ -294,7 +294,7 @@
{48, 48, 48, 112, 112, 112, 96, 96, 96, 224, 224, 224};
float input[kNumInputChannels][kNumFrames];
- for (int i = 0; i < kNumInputChannels; ++i) {
+ for (size_t i = 0; i < kNumInputChannels; ++i) {
for (size_t j = 0; j < kNumFrames; ++j) {
input[i][j] = i + 1;
}
diff --git a/webrtc/common_audio/channel_buffer.cc b/webrtc/common_audio/channel_buffer.cc
index d3dc7c0..44520c6 100644
--- a/webrtc/common_audio/channel_buffer.cc
+++ b/webrtc/common_audio/channel_buffer.cc
@@ -13,7 +13,7 @@
namespace webrtc {
IFChannelBuffer::IFChannelBuffer(size_t num_frames,
- int num_channels,
+ size_t num_channels,
size_t num_bands)
: ivalid_(true),
ibuf_(num_frames, num_channels, num_bands),
@@ -47,7 +47,7 @@
assert(ivalid_);
const int16_t* const* int_channels = ibuf_.channels();
float* const* float_channels = fbuf_.channels();
- for (int i = 0; i < ibuf_.num_channels(); ++i) {
+ for (size_t i = 0; i < ibuf_.num_channels(); ++i) {
for (size_t j = 0; j < ibuf_.num_frames(); ++j) {
float_channels[i][j] = int_channels[i][j];
}
@@ -61,7 +61,7 @@
assert(fvalid_);
int16_t* const* int_channels = ibuf_.channels();
const float* const* float_channels = fbuf_.channels();
- for (int i = 0; i < ibuf_.num_channels(); ++i) {
+ for (size_t i = 0; i < ibuf_.num_channels(); ++i) {
FloatS16ToS16(float_channels[i],
ibuf_.num_frames(),
int_channels[i]);
diff --git a/webrtc/common_audio/channel_buffer.h b/webrtc/common_audio/channel_buffer.h
index 6050090..d906916 100644
--- a/webrtc/common_audio/channel_buffer.h
+++ b/webrtc/common_audio/channel_buffer.h
@@ -40,7 +40,7 @@
class ChannelBuffer {
public:
ChannelBuffer(size_t num_frames,
- int num_channels,
+ size_t num_channels,
size_t num_bands = 1)
: data_(new T[num_frames * num_channels]()),
channels_(new T*[num_channels * num_bands]),
@@ -49,7 +49,7 @@
num_frames_per_band_(num_frames / num_bands),
num_channels_(num_channels),
num_bands_(num_bands) {
- for (int i = 0; i < num_channels_; ++i) {
+ for (size_t i = 0; i < num_channels_; ++i) {
for (size_t j = 0; j < num_bands_; ++j) {
channels_[j * num_channels_ + i] =
&data_[i * num_frames_ + j * num_frames_per_band_];
@@ -90,12 +90,12 @@
// 0 <= channel < |num_channels_|
// 0 <= band < |num_bands_|
// 0 <= sample < |num_frames_per_band_|
- const T* const* bands(int channel) const {
+ const T* const* bands(size_t channel) const {
RTC_DCHECK_LT(channel, num_channels_);
- RTC_DCHECK_GE(channel, 0);
+ RTC_DCHECK_GE(channel, 0u);
return &bands_[channel * num_bands_];
}
- T* const* bands(int channel) {
+ T* const* bands(size_t channel) {
const ChannelBuffer<T>* t = this;
return const_cast<T* const*>(t->bands(channel));
}
@@ -104,7 +104,7 @@
// Returns |slice| for convenience.
const T* const* Slice(T** slice, size_t start_frame) const {
RTC_DCHECK_LT(start_frame, num_frames_);
- for (int i = 0; i < num_channels_; ++i)
+ for (size_t i = 0; i < num_channels_; ++i)
slice[i] = &channels_[i][start_frame];
return slice;
}
@@ -115,7 +115,7 @@
size_t num_frames() const { return num_frames_; }
size_t num_frames_per_band() const { return num_frames_per_band_; }
- int num_channels() const { return num_channels_; }
+ size_t num_channels() const { return num_channels_; }
size_t num_bands() const { return num_bands_; }
size_t size() const {return num_frames_ * num_channels_; }
@@ -130,7 +130,7 @@
rtc::scoped_ptr<T* []> bands_;
const size_t num_frames_;
const size_t num_frames_per_band_;
- const int num_channels_;
+ const size_t num_channels_;
const size_t num_bands_;
};
@@ -142,7 +142,7 @@
// fbuf() until the next call to any of the other functions.
class IFChannelBuffer {
public:
- IFChannelBuffer(size_t num_frames, int num_channels, size_t num_bands = 1);
+ IFChannelBuffer(size_t num_frames, size_t num_channels, size_t num_bands = 1);
ChannelBuffer<int16_t>* ibuf();
ChannelBuffer<float>* fbuf();
@@ -151,7 +151,7 @@
size_t num_frames() const { return ibuf_.num_frames(); }
size_t num_frames_per_band() const { return ibuf_.num_frames_per_band(); }
- int num_channels() const { return ibuf_.num_channels(); }
+ size_t num_channels() const { return ibuf_.num_channels(); }
size_t num_bands() const { return ibuf_.num_bands(); }
private:
diff --git a/webrtc/common_audio/include/audio_util.h b/webrtc/common_audio/include/audio_util.h
index 2c0028c..55dfc06 100644
--- a/webrtc/common_audio/include/audio_util.h
+++ b/webrtc/common_audio/include/audio_util.h
@@ -87,11 +87,11 @@
template <typename T>
void Deinterleave(const T* interleaved,
size_t samples_per_channel,
- int num_channels,
+ size_t num_channels,
T* const* deinterleaved) {
- for (int i = 0; i < num_channels; ++i) {
+ for (size_t i = 0; i < num_channels; ++i) {
T* channel = deinterleaved[i];
- int interleaved_idx = i;
+ size_t interleaved_idx = i;
for (size_t j = 0; j < samples_per_channel; ++j) {
channel[j] = interleaved[interleaved_idx];
interleaved_idx += num_channels;
@@ -105,11 +105,11 @@
template <typename T>
void Interleave(const T* const* deinterleaved,
size_t samples_per_channel,
- int num_channels,
+ size_t num_channels,
T* interleaved) {
- for (int i = 0; i < num_channels; ++i) {
+ for (size_t i = 0; i < num_channels; ++i) {
const T* channel = deinterleaved[i];
- int interleaved_idx = i;
+ size_t interleaved_idx = i;
for (size_t j = 0; j < samples_per_channel; ++j) {
interleaved[interleaved_idx] = channel[j];
interleaved_idx += num_channels;
diff --git a/webrtc/common_audio/lapped_transform.cc b/webrtc/common_audio/lapped_transform.cc
index c01f1d9..5ab1db1 100644
--- a/webrtc/common_audio/lapped_transform.cc
+++ b/webrtc/common_audio/lapped_transform.cc
@@ -21,14 +21,14 @@
void LappedTransform::BlockThunk::ProcessBlock(const float* const* input,
size_t num_frames,
- int num_input_channels,
- int num_output_channels,
+ size_t num_input_channels,
+ size_t num_output_channels,
float* const* output) {
RTC_CHECK_EQ(num_input_channels, parent_->num_in_channels_);
RTC_CHECK_EQ(num_output_channels, parent_->num_out_channels_);
RTC_CHECK_EQ(parent_->block_length_, num_frames);
- for (int i = 0; i < num_input_channels; ++i) {
+ for (size_t i = 0; i < num_input_channels; ++i) {
memcpy(parent_->real_buf_.Row(i), input[i],
num_frames * sizeof(*input[0]));
parent_->fft_->Forward(parent_->real_buf_.Row(i),
@@ -44,7 +44,7 @@
num_output_channels,
parent_->cplx_post_.Array());
- for (int i = 0; i < num_output_channels; ++i) {
+ for (size_t i = 0; i < num_output_channels; ++i) {
parent_->fft_->Inverse(parent_->cplx_post_.Row(i),
parent_->real_buf_.Row(i));
memcpy(output[i], parent_->real_buf_.Row(i),
@@ -52,8 +52,8 @@
}
}
-LappedTransform::LappedTransform(int num_in_channels,
- int num_out_channels,
+LappedTransform::LappedTransform(size_t num_in_channels,
+ size_t num_out_channels,
size_t chunk_length,
const float* window,
size_t block_length,
diff --git a/webrtc/common_audio/lapped_transform.h b/webrtc/common_audio/lapped_transform.h
index 21e10e3..1373ca1 100644
--- a/webrtc/common_audio/lapped_transform.h
+++ b/webrtc/common_audio/lapped_transform.h
@@ -35,8 +35,8 @@
virtual ~Callback() {}
virtual void ProcessAudioBlock(const std::complex<float>* const* in_block,
- int num_in_channels, size_t frames,
- int num_out_channels,
+ size_t num_in_channels, size_t frames,
+ size_t num_out_channels,
std::complex<float>* const* out_block) = 0;
};
@@ -46,8 +46,8 @@
// |block_length| defines the length of a block, in samples.
// |shift_amount| is in samples. |callback| is the caller-owned audio
// processing function called for each block of the input chunk.
- LappedTransform(int num_in_channels,
- int num_out_channels,
+ LappedTransform(size_t num_in_channels,
+ size_t num_out_channels,
size_t chunk_length,
const float* window,
size_t block_length,
@@ -75,7 +75,7 @@
// in_chunk.
//
// Returns the same num_in_channels passed to the LappedTransform constructor.
- int num_in_channels() const { return num_in_channels_; }
+ size_t num_in_channels() const { return num_in_channels_; }
// Get the number of output channels.
//
@@ -84,7 +84,7 @@
//
// Returns the same num_out_channels passed to the LappedTransform
// constructor.
- int num_out_channels() const { return num_out_channels_; }
+ size_t num_out_channels() const { return num_out_channels_; }
private:
// Internal middleware callback, given to the blocker. Transforms each block
@@ -93,16 +93,18 @@
public:
explicit BlockThunk(LappedTransform* parent) : parent_(parent) {}
- virtual void ProcessBlock(const float* const* input, size_t num_frames,
- int num_input_channels, int num_output_channels,
+ virtual void ProcessBlock(const float* const* input,
+ size_t num_frames,
+ size_t num_input_channels,
+ size_t num_output_channels,
float* const* output);
private:
LappedTransform* const parent_;
} blocker_callback_;
- const int num_in_channels_;
- const int num_out_channels_;
+ const size_t num_in_channels_;
+ const size_t num_out_channels_;
const size_t block_length_;
const size_t chunk_length_;
diff --git a/webrtc/common_audio/lapped_transform_unittest.cc b/webrtc/common_audio/lapped_transform_unittest.cc
index eb1c80f..a78488e 100644
--- a/webrtc/common_audio/lapped_transform_unittest.cc
+++ b/webrtc/common_audio/lapped_transform_unittest.cc
@@ -25,12 +25,12 @@
NoopCallback() : block_num_(0) {}
virtual void ProcessAudioBlock(const complex<float>* const* in_block,
- int in_channels,
+ size_t in_channels,
size_t frames,
- int out_channels,
+ size_t out_channels,
complex<float>* const* out_block) {
RTC_CHECK_EQ(in_channels, out_channels);
- for (int i = 0; i < out_channels; ++i) {
+ for (size_t i = 0; i < out_channels; ++i) {
memcpy(out_block[i], in_block[i], sizeof(**in_block) * frames);
}
++block_num_;
@@ -49,9 +49,9 @@
FftCheckerCallback() : block_num_(0) {}
virtual void ProcessAudioBlock(const complex<float>* const* in_block,
- int in_channels,
+ size_t in_channels,
size_t frames,
- int out_channels,
+ size_t out_channels,
complex<float>* const* out_block) {
RTC_CHECK_EQ(in_channels, out_channels);
@@ -90,7 +90,7 @@
namespace webrtc {
TEST(LappedTransformTest, Windowless) {
- const int kChannels = 3;
+ const size_t kChannels = 3;
const size_t kChunkLength = 512;
const size_t kBlockLength = 64;
const size_t kShiftAmount = 64;
@@ -118,7 +118,7 @@
trans.ProcessChunk(in_chunk, out_chunk);
- for (int i = 0; i < kChannels; ++i) {
+ for (size_t i = 0; i < kChannels; ++i) {
for (size_t j = 0; j < kChunkLength; ++j) {
ASSERT_NEAR(out_chunk[i][j], 2.0f, 1e-5f);
}
diff --git a/webrtc/common_audio/resampler/include/push_resampler.h b/webrtc/common_audio/resampler/include/push_resampler.h
index b5c0003..eeda790 100644
--- a/webrtc/common_audio/resampler/include/push_resampler.h
+++ b/webrtc/common_audio/resampler/include/push_resampler.h
@@ -29,7 +29,7 @@
// Must be called whenever the parameters change. Free to be called at any
// time as it is a no-op if parameters have not changed since the last call.
int InitializeIfNeeded(int src_sample_rate_hz, int dst_sample_rate_hz,
- int num_channels);
+ size_t num_channels);
// Returns the total number of samples provided in destination (e.g. 32 kHz,
// 2 channel audio gives 640 samples).
@@ -40,7 +40,7 @@
rtc::scoped_ptr<PushSincResampler> sinc_resampler_right_;
int src_sample_rate_hz_;
int dst_sample_rate_hz_;
- int num_channels_;
+ size_t num_channels_;
rtc::scoped_ptr<T[]> src_left_;
rtc::scoped_ptr<T[]> src_right_;
rtc::scoped_ptr<T[]> dst_left_;
diff --git a/webrtc/common_audio/resampler/include/resampler.h b/webrtc/common_audio/resampler/include/resampler.h
index 0d4c1af..e26ac90 100644
--- a/webrtc/common_audio/resampler/include/resampler.h
+++ b/webrtc/common_audio/resampler/include/resampler.h
@@ -28,14 +28,14 @@
public:
Resampler();
- Resampler(int inFreq, int outFreq, int num_channels);
+ Resampler(int inFreq, int outFreq, size_t num_channels);
~Resampler();
// Reset all states
- int Reset(int inFreq, int outFreq, int num_channels);
+ int Reset(int inFreq, int outFreq, size_t num_channels);
// Reset all states if any parameter has changed
- int ResetIfNeeded(int inFreq, int outFreq, int num_channels);
+ int ResetIfNeeded(int inFreq, int outFreq, size_t num_channels);
// Resample samplesIn to samplesOut.
int Push(const int16_t* samplesIn, size_t lengthIn, int16_t* samplesOut,
@@ -83,7 +83,7 @@
int my_in_frequency_khz_;
int my_out_frequency_khz_;
ResamplerMode my_mode_;
- int num_channels_;
+ size_t num_channels_;
// Extra instance for stereo
Resampler* slave_left_;
diff --git a/webrtc/common_audio/resampler/push_resampler.cc b/webrtc/common_audio/resampler/push_resampler.cc
index 566acde..f654e9a 100644
--- a/webrtc/common_audio/resampler/push_resampler.cc
+++ b/webrtc/common_audio/resampler/push_resampler.cc
@@ -32,7 +32,7 @@
template <typename T>
int PushResampler<T>::InitializeIfNeeded(int src_sample_rate_hz,
int dst_sample_rate_hz,
- int num_channels) {
+ size_t num_channels) {
if (src_sample_rate_hz == src_sample_rate_hz_ &&
dst_sample_rate_hz == dst_sample_rate_hz_ &&
num_channels == num_channels_)
@@ -68,10 +68,8 @@
template <typename T>
int PushResampler<T>::Resample(const T* src, size_t src_length, T* dst,
size_t dst_capacity) {
- const size_t src_size_10ms =
- static_cast<size_t>(src_sample_rate_hz_ * num_channels_ / 100);
- const size_t dst_size_10ms =
- static_cast<size_t>(dst_sample_rate_hz_ * num_channels_ / 100);
+ const size_t src_size_10ms = src_sample_rate_hz_ * num_channels_ / 100;
+ const size_t dst_size_10ms = dst_sample_rate_hz_ * num_channels_ / 100;
if (src_length != src_size_10ms || dst_capacity < dst_size_10ms)
return -1;
diff --git a/webrtc/common_audio/resampler/resampler.cc b/webrtc/common_audio/resampler/resampler.cc
index c9e7a1f..7c690fc 100644
--- a/webrtc/common_audio/resampler/resampler.cc
+++ b/webrtc/common_audio/resampler/resampler.cc
@@ -39,7 +39,7 @@
slave_right_(nullptr) {
}
-Resampler::Resampler(int inFreq, int outFreq, int num_channels)
+Resampler::Resampler(int inFreq, int outFreq, size_t num_channels)
: Resampler() {
Reset(inFreq, outFreq, num_channels);
}
@@ -76,7 +76,7 @@
}
}
-int Resampler::ResetIfNeeded(int inFreq, int outFreq, int num_channels)
+int Resampler::ResetIfNeeded(int inFreq, int outFreq, size_t num_channels)
{
int tmpInFreq_kHz = inFreq / 1000;
int tmpOutFreq_kHz = outFreq / 1000;
@@ -91,7 +91,7 @@
}
}
-int Resampler::Reset(int inFreq, int outFreq, int num_channels)
+int Resampler::Reset(int inFreq, int outFreq, size_t num_channels)
{
if (num_channels != 1 && num_channels != 2) {
return -1;
diff --git a/webrtc/common_audio/wav_file.cc b/webrtc/common_audio/wav_file.cc
index 27d335c..94b7a3c 100644
--- a/webrtc/common_audio/wav_file.cc
+++ b/webrtc/common_audio/wav_file.cc
@@ -99,7 +99,7 @@
}
WavWriter::WavWriter(const std::string& filename, int sample_rate,
- int num_channels)
+ size_t num_channels)
: sample_rate_(sample_rate),
num_channels_(num_channels),
num_samples_(0),
@@ -153,7 +153,7 @@
rtc_WavWriter* rtc_WavOpen(const char* filename,
int sample_rate,
- int num_channels) {
+ size_t num_channels) {
return reinterpret_cast<rtc_WavWriter*>(
new webrtc::WavWriter(filename, sample_rate, num_channels));
}
@@ -172,7 +172,7 @@
return reinterpret_cast<const webrtc::WavWriter*>(wf)->sample_rate();
}
-int rtc_WavNumChannels(const rtc_WavWriter* wf) {
+size_t rtc_WavNumChannels(const rtc_WavWriter* wf) {
return reinterpret_cast<const webrtc::WavWriter*>(wf)->num_channels();
}
diff --git a/webrtc/common_audio/wav_file.h b/webrtc/common_audio/wav_file.h
index eb2ce1e..e656eb8 100644
--- a/webrtc/common_audio/wav_file.h
+++ b/webrtc/common_audio/wav_file.h
@@ -27,7 +27,7 @@
virtual ~WavFile() {}
virtual int sample_rate() const = 0;
- virtual int num_channels() const = 0;
+ virtual size_t num_channels() const = 0;
virtual size_t num_samples() const = 0;
// Returns a human-readable string containing the audio format.
@@ -39,7 +39,7 @@
class WavWriter final : public WavFile {
public:
// Open a new WAV file for writing.
- WavWriter(const std::string& filename, int sample_rate, int num_channels);
+ WavWriter(const std::string& filename, int sample_rate, size_t num_channels);
// Close the WAV file, after writing its header.
~WavWriter();
@@ -51,13 +51,13 @@
void WriteSamples(const int16_t* samples, size_t num_samples);
int sample_rate() const override { return sample_rate_; }
- int num_channels() const override { return num_channels_; }
+ size_t num_channels() const override { return num_channels_; }
size_t num_samples() const override { return num_samples_; }
private:
void Close();
const int sample_rate_;
- const int num_channels_;
+ const size_t num_channels_;
size_t num_samples_; // Total number of samples written to file.
FILE* file_handle_; // Output file, owned by this class
@@ -79,13 +79,13 @@
size_t ReadSamples(size_t num_samples, int16_t* samples);
int sample_rate() const override { return sample_rate_; }
- int num_channels() const override { return num_channels_; }
+ size_t num_channels() const override { return num_channels_; }
size_t num_samples() const override { return num_samples_; }
private:
void Close();
int sample_rate_;
- int num_channels_;
+ size_t num_channels_;
size_t num_samples_; // Total number of samples in the file.
size_t num_samples_remaining_;
FILE* file_handle_; // Input file, owned by this class.
@@ -102,13 +102,13 @@
typedef struct rtc_WavWriter rtc_WavWriter;
rtc_WavWriter* rtc_WavOpen(const char* filename,
int sample_rate,
- int num_channels);
+ size_t num_channels);
void rtc_WavClose(rtc_WavWriter* wf);
void rtc_WavWriteSamples(rtc_WavWriter* wf,
const float* samples,
size_t num_samples);
int rtc_WavSampleRate(const rtc_WavWriter* wf);
-int rtc_WavNumChannels(const rtc_WavWriter* wf);
+size_t rtc_WavNumChannels(const rtc_WavWriter* wf);
size_t rtc_WavNumSamples(const rtc_WavWriter* wf);
#ifdef __cplusplus
diff --git a/webrtc/common_audio/wav_file_unittest.cc b/webrtc/common_audio/wav_file_unittest.cc
index 3732079..ba1db1c 100644
--- a/webrtc/common_audio/wav_file_unittest.cc
+++ b/webrtc/common_audio/wav_file_unittest.cc
@@ -30,7 +30,7 @@
{
WavWriter w(outfile, 14099, 1);
EXPECT_EQ(14099, w.sample_rate());
- EXPECT_EQ(1, w.num_channels());
+ EXPECT_EQ(1u, w.num_channels());
EXPECT_EQ(0u, w.num_samples());
w.WriteSamples(kSamples, kNumSamples);
EXPECT_EQ(kNumSamples, w.num_samples());
@@ -78,7 +78,7 @@
{
WavReader r(outfile);
EXPECT_EQ(14099, r.sample_rate());
- EXPECT_EQ(1, r.num_channels());
+ EXPECT_EQ(1u, r.num_channels());
EXPECT_EQ(kNumSamples, r.num_samples());
static const float kTruncatedSamples[] = {0.0, 10.0, 32767.0};
float samples[kNumSamples];
@@ -93,7 +93,7 @@
const std::string outfile = test::OutputPath() + "wavtest2.wav";
rtc_WavWriter* w = rtc_WavOpen(outfile.c_str(), 11904, 2);
EXPECT_EQ(11904, rtc_WavSampleRate(w));
- EXPECT_EQ(2, rtc_WavNumChannels(w));
+ EXPECT_EQ(2u, rtc_WavNumChannels(w));
EXPECT_EQ(0u, rtc_WavNumSamples(w));
static const size_t kNumSamples = 4;
rtc_WavWriteSamples(w, &kSamples[0], 2);
@@ -136,7 +136,7 @@
TEST(WavWriterTest, LargeFile) {
std::string outfile = test::OutputPath() + "wavtest3.wav";
static const int kSampleRate = 8000;
- static const int kNumChannels = 2;
+ static const size_t kNumChannels = 2;
static const size_t kNumSamples = 3 * kSampleRate * kNumChannels;
float samples[kNumSamples];
for (size_t i = 0; i < kNumSamples; i += kNumChannels) {
diff --git a/webrtc/common_audio/wav_header.cc b/webrtc/common_audio/wav_header.cc
index d2aa426..402ea17 100644
--- a/webrtc/common_audio/wav_header.cc
+++ b/webrtc/common_audio/wav_header.cc
@@ -59,7 +59,7 @@
} // namespace
-bool CheckWavParameters(int num_channels,
+bool CheckWavParameters(size_t num_channels,
int sample_rate,
WavFormat format,
size_t bytes_per_sample,
@@ -67,12 +67,11 @@
// num_channels, sample_rate, and bytes_per_sample must be positive, must fit
// in their respective fields, and their product must fit in the 32-bit
// ByteRate field.
- if (num_channels <= 0 || sample_rate <= 0 || bytes_per_sample == 0)
+ if (num_channels == 0 || sample_rate <= 0 || bytes_per_sample == 0)
return false;
if (static_cast<uint64_t>(sample_rate) > std::numeric_limits<uint32_t>::max())
return false;
- if (static_cast<uint64_t>(num_channels) >
- std::numeric_limits<uint16_t>::max())
+ if (num_channels > std::numeric_limits<uint16_t>::max())
return false;
if (static_cast<uint64_t>(bytes_per_sample) * 8 >
std::numeric_limits<uint16_t>::max())
@@ -136,17 +135,18 @@
bytes_in_payload + kWavHeaderSize - sizeof(ChunkHeader));
}
-static inline uint32_t ByteRate(int num_channels, int sample_rate,
+static inline uint32_t ByteRate(size_t num_channels, int sample_rate,
size_t bytes_per_sample) {
return static_cast<uint32_t>(num_channels * sample_rate * bytes_per_sample);
}
-static inline uint16_t BlockAlign(int num_channels, size_t bytes_per_sample) {
+static inline uint16_t BlockAlign(size_t num_channels,
+ size_t bytes_per_sample) {
return static_cast<uint16_t>(num_channels * bytes_per_sample);
}
void WriteWavHeader(uint8_t* buf,
- int num_channels,
+ size_t num_channels,
int sample_rate,
WavFormat format,
size_t bytes_per_sample,
@@ -181,7 +181,7 @@
}
bool ReadWavHeader(ReadableWav* readable,
- int* num_channels,
+ size_t* num_channels,
int* sample_rate,
WavFormat* format,
size_t* bytes_per_sample,
diff --git a/webrtc/common_audio/wav_header.h b/webrtc/common_audio/wav_header.h
index 65b7792..6844306 100644
--- a/webrtc/common_audio/wav_header.h
+++ b/webrtc/common_audio/wav_header.h
@@ -32,7 +32,7 @@
};
// Return true if the given parameters will make a well-formed WAV header.
-bool CheckWavParameters(int num_channels,
+bool CheckWavParameters(size_t num_channels,
int sample_rate,
WavFormat format,
size_t bytes_per_sample,
@@ -43,7 +43,7 @@
// channels and contain the specified total number of samples of the specified
// type. CHECKs the input parameters for validity.
void WriteWavHeader(uint8_t* buf,
- int num_channels,
+ size_t num_channels,
int sample_rate,
WavFormat format,
size_t bytes_per_sample,
@@ -53,7 +53,7 @@
// the provided output parameters. ReadableWav is used because the header can
// be variably sized. Returns false if the header is invalid.
bool ReadWavHeader(ReadableWav* readable,
- int* num_channels,
+ size_t* num_channels,
int* sample_rate,
WavFormat* format,
size_t* bytes_per_sample,
diff --git a/webrtc/common_audio/wav_header_unittest.cc b/webrtc/common_audio/wav_header_unittest.cc
index 226d788..8527939 100644
--- a/webrtc/common_audio/wav_header_unittest.cc
+++ b/webrtc/common_audio/wav_header_unittest.cc
@@ -91,7 +91,7 @@
}
TEST(WavHeaderTest, ReadWavHeaderWithErrors) {
- int num_channels = 0;
+ size_t num_channels = 0;
int sample_rate = 0;
WavFormat format = kWavFormatPcm;
size_t bytes_per_sample = 0;
@@ -268,7 +268,7 @@
static_assert(sizeof(kExpectedBuf) == kSize, "buffer size");
EXPECT_EQ(0, memcmp(kExpectedBuf, buf, kSize));
- int num_channels = 0;
+ size_t num_channels = 0;
int sample_rate = 0;
WavFormat format = kWavFormatPcm;
size_t bytes_per_sample = 0;
@@ -277,7 +277,7 @@
EXPECT_TRUE(
ReadWavHeader(&r, &num_channels, &sample_rate, &format,
&bytes_per_sample, &num_samples));
- EXPECT_EQ(17, num_channels);
+ EXPECT_EQ(17u, num_channels);
EXPECT_EQ(12345, sample_rate);
EXPECT_EQ(kWavFormatALaw, format);
EXPECT_EQ(1u, bytes_per_sample);
@@ -304,7 +304,7 @@
0x99, 0xd0, 0x5b, 0x07, // size of payload: 123457689
};
- int num_channels = 0;
+ size_t num_channels = 0;
int sample_rate = 0;
WavFormat format = kWavFormatPcm;
size_t bytes_per_sample = 0;
@@ -313,7 +313,7 @@
EXPECT_TRUE(
ReadWavHeader(&r, &num_channels, &sample_rate, &format,
&bytes_per_sample, &num_samples));
- EXPECT_EQ(17, num_channels);
+ EXPECT_EQ(17u, num_channels);
EXPECT_EQ(12345, sample_rate);
EXPECT_EQ(kWavFormatALaw, format);
EXPECT_EQ(1u, bytes_per_sample);
diff --git a/webrtc/common_types.h b/webrtc/common_types.h
index 6d1886b..444ef92 100644
--- a/webrtc/common_types.h
+++ b/webrtc/common_types.h
@@ -291,7 +291,7 @@
char plname[RTP_PAYLOAD_NAME_SIZE];
int plfreq;
int pacsize;
- int channels;
+ size_t channels;
int rate; // bits/sec unlike {start,min,max}Bitrate elsewhere in this file!
bool operator==(const CodecInst& other) const {
diff --git a/webrtc/modules/audio_coding/acm2/acm_codec_database.cc b/webrtc/modules/audio_coding/acm2/acm_codec_database.cc
index bbd4509..5f3c078 100644
--- a/webrtc/modules/audio_coding/acm2/acm_codec_database.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_codec_database.cc
@@ -292,7 +292,9 @@
codec_inst.channels));
}
-int ACMCodecDB::CodecId(const char* payload_name, int frequency, int channels) {
+int ACMCodecDB::CodecId(const char* payload_name,
+ int frequency,
+ size_t channels) {
for (const CodecInst& ci : RentACodec::Database()) {
bool name_match = false;
bool frequency_match = false;
diff --git a/webrtc/modules/audio_coding/acm2/acm_codec_database.h b/webrtc/modules/audio_coding/acm2/acm_codec_database.h
index 9e87238..6c2db9c 100644
--- a/webrtc/modules/audio_coding/acm2/acm_codec_database.h
+++ b/webrtc/modules/audio_coding/acm2/acm_codec_database.h
@@ -48,7 +48,7 @@
int num_packet_sizes;
int packet_sizes_samples[kMaxNumPacketSize];
int basic_block_samples;
- int channel_support;
+ size_t channel_support;
};
// Returns codec id from database, given the information received in the input
@@ -60,7 +60,7 @@
// codec id if successful, otherwise < 0.
static int CodecNumber(const CodecInst& codec_inst);
static int CodecId(const CodecInst& codec_inst);
- static int CodecId(const char* payload_name, int frequency, int channels);
+ static int CodecId(const char* payload_name, int frequency, size_t channels);
static int ReceiverCodecNumber(const CodecInst& codec_inst);
// Databases with information about the supported codecs
diff --git a/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc b/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
index 8ca77ec..855a39e 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
@@ -55,7 +55,7 @@
// G.722 = 94
bool RemapPltypeAndUseThisCodec(const char* plname,
int plfreq,
- int channels,
+ size_t channels,
int* pltype) {
if (channels != 1)
return false; // Don't use non-mono codecs.
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
index 335c2d6..f45d5d3 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
@@ -213,7 +213,7 @@
int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
enum NetEqOutputType type;
size_t samples_per_channel;
- int num_channels;
+ size_t num_channels;
// Accessing members, take the lock.
CriticalSectionScoped lock(crit_sect_.get());
@@ -301,7 +301,7 @@
int32_t AcmReceiver::AddCodec(int acm_codec_id,
uint8_t payload_type,
- int channels,
+ size_t channels,
int sample_rate_hz,
AudioDecoder* audio_decoder,
const std::string& name) {
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.h b/webrtc/modules/audio_coding/acm2/acm_receiver.h
index 86fd927..b150612 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.h
@@ -44,7 +44,7 @@
uint8_t payload_type;
// This field is meaningful for codecs where both mono and
// stereo versions are registered under the same ID.
- int channels;
+ size_t channels;
int sample_rate_hz;
};
@@ -116,7 +116,7 @@
//
int AddCodec(int acm_codec_id,
uint8_t payload_type,
- int channels,
+ size_t channels,
int sample_rate_hz,
AudioDecoder* audio_decoder,
const std::string& name);
diff --git a/webrtc/modules/audio_coding/acm2/acm_resampler.cc b/webrtc/modules/audio_coding/acm2/acm_resampler.cc
index d7ceb8a..dfc3ef7 100644
--- a/webrtc/modules/audio_coding/acm2/acm_resampler.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_resampler.cc
@@ -28,10 +28,10 @@
int ACMResampler::Resample10Msec(const int16_t* in_audio,
int in_freq_hz,
int out_freq_hz,
- int num_audio_channels,
+ size_t num_audio_channels,
size_t out_capacity_samples,
int16_t* out_audio) {
- size_t in_length = static_cast<size_t>(in_freq_hz * num_audio_channels / 100);
+ size_t in_length = in_freq_hz * num_audio_channels / 100;
if (in_freq_hz == out_freq_hz) {
if (out_capacity_samples < in_length) {
assert(false);
@@ -56,7 +56,7 @@
return -1;
}
- return out_length / num_audio_channels;
+ return static_cast<int>(out_length / num_audio_channels);
}
} // namespace acm2
diff --git a/webrtc/modules/audio_coding/acm2/acm_resampler.h b/webrtc/modules/audio_coding/acm2/acm_resampler.h
index 700fefa..268db8b 100644
--- a/webrtc/modules/audio_coding/acm2/acm_resampler.h
+++ b/webrtc/modules/audio_coding/acm2/acm_resampler.h
@@ -25,7 +25,7 @@
int Resample10Msec(const int16_t* in_audio,
int in_freq_hz,
int out_freq_hz,
- int num_audio_channels,
+ size_t num_audio_channels,
size_t out_capacity_samples,
int16_t* out_audio);
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
index 034de32..c4dd349 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
@@ -56,7 +56,7 @@
int AudioCodingModule::Codec(const char* payload_name,
CodecInst* codec,
int sampling_freq_hz,
- int channels) {
+ size_t channels) {
rtc::Optional<CodecInst> ci = acm2::RentACodec::CodecInstByParams(
payload_name, sampling_freq_hz, channels);
if (ci) {
@@ -76,7 +76,7 @@
int AudioCodingModule::Codec(const char* payload_name,
int sampling_freq_hz,
- int channels) {
+ size_t channels) {
rtc::Optional<acm2::RentACodec::CodecId> ci =
acm2::RentACodec::CodecIdByParams(payload_name, sampling_freq_hz,
channels);
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
index b434da2..ac302f0 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
@@ -324,7 +324,7 @@
}
// Check whether we need an up-mix or down-mix?
- const int current_num_channels =
+ const size_t current_num_channels =
rent_a_codec_.GetEncoderStack()->NumChannels();
const bool same_num_channels =
ptr_frame->num_channels_ == current_num_channels;
@@ -589,7 +589,7 @@
int AudioCodingModuleImpl::RegisterReceiveCodec(const CodecInst& codec) {
CriticalSectionScoped lock(acm_crit_sect_.get());
RTC_DCHECK(receiver_initialized_);
- if (codec.channels > 2 || codec.channels < 0) {
+ if (codec.channels > 2) {
LOG_F(LS_ERROR) << "Unsupported number of channels: " << codec.channels;
return -1;
}
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
index 337ff13..926671f 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
@@ -189,7 +189,7 @@
uint32_t input_timestamp;
const int16_t* audio;
size_t length_per_channel;
- uint8_t audio_channel;
+ size_t audio_channel;
// If a re-mix is required (up or down), this buffer will store a re-mixed
// version of the input.
int16_t buffer[WEBRTC_10MS_PCM_AUDIO];
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
index ef48a48..6f82a96 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
@@ -300,7 +300,7 @@
EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame));
EXPECT_EQ(id_, audio_frame.id_);
EXPECT_EQ(0u, audio_frame.timestamp_);
- EXPECT_GT(audio_frame.num_channels_, 0);
+ EXPECT_GT(audio_frame.num_channels_, 0u);
EXPECT_EQ(static_cast<size_t>(kSampleRateHz / 100),
audio_frame.samples_per_channel_);
EXPECT_EQ(kSampleRateHz, audio_frame.sample_rate_hz_);
diff --git a/webrtc/modules/audio_coding/acm2/codec_manager.cc b/webrtc/modules/audio_coding/acm2/codec_manager.cc
index d8ef2bf..ad67377 100644
--- a/webrtc/modules/audio_coding/acm2/codec_manager.cc
+++ b/webrtc/modules/audio_coding/acm2/codec_manager.cc
@@ -11,6 +11,7 @@
#include "webrtc/modules/audio_coding/acm2/codec_manager.h"
#include "webrtc/base/checks.h"
+#include "webrtc/base/format_macros.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
#include "webrtc/system_wrappers/include/trace.h"
@@ -25,8 +26,8 @@
int dummy_id = 0;
if ((send_codec.channels != 1) && (send_codec.channels != 2)) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
- "Wrong number of channels (%d, only mono and stereo are "
- "supported)",
+ "Wrong number of channels (%" PRIuS ", only mono and stereo "
+ "are supported)",
send_codec.channels);
return -1;
}
@@ -48,7 +49,7 @@
if (!RentACodec::IsSupportedNumChannels(*maybe_codec_id, send_codec.channels)
.value_or(false)) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
- "%d number of channels not supportedn for %s.",
+ "%" PRIuS " number of channels not supportedn for %s.",
send_codec.channels, send_codec.plname);
return -1;
}
diff --git a/webrtc/modules/audio_coding/acm2/rent_a_codec.cc b/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
index 14302e4..5695fd6 100644
--- a/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
+++ b/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
@@ -45,7 +45,7 @@
rtc::Optional<RentACodec::CodecId> RentACodec::CodecIdByParams(
const char* payload_name,
int sampling_freq_hz,
- int channels) {
+ size_t channels) {
return CodecIdFromIndex(
ACMCodecDB::CodecId(payload_name, sampling_freq_hz, channels));
}
@@ -63,7 +63,7 @@
rtc::Optional<CodecInst> RentACodec::CodecInstByParams(const char* payload_name,
int sampling_freq_hz,
- int channels) {
+ size_t channels) {
rtc::Optional<CodecId> codec_id =
CodecIdByParams(payload_name, sampling_freq_hz, channels);
if (!codec_id)
@@ -83,7 +83,7 @@
}
rtc::Optional<bool> RentACodec::IsSupportedNumChannels(CodecId codec_id,
- int num_channels) {
+ size_t num_channels) {
auto i = CodecIndexFromId(codec_id);
return i ? rtc::Optional<bool>(
ACMCodecDB::codec_settings_[*i].channel_support >=
@@ -98,7 +98,7 @@
rtc::Optional<NetEqDecoder> RentACodec::NetEqDecoderFromCodecId(
CodecId codec_id,
- int num_channels) {
+ size_t num_channels) {
rtc::Optional<int> i = CodecIndexFromId(codec_id);
if (!i)
return rtc::Optional<NetEqDecoder>();
diff --git a/webrtc/modules/audio_coding/acm2/rent_a_codec.h b/webrtc/modules/audio_coding/acm2/rent_a_codec.h
index cf6891a..b1dcc91 100644
--- a/webrtc/modules/audio_coding/acm2/rent_a_codec.h
+++ b/webrtc/modules/audio_coding/acm2/rent_a_codec.h
@@ -162,12 +162,12 @@
static rtc::Optional<CodecId> CodecIdByParams(const char* payload_name,
int sampling_freq_hz,
- int channels);
+ size_t channels);
static rtc::Optional<CodecInst> CodecInstById(CodecId codec_id);
static rtc::Optional<CodecId> CodecIdByInst(const CodecInst& codec_inst);
static rtc::Optional<CodecInst> CodecInstByParams(const char* payload_name,
int sampling_freq_hz,
- int channels);
+ size_t channels);
static bool IsCodecValid(const CodecInst& codec_inst);
static inline bool IsPayloadTypeValid(int payload_type) {
@@ -177,10 +177,11 @@
static rtc::ArrayView<const CodecInst> Database();
static rtc::Optional<bool> IsSupportedNumChannels(CodecId codec_id,
- int num_channels);
+ size_t num_channels);
- static rtc::Optional<NetEqDecoder> NetEqDecoderFromCodecId(CodecId codec_id,
- int num_channels);
+ static rtc::Optional<NetEqDecoder> NetEqDecoderFromCodecId(
+ CodecId codec_id,
+ size_t num_channels);
// Parse codec_inst and extract payload types. If the given CodecInst was for
// the wrong sort of codec, return kSkip; otherwise, if the rate was illegal,
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h
index ebb5d36..a46b0e8 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -61,7 +61,7 @@
// Returns the input sample rate in Hz and the number of input channels.
// These are constants set at instantiation time.
virtual int SampleRateHz() const = 0;
- virtual int NumChannels() const = 0;
+ virtual size_t NumChannels() const = 0;
// Returns the rate at which the RTP timestamps are updated. The default
// implementation returns SampleRateHz().
diff --git a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
index 3901c30..180166c 100644
--- a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
+++ b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
@@ -75,7 +75,7 @@
return speech_encoder_->SampleRateHz();
}
-int AudioEncoderCng::NumChannels() const {
+size_t AudioEncoderCng::NumChannels() const {
return 1;
}
diff --git a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
index d5d00ac..87383e2 100644
--- a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
+++ b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
@@ -32,7 +32,7 @@
struct Config {
bool IsOk() const;
- int num_channels = 1;
+ size_t num_channels = 1;
int payload_type = 13;
// Caller keeps ownership of the AudioEncoder object.
AudioEncoder* speech_encoder = nullptr;
@@ -51,7 +51,7 @@
size_t MaxEncodedBytes() const override;
int SampleRateHz() const override;
- int NumChannels() const override;
+ size_t NumChannels() const override;
int RtpTimestampRateHz() const override;
size_t Num10MsFramesInNextPacket() const override;
size_t Max10MsFramesInAPacket() const override;
diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
index 26c7838..ff61db8 100644
--- a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
+++ b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
@@ -20,15 +20,6 @@
namespace {
-int16_t NumSamplesPerFrame(int num_channels,
- int frame_size_ms,
- int sample_rate_hz) {
- int samples_per_frame = num_channels * frame_size_ms * sample_rate_hz / 1000;
- RTC_CHECK_LE(samples_per_frame, std::numeric_limits<int16_t>::max())
- << "Frame size too large.";
- return static_cast<int16_t>(samples_per_frame);
-}
-
template <typename T>
typename T::Config CreateConfig(const CodecInst& codec_inst) {
typename T::Config config;
@@ -50,9 +41,8 @@
payload_type_(config.payload_type),
num_10ms_frames_per_packet_(
static_cast<size_t>(config.frame_size_ms / 10)),
- full_frame_samples_(NumSamplesPerFrame(config.num_channels,
- config.frame_size_ms,
- sample_rate_hz_)),
+ full_frame_samples_(
+ config.num_channels * config.frame_size_ms * sample_rate_hz / 1000),
first_timestamp_in_buffer_(0) {
RTC_CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz";
RTC_CHECK_EQ(config.frame_size_ms % 10, 0)
@@ -70,7 +60,7 @@
return sample_rate_hz_;
}
-int AudioEncoderPcm::NumChannels() const {
+size_t AudioEncoderPcm::NumChannels() const {
return num_channels_;
}
diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
index 6891cbd..b839488 100644
--- a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
+++ b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
@@ -25,7 +25,7 @@
bool IsOk() const;
int frame_size_ms;
- int num_channels;
+ size_t num_channels;
int payload_type;
protected:
@@ -37,7 +37,7 @@
size_t MaxEncodedBytes() const override;
int SampleRateHz() const override;
- int NumChannels() const override;
+ size_t NumChannels() const override;
size_t Num10MsFramesInNextPacket() const override;
size_t Max10MsFramesInAPacket() const override;
int GetTargetBitrate() const override;
@@ -58,7 +58,7 @@
private:
const int sample_rate_hz_;
- const int num_channels_;
+ const size_t num_channels_;
const int payload_type_;
const size_t num_10ms_frames_per_packet_;
const size_t full_frame_samples_;
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 fa476e8..d7203b9 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 @@
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 @@
return kSampleRateHz;
}
-int AudioEncoderG722::NumChannels() const {
+size_t AudioEncoderG722::NumChannels() const {
return num_channels_;
}
@@ -88,7 +88,7 @@
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 @@
// 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 @@
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 @@
// 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 @@
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));
}
diff --git a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
index 5f1b762..07d767e 100644
--- a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
+++ b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
@@ -27,7 +27,7 @@
int payload_type = 9;
int frame_size_ms = 20;
- int num_channels = 1;
+ size_t num_channels = 1;
};
explicit AudioEncoderG722(const Config& config);
@@ -36,7 +36,7 @@
size_t MaxEncodedBytes() const override;
int SampleRateHz() const override;
- int NumChannels() const override;
+ size_t NumChannels() const override;
int RtpTimestampRateHz() const override;
size_t Num10MsFramesInNextPacket() const override;
size_t Max10MsFramesInAPacket() const override;
@@ -59,7 +59,7 @@
size_t SamplesPerChannel() const;
- const int num_channels_;
+ const size_t num_channels_;
const int payload_type_;
const size_t num_10ms_frames_per_packet_;
size_t num_10ms_frames_buffered_;
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
index 15c4149..ddd6dde 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
+++ b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
@@ -64,7 +64,7 @@
return kSampleRateHz;
}
-int AudioEncoderIlbc::NumChannels() const {
+size_t AudioEncoderIlbc::NumChannels() const {
return 1;
}
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
index 0800c0f..102a274 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
@@ -36,7 +36,7 @@
size_t MaxEncodedBytes() const override;
int SampleRateHz() const override;
- int NumChannels() const override;
+ size_t NumChannels() const override;
size_t Num10MsFramesInNextPacket() const override;
size_t Max10MsFramesInAPacket() const override;
int GetTargetBitrate() const override;
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
index 3226877..321dac3 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
@@ -56,7 +56,7 @@
size_t MaxEncodedBytes() const override;
int SampleRateHz() const override;
- int NumChannels() const override;
+ size_t NumChannels() const override;
size_t Num10MsFramesInNextPacket() const override;
size_t Max10MsFramesInAPacket() const override;
int GetTargetBitrate() const override;
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
index 4cfd782..d4438cc 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
@@ -88,7 +88,7 @@
}
template <typename T>
-int AudioEncoderIsacT<T>::NumChannels() const {
+size_t AudioEncoderIsacT<T>::NumChannels() const {
return 1;
}
diff --git a/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h b/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h
index 29cba8f..66adde4 100644
--- a/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h
@@ -24,7 +24,7 @@
MOCK_METHOD1(Mark, void(std::string desc));
MOCK_CONST_METHOD0(MaxEncodedBytes, size_t());
MOCK_CONST_METHOD0(SampleRateHz, int());
- MOCK_CONST_METHOD0(NumChannels, int());
+ MOCK_CONST_METHOD0(NumChannels, size_t());
MOCK_CONST_METHOD0(RtpTimestampRateHz, int());
MOCK_CONST_METHOD0(Num10MsFramesInNextPacket, size_t());
MOCK_CONST_METHOD0(Max10MsFramesInAPacket, size_t());
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
index df0e79b..f64e811 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
@@ -17,7 +17,7 @@
AudioDecoderOpus::AudioDecoderOpus(size_t num_channels)
: channels_(num_channels) {
RTC_DCHECK(num_channels == 1 || num_channels == 2);
- WebRtcOpus_DecoderCreate(&dec_state_, static_cast<int>(channels_));
+ WebRtcOpus_DecoderCreate(&dec_state_, channels_);
WebRtcOpus_DecoderInit(dec_state_);
}
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index 0806bb8..707d6c2 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -114,7 +114,7 @@
return kSampleRateHz;
}
-int AudioEncoderOpus::NumChannels() const {
+size_t AudioEncoderOpus::NumChannels() const {
return config_.num_channels;
}
@@ -147,8 +147,7 @@
Num10msFramesPerPacket() * SamplesPer10msFrame());
int status = WebRtcOpus_Encode(
inst_, &input_buffer_[0],
- rtc::CheckedDivExact(input_buffer_.size(),
- static_cast<size_t>(config_.num_channels)),
+ rtc::CheckedDivExact(input_buffer_.size(), config_.num_channels),
rtc::saturated_cast<int16_t>(max_encoded_bytes), encoded);
RTC_CHECK_GE(status, 0); // Fails only if fed invalid data.
input_buffer_.clear();
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
index f37e344..59c8f79 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
@@ -31,7 +31,7 @@
struct Config {
bool IsOk() const;
int frame_size_ms = 20;
- int num_channels = 1;
+ size_t num_channels = 1;
int payload_type = 120;
ApplicationMode application = kVoip;
int bitrate_bps = 64000;
@@ -56,7 +56,7 @@
size_t MaxEncodedBytes() const override;
int SampleRateHz() const override;
- int NumChannels() const override;
+ size_t NumChannels() const override;
size_t Num10MsFramesInNextPacket() const override;
size_t Max10MsFramesInAPacket() const override;
int GetTargetBitrate() const override;
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc b/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
index cb895a5..4f9f7ff 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
@@ -9,6 +9,7 @@
*/
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/format_macros.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
#include "webrtc/test/testsupport/fileutils.h"
@@ -21,7 +22,7 @@
namespace webrtc {
// Define coding parameter as <channels, bit_rate, filename, extension>.
-typedef tuple<int, int, string, string> coding_param;
+typedef tuple<size_t, int, string, string> coding_param;
typedef struct mode mode;
struct mode {
@@ -47,7 +48,7 @@
int sampling_khz_;
size_t block_length_sample_;
- int channels_;
+ size_t channels_;
int bit_rate_;
size_t data_pointer_;
@@ -68,7 +69,7 @@
void OpusFecTest::SetUp() {
channels_ = get<0>(GetParam());
bit_rate_ = get<1>(GetParam());
- printf("Coding %d channel signal at %d bps.\n", channels_, bit_rate_);
+ printf("Coding %" PRIuS " channel signal at %d bps.\n", channels_, bit_rate_);
in_filename_ = test::ResourcePath(get<2>(GetParam()), get<3>(GetParam()));
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_inst.h b/webrtc/modules/audio_coding/codecs/opus/opus_inst.h
index 662c6fa..8d032ba 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_inst.h
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_inst.h
@@ -17,7 +17,7 @@
struct WebRtcOpusEncInst {
OpusEncoder* encoder;
- int channels;
+ size_t channels;
int in_dtx_mode;
// When Opus is in DTX mode, we use |zero_counts| to count consecutive zeros
// to break long zero segment so as to prevent DTX from going wrong. We use
@@ -30,7 +30,7 @@
struct WebRtcOpusDecInst {
OpusDecoder* decoder;
int prev_decoded_samples;
- int channels;
+ size_t channels;
int in_dtx_mode;
};
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
index e8cdd64..9dc7ef9 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
@@ -42,7 +42,7 @@
};
int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst,
- int32_t channels,
+ size_t channels,
int32_t application) {
int opus_app;
if (!inst)
@@ -67,7 +67,7 @@
assert(state->zero_counts);
int error;
- state->encoder = opus_encoder_create(48000, channels, opus_app,
+ state->encoder = opus_encoder_create(48000, (int)channels, opus_app,
&error);
if (error != OPUS_OK || !state->encoder) {
WebRtcOpus_EncoderFree(state);
@@ -99,7 +99,7 @@
uint8_t* encoded) {
int res;
size_t i;
- int c;
+ size_t c;
int16_t buffer[2 * 48 * kWebRtcOpusMaxEncodeFrameSizeMs];
@@ -107,7 +107,7 @@
return -1;
}
- const int channels = inst->channels;
+ const size_t channels = inst->channels;
int use_buffer = 0;
// Break long consecutive zeros by forcing a "1" every |kZeroBreakCount|
@@ -248,7 +248,7 @@
}
}
-int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels) {
+int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, size_t channels) {
int error;
OpusDecInst* state;
@@ -260,7 +260,7 @@
}
/* Create new memory, always at 48000 Hz. */
- state->decoder = opus_decoder_create(48000, channels, &error);
+ state->decoder = opus_decoder_create(48000, (int)channels, &error);
if (error == OPUS_OK && state->decoder != NULL) {
/* Creation of memory all ok. */
state->channels = channels;
@@ -289,7 +289,7 @@
}
}
-int WebRtcOpus_DecoderChannels(OpusDecInst* inst) {
+size_t WebRtcOpus_DecoderChannels(OpusDecInst* inst) {
return inst->channels;
}
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_interface.h b/webrtc/modules/audio_coding/codecs/opus/opus_interface.h
index b73e1fb..754b49c 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_interface.h
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_interface.h
@@ -43,7 +43,7 @@
* -1 - Error
*/
int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst,
- int32_t channels,
+ size_t channels,
int32_t application);
int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst);
@@ -195,7 +195,7 @@
*/
int16_t WebRtcOpus_SetComplexity(OpusEncInst* inst, int32_t complexity);
-int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels);
+int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, size_t channels);
int16_t WebRtcOpus_DecoderFree(OpusDecInst* inst);
/****************************************************************************
@@ -203,7 +203,7 @@
*
* This function returns the number of channels created for Opus decoder.
*/
-int WebRtcOpus_DecoderChannels(OpusDecInst* inst);
+size_t WebRtcOpus_DecoderChannels(OpusDecInst* inst);
/****************************************************************************
* WebRtcOpus_DecoderInit(...)
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc b/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
index b29501e..c82b184 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
@@ -42,7 +42,9 @@
// After preparation, |speech_data_.GetNextBlock()| returns a pointer to a
// block of |block_length_ms| milliseconds. The data is looped every
// |loop_length_ms| milliseconds.
- void PrepareSpeechData(int channel, int block_length_ms, int loop_length_ms);
+ void PrepareSpeechData(size_t channel,
+ int block_length_ms,
+ int loop_length_ms);
int EncodeDecode(WebRtcOpusEncInst* encoder,
rtc::ArrayView<const int16_t> input_audio,
@@ -53,7 +55,7 @@
void SetMaxPlaybackRate(WebRtcOpusEncInst* encoder,
opus_int32 expect, int32_t set);
- void CheckAudioBounded(const int16_t* audio, size_t samples, int channels,
+ void CheckAudioBounded(const int16_t* audio, size_t samples, size_t channels,
uint16_t bound) const;
WebRtcOpusEncInst* opus_encoder_;
@@ -62,7 +64,7 @@
AudioLoop speech_data_;
uint8_t bitstream_[kMaxBytes];
size_t encoded_bytes_;
- int channels_;
+ size_t channels_;
int application_;
};
@@ -70,11 +72,11 @@
: opus_encoder_(NULL),
opus_decoder_(NULL),
encoded_bytes_(0),
- channels_(::testing::get<0>(GetParam())),
+ channels_(static_cast<size_t>(::testing::get<0>(GetParam()))),
application_(::testing::get<1>(GetParam())) {
}
-void OpusTest::PrepareSpeechData(int channel, int block_length_ms,
+void OpusTest::PrepareSpeechData(size_t channel, int block_length_ms,
int loop_length_ms) {
const std::string file_name =
webrtc::test::ResourcePath((channel == 1) ?
@@ -99,9 +101,9 @@
}
void OpusTest::CheckAudioBounded(const int16_t* audio, size_t samples,
- int channels, uint16_t bound) const {
+ size_t channels, uint16_t bound) const {
for (size_t i = 0; i < samples; ++i) {
- for (int c = 0; c < channels; ++c) {
+ for (size_t c = 0; c < channels; ++c) {
ASSERT_GE(audio[i * channels + c], -bound);
ASSERT_LE(audio[i * channels + c], bound);
}
@@ -115,7 +117,7 @@
int16_t* audio_type) {
int encoded_bytes_int = WebRtcOpus_Encode(
encoder, input_audio.data(),
- rtc::CheckedDivExact(input_audio.size(), static_cast<size_t>(channels_)),
+ rtc::CheckedDivExact(input_audio.size(), channels_),
kMaxBytes, bitstream_);
EXPECT_GE(encoded_bytes_int, 0);
encoded_bytes_ = static_cast<size_t>(encoded_bytes_int);
@@ -588,8 +590,7 @@
auto speech_block = speech_data_.GetNextBlock();
int encoded_bytes_int = WebRtcOpus_Encode(
opus_encoder_, speech_block.data(),
- rtc::CheckedDivExact(speech_block.size(),
- 2 * static_cast<size_t>(channels_)),
+ rtc::CheckedDivExact(speech_block.size(), 2 * channels_),
kMaxBytes, bitstream_);
EXPECT_GE(encoded_bytes_int, 0);
EXPECT_EQ(kOpus10msFrameSamples,
@@ -601,7 +602,7 @@
speech_block = speech_data_.GetNextBlock();
encoded_bytes_int = WebRtcOpus_Encode(
opus_encoder_, speech_block.data(),
- rtc::CheckedDivExact(speech_block.size(), static_cast<size_t>(channels_)),
+ rtc::CheckedDivExact(speech_block.size(), channels_),
kMaxBytes, bitstream_);
EXPECT_GE(encoded_bytes_int, 0);
EXPECT_EQ(kOpus20msFrameSamples,
@@ -643,8 +644,7 @@
auto speech_block = speech_data_.GetNextBlock();
encoded_bytes_ =
WebRtcOpus_Encode(opus_encoder_, speech_block.data(),
- rtc::CheckedDivExact(speech_block.size(),
- static_cast<size_t>(channels_)),
+ rtc::CheckedDivExact(speech_block.size(), channels_),
kMaxBytes, bitstream_);
EXPECT_EQ(OPUS_OK, opus_repacketizer_cat(rp, bitstream_, encoded_bytes_));
}
diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
index 177c19a..7ef1ce0 100644
--- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
+++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
@@ -32,7 +32,7 @@
return speech_encoder_->SampleRateHz();
}
-int AudioEncoderCopyRed::NumChannels() const {
+size_t AudioEncoderCopyRed::NumChannels() const {
return speech_encoder_->NumChannels();
}
diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
index d7d3a66..2f53765 100644
--- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
+++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
@@ -38,7 +38,7 @@
size_t MaxEncodedBytes() const override;
int SampleRateHz() const override;
- int NumChannels() const override;
+ size_t NumChannels() const override;
int RtpTimestampRateHz() const override;
size_t Num10MsFramesInNextPacket() const override;
size_t Max10MsFramesInAPacket() const override;
diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
index c4c3910..22601b6 100644
--- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
@@ -42,7 +42,7 @@
config.speech_encoder = &mock_encoder_;
red_.reset(new AudioEncoderCopyRed(config));
memset(audio_, 0, sizeof(audio_));
- EXPECT_CALL(mock_encoder_, NumChannels()).WillRepeatedly(Return(1));
+ EXPECT_CALL(mock_encoder_, NumChannels()).WillRepeatedly(Return(1U));
EXPECT_CALL(mock_encoder_, SampleRateHz())
.WillRepeatedly(Return(sample_rate_hz_));
EXPECT_CALL(mock_encoder_, MaxEncodedBytes())
@@ -110,8 +110,8 @@
}
TEST_F(AudioEncoderCopyRedTest, CheckNumChannelsPropagation) {
- EXPECT_CALL(mock_encoder_, NumChannels()).WillOnce(Return(17));
- EXPECT_EQ(17, red_->NumChannels());
+ EXPECT_CALL(mock_encoder_, NumChannels()).WillOnce(Return(17U));
+ EXPECT_EQ(17U, red_->NumChannels());
}
TEST_F(AudioEncoderCopyRedTest, CheckFrameSizePropagation) {
diff --git a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc b/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
index 07a15ff..3dc6654 100644
--- a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
+++ b/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
@@ -11,6 +11,7 @@
#include "webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/format_macros.h"
#include "webrtc/test/testsupport/fileutils.h"
using ::std::tr1::get;
@@ -99,7 +100,7 @@
size_t time_now_ms = 0;
float time_ms;
- printf("Coding %d kHz-sampled %d-channel audio at %d bps ...\n",
+ printf("Coding %d kHz-sampled %" PRIuS "-channel audio at %d bps ...\n",
input_sampling_khz_, channels_, bit_rate_);
while (time_now_ms < audio_duration_sec * 1000) {
diff --git a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h b/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
index b5aef75..fb7b3e5 100644
--- a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
+++ b/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
@@ -20,7 +20,8 @@
// Define coding parameter as
// <channels, bit_rate, file_name, extension, if_save_output>.
-typedef std::tr1::tuple<int, int, std::string, std::string, bool> coding_param;
+typedef std::tr1::tuple<size_t, int, std::string, std::string, bool>
+ coding_param;
class AudioCodecSpeedTest : public testing::TestWithParam<coding_param> {
protected:
@@ -74,7 +75,7 @@
float decoding_time_ms_;
FILE* out_file_;
- int channels_;
+ size_t channels_;
// Bit rate is in bit-per-second.
int bit_rate_;
diff --git a/webrtc/modules/audio_coding/include/audio_coding_module.h b/webrtc/modules/audio_coding/include/audio_coding_module.h
index 52fe383..9e7991f 100644
--- a/webrtc/modules/audio_coding/include/audio_coding_module.h
+++ b/webrtc/modules/audio_coding/include/audio_coding_module.h
@@ -134,7 +134,7 @@
// 0 if succeeded.
//
static int Codec(const char* payload_name, CodecInst* codec,
- int sampling_freq_hz, int channels);
+ int sampling_freq_hz, size_t channels);
///////////////////////////////////////////////////////////////////////////
// int32_t Codec()
@@ -153,7 +153,7 @@
// -1 if the codec is not found.
//
static int Codec(const char* payload_name, int sampling_freq_hz,
- int channels);
+ size_t channels);
///////////////////////////////////////////////////////////////////////////
// bool IsCodecValid()
diff --git a/webrtc/modules/audio_coding/neteq/include/neteq.h b/webrtc/modules/audio_coding/neteq/include/neteq.h
index abe09a3..1322223 100644
--- a/webrtc/modules/audio_coding/neteq/include/neteq.h
+++ b/webrtc/modules/audio_coding/neteq/include/neteq.h
@@ -171,7 +171,7 @@
// The speech type is written to |type|, if |type| is not NULL.
// Returns kOK on success, or kFail in case of an error.
virtual int GetAudio(size_t max_length, int16_t* output_audio,
- size_t* samples_per_channel, int* num_channels,
+ size_t* samples_per_channel, size_t* num_channels,
NetEqOutputType* type) = 0;
// Associates |rtp_payload_type| with |codec| and |codec_name|, and stores the
diff --git a/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
index 7bf9499..c03fbb7 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
@@ -188,7 +188,7 @@
void GetAndVerifyOutput() override {
NetEqOutputType output_type;
size_t samples_per_channel;
- int num_channels;
+ size_t num_channels;
// Get audio from internal decoder instance.
EXPECT_EQ(NetEq::kOK,
neteq_internal_->GetAudio(kMaxBlockSize,
@@ -196,7 +196,7 @@
&samples_per_channel,
&num_channels,
&output_type));
- EXPECT_EQ(1, num_channels);
+ EXPECT_EQ(1u, num_channels);
EXPECT_EQ(static_cast<size_t>(kOutputLengthMs * sample_rate_hz_ / 1000),
samples_per_channel);
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index ac84ce9..6c07da4 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -151,7 +151,7 @@
}
int NetEqImpl::GetAudio(size_t max_length, int16_t* output_audio,
- size_t* samples_per_channel, int* num_channels,
+ size_t* samples_per_channel, size_t* num_channels,
NetEqOutputType* type) {
TRACE_EVENT0("webrtc", "NetEqImpl::GetAudio");
CriticalSectionScoped lock(crit_sect_.get());
@@ -744,7 +744,7 @@
int NetEqImpl::GetAudioInternal(size_t max_length,
int16_t* output,
size_t* samples_per_channel,
- int* num_channels) {
+ size_t* num_channels) {
PacketList packet_list;
DtmfEvent dtmf_event;
Operations operation;
@@ -868,7 +868,7 @@
const size_t samples_from_sync =
sync_buffer_->GetNextAudioInterleaved(num_output_samples_per_channel,
output);
- *num_channels = static_cast<int>(sync_buffer_->Channels());
+ *num_channels = sync_buffer_->Channels();
if (sync_buffer_->FutureLength() < expand_->overlap_length()) {
// The sync buffer should always contain |overlap_length| samples, but now
// too many samples have been extracted. Reinstall the |overlap_length|
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.h b/webrtc/modules/audio_coding/neteq/neteq_impl.h
index 093c8d5..940dead 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.h
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.h
@@ -107,7 +107,7 @@
int GetAudio(size_t max_length,
int16_t* output_audio,
size_t* samples_per_channel,
- int* num_channels,
+ size_t* num_channels,
NetEqOutputType* type) override;
int RegisterPayloadType(NetEqDecoder codec,
@@ -220,7 +220,8 @@
int GetAudioInternal(size_t max_length,
int16_t* output,
size_t* samples_per_channel,
- int* num_channels) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+ size_t* num_channels)
+ EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// Provides a decision to the GetAudioInternal method. The decision what to
// do is written to |operation|. Packets to decode are written to
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
index d3b304a..f734883 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -466,14 +466,14 @@
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
int16_t output[kMaxOutputSize];
size_t samples_per_channel;
- int num_channels;
+ size_t num_channels;
NetEqOutputType type;
EXPECT_EQ(
NetEq::kOK,
neteq_->GetAudio(
kMaxOutputSize, output, &samples_per_channel, &num_channels, &type));
ASSERT_EQ(kMaxOutputSize, samples_per_channel);
- EXPECT_EQ(1, num_channels);
+ EXPECT_EQ(1u, num_channels);
EXPECT_EQ(kOutputNormal, type);
// Start with a simple check that the fake decoder is behaving as expected.
@@ -545,14 +545,14 @@
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
int16_t output[kMaxOutputSize];
size_t samples_per_channel;
- int num_channels;
+ size_t num_channels;
NetEqOutputType type;
EXPECT_EQ(
NetEq::kOK,
neteq_->GetAudio(
kMaxOutputSize, output, &samples_per_channel, &num_channels, &type));
ASSERT_EQ(kMaxOutputSize, samples_per_channel);
- EXPECT_EQ(1, num_channels);
+ EXPECT_EQ(1u, num_channels);
EXPECT_EQ(kOutputNormal, type);
// Insert two more packets. The first one is out of order, and is already too
@@ -583,7 +583,7 @@
neteq_->GetAudio(
kMaxOutputSize, output, &samples_per_channel, &num_channels, &type));
ASSERT_EQ(kMaxOutputSize, samples_per_channel);
- EXPECT_EQ(1, num_channels);
+ EXPECT_EQ(1u, num_channels);
EXPECT_EQ(kOutputNormal, type);
// Now check the packet buffer, and make sure it is empty, since the
@@ -622,14 +622,14 @@
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
int16_t output[kMaxOutputSize];
size_t samples_per_channel;
- int num_channels;
+ size_t num_channels;
NetEqOutputType type;
EXPECT_EQ(NetEq::kOK,
neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
&num_channels, &type));
ASSERT_LE(samples_per_channel, kMaxOutputSize);
EXPECT_EQ(kMaxOutputSize, samples_per_channel);
- EXPECT_EQ(1, num_channels);
+ EXPECT_EQ(1u, num_channels);
EXPECT_EQ(kOutputPLC, type);
// Register the payload type.
@@ -652,7 +652,7 @@
&num_channels, &type));
ASSERT_LE(samples_per_channel, kMaxOutputSize);
EXPECT_EQ(kMaxOutputSize, samples_per_channel);
- EXPECT_EQ(1, num_channels);
+ EXPECT_EQ(1u, num_channels);
EXPECT_EQ(kOutputNormal, type)
<< "NetEq did not decode the packets as expected.";
}
@@ -734,7 +734,7 @@
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateKhz);
int16_t output[kMaxOutputSize];
size_t samples_per_channel;
- int num_channels;
+ size_t num_channels;
uint32_t timestamp;
uint32_t last_timestamp;
NetEqOutputType type;
@@ -759,7 +759,7 @@
for (size_t i = 1; i < 6; ++i) {
ASSERT_EQ(kMaxOutputSize, samples_per_channel);
- EXPECT_EQ(1, num_channels);
+ EXPECT_EQ(1u, num_channels);
EXPECT_EQ(expected_type[i - 1], type);
EXPECT_TRUE(neteq_->GetPlayoutTimestamp(×tamp));
EXPECT_EQ(NetEq::kOK,
@@ -779,7 +779,7 @@
for (size_t i = 6; i < 8; ++i) {
ASSERT_EQ(kMaxOutputSize, samples_per_channel);
- EXPECT_EQ(1, num_channels);
+ EXPECT_EQ(1u, num_channels);
EXPECT_EQ(expected_type[i - 1], type);
EXPECT_EQ(NetEq::kOK,
neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
@@ -799,7 +799,7 @@
UseNoMocks();
CreateInstance();
static const size_t kNetEqMaxFrameSize = 2880; // 60 ms @ 48 kHz.
- static const int kChannels = 2;
+ static const size_t kChannels = 2;
const uint8_t kPayloadType = 17; // Just an arbitrary number.
const uint32_t kReceiveTime = 17; // Value doesn't matter for this test.
@@ -871,11 +871,10 @@
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
- const size_t kMaxOutputSize =
- static_cast<size_t>(10 * kSampleRateHz / 1000 * kChannels);
+ const size_t kMaxOutputSize = 10 * kSampleRateHz / 1000 * kChannels;
int16_t output[kMaxOutputSize];
size_t samples_per_channel;
- int num_channels;
+ size_t num_channels;
NetEqOutputType type;
EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(kMaxOutputSize, output,
@@ -981,13 +980,13 @@
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
int16_t output[kMaxOutputSize];
size_t samples_per_channel;
- int num_channels;
+ size_t num_channels;
NetEqOutputType type;
EXPECT_EQ(NetEq::kOK,
neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
&num_channels, &type));
ASSERT_EQ(kMaxOutputSize, samples_per_channel);
- EXPECT_EQ(1, num_channels);
+ EXPECT_EQ(1u, num_channels);
EXPECT_EQ(kOutputNormal, type);
EXPECT_CALL(mock_decoder, Die());
@@ -1078,13 +1077,13 @@
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
int16_t output[kMaxOutputSize];
size_t samples_per_channel;
- int num_channels;
+ size_t num_channels;
NetEqOutputType type;
EXPECT_EQ(NetEq::kOK,
neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
&num_channels, &type));
EXPECT_EQ(kMaxOutputSize, samples_per_channel);
- EXPECT_EQ(1, num_channels);
+ EXPECT_EQ(1u, num_channels);
EXPECT_EQ(kOutputNormal, type);
// Pull audio again. Decoder fails.
@@ -1094,7 +1093,7 @@
EXPECT_EQ(NetEq::kDecoderErrorCode, neteq_->LastError());
EXPECT_EQ(kDecoderErrorCode, neteq_->LastDecoderError());
EXPECT_EQ(kMaxOutputSize, samples_per_channel);
- EXPECT_EQ(1, num_channels);
+ EXPECT_EQ(1u, num_channels);
// TODO(minyue): should NetEq better give kOutputPLC, since it is actually an
// expansion.
EXPECT_EQ(kOutputNormal, type);
@@ -1104,7 +1103,7 @@
neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
&num_channels, &type));
EXPECT_EQ(kMaxOutputSize, samples_per_channel);
- EXPECT_EQ(1, num_channels);
+ EXPECT_EQ(1u, num_channels);
EXPECT_EQ(kOutputPLC, type);
// Pull audio again, should behave normal.
@@ -1112,7 +1111,7 @@
neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
&num_channels, &type));
EXPECT_EQ(kMaxOutputSize, samples_per_channel);
- EXPECT_EQ(1, num_channels);
+ EXPECT_EQ(1u, num_channels);
EXPECT_EQ(kOutputNormal, type);
EXPECT_CALL(mock_decoder, Die());
@@ -1199,13 +1198,13 @@
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
int16_t output[kMaxOutputSize];
size_t samples_per_channel;
- int num_channels;
+ size_t num_channels;
NetEqOutputType type;
EXPECT_EQ(NetEq::kOK,
neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
&num_channels, &type));
EXPECT_EQ(kMaxOutputSize, samples_per_channel);
- EXPECT_EQ(1, num_channels);
+ EXPECT_EQ(1u, num_channels);
EXPECT_EQ(kOutputCNG, type);
// Pull audio again. Decoder fails.
@@ -1215,7 +1214,7 @@
EXPECT_EQ(NetEq::kDecoderErrorCode, neteq_->LastError());
EXPECT_EQ(kDecoderErrorCode, neteq_->LastDecoderError());
EXPECT_EQ(kMaxOutputSize, samples_per_channel);
- EXPECT_EQ(1, num_channels);
+ EXPECT_EQ(1u, num_channels);
// TODO(minyue): should NetEq better give kOutputPLC, since it is actually an
// expansion.
EXPECT_EQ(kOutputCNG, type);
@@ -1225,7 +1224,7 @@
neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
&num_channels, &type));
EXPECT_EQ(kMaxOutputSize, samples_per_channel);
- EXPECT_EQ(1, num_channels);
+ EXPECT_EQ(1u, num_channels);
EXPECT_EQ(kOutputCNG, type);
EXPECT_CALL(mock_decoder, Die());
diff --git a/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
index 1ddc7f2..d3f59ec 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
@@ -27,7 +27,7 @@
struct TestParameters {
int frame_size;
int sample_rate;
- int num_channels;
+ size_t num_channels;
};
// This is a parameterized test. The test parameters are supplied through a
@@ -163,7 +163,7 @@
void VerifyOutput(size_t num_samples) {
for (size_t i = 0; i < num_samples; ++i) {
- for (int j = 0; j < num_channels_; ++j) {
+ for (size_t j = 0; j < num_channels_; ++j) {
ASSERT_EQ(output_[i], output_multi_channel_[i * num_channels_ + j]) <<
"Diff in sample " << i << ", channel " << j << ".";
}
@@ -214,12 +214,12 @@
NetEqOutputType output_type;
// Get audio from mono instance.
size_t samples_per_channel;
- int num_channels;
+ size_t num_channels;
EXPECT_EQ(NetEq::kOK,
neteq_mono_->GetAudio(kMaxBlockSize, output_,
&samples_per_channel, &num_channels,
&output_type));
- EXPECT_EQ(1, num_channels);
+ EXPECT_EQ(1u, num_channels);
EXPECT_EQ(output_size_samples_, samples_per_channel);
// Get audio from multi-channel instance.
ASSERT_EQ(NetEq::kOK,
@@ -239,7 +239,7 @@
}
}
- const int num_channels_;
+ const size_t num_channels_;
const int sample_rate_hz_;
const int samples_per_ms_;
const int frame_size_ms_;
diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
index a20d712..8d52c61 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
@@ -425,7 +425,7 @@
// Get audio from NetEq.
NetEqOutputType type;
- int num_channels;
+ size_t num_channels;
ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, out_len,
&num_channels, &type));
ASSERT_TRUE((*out_len == kBlockSize8kHz) ||
@@ -608,7 +608,7 @@
// Pull out all data.
for (size_t i = 0; i < num_frames; ++i) {
size_t out_len;
- int num_channels;
+ size_t num_channels;
NetEqOutputType type;
ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
&num_channels, &type));
@@ -653,7 +653,7 @@
// Pull out data once.
size_t out_len;
- int num_channels;
+ size_t num_channels;
NetEqOutputType type;
ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
&num_channels, &type));
@@ -684,7 +684,7 @@
// Pull out data once.
size_t out_len;
- int num_channels;
+ size_t num_channels;
NetEqOutputType type;
ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
&num_channels, &type));
@@ -709,7 +709,7 @@
double next_input_time_ms = 0.0;
double t_ms;
size_t out_len;
- int num_channels;
+ size_t num_channels;
NetEqOutputType type;
// Insert speech for 5 seconds.
@@ -948,7 +948,7 @@
for (size_t i = 0; i < kMaxBlockSize; ++i) {
out_data_[i] = 1;
}
- int num_channels;
+ size_t num_channels;
size_t samples_per_channel;
EXPECT_EQ(NetEq::kFail,
neteq_->GetAudio(kMaxBlockSize, out_data_,
@@ -982,7 +982,7 @@
for (size_t i = 0; i < kMaxBlockSize; ++i) {
out_data_[i] = 1;
}
- int num_channels;
+ size_t num_channels;
size_t samples_per_channel;
EXPECT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_,
&samples_per_channel,
@@ -1038,7 +1038,7 @@
PopulateRtpInfo(0, 0, &rtp_info);
rtp_info.header.payloadType = payload_type;
- int number_channels = 0;
+ size_t number_channels = 0;
size_t samples_per_channel = 0;
uint32_t receive_timestamp = 0;
@@ -1060,7 +1060,7 @@
&samples_per_channel,
&number_channels,
&type));
- ASSERT_EQ(1, number_channels);
+ ASSERT_EQ(1u, number_channels);
ASSERT_EQ(expected_samples_per_channel, samples_per_channel);
ASSERT_EQ(kOutputNormal, type);
@@ -1082,7 +1082,7 @@
&samples_per_channel,
&number_channels,
&type));
- ASSERT_EQ(1, number_channels);
+ ASSERT_EQ(1u, number_channels);
ASSERT_EQ(expected_samples_per_channel, samples_per_channel);
// To be able to test the fading of background noise we need at lease to
@@ -1103,7 +1103,7 @@
&samples_per_channel,
&number_channels,
&type));
- ASSERT_EQ(1, number_channels);
+ ASSERT_EQ(1u, number_channels);
ASSERT_EQ(expected_samples_per_channel, samples_per_channel);
if (type == kOutputPLCtoCNG) {
plc_to_cng = true;
@@ -1272,7 +1272,7 @@
// Insert some packets which decode to noise. We are not interested in
// actual decoded values.
NetEqOutputType output_type;
- int num_channels;
+ size_t num_channels;
size_t samples_per_channel;
uint32_t receive_timestamp = 0;
for (int n = 0; n < 100; ++n) {
@@ -1281,7 +1281,7 @@
&samples_per_channel, &num_channels,
&output_type));
ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
- ASSERT_EQ(1, num_channels);
+ ASSERT_EQ(1u, num_channels);
rtp_info.header.sequenceNumber++;
rtp_info.header.timestamp += kBlockSize16kHz;
@@ -1299,7 +1299,7 @@
&samples_per_channel, &num_channels,
&output_type));
ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
- ASSERT_EQ(1, num_channels);
+ ASSERT_EQ(1u, num_channels);
if (n > algorithmic_frame_delay) {
EXPECT_TRUE(IsAllZero(decoded, samples_per_channel * num_channels));
}
@@ -1348,7 +1348,7 @@
// Insert some packets which decode to noise. We are not interested in
// actual decoded values.
NetEqOutputType output_type;
- int num_channels;
+ size_t num_channels;
size_t samples_per_channel;
uint32_t receive_timestamp = 0;
int algorithmic_frame_delay = algorithmic_delay_ms_ / 10 + 1;
@@ -1358,7 +1358,7 @@
&samples_per_channel, &num_channels,
&output_type));
ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
- ASSERT_EQ(1, num_channels);
+ ASSERT_EQ(1u, num_channels);
rtp_info.header.sequenceNumber++;
rtp_info.header.timestamp += kBlockSize16kHz;
receive_timestamp += kBlockSize16kHz;
@@ -1397,7 +1397,7 @@
&samples_per_channel, &num_channels,
&output_type));
ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
- ASSERT_EQ(1, num_channels);
+ ASSERT_EQ(1u, num_channels);
EXPECT_TRUE(IsAllNonZero(decoded, samples_per_channel * num_channels));
}
}
@@ -1415,7 +1415,7 @@
const size_t kPayloadBytes = kSamples * sizeof(int16_t);
double next_input_time_ms = 0.0;
int16_t decoded[kBlockSize16kHz];
- int num_channels;
+ size_t num_channels;
size_t samples_per_channel;
NetEqOutputType output_type;
uint32_t receive_timestamp = 0;
@@ -1468,7 +1468,7 @@
&samples_per_channel, &num_channels,
&output_type));
ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
- ASSERT_EQ(1, num_channels);
+ ASSERT_EQ(1u, num_channels);
// Expect delay (in samples) to be less than 2 packets.
EXPECT_LE(timestamp - PlayoutTimestamp(),
@@ -1519,7 +1519,7 @@
// Insert three speech packets. Three are needed to get the frame length
// correct.
size_t out_len;
- int num_channels;
+ size_t num_channels;
NetEqOutputType type;
uint8_t payload[kPayloadBytes] = {0};
WebRtcRTPHeader rtp_info;
@@ -1622,7 +1622,7 @@
// Pull audio once and make sure CNG is played.
size_t out_len;
- int num_channels;
+ size_t num_channels;
NetEqOutputType type;
ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
&num_channels, &type));
diff --git a/webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc b/webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
index 05d559d..0c09e92 100644
--- a/webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
+++ b/webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
@@ -50,7 +50,7 @@
NetEqDecoder::kDecoderILBC) {}
void SetUp() override {
- ASSERT_EQ(1, channels_) << "iLBC supports only mono audio.";
+ ASSERT_EQ(1u, channels_) << "iLBC supports only mono audio.";
AudioEncoderIlbc::Config config;
config.frame_size_ms = FLAGS_frame_size_ms;
encoder_.reset(new AudioEncoderIlbc(config));
diff --git a/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc b/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
index 66b0903..4ccebb3 100644
--- a/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
+++ b/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
@@ -59,7 +59,7 @@
bit_rate_kbps_(FLAGS_bit_rate_kbps) {}
void NetEqIsacQualityTest::SetUp() {
- ASSERT_EQ(1, channels_) << "iSAC supports only mono audio.";
+ ASSERT_EQ(1u, channels_) << "iSAC supports only mono audio.";
// Create encoder memory.
WebRtcIsacfix_Create(&isac_encoder_);
ASSERT_TRUE(isac_encoder_ != NULL);
diff --git a/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc b/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
index 6e1d86c..ac478ab 100644
--- a/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
+++ b/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
@@ -50,7 +50,7 @@
NetEqDecoder::kDecoderPCMu) {}
void SetUp() override {
- ASSERT_EQ(1, channels_) << "PCMu supports only mono audio.";
+ ASSERT_EQ(1u, channels_) << "PCMu supports only mono audio.";
AudioEncoderPcmU::Config config;
config.frame_size_ms = FLAGS_frame_size_ms;
encoder_.reset(new AudioEncoderPcmU(config));
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
index 6becb86..694b9ed 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
@@ -12,6 +12,7 @@
#include "webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/format_macros.h"
namespace webrtc {
namespace test {
@@ -21,11 +22,11 @@
: codec_(codec),
decoder_(decoder),
sample_rate_hz_(CodecSampleRateHz(codec_)),
- channels_(static_cast<int>(decoder_->Channels())) {
+ channels_(decoder_->Channels()) {
NetEq::Config config;
config.sample_rate_hz = sample_rate_hz_;
neteq_.reset(NetEq::Create(config));
- printf("%d\n", channels_);
+ printf("%" PRIuS "\n", channels_);
}
void NetEqExternalDecoderTest::Init() {
@@ -47,7 +48,7 @@
NetEqOutputType* output_type) {
// Get audio from regular instance.
size_t samples_per_channel;
- int num_channels;
+ size_t num_channels;
EXPECT_EQ(NetEq::kOK,
neteq_->GetAudio(max_length,
output,
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h b/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
index 383b43a..d7b01fe 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
@@ -54,7 +54,7 @@
std::string name_ = "dummy name";
AudioDecoder* decoder_;
int sample_rate_hz_;
- int channels_;
+ size_t channels_;
rtc::scoped_ptr<NetEq> neteq_;
};
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
index a979934..7d1f9f9 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
@@ -109,7 +109,7 @@
static const size_t kOutDataLen =
kOutputBlockSizeMs * kMaxSamplesPerMs * kMaxChannels;
int16_t out_data[kOutDataLen];
- int num_channels;
+ size_t num_channels;
size_t samples_per_channel;
int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel,
&num_channels, NULL);
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
index f9b0db3..9c64e0f 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
@@ -210,7 +210,7 @@
int out_sampling_khz,
NetEqDecoder decoder_type)
: decoder_type_(decoder_type),
- channels_(FLAGS_channels),
+ channels_(static_cast<size_t>(FLAGS_channels)),
decoded_time_ms_(0),
decodable_time_ms_(0),
drift_factor_(FLAGS_drift_factor),
@@ -394,7 +394,7 @@
}
int NetEqQualityTest::DecodeBlock() {
- int channels;
+ size_t channels;
size_t samples;
int ret = neteq_->GetAudio(out_size_samples_ * channels_, &out_data_[0],
&samples, &channels, NULL);
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h b/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
index e20be57..c2b2eff 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
@@ -99,7 +99,7 @@
std::ofstream& Log();
NetEqDecoder decoder_type_;
- const int channels_;
+ const size_t channels_;
private:
int decoded_time_ms_;
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
index f1748d5..3d79e5b 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -609,7 +609,7 @@
static const size_t kOutDataLen =
kOutputBlockSizeMs * kMaxSamplesPerMs * kMaxChannels;
int16_t out_data[kOutDataLen];
- int num_channels;
+ size_t num_channels;
size_t samples_per_channel;
int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel,
&num_channels, NULL);
diff --git a/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc b/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
index e0222af..ba3c8d9 100644
--- a/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -52,7 +52,7 @@
}
void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
- std::string in_file_name, int sample_rate, int channels) {
+ std::string in_file_name, int sample_rate, size_t channels) {
struct CodecInst sendCodec;
int noOfCodecs = acm->NumberOfCodecs();
int codecNo;
@@ -123,7 +123,7 @@
}
void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
- std::string out_file_name, int channels) {
+ std::string out_file_name, size_t channels) {
struct CodecInst recvCodec = CodecInst();
int noOfCodecs;
EXPECT_EQ(0, acm->InitializeReceiver());
diff --git a/webrtc/modules/audio_coding/test/EncodeDecodeTest.h b/webrtc/modules/audio_coding/test/EncodeDecodeTest.h
index 3881062..f9a9a5b 100644
--- a/webrtc/modules/audio_coding/test/EncodeDecodeTest.h
+++ b/webrtc/modules/audio_coding/test/EncodeDecodeTest.h
@@ -48,7 +48,7 @@
public:
Sender();
void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
- std::string in_file_name, int sample_rate, int channels);
+ std::string in_file_name, int sample_rate, size_t channels);
void Teardown();
void Run();
bool Add10MsData();
@@ -71,7 +71,7 @@
Receiver();
virtual ~Receiver() {};
void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
- std::string out_file_name, int channels);
+ std::string out_file_name, size_t channels);
void Teardown();
void Run();
virtual bool IncomingPacket();
diff --git a/webrtc/modules/audio_coding/test/opus_test.cc b/webrtc/modules/audio_coding/test/opus_test.cc
index 466db9f..104b5e5 100644
--- a/webrtc/modules/audio_coding/test/opus_test.cc
+++ b/webrtc/modules/audio_coding/test/opus_test.cc
@@ -62,7 +62,7 @@
return;
#else
uint16_t frequency_hz;
- int audio_channels;
+ size_t audio_channels;
int16_t test_cntr = 0;
// Open both mono and stereo test files in 32 kHz.
@@ -205,7 +205,7 @@
#endif
}
-void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
+void OpusTest::Run(TestPackStereo* channel, size_t channels, int bitrate,
size_t frame_length, int percent_loss) {
AudioFrame audio_frame;
int32_t out_freq_hz_b = out_file_.SamplingFrequency();
diff --git a/webrtc/modules/audio_coding/test/opus_test.h b/webrtc/modules/audio_coding/test/opus_test.h
index 88ef0ec..93c9ffb 100644
--- a/webrtc/modules/audio_coding/test/opus_test.h
+++ b/webrtc/modules/audio_coding/test/opus_test.h
@@ -32,7 +32,7 @@
private:
void Run(TestPackStereo* channel,
- int channels,
+ size_t channels,
int bitrate,
size_t frame_length,
int percent_loss = 0);
diff --git a/webrtc/modules/audio_coding/test/target_delay_unittest.cc b/webrtc/modules/audio_coding/test/target_delay_unittest.cc
index 97471bb..195e9d8 100644
--- a/webrtc/modules/audio_coding/test/target_delay_unittest.cc
+++ b/webrtc/modules/audio_coding/test/target_delay_unittest.cc
@@ -153,7 +153,7 @@
ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &frame));
// Had to use ASSERT_TRUE, ASSERT_EQ generated error.
ASSERT_TRUE(kSampleRateHz == frame.sample_rate_hz_);
- ASSERT_EQ(1, frame.num_channels_);
+ ASSERT_EQ(1u, frame.num_channels_);
ASSERT_TRUE(kSampleRateHz / 100 == frame.samples_per_channel_);
}
}
diff --git a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
index 0ac9eae..afb060f 100644
--- a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
+++ b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
@@ -50,8 +50,8 @@
}
// Return the max number of channels from a |list| composed of AudioFrames.
-int MaxNumChannels(const AudioFrameList* list) {
- int max_num_channels = 1;
+size_t MaxNumChannels(const AudioFrameList* list) {
+ size_t max_num_channels = 1;
for (AudioFrameList::const_iterator iter = list->begin();
iter != list->end();
++iter) {
@@ -278,7 +278,7 @@
// with an API instead of dynamically.
// Find the max channels over all mixing lists.
- const int num_mixed_channels = std::max(MaxNumChannels(&mixList),
+ const size_t num_mixed_channels = std::max(MaxNumChannels(&mixList),
std::max(MaxNumChannels(&additionalFramesList),
MaxNumChannels(&rampOutList)));
diff --git a/webrtc/modules/audio_device/android/audio_device_unittest.cc b/webrtc/modules/audio_device/android/audio_device_unittest.cc
index 7b2d635..768047d 100644
--- a/webrtc/modules/audio_device/android/audio_device_unittest.cc
+++ b/webrtc/modules/audio_device/android/audio_device_unittest.cc
@@ -383,7 +383,7 @@
int32_t(const void* audioSamples,
const size_t nSamples,
const size_t nBytesPerSample,
- const uint8_t nChannels,
+ const size_t nChannels,
const uint32_t samplesPerSec,
const uint32_t totalDelayMS,
const int32_t clockDrift,
@@ -393,7 +393,7 @@
MOCK_METHOD8(NeedMorePlayData,
int32_t(const size_t nSamples,
const size_t nBytesPerSample,
- const uint8_t nChannels,
+ const size_t nChannels,
const uint32_t samplesPerSec,
void* audioSamples,
size_t& nSamplesOut,
@@ -423,7 +423,7 @@
int32_t RealRecordedDataIsAvailable(const void* audioSamples,
const size_t nSamples,
const size_t nBytesPerSample,
- const uint8_t nChannels,
+ const size_t nChannels,
const uint32_t samplesPerSec,
const uint32_t totalDelayMS,
const int32_t clockDrift,
@@ -445,7 +445,7 @@
int32_t RealNeedMorePlayData(const size_t nSamples,
const size_t nBytesPerSample,
- const uint8_t nChannels,
+ const size_t nChannels,
const uint32_t samplesPerSec,
void* audioSamples,
size_t& nSamplesOut,
@@ -521,10 +521,10 @@
int record_sample_rate() const {
return record_parameters_.sample_rate();
}
- int playout_channels() const {
+ size_t playout_channels() const {
return playout_parameters_.channels();
}
- int record_channels() const {
+ size_t record_channels() const {
return record_parameters_.channels();
}
size_t playout_frames_per_10ms_buffer() const {
@@ -931,7 +931,7 @@
// not contain any explicit verification that the audio quality is perfect.
TEST_F(AudioDeviceTest, RunPlayoutWithFileAsSource) {
// TODO(henrika): extend test when mono output is supported.
- EXPECT_EQ(1, playout_channels());
+ EXPECT_EQ(1u, playout_channels());
NiceMock<MockAudioTransport> mock(kPlayout);
const int num_callbacks = kFilePlayTimeInSec * kNumCallbacksPerSecond;
std::string file_name = GetFileName(playout_sample_rate());
diff --git a/webrtc/modules/audio_device/android/audio_manager.cc b/webrtc/modules/audio_device/android/audio_manager.cc
index 5cca52d..1d08a6a 100644
--- a/webrtc/modules/audio_device/android/audio_manager.cc
+++ b/webrtc/modules/audio_device/android/audio_manager.cc
@@ -214,9 +214,9 @@
hardware_ns_ = hardware_ns;
low_latency_playout_ = low_latency_output;
// TODO(henrika): add support for stereo output.
- playout_parameters_.reset(sample_rate, channels,
+ playout_parameters_.reset(sample_rate, static_cast<size_t>(channels),
static_cast<size_t>(output_buffer_size));
- record_parameters_.reset(sample_rate, channels,
+ record_parameters_.reset(sample_rate, static_cast<size_t>(channels),
static_cast<size_t>(input_buffer_size));
}
diff --git a/webrtc/modules/audio_device/android/audio_manager_unittest.cc b/webrtc/modules/audio_device/android/audio_manager_unittest.cc
index a5bc840..ddae730 100644
--- a/webrtc/modules/audio_device/android/audio_manager_unittest.cc
+++ b/webrtc/modules/audio_device/android/audio_manager_unittest.cc
@@ -82,14 +82,14 @@
PRINT("%saudio layer: %s\n", kTag,
low_latency_out ? "Low latency OpenSL" : "Java/JNI based AudioTrack");
PRINT("%ssample rate: %d Hz\n", kTag, playout_parameters_.sample_rate());
- PRINT("%schannels: %d\n", kTag, playout_parameters_.channels());
+ PRINT("%schannels: %" PRIuS "\n", kTag, playout_parameters_.channels());
PRINT("%sframes per buffer: %" PRIuS " <=> %.2f ms\n", kTag,
playout_parameters_.frames_per_buffer(),
playout_parameters_.GetBufferSizeInMilliseconds());
PRINT("RECORD: \n");
PRINT("%saudio layer: %s\n", kTag, "Java/JNI based AudioRecord");
PRINT("%ssample rate: %d Hz\n", kTag, record_parameters_.sample_rate());
- PRINT("%schannels: %d\n", kTag, record_parameters_.channels());
+ PRINT("%schannels: %" PRIuS "\n", kTag, record_parameters_.channels());
PRINT("%sframes per buffer: %" PRIuS " <=> %.2f ms\n", kTag,
record_parameters_.frames_per_buffer(),
record_parameters_.GetBufferSizeInMilliseconds());
@@ -119,7 +119,7 @@
AudioParameters params;
EXPECT_FALSE(params.is_valid());
EXPECT_EQ(0, params.sample_rate());
- EXPECT_EQ(0, params.channels());
+ EXPECT_EQ(0U, params.channels());
EXPECT_EQ(0U, params.frames_per_buffer());
EXPECT_EQ(0U, params.frames_per_10ms_buffer());
EXPECT_EQ(0U, params.GetBytesPerFrame());
@@ -131,7 +131,7 @@
// Basic test of the AudioParameters class using non default construction.
TEST_F(AudioManagerTest, AudioParametersWithNonDefaultConstruction) {
const int kSampleRate = 48000;
- const int kChannels = 1;
+ const size_t kChannels = 1;
const size_t kFramesPerBuffer = 480;
const size_t kFramesPer10msBuffer = 480;
const size_t kBytesPerFrame = 2;
diff --git a/webrtc/modules/audio_device/android/audio_record_jni.cc b/webrtc/modules/audio_device/android/audio_record_jni.cc
index 4a63197..5dda724 100644
--- a/webrtc/modules/audio_device/android/audio_record_jni.cc
+++ b/webrtc/modules/audio_device/android/audio_record_jni.cc
@@ -43,7 +43,7 @@
AudioRecordJni::JavaAudioRecord::~JavaAudioRecord() {}
int AudioRecordJni::JavaAudioRecord::InitRecording(
- int sample_rate, int channels) {
+ int sample_rate, size_t channels) {
return audio_record_->CallIntMethod(init_recording_,
static_cast<jint>(sample_rate),
static_cast<jint>(channels));
@@ -185,8 +185,8 @@
const int sample_rate_hz = audio_parameters_.sample_rate();
ALOGD("SetRecordingSampleRate(%d)", sample_rate_hz);
audio_device_buffer_->SetRecordingSampleRate(sample_rate_hz);
- const int channels = audio_parameters_.channels();
- ALOGD("SetRecordingChannels(%d)", channels);
+ const size_t channels = audio_parameters_.channels();
+ ALOGD("SetRecordingChannels(%" PRIuS ")", channels);
audio_device_buffer_->SetRecordingChannels(channels);
total_delay_in_milliseconds_ =
audio_manager_->GetDelayEstimateInMilliseconds();
diff --git a/webrtc/modules/audio_device/android/audio_record_jni.h b/webrtc/modules/audio_device/android/audio_record_jni.h
index a847180..766316a 100644
--- a/webrtc/modules/audio_device/android/audio_record_jni.h
+++ b/webrtc/modules/audio_device/android/audio_record_jni.h
@@ -49,7 +49,7 @@
rtc::scoped_ptr<GlobalRef> audio_track);
~JavaAudioRecord();
- int InitRecording(int sample_rate, int channels);
+ int InitRecording(int sample_rate, size_t channels);
bool StartRecording();
bool StopRecording();
bool EnableBuiltInAEC(bool enable);
diff --git a/webrtc/modules/audio_device/android/audio_track_jni.cc b/webrtc/modules/audio_device/android/audio_track_jni.cc
index c660868..057e016 100644
--- a/webrtc/modules/audio_device/android/audio_track_jni.cc
+++ b/webrtc/modules/audio_device/android/audio_track_jni.cc
@@ -202,8 +202,8 @@
const int sample_rate_hz = audio_parameters_.sample_rate();
ALOGD("SetPlayoutSampleRate(%d)", sample_rate_hz);
audio_device_buffer_->SetPlayoutSampleRate(sample_rate_hz);
- const int channels = audio_parameters_.channels();
- ALOGD("SetPlayoutChannels(%d)", channels);
+ const size_t channels = audio_parameters_.channels();
+ ALOGD("SetPlayoutChannels(%" PRIuS ")", channels);
audio_device_buffer_->SetPlayoutChannels(channels);
}
diff --git a/webrtc/modules/audio_device/android/opensles_player.cc b/webrtc/modules/audio_device/android/opensles_player.cc
index 9dc001c..d2bff49 100644
--- a/webrtc/modules/audio_device/android/opensles_player.cc
+++ b/webrtc/modules/audio_device/android/opensles_player.cc
@@ -179,15 +179,15 @@
const int sample_rate_hz = audio_parameters_.sample_rate();
ALOGD("SetPlayoutSampleRate(%d)", sample_rate_hz);
audio_device_buffer_->SetPlayoutSampleRate(sample_rate_hz);
- const int channels = audio_parameters_.channels();
- ALOGD("SetPlayoutChannels(%d)", channels);
+ const size_t channels = audio_parameters_.channels();
+ ALOGD("SetPlayoutChannels(%" PRIuS ")", channels);
audio_device_buffer_->SetPlayoutChannels(channels);
RTC_CHECK(audio_device_buffer_);
AllocateDataBuffers();
}
SLDataFormat_PCM OpenSLESPlayer::CreatePCMConfiguration(
- int channels,
+ size_t channels,
int sample_rate,
size_t bits_per_sample) {
ALOGD("CreatePCMConfiguration");
diff --git a/webrtc/modules/audio_device/android/opensles_player.h b/webrtc/modules/audio_device/android/opensles_player.h
index c9aa086..fa9e931 100644
--- a/webrtc/modules/audio_device/android/opensles_player.h
+++ b/webrtc/modules/audio_device/android/opensles_player.h
@@ -94,7 +94,7 @@
void EnqueuePlayoutData();
// Configures the SL_DATAFORMAT_PCM structure.
- SLDataFormat_PCM CreatePCMConfiguration(int channels,
+ SLDataFormat_PCM CreatePCMConfiguration(size_t channels,
int sample_rate,
size_t bits_per_sample);
diff --git a/webrtc/modules/audio_device/audio_device_buffer.cc b/webrtc/modules/audio_device/audio_device_buffer.cc
index e7b487d..48ae88e 100644
--- a/webrtc/modules/audio_device/audio_device_buffer.cc
+++ b/webrtc/modules/audio_device/audio_device_buffer.cc
@@ -169,7 +169,7 @@
// SetRecordingChannels
// ----------------------------------------------------------------------------
-int32_t AudioDeviceBuffer::SetRecordingChannels(uint8_t channels)
+int32_t AudioDeviceBuffer::SetRecordingChannels(size_t channels)
{
CriticalSectionScoped lock(&_critSect);
_recChannels = channels;
@@ -181,7 +181,7 @@
// SetPlayoutChannels
// ----------------------------------------------------------------------------
-int32_t AudioDeviceBuffer::SetPlayoutChannels(uint8_t channels)
+int32_t AudioDeviceBuffer::SetPlayoutChannels(size_t channels)
{
CriticalSectionScoped lock(&_critSect);
_playChannels = channels;
@@ -239,7 +239,7 @@
// RecordingChannels
// ----------------------------------------------------------------------------
-uint8_t AudioDeviceBuffer::RecordingChannels() const
+size_t AudioDeviceBuffer::RecordingChannels() const
{
return _recChannels;
}
@@ -248,7 +248,7 @@
// PlayoutChannels
// ----------------------------------------------------------------------------
-uint8_t AudioDeviceBuffer::PlayoutChannels() const
+size_t AudioDeviceBuffer::PlayoutChannels() const
{
return _playChannels;
}
@@ -487,7 +487,7 @@
{
uint32_t playSampleRate = 0;
size_t playBytesPerSample = 0;
- uint8_t playChannels = 0;
+ size_t playChannels = 0;
{
CriticalSectionScoped lock(&_critSect);
diff --git a/webrtc/modules/audio_device/audio_device_buffer.h b/webrtc/modules/audio_device/audio_device_buffer.h
index 2ab7ff5..1095971 100644
--- a/webrtc/modules/audio_device/audio_device_buffer.h
+++ b/webrtc/modules/audio_device/audio_device_buffer.h
@@ -40,10 +40,10 @@
int32_t RecordingSampleRate() const;
int32_t PlayoutSampleRate() const;
- virtual int32_t SetRecordingChannels(uint8_t channels);
- virtual int32_t SetPlayoutChannels(uint8_t channels);
- uint8_t RecordingChannels() const;
- uint8_t PlayoutChannels() const;
+ virtual int32_t SetRecordingChannels(size_t channels);
+ virtual int32_t SetPlayoutChannels(size_t channels);
+ size_t RecordingChannels() const;
+ size_t PlayoutChannels() const;
int32_t SetRecordingChannel(
const AudioDeviceModule::ChannelType channel);
int32_t RecordingChannel(
@@ -80,8 +80,8 @@
uint32_t _recSampleRate;
uint32_t _playSampleRate;
- uint8_t _recChannels;
- uint8_t _playChannels;
+ size_t _recChannels;
+ size_t _playChannels;
// selected recording channel (left/right/both)
AudioDeviceModule::ChannelType _recChannel;
diff --git a/webrtc/modules/audio_device/dummy/file_audio_device.cc b/webrtc/modules/audio_device/dummy/file_audio_device.cc
index ea61ce6..aac0962 100644
--- a/webrtc/modules/audio_device/dummy/file_audio_device.cc
+++ b/webrtc/modules/audio_device/dummy/file_audio_device.cc
@@ -14,9 +14,9 @@
namespace webrtc {
const int kRecordingFixedSampleRate = 48000;
-const int kRecordingNumChannels = 2;
+const size_t kRecordingNumChannels = 2;
const int kPlayoutFixedSampleRate = 48000;
-const int kPlayoutNumChannels = 2;
+const size_t kPlayoutNumChannels = 2;
const size_t kPlayoutBufferSize =
kPlayoutFixedSampleRate / 100 * kPlayoutNumChannels * 2;
const size_t kRecordingBufferSize =
diff --git a/webrtc/modules/audio_device/include/audio_device_defines.h b/webrtc/modules/audio_device/include/audio_device_defines.h
index 3ebbd23..b847729 100644
--- a/webrtc/modules/audio_device/include/audio_device_defines.h
+++ b/webrtc/modules/audio_device/include/audio_device_defines.h
@@ -49,7 +49,7 @@
virtual int32_t RecordedDataIsAvailable(const void* audioSamples,
const size_t nSamples,
const size_t nBytesPerSample,
- const uint8_t nChannels,
+ const size_t nChannels,
const uint32_t samplesPerSec,
const uint32_t totalDelayMS,
const int32_t clockDrift,
@@ -59,7 +59,7 @@
virtual int32_t NeedMorePlayData(const size_t nSamples,
const size_t nBytesPerSample,
- const uint8_t nChannels,
+ const size_t nChannels,
const uint32_t samplesPerSec,
void* audioSamples,
size_t& nSamplesOut,
@@ -82,10 +82,10 @@
// TODO(xians): Remove this interface after Chrome and Libjingle switches
// to OnData().
virtual int OnDataAvailable(const int voe_channels[],
- int number_of_voe_channels,
+ size_t number_of_voe_channels,
const int16_t* audio_data,
int sample_rate,
- int number_of_channels,
+ size_t number_of_channels,
size_t number_of_frames,
int audio_delay_milliseconds,
int current_volume,
@@ -103,7 +103,7 @@
const void* audio_data,
int bits_per_sample,
int sample_rate,
- int number_of_channels,
+ size_t number_of_channels,
size_t number_of_frames) {}
// Method to push the captured audio data to the specific VoE channel.
@@ -116,7 +116,7 @@
const void* audio_data,
int bits_per_sample,
int sample_rate,
- int number_of_channels,
+ size_t number_of_channels,
size_t number_of_frames) {}
// Method to pull mixed render audio data from all active VoE channels.
@@ -125,7 +125,7 @@
// channel.
virtual void PullRenderData(int bits_per_sample,
int sample_rate,
- int number_of_channels,
+ size_t number_of_channels,
size_t number_of_frames,
void* audio_data,
int64_t* elapsed_time_ms,
@@ -149,27 +149,27 @@
channels_(0),
frames_per_buffer_(0),
frames_per_10ms_buffer_(0) {}
- AudioParameters(int sample_rate, int channels, size_t frames_per_buffer)
+ AudioParameters(int sample_rate, size_t channels, size_t frames_per_buffer)
: sample_rate_(sample_rate),
channels_(channels),
frames_per_buffer_(frames_per_buffer),
frames_per_10ms_buffer_(static_cast<size_t>(sample_rate / 100)) {}
- void reset(int sample_rate, int channels, size_t frames_per_buffer) {
+ void reset(int sample_rate, size_t channels, size_t frames_per_buffer) {
sample_rate_ = sample_rate;
channels_ = channels;
frames_per_buffer_ = frames_per_buffer;
frames_per_10ms_buffer_ = static_cast<size_t>(sample_rate / 100);
}
size_t bits_per_sample() const { return kBitsPerSample; }
- void reset(int sample_rate, int channels, double ms_per_buffer) {
+ void reset(int sample_rate, size_t channels, double ms_per_buffer) {
reset(sample_rate, channels,
static_cast<size_t>(sample_rate * ms_per_buffer + 0.5));
}
- void reset(int sample_rate, int channels) {
+ void reset(int sample_rate, size_t channels) {
reset(sample_rate, channels, static_cast<size_t>(0));
}
int sample_rate() const { return sample_rate_; }
- int channels() const { return channels_; }
+ size_t channels() const { return channels_; }
size_t frames_per_buffer() const { return frames_per_buffer_; }
size_t frames_per_10ms_buffer() const { return frames_per_10ms_buffer_; }
size_t GetBytesPerFrame() const { return channels_ * kBitsPerSample / 8; }
@@ -200,7 +200,7 @@
private:
int sample_rate_;
- int channels_;
+ size_t channels_;
size_t frames_per_buffer_;
size_t frames_per_10ms_buffer_;
};
diff --git a/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc b/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc
index 1c9746f..076a674 100644
--- a/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc
+++ b/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc
@@ -373,7 +373,7 @@
int32_t(const void* audioSamples,
const size_t nSamples,
const size_t nBytesPerSample,
- const uint8_t nChannels,
+ const size_t nChannels,
const uint32_t samplesPerSec,
const uint32_t totalDelayMS,
const int32_t clockDrift,
@@ -383,7 +383,7 @@
MOCK_METHOD8(NeedMorePlayData,
int32_t(const size_t nSamples,
const size_t nBytesPerSample,
- const uint8_t nChannels,
+ const size_t nChannels,
const uint32_t samplesPerSec,
void* audioSamples,
size_t& nSamplesOut,
@@ -413,7 +413,7 @@
int32_t RealRecordedDataIsAvailable(const void* audioSamples,
const size_t nSamples,
const size_t nBytesPerSample,
- const uint8_t nChannels,
+ const size_t nChannels,
const uint32_t samplesPerSec,
const uint32_t totalDelayMS,
const int32_t clockDrift,
@@ -437,7 +437,7 @@
int32_t RealNeedMorePlayData(const size_t nSamples,
const size_t nBytesPerSample,
- const uint8_t nChannels,
+ const size_t nChannels,
const uint32_t samplesPerSec,
void* audioSamples,
size_t& nSamplesOut,
diff --git a/webrtc/modules/audio_device/test/audio_device_test_api.cc b/webrtc/modules/audio_device/test/audio_device_test_api.cc
index 2bbdfed..26a2dcd 100644
--- a/webrtc/modules/audio_device/test/audio_device_test_api.cc
+++ b/webrtc/modules/audio_device/test/audio_device_test_api.cc
@@ -85,7 +85,7 @@
int32_t RecordedDataIsAvailable(const void* audioSamples,
const size_t nSamples,
const size_t nBytesPerSample,
- const uint8_t nChannels,
+ const size_t nChannels,
const uint32_t sampleRate,
const uint32_t totalDelay,
const int32_t clockSkew,
@@ -110,7 +110,7 @@
int32_t NeedMorePlayData(const size_t nSamples,
const size_t nBytesPerSample,
- const uint8_t nChannels,
+ const size_t nChannels,
const uint32_t sampleRate,
void* audioSamples,
size_t& nSamplesOut,
diff --git a/webrtc/modules/audio_device/test/func_test_manager.cc b/webrtc/modules/audio_device/test/func_test_manager.cc
index b358e50..0a2963e 100644
--- a/webrtc/modules/audio_device/test/func_test_manager.cc
+++ b/webrtc/modules/audio_device/test/func_test_manager.cc
@@ -194,7 +194,7 @@
const void* audioSamples,
const size_t nSamples,
const size_t nBytesPerSample,
- const uint8_t nChannels,
+ const size_t nChannels,
const uint32_t samplesPerSec,
const uint32_t totalDelayMS,
const int32_t clockDrift,
@@ -339,7 +339,7 @@
int32_t AudioTransportImpl::NeedMorePlayData(
const size_t nSamples,
const size_t nBytesPerSample,
- const uint8_t nChannels,
+ const size_t nChannels,
const uint32_t samplesPerSec,
void* audioSamples,
size_t& nSamplesOut,
@@ -365,7 +365,7 @@
int16_t* ptr16Out = NULL;
const size_t nSamplesIn = packet->nSamples;
- const uint8_t nChannelsIn = packet->nChannels;
+ const size_t nChannelsIn = packet->nChannels;
const uint32_t samplesPerSecIn = packet->samplesPerSec;
const size_t nBytesPerSampleIn = packet->nBytesPerSample;
diff --git a/webrtc/modules/audio_device/test/func_test_manager.h b/webrtc/modules/audio_device/test/func_test_manager.h
index 1870709..b7cc81c 100644
--- a/webrtc/modules/audio_device/test/func_test_manager.h
+++ b/webrtc/modules/audio_device/test/func_test_manager.h
@@ -49,7 +49,7 @@
uint8_t dataBuffer[4 * 960];
size_t nSamples;
size_t nBytesPerSample;
- uint8_t nChannels;
+ size_t nChannels;
uint32_t samplesPerSec;
};
@@ -88,7 +88,7 @@
int32_t RecordedDataIsAvailable(const void* audioSamples,
const size_t nSamples,
const size_t nBytesPerSample,
- const uint8_t nChannels,
+ const size_t nChannels,
const uint32_t samplesPerSec,
const uint32_t totalDelayMS,
const int32_t clockDrift,
@@ -98,7 +98,7 @@
int32_t NeedMorePlayData(const size_t nSamples,
const size_t nBytesPerSample,
- const uint8_t nChannels,
+ const size_t nChannels,
const uint32_t samplesPerSec,
void* audioSamples,
size_t& nSamplesOut,
diff --git a/webrtc/modules/audio_processing/audio_buffer.cc b/webrtc/modules/audio_processing/audio_buffer.cc
index 77bda79..ff64267 100644
--- a/webrtc/modules/audio_processing/audio_buffer.cc
+++ b/webrtc/modules/audio_processing/audio_buffer.cc
@@ -44,9 +44,9 @@
} // namespace
AudioBuffer::AudioBuffer(size_t input_num_frames,
- int num_input_channels,
+ size_t num_input_channels,
size_t process_num_frames,
- int num_process_channels,
+ size_t num_process_channels,
size_t output_num_frames)
: input_num_frames_(input_num_frames),
num_input_channels_(num_input_channels),
@@ -74,7 +74,7 @@
num_proc_channels_));
if (input_num_frames_ != proc_num_frames_) {
- for (int i = 0; i < num_proc_channels_; ++i) {
+ for (size_t i = 0; i < num_proc_channels_; ++i) {
input_resamplers_.push_back(
new PushSincResampler(input_num_frames_,
proc_num_frames_));
@@ -82,7 +82,7 @@
}
if (output_num_frames_ != proc_num_frames_) {
- for (int i = 0; i < num_proc_channels_; ++i) {
+ for (size_t i = 0; i < num_proc_channels_; ++i) {
output_resamplers_.push_back(
new PushSincResampler(proc_num_frames_,
output_num_frames_));
@@ -130,7 +130,7 @@
// Resample.
if (input_num_frames_ != proc_num_frames_) {
- for (int i = 0; i < num_proc_channels_; ++i) {
+ for (size_t i = 0; i < num_proc_channels_; ++i) {
input_resamplers_[i]->Resample(data_ptr[i],
input_num_frames_,
process_buffer_->channels()[i],
@@ -140,7 +140,7 @@
}
// Convert to the S16 range.
- for (int i = 0; i < num_proc_channels_; ++i) {
+ for (size_t i = 0; i < num_proc_channels_; ++i) {
FloatToFloatS16(data_ptr[i],
proc_num_frames_,
data_->fbuf()->channels()[i]);
@@ -158,7 +158,7 @@
// Convert to an intermediate buffer for subsequent resampling.
data_ptr = process_buffer_->channels();
}
- for (int i = 0; i < num_channels_; ++i) {
+ for (size_t i = 0; i < num_channels_; ++i) {
FloatS16ToFloat(data_->fbuf()->channels()[i],
proc_num_frames_,
data_ptr[i]);
@@ -166,7 +166,7 @@
// Resample.
if (output_num_frames_ != proc_num_frames_) {
- for (int i = 0; i < num_channels_; ++i) {
+ for (size_t i = 0; i < num_channels_; ++i) {
output_resamplers_[i]->Resample(data_ptr[i],
proc_num_frames_,
data[i],
@@ -175,7 +175,7 @@
}
// Upmix.
- for (int i = num_channels_; i < stream_config.num_channels(); ++i) {
+ for (size_t i = num_channels_; i < stream_config.num_channels(); ++i) {
memcpy(data[i], data[0], output_num_frames_ * sizeof(**data));
}
}
@@ -197,13 +197,13 @@
return data_->ibuf()->channels();
}
-const int16_t* const* AudioBuffer::split_bands_const(int channel) const {
+const int16_t* const* AudioBuffer::split_bands_const(size_t channel) const {
return split_data_.get() ?
split_data_->ibuf_const()->bands(channel) :
data_->ibuf_const()->bands(channel);
}
-int16_t* const* AudioBuffer::split_bands(int channel) {
+int16_t* const* AudioBuffer::split_bands(size_t channel) {
mixed_low_pass_valid_ = false;
return split_data_.get() ?
split_data_->ibuf()->bands(channel) :
@@ -254,13 +254,13 @@
return data_->fbuf()->channels();
}
-const float* const* AudioBuffer::split_bands_const_f(int channel) const {
+const float* const* AudioBuffer::split_bands_const_f(size_t channel) const {
return split_data_.get() ?
split_data_->fbuf_const()->bands(channel) :
data_->fbuf_const()->bands(channel);
}
-float* const* AudioBuffer::split_bands_f(int channel) {
+float* const* AudioBuffer::split_bands_f(size_t channel) {
mixed_low_pass_valid_ = false;
return split_data_.get() ?
split_data_->fbuf()->bands(channel) :
@@ -341,11 +341,11 @@
return activity_;
}
-int AudioBuffer::num_channels() const {
+size_t AudioBuffer::num_channels() const {
return num_channels_;
}
-void AudioBuffer::set_num_channels(int num_channels) {
+void AudioBuffer::set_num_channels(size_t num_channels) {
num_channels_ = num_channels;
}
@@ -398,7 +398,7 @@
// Resample.
if (input_num_frames_ != proc_num_frames_) {
- for (int i = 0; i < num_proc_channels_; ++i) {
+ for (size_t i = 0; i < num_proc_channels_; ++i) {
input_resamplers_[i]->Resample(input_buffer_->fbuf_const()->channels()[i],
input_num_frames_,
data_->fbuf()->channels()[i],
@@ -423,7 +423,7 @@
output_buffer_.reset(
new IFChannelBuffer(output_num_frames_, num_channels_));
}
- for (int i = 0; i < num_channels_; ++i) {
+ for (size_t i = 0; i < num_channels_; ++i) {
output_resamplers_[i]->Resample(
data_->fbuf()->channels()[i], proc_num_frames_,
output_buffer_->fbuf()->channels()[i], output_num_frames_);
@@ -448,7 +448,7 @@
new ChannelBuffer<int16_t>(num_split_frames_,
num_proc_channels_));
}
- for (int i = 0; i < num_proc_channels_; i++) {
+ for (size_t i = 0; i < num_proc_channels_; i++) {
memcpy(low_pass_reference_channels_->channels()[i],
split_bands_const(i)[kBand0To8kHz],
low_pass_reference_channels_->num_frames_per_band() *
diff --git a/webrtc/modules/audio_processing/audio_buffer.h b/webrtc/modules/audio_processing/audio_buffer.h
index 48c9488..ff12ca2 100644
--- a/webrtc/modules/audio_processing/audio_buffer.h
+++ b/webrtc/modules/audio_processing/audio_buffer.h
@@ -34,14 +34,14 @@
public:
// TODO(ajm): Switch to take ChannelLayouts.
AudioBuffer(size_t input_num_frames,
- int num_input_channels,
+ size_t num_input_channels,
size_t process_num_frames,
- int num_process_channels,
+ size_t num_process_channels,
size_t output_num_frames);
virtual ~AudioBuffer();
- int num_channels() const;
- void set_num_channels(int num_channels);
+ size_t num_channels() const;
+ void set_num_channels(size_t num_channels);
size_t num_frames() const;
size_t num_frames_per_band() const;
size_t num_keyboard_frames() const;
@@ -65,10 +65,10 @@
// 0 <= channel < |num_proc_channels_|
// 0 <= band < |num_bands_|
// 0 <= sample < |num_split_frames_|
- int16_t* const* split_bands(int channel);
- const int16_t* const* split_bands_const(int channel) const;
- float* const* split_bands_f(int channel);
- const float* const* split_bands_const_f(int channel) const;
+ int16_t* const* split_bands(size_t channel);
+ const int16_t* const* split_bands_const(size_t channel) const;
+ float* const* split_bands_f(size_t channel);
+ const float* const* split_bands_const_f(size_t channel) const;
// Returns a pointer array to the channels for a specific band.
// Usage:
@@ -128,16 +128,16 @@
// The audio is passed into DeinterleaveFrom() or CopyFrom() with input
// format (samples per channel and number of channels).
const size_t input_num_frames_;
- const int num_input_channels_;
+ const size_t num_input_channels_;
// The audio is stored by DeinterleaveFrom() or CopyFrom() with processing
// format.
const size_t proc_num_frames_;
- const int num_proc_channels_;
+ const size_t num_proc_channels_;
// The audio is returned by InterleaveTo() and CopyTo() with output samples
// per channels and the current number of channels. This last one can be
// changed at any time using set_num_channels().
const size_t output_num_frames_;
- int num_channels_;
+ size_t num_channels_;
size_t num_bands_;
size_t num_split_frames_;
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index fea5785..744309c 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -410,16 +410,13 @@
int AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) {
for (const auto& stream : config.streams) {
- if (stream.num_channels() < 0) {
- return kBadNumberChannelsError;
- }
if (stream.num_channels() > 0 && stream.sample_rate_hz() <= 0) {
return kBadSampleRateError;
}
}
- const int num_in_channels = config.input_stream().num_channels();
- const int num_out_channels = config.output_stream().num_channels();
+ const size_t num_in_channels = config.input_stream().num_channels();
+ const size_t num_out_channels = config.output_stream().num_channels();
// Need at least one input channel.
// Need either one output channel or as many outputs as there are inputs.
@@ -429,7 +426,7 @@
}
if (capture_nonlocked_.beamformer_enabled &&
- static_cast<size_t>(num_in_channels) != capture_.array_geometry.size()) {
+ num_in_channels != capture_.array_geometry.size()) {
return kBadNumberChannelsError;
}
@@ -527,22 +524,22 @@
return capture_nonlocked_.split_rate;
}
-int AudioProcessingImpl::num_reverse_channels() const {
+size_t AudioProcessingImpl::num_reverse_channels() const {
// Used as callback from submodules, hence locking is not allowed.
return formats_.rev_proc_format.num_channels();
}
-int AudioProcessingImpl::num_input_channels() const {
+size_t AudioProcessingImpl::num_input_channels() const {
// Used as callback from submodules, hence locking is not allowed.
return formats_.api_format.input_stream().num_channels();
}
-int AudioProcessingImpl::num_proc_channels() const {
+size_t AudioProcessingImpl::num_proc_channels() const {
// Used as callback from submodules, hence locking is not allowed.
return capture_nonlocked_.beamformer_enabled ? 1 : num_output_channels();
}
-int AudioProcessingImpl::num_output_channels() const {
+size_t AudioProcessingImpl::num_output_channels() const {
// Used as callback from submodules, hence locking is not allowed.
return formats_.api_format.output_stream().num_channels();
}
@@ -631,7 +628,8 @@
audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
const size_t channel_size =
sizeof(float) * formats_.api_format.input_stream().num_frames();
- for (int i = 0; i < formats_.api_format.input_stream().num_channels(); ++i)
+ for (size_t i = 0; i < formats_.api_format.input_stream().num_channels();
+ ++i)
msg->add_input_channel(src[i], channel_size);
}
#endif
@@ -645,7 +643,8 @@
audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
const size_t channel_size =
sizeof(float) * formats_.api_format.output_stream().num_frames();
- for (int i = 0; i < formats_.api_format.output_stream().num_channels(); ++i)
+ for (size_t i = 0; i < formats_.api_format.output_stream().num_channels();
+ ++i)
msg->add_output_channel(dest[i], channel_size);
RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
&crit_debug_, &debug_dump_.capture));
@@ -879,7 +878,7 @@
return kNullPointerError;
}
- if (reverse_input_config.num_channels() <= 0) {
+ if (reverse_input_config.num_channels() == 0) {
return kBadNumberChannelsError;
}
@@ -898,7 +897,7 @@
debug_dump_.render.event_msg->mutable_reverse_stream();
const size_t channel_size =
sizeof(float) * formats_.api_format.reverse_input_stream().num_frames();
- for (int i = 0;
+ for (size_t i = 0;
i < formats_.api_format.reverse_input_stream().num_channels(); ++i)
msg->add_channel(src[i], channel_size);
RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
@@ -1455,12 +1454,12 @@
audioproc::Init* msg = debug_dump_.capture.event_msg->mutable_init();
msg->set_sample_rate(formats_.api_format.input_stream().sample_rate_hz());
- msg->set_num_input_channels(
- formats_.api_format.input_stream().num_channels());
- msg->set_num_output_channels(
- formats_.api_format.output_stream().num_channels());
- msg->set_num_reverse_channels(
- formats_.api_format.reverse_input_stream().num_channels());
+ msg->set_num_input_channels(static_cast<google::protobuf::int32>(
+ formats_.api_format.input_stream().num_channels()));
+ msg->set_num_output_channels(static_cast<google::protobuf::int32>(
+ formats_.api_format.output_stream().num_channels()));
+ msg->set_num_reverse_channels(static_cast<google::protobuf::int32>(
+ formats_.api_format.reverse_input_stream().num_channels()));
msg->set_reverse_sample_rate(
formats_.api_format.reverse_input_stream().sample_rate_hz());
msg->set_output_sample_rate(
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h
index 6cb9e8c..b310896 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.h
+++ b/webrtc/modules/audio_processing/audio_processing_impl.h
@@ -101,10 +101,10 @@
// Hence there is no need for locks in these.
int proc_sample_rate_hz() const override;
int proc_split_sample_rate_hz() const override;
- int num_input_channels() const override;
- int num_proc_channels() const override;
- int num_output_channels() const override;
- int num_reverse_channels() const override;
+ size_t num_input_channels() const override;
+ size_t num_proc_channels() const override;
+ size_t num_output_channels() const override;
+ size_t num_reverse_channels() const override;
int stream_delay_ms() const override;
bool was_stream_delay_set() const override
EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
diff --git a/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc b/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
index 7e96774..e1e6a31 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
@@ -477,7 +477,7 @@
RandomGenerator* rand_gen) {
ASSERT_GT(amplitude, 0);
ASSERT_LE(amplitude, 32767);
- for (int ch = 0; ch < frame->num_channels_; ch++) {
+ for (size_t ch = 0; ch < frame->num_channels_; ch++) {
for (size_t k = 0; k < frame->samples_per_channel_; k++) {
// Store random 16 bit number between -(amplitude+1) and
// amplitude.
diff --git a/webrtc/modules/audio_processing/beamformer/complex_matrix.h b/webrtc/modules/audio_processing/beamformer/complex_matrix.h
index bfa3563..707c515 100644
--- a/webrtc/modules/audio_processing/beamformer/complex_matrix.h
+++ b/webrtc/modules/audio_processing/beamformer/complex_matrix.h
@@ -27,10 +27,10 @@
public:
ComplexMatrix() : Matrix<complex<T> >() {}
- ComplexMatrix(int num_rows, int num_columns)
+ ComplexMatrix(size_t num_rows, size_t num_columns)
: Matrix<complex<T> >(num_rows, num_columns) {}
- ComplexMatrix(const complex<T>* data, int num_rows, int num_columns)
+ ComplexMatrix(const complex<T>* data, size_t num_rows, size_t num_columns)
: Matrix<complex<T> >(data, num_rows, num_columns) {}
// Complex Matrix operations.
@@ -51,7 +51,7 @@
ComplexMatrix& ConjugateTranspose() {
this->CopyDataToScratch();
- int num_rows = this->num_rows();
+ size_t num_rows = this->num_rows();
this->SetNumRows(this->num_columns());
this->SetNumColumns(num_rows);
this->Resize();
@@ -82,8 +82,8 @@
private:
ComplexMatrix& ConjugateTranspose(const complex<T>* const* src) {
complex<T>* const* elements = this->elements();
- for (int i = 0; i < this->num_rows(); ++i) {
- for (int j = 0; j < this->num_columns(); ++j) {
+ for (size_t i = 0; i < this->num_rows(); ++i) {
+ for (size_t j = 0; j < this->num_columns(); ++j) {
elements[i][j] = conj(src[j][i]);
}
}
diff --git a/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.cc b/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.cc
index d072832..78f4df5 100644
--- a/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.cc
+++ b/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.cc
@@ -27,7 +27,7 @@
// Calculates the Euclidean norm for a row vector.
float Norm(const ComplexMatrix<float>& x) {
- RTC_CHECK_EQ(1, x.num_rows());
+ RTC_CHECK_EQ(1u, x.num_rows());
const size_t length = x.num_columns();
const complex<float>* elems = x.elements()[0];
float result = 0.f;
@@ -43,8 +43,8 @@
float wave_number,
const std::vector<Point>& geometry,
ComplexMatrix<float>* mat) {
- RTC_CHECK_EQ(static_cast<int>(geometry.size()), mat->num_rows());
- RTC_CHECK_EQ(static_cast<int>(geometry.size()), mat->num_columns());
+ RTC_CHECK_EQ(geometry.size(), mat->num_rows());
+ RTC_CHECK_EQ(geometry.size(), mat->num_columns());
complex<float>* const* mat_els = mat->elements();
for (size_t i = 0; i < geometry.size(); ++i) {
@@ -68,8 +68,8 @@
int sample_rate,
const std::vector<Point>& geometry,
ComplexMatrix<float>* mat) {
- RTC_CHECK_EQ(static_cast<int>(geometry.size()), mat->num_rows());
- RTC_CHECK_EQ(static_cast<int>(geometry.size()), mat->num_columns());
+ RTC_CHECK_EQ(geometry.size(), mat->num_rows());
+ RTC_CHECK_EQ(geometry.size(), mat->num_columns());
ComplexMatrix<float> interf_cov_vector(1, geometry.size());
ComplexMatrix<float> interf_cov_vector_transposed(geometry.size(), 1);
@@ -94,8 +94,8 @@
const std::vector<Point>& geometry,
float angle,
ComplexMatrix<float>* mat) {
- RTC_CHECK_EQ(1, mat->num_rows());
- RTC_CHECK_EQ(static_cast<int>(geometry.size()), mat->num_columns());
+ RTC_CHECK_EQ(1u, mat->num_rows());
+ RTC_CHECK_EQ(geometry.size(), mat->num_columns());
float freq_in_hertz =
(static_cast<float>(frequency_bin) / fft_size) * sample_rate;
diff --git a/webrtc/modules/audio_processing/beamformer/matrix.h b/webrtc/modules/audio_processing/beamformer/matrix.h
index 162aef1..51c1cece9 100644
--- a/webrtc/modules/audio_processing/beamformer/matrix.h
+++ b/webrtc/modules/audio_processing/beamformer/matrix.h
@@ -67,7 +67,7 @@
Matrix() : num_rows_(0), num_columns_(0) {}
// Allocates space for the elements and initializes all values to zero.
- Matrix(int num_rows, int num_columns)
+ Matrix(size_t num_rows, size_t num_columns)
: num_rows_(num_rows), num_columns_(num_columns) {
Resize();
scratch_data_.resize(num_rows_ * num_columns_);
@@ -75,7 +75,7 @@
}
// Copies |data| into the new Matrix.
- Matrix(const T* data, int num_rows, int num_columns)
+ Matrix(const T* data, size_t num_rows, size_t num_columns)
: num_rows_(0), num_columns_(0) {
CopyFrom(data, num_rows, num_columns);
scratch_data_.resize(num_rows_ * num_columns_);
@@ -90,23 +90,23 @@
}
// Copy |data| into the Matrix. The current data is lost.
- void CopyFrom(const T* const data, int num_rows, int num_columns) {
+ void CopyFrom(const T* const data, size_t num_rows, size_t num_columns) {
Resize(num_rows, num_columns);
memcpy(&data_[0], data, num_rows_ * num_columns_ * sizeof(data_[0]));
}
Matrix& CopyFromColumn(const T* const* src,
size_t column_index,
- int num_rows) {
+ size_t num_rows) {
Resize(1, num_rows);
- for (int i = 0; i < num_columns_; ++i) {
+ for (size_t i = 0; i < num_columns_; ++i) {
data_[i] = src[i][column_index];
}
return *this;
}
- void Resize(int num_rows, int num_columns) {
+ void Resize(size_t num_rows, size_t num_columns) {
if (num_rows != num_rows_ || num_columns != num_columns_) {
num_rows_ = num_rows;
num_columns_ = num_columns;
@@ -115,8 +115,8 @@
}
// Accessors and mutators.
- int num_rows() const { return num_rows_; }
- int num_columns() const { return num_columns_; }
+ size_t num_rows() const { return num_rows_; }
+ size_t num_columns() const { return num_columns_; }
T* const* elements() { return &elements_[0]; }
const T* const* elements() const { return &elements_[0]; }
@@ -124,7 +124,7 @@
RTC_CHECK_EQ(num_rows_, num_columns_);
T trace = 0;
- for (int i = 0; i < num_rows_; ++i) {
+ for (size_t i = 0; i < num_rows_; ++i) {
trace += elements_[i][i];
}
return trace;
@@ -282,8 +282,8 @@
std::ostringstream ss;
ss << std::endl << "Matrix" << std::endl;
- for (int i = 0; i < num_rows_; ++i) {
- for (int j = 0; j < num_columns_; ++j) {
+ for (size_t i = 0; i < num_rows_; ++i) {
+ for (size_t j = 0; j < num_columns_; ++j) {
ss << elements_[i][j] << " ";
}
ss << std::endl;
@@ -294,8 +294,8 @@
}
protected:
- void SetNumRows(const int num_rows) { num_rows_ = num_rows; }
- void SetNumColumns(const int num_columns) { num_columns_ = num_columns; }
+ void SetNumRows(const size_t num_rows) { num_rows_ = num_rows; }
+ void SetNumColumns(const size_t num_columns) { num_columns_ = num_columns; }
T* data() { return &data_[0]; }
const T* data() const { return &data_[0]; }
const T* const* scratch_elements() const { return &scratch_elements_[0]; }
@@ -307,7 +307,7 @@
data_.resize(size);
elements_.resize(num_rows_);
- for (int i = 0; i < num_rows_; ++i) {
+ for (size_t i = 0; i < num_rows_; ++i) {
elements_[i] = &data_[i * num_columns_];
}
}
@@ -317,14 +317,14 @@
scratch_data_ = data_;
scratch_elements_.resize(num_rows_);
- for (int i = 0; i < num_rows_; ++i) {
+ for (size_t i = 0; i < num_rows_; ++i) {
scratch_elements_[i] = &scratch_data_[i * num_columns_];
}
}
private:
- int num_rows_;
- int num_columns_;
+ size_t num_rows_;
+ size_t num_columns_;
std::vector<T> data_;
std::vector<T*> elements_;
@@ -336,8 +336,8 @@
// Helpers for Transpose and Multiply operations that unify in-place and
// out-of-place solutions.
Matrix& Transpose(const T* const* src) {
- for (int i = 0; i < num_rows_; ++i) {
- for (int j = 0; j < num_columns_; ++j) {
+ for (size_t i = 0; i < num_rows_; ++i) {
+ for (size_t j = 0; j < num_columns_; ++j) {
elements_[i][j] = src[j][i];
}
}
@@ -345,11 +345,13 @@
return *this;
}
- Matrix& Multiply(const T* const* lhs, int num_rows_rhs, const T* const* rhs) {
- for (int row = 0; row < num_rows_; ++row) {
- for (int col = 0; col < num_columns_; ++col) {
+ Matrix& Multiply(const T* const* lhs,
+ size_t num_rows_rhs,
+ const T* const* rhs) {
+ for (size_t row = 0; row < num_rows_; ++row) {
+ for (size_t col = 0; col < num_columns_; ++col) {
T cur_element = 0;
- for (int i = 0; i < num_rows_rhs; ++i) {
+ for (size_t i = 0; i < num_rows_rhs; ++i) {
cur_element += lhs[row][i] * rhs[i][col];
}
diff --git a/webrtc/modules/audio_processing/beamformer/matrix_test_helpers.h b/webrtc/modules/audio_processing/beamformer/matrix_test_helpers.h
index 7c58670..9891a82 100644
--- a/webrtc/modules/audio_processing/beamformer/matrix_test_helpers.h
+++ b/webrtc/modules/audio_processing/beamformer/matrix_test_helpers.h
@@ -34,8 +34,8 @@
const T* const* expected_elements = expected.elements();
const T* const* actual_elements = actual.elements();
- for (int i = 0; i < expected.num_rows(); ++i) {
- for (int j = 0; j < expected.num_columns(); ++j) {
+ for (size_t i = 0; i < expected.num_rows(); ++i) {
+ for (size_t j = 0; j < expected.num_columns(); ++j) {
EXPECT_EQ(expected_elements[i][j], actual_elements[i][j]);
}
}
@@ -48,8 +48,8 @@
const float* const* expected_elements = expected.elements();
const float* const* actual_elements = actual.elements();
- for (int i = 0; i < expected.num_rows(); ++i) {
- for (int j = 0; j < expected.num_columns(); ++j) {
+ for (size_t i = 0; i < expected.num_rows(); ++i) {
+ for (size_t j = 0; j < expected.num_columns(); ++j) {
EXPECT_NEAR(expected_elements[i][j], actual_elements[i][j], kTolerance);
}
}
@@ -63,8 +63,8 @@
const complex<float>* const* expected_elements = expected.elements();
const complex<float>* const* actual_elements = actual.elements();
- for (int i = 0; i < expected.num_rows(); ++i) {
- for (int j = 0; j < expected.num_columns(); ++j) {
+ for (size_t i = 0; i < expected.num_rows(); ++i) {
+ for (size_t j = 0; j < expected.num_columns(); ++j) {
EXPECT_NEAR(expected_elements[i][j].real(),
actual_elements[i][j].real(),
kTolerance);
@@ -84,8 +84,8 @@
const complex<float>* const* expected_elements = expected.elements();
const complex<float>* const* actual_elements = actual.elements();
- for (int i = 0; i < expected.num_rows(); ++i) {
- for (int j = 0; j < expected.num_columns(); ++j) {
+ for (size_t i = 0; i < expected.num_rows(); ++i) {
+ for (size_t j = 0; j < expected.num_columns(); ++j) {
EXPECT_NEAR(expected_elements[i][j].real(),
actual_elements[i][j].real(),
tolerance);
diff --git a/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.cc b/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.cc
index 0544104..6ea7234 100644
--- a/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.cc
+++ b/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.cc
@@ -79,7 +79,7 @@
// The returned norm is clamped to be non-negative.
float Norm(const ComplexMatrix<float>& mat,
const ComplexMatrix<float>& norm_mat) {
- RTC_CHECK_EQ(norm_mat.num_rows(), 1);
+ RTC_CHECK_EQ(1u, norm_mat.num_rows());
RTC_CHECK_EQ(norm_mat.num_columns(), mat.num_rows());
RTC_CHECK_EQ(norm_mat.num_columns(), mat.num_columns());
@@ -89,8 +89,8 @@
const complex<float>* const* mat_els = mat.elements();
const complex<float>* const* norm_mat_els = norm_mat.elements();
- for (int i = 0; i < norm_mat.num_columns(); ++i) {
- for (int j = 0; j < norm_mat.num_columns(); ++j) {
+ for (size_t i = 0; i < norm_mat.num_columns(); ++i) {
+ for (size_t j = 0; j < norm_mat.num_columns(); ++j) {
first_product += conj(norm_mat_els[0][j]) * mat_els[j][i];
}
second_product += first_product * norm_mat_els[0][i];
@@ -102,15 +102,15 @@
// Does conjugate(|lhs|) * |rhs| for row vectors |lhs| and |rhs|.
complex<float> ConjugateDotProduct(const ComplexMatrix<float>& lhs,
const ComplexMatrix<float>& rhs) {
- RTC_CHECK_EQ(lhs.num_rows(), 1);
- RTC_CHECK_EQ(rhs.num_rows(), 1);
+ RTC_CHECK_EQ(1u, lhs.num_rows());
+ RTC_CHECK_EQ(1u, rhs.num_rows());
RTC_CHECK_EQ(lhs.num_columns(), rhs.num_columns());
const complex<float>* const* lhs_elements = lhs.elements();
const complex<float>* const* rhs_elements = rhs.elements();
complex<float> result = complex<float>(0.f, 0.f);
- for (int i = 0; i < lhs.num_columns(); ++i) {
+ for (size_t i = 0; i < lhs.num_columns(); ++i) {
result += conj(lhs_elements[0][i]) * rhs_elements[0][i];
}
@@ -126,8 +126,8 @@
float SumAbs(const ComplexMatrix<float>& mat) {
float sum_abs = 0.f;
const complex<float>* const* mat_els = mat.elements();
- for (int i = 0; i < mat.num_rows(); ++i) {
- for (int j = 0; j < mat.num_columns(); ++j) {
+ for (size_t i = 0; i < mat.num_rows(); ++i) {
+ for (size_t j = 0; j < mat.num_columns(); ++j) {
sum_abs += std::abs(mat_els[i][j]);
}
}
@@ -138,8 +138,8 @@
float SumSquares(const ComplexMatrix<float>& mat) {
float sum_squares = 0.f;
const complex<float>* const* mat_els = mat.elements();
- for (int i = 0; i < mat.num_rows(); ++i) {
- for (int j = 0; j < mat.num_columns(); ++j) {
+ for (size_t i = 0; i < mat.num_rows(); ++i) {
+ for (size_t j = 0; j < mat.num_columns(); ++j) {
float abs_value = std::abs(mat_els[i][j]);
sum_squares += abs_value * abs_value;
}
@@ -150,13 +150,13 @@
// Does |out| = |in|.' * conj(|in|) for row vector |in|.
void TransposedConjugatedProduct(const ComplexMatrix<float>& in,
ComplexMatrix<float>* out) {
- RTC_CHECK_EQ(in.num_rows(), 1);
+ RTC_CHECK_EQ(1u, in.num_rows());
RTC_CHECK_EQ(out->num_rows(), in.num_columns());
RTC_CHECK_EQ(out->num_columns(), in.num_columns());
const complex<float>* in_elements = in.elements()[0];
complex<float>* const* out_elements = out->elements();
- for (int i = 0; i < out->num_rows(); ++i) {
- for (int j = 0; j < out->num_columns(); ++j) {
+ for (size_t i = 0; i < out->num_rows(); ++i) {
+ for (size_t j = 0; j < out->num_columns(); ++j) {
out_elements[i][j] = in_elements[i] * conj(in_elements[j]);
}
}
@@ -408,13 +408,13 @@
}
void NonlinearBeamformer::ProcessAudioBlock(const complex_f* const* input,
- int num_input_channels,
+ size_t num_input_channels,
size_t num_freq_bins,
- int num_output_channels,
+ size_t num_output_channels,
complex_f* const* output) {
RTC_CHECK_EQ(kNumFreqBins, num_freq_bins);
RTC_CHECK_EQ(num_input_channels_, num_input_channels);
- RTC_CHECK_EQ(1, num_output_channels);
+ RTC_CHECK_EQ(1u, num_output_channels);
// Calculating the post-filter masks. Note that we need two for each
// frequency bin to account for the positive and negative interferer
@@ -483,7 +483,7 @@
const complex_f* delay_sum_mask_els =
normalized_delay_sum_masks_[f_ix].elements()[0];
- for (int c_ix = 0; c_ix < num_input_channels_; ++c_ix) {
+ for (size_t c_ix = 0; c_ix < num_input_channels_; ++c_ix) {
output_channel[f_ix] += input[c_ix][f_ix] * delay_sum_mask_els[c_ix];
}
diff --git a/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h b/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h
index b20d938..29c416c 100644
--- a/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h
+++ b/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h
@@ -67,9 +67,9 @@
// Process one frequency-domain block of audio. This is where the fun
// happens. Implements LappedTransform::Callback.
void ProcessAudioBlock(const complex<float>* const* input,
- int num_input_channels,
+ size_t num_input_channels,
size_t num_freq_bins,
- int num_output_channels,
+ size_t num_output_channels,
complex<float>* const* output) override;
private:
@@ -129,7 +129,7 @@
float window_[kFftSize];
// Parameters exposed to the user.
- const int num_input_channels_;
+ const size_t num_input_channels_;
int sample_rate_hz_;
const std::vector<Point> array_geometry_;
diff --git a/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer_test.cc b/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer_test.cc
index cc75248..d187552 100644
--- a/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer_test.cc
+++ b/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer_test.cc
@@ -12,6 +12,7 @@
#include "gflags/gflags.h"
#include "webrtc/base/checks.h"
+#include "webrtc/base/format_macros.h"
#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/common_audio/wav_file.h"
#include "webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h"
@@ -52,9 +53,9 @@
NonlinearBeamformer bf(array_geometry);
bf.Initialize(kChunkSizeMs, in_file.sample_rate());
- printf("Input file: %s\nChannels: %d, Sample rate: %d Hz\n\n",
+ printf("Input file: %s\nChannels: %" PRIuS ", Sample rate: %d Hz\n\n",
FLAGS_i.c_str(), in_file.num_channels(), in_file.sample_rate());
- printf("Output file: %s\nChannels: %d, Sample rate: %d Hz\n\n",
+ printf("Output file: %s\nChannels: %" PRIuS ", Sample rate: %d Hz\n\n",
FLAGS_o.c_str(), out_file.num_channels(), out_file.sample_rate());
ChannelBuffer<float> in_buf(
diff --git a/webrtc/modules/audio_processing/common.h b/webrtc/modules/audio_processing/common.h
index cf0d8b7..d4ddb92 100644
--- a/webrtc/modules/audio_processing/common.h
+++ b/webrtc/modules/audio_processing/common.h
@@ -17,7 +17,7 @@
namespace webrtc {
-static inline int ChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
+static inline size_t ChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
switch (layout) {
case AudioProcessing::kMono:
case AudioProcessing::kMonoAndKeyboard:
diff --git a/webrtc/modules/audio_processing/echo_cancellation_impl.cc b/webrtc/modules/audio_processing/echo_cancellation_impl.cc
index 13e71bc..debc597 100644
--- a/webrtc/modules/audio_processing/echo_cancellation_impl.cc
+++ b/webrtc/modules/audio_processing/echo_cancellation_impl.cc
@@ -99,8 +99,8 @@
// The ordering convention must be followed to pass to the correct AEC.
size_t handle_index = 0;
render_queue_buffer_.clear();
- for (int i = 0; i < apm_->num_output_channels(); i++) {
- for (int j = 0; j < audio->num_channels(); j++) {
+ for (size_t i = 0; i < apm_->num_output_channels(); i++) {
+ for (size_t j = 0; j < audio->num_channels(); j++) {
Handle* my_handle = static_cast<Handle*>(handle(handle_index));
// Retrieve any error code produced by the buffering of the farend
// signal
@@ -146,8 +146,8 @@
const size_t num_frames_per_band =
capture_queue_buffer_.size() /
(apm_->num_output_channels() * apm_->num_reverse_channels());
- for (int i = 0; i < apm_->num_output_channels(); i++) {
- for (int j = 0; j < apm_->num_reverse_channels(); j++) {
+ for (size_t i = 0; i < apm_->num_output_channels(); i++) {
+ for (size_t j = 0; j < apm_->num_reverse_channels(); j++) {
Handle* my_handle = static_cast<Handle*>(handle(handle_index));
WebRtcAec_BufferFarend(my_handle, &capture_queue_buffer_[buffer_index],
num_frames_per_band);
@@ -181,8 +181,8 @@
// The ordering convention must be followed to pass to the correct AEC.
size_t handle_index = 0;
stream_has_echo_ = false;
- for (int i = 0; i < audio->num_channels(); i++) {
- for (int j = 0; j < apm_->num_reverse_channels(); j++) {
+ for (size_t i = 0; i < audio->num_channels(); i++) {
+ for (size_t j = 0; j < apm_->num_reverse_channels(); j++) {
Handle* my_handle = handle(handle_index);
err = WebRtcAec_Process(my_handle, audio->split_bands_const_f(i),
audio->num_bands(), audio->split_bands_f(i),
@@ -489,7 +489,7 @@
return WebRtcAec_set_config(static_cast<Handle*>(handle), config);
}
-int EchoCancellationImpl::num_handles_required() const {
+size_t EchoCancellationImpl::num_handles_required() const {
// Not locked as it only relies on APM public API which is threadsafe.
return apm_->num_output_channels() * apm_->num_reverse_channels();
}
diff --git a/webrtc/modules/audio_processing/echo_cancellation_impl.h b/webrtc/modules/audio_processing/echo_cancellation_impl.h
index 9418fbf..a40a267 100644
--- a/webrtc/modules/audio_processing/echo_cancellation_impl.h
+++ b/webrtc/modules/audio_processing/echo_cancellation_impl.h
@@ -72,7 +72,7 @@
int InitializeHandle(void* handle) const override;
int ConfigureHandle(void* handle) const override;
void DestroyHandle(void* handle) const override;
- int num_handles_required() const override;
+ size_t num_handles_required() const override;
int GetHandleError(void* handle) const override;
void AllocateRenderQueue();
diff --git a/webrtc/modules/audio_processing/echo_control_mobile_impl.cc b/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
index a39528e..f2df5f7 100644
--- a/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
+++ b/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
@@ -104,8 +104,8 @@
// The ordering convention must be followed to pass to the correct AECM.
size_t handle_index = 0;
render_queue_buffer_.clear();
- for (int i = 0; i < apm_->num_output_channels(); i++) {
- for (int j = 0; j < audio->num_channels(); j++) {
+ for (size_t i = 0; i < apm_->num_output_channels(); i++) {
+ for (size_t j = 0; j < audio->num_channels(); j++) {
Handle* my_handle = static_cast<Handle*>(handle(handle_index));
err = WebRtcAecm_GetBufferFarendError(
my_handle, audio->split_bands_const(j)[kBand0To8kHz],
@@ -151,8 +151,8 @@
const size_t num_frames_per_band =
capture_queue_buffer_.size() /
(apm_->num_output_channels() * apm_->num_reverse_channels());
- for (int i = 0; i < apm_->num_output_channels(); i++) {
- for (int j = 0; j < apm_->num_reverse_channels(); j++) {
+ for (size_t i = 0; i < apm_->num_output_channels(); i++) {
+ for (size_t j = 0; j < apm_->num_reverse_channels(); j++) {
Handle* my_handle = static_cast<Handle*>(handle(handle_index));
WebRtcAecm_BufferFarend(my_handle, &capture_queue_buffer_[buffer_index],
num_frames_per_band);
@@ -182,7 +182,7 @@
// The ordering convention must be followed to pass to the correct AECM.
size_t handle_index = 0;
- for (int i = 0; i < audio->num_channels(); i++) {
+ for (size_t i = 0; i < audio->num_channels(); i++) {
// TODO(ajm): improve how this works, possibly inside AECM.
// This is kind of hacked up.
const int16_t* noisy = audio->low_pass_reference(i);
@@ -191,7 +191,7 @@
noisy = clean;
clean = NULL;
}
- for (int j = 0; j < apm_->num_reverse_channels(); j++) {
+ for (size_t j = 0; j < apm_->num_reverse_channels(); j++) {
Handle* my_handle = static_cast<Handle*>(handle(handle_index));
err = WebRtcAecm_Process(
my_handle,
@@ -394,7 +394,7 @@
return WebRtcAecm_set_config(static_cast<Handle*>(handle), config);
}
-int EchoControlMobileImpl::num_handles_required() const {
+size_t EchoControlMobileImpl::num_handles_required() const {
// Not locked as it only relies on APM public API which is threadsafe.
return apm_->num_output_channels() * apm_->num_reverse_channels();
}
diff --git a/webrtc/modules/audio_processing/echo_control_mobile_impl.h b/webrtc/modules/audio_processing/echo_control_mobile_impl.h
index 3b5dbf3..4d6529d 100644
--- a/webrtc/modules/audio_processing/echo_control_mobile_impl.h
+++ b/webrtc/modules/audio_processing/echo_control_mobile_impl.h
@@ -58,7 +58,7 @@
int InitializeHandle(void* handle) const override;
int ConfigureHandle(void* handle) const override;
void DestroyHandle(void* handle) const override;
- int num_handles_required() const override;
+ size_t num_handles_required() const override;
int GetHandleError(void* handle) const override;
void AllocateRenderQueue();
diff --git a/webrtc/modules/audio_processing/gain_control_impl.cc b/webrtc/modules/audio_processing/gain_control_impl.cc
index 7b284e8..04a6c7b 100644
--- a/webrtc/modules/audio_processing/gain_control_impl.cc
+++ b/webrtc/modules/audio_processing/gain_control_impl.cc
@@ -75,7 +75,7 @@
assert(audio->num_frames_per_band() <= 160);
render_queue_buffer_.resize(0);
- for (int i = 0; i < num_handles(); i++) {
+ for (size_t i = 0; i < num_handles(); i++) {
Handle* my_handle = static_cast<Handle*>(handle(i));
int err =
WebRtcAgc_GetAddFarendError(my_handle, audio->num_frames_per_band());
@@ -114,7 +114,7 @@
size_t buffer_index = 0;
const size_t num_frames_per_band =
capture_queue_buffer_.size() / num_handles();
- for (int i = 0; i < num_handles(); i++) {
+ for (size_t i = 0; i < num_handles(); i++) {
Handle* my_handle = static_cast<Handle*>(handle(i));
WebRtcAgc_AddFarend(my_handle, &capture_queue_buffer_[buffer_index],
num_frames_per_band);
@@ -138,7 +138,7 @@
if (mode_ == kAdaptiveAnalog) {
capture_levels_.assign(num_handles(), analog_capture_level_);
- for (int i = 0; i < num_handles(); i++) {
+ for (size_t i = 0; i < num_handles(); i++) {
Handle* my_handle = static_cast<Handle*>(handle(i));
err = WebRtcAgc_AddMic(
my_handle,
@@ -152,7 +152,7 @@
}
} else if (mode_ == kAdaptiveDigital) {
- for (int i = 0; i < num_handles(); i++) {
+ for (size_t i = 0; i < num_handles(); i++) {
Handle* my_handle = static_cast<Handle*>(handle(i));
int32_t capture_level_out = 0;
@@ -191,7 +191,7 @@
assert(audio->num_channels() == num_handles());
stream_is_saturated_ = false;
- for (int i = 0; i < num_handles(); i++) {
+ for (size_t i = 0; i < num_handles(); i++) {
Handle* my_handle = static_cast<Handle*>(handle(i));
int32_t capture_level_out = 0;
uint8_t saturation_warning = 0;
@@ -222,7 +222,7 @@
if (mode_ == kAdaptiveAnalog) {
// Take the analog level to be the average across the handles.
analog_capture_level_ = 0;
- for (int i = 0; i < num_handles(); i++) {
+ for (size_t i = 0; i < num_handles(); i++) {
analog_capture_level_ += capture_levels_[i];
}
@@ -433,7 +433,7 @@
return WebRtcAgc_set_config(static_cast<Handle*>(handle), config);
}
-int GainControlImpl::num_handles_required() const {
+size_t GainControlImpl::num_handles_required() const {
// Not locked as it only relies on APM public API which is threadsafe.
return apm_->num_proc_channels();
}
diff --git a/webrtc/modules/audio_processing/gain_control_impl.h b/webrtc/modules/audio_processing/gain_control_impl.h
index b531de9..72789ba 100644
--- a/webrtc/modules/audio_processing/gain_control_impl.h
+++ b/webrtc/modules/audio_processing/gain_control_impl.h
@@ -68,7 +68,7 @@
int InitializeHandle(void* handle) const override;
int ConfigureHandle(void* handle) const override;
void DestroyHandle(void* handle) const override;
- int num_handles_required() const override;
+ size_t num_handles_required() const override;
int GetHandleError(void* handle) const override;
void AllocateRenderQueue();
diff --git a/webrtc/modules/audio_processing/high_pass_filter_impl.cc b/webrtc/modules/audio_processing/high_pass_filter_impl.cc
index facf1e4..375d58f 100644
--- a/webrtc/modules/audio_processing/high_pass_filter_impl.cc
+++ b/webrtc/modules/audio_processing/high_pass_filter_impl.cc
@@ -92,10 +92,9 @@
HighPassFilterImpl::~HighPassFilterImpl() {}
-void HighPassFilterImpl::Initialize(int channels, int sample_rate_hz) {
- RTC_DCHECK_LE(0, channels);
+void HighPassFilterImpl::Initialize(size_t channels, int sample_rate_hz) {
std::vector<rtc::scoped_ptr<BiquadFilter>> new_filters(channels);
- for (int i = 0; i < channels; i++) {
+ for (size_t i = 0; i < channels; i++) {
new_filters[i].reset(new BiquadFilter(sample_rate_hz));
}
rtc::CritScope cs(crit_);
@@ -110,7 +109,7 @@
}
RTC_DCHECK_GE(160u, audio->num_frames_per_band());
- RTC_DCHECK_EQ(filters_.size(), static_cast<size_t>(audio->num_channels()));
+ RTC_DCHECK_EQ(filters_.size(), audio->num_channels());
for (size_t i = 0; i < filters_.size(); i++) {
filters_[i]->Process(audio->split_bands(i)[kBand0To8kHz],
audio->num_frames_per_band());
diff --git a/webrtc/modules/audio_processing/high_pass_filter_impl.h b/webrtc/modules/audio_processing/high_pass_filter_impl.h
index d6e84fd..0e985ba 100644
--- a/webrtc/modules/audio_processing/high_pass_filter_impl.h
+++ b/webrtc/modules/audio_processing/high_pass_filter_impl.h
@@ -26,7 +26,7 @@
~HighPassFilterImpl() override;
// TODO(peah): Fold into ctor, once public API is removed.
- void Initialize(int channels, int sample_rate_hz);
+ void Initialize(size_t channels, int sample_rate_hz);
void ProcessCaptureAudio(AudioBuffer* audio);
// HighPassFilter implementation.
diff --git a/webrtc/modules/audio_processing/include/audio_processing.h b/webrtc/modules/audio_processing/include/audio_processing.h
index d39d27e..5dd63eb 100644
--- a/webrtc/modules/audio_processing/include/audio_processing.h
+++ b/webrtc/modules/audio_processing/include/audio_processing.h
@@ -287,10 +287,10 @@
// necessary classes?
virtual int proc_sample_rate_hz() const = 0;
virtual int proc_split_sample_rate_hz() const = 0;
- virtual int num_input_channels() const = 0;
- virtual int num_proc_channels() const = 0;
- virtual int num_output_channels() const = 0;
- virtual int num_reverse_channels() const = 0;
+ virtual size_t num_input_channels() const = 0;
+ virtual size_t num_proc_channels() const = 0;
+ virtual size_t num_output_channels() const = 0;
+ virtual size_t num_reverse_channels() const = 0;
// Set to true when the output of AudioProcessing will be muted or in some
// other way not used. Ideally, the captured audio would still be processed,
@@ -502,7 +502,7 @@
// is true, the last channel in any corresponding list of
// channels is the keyboard channel.
StreamConfig(int sample_rate_hz = 0,
- int num_channels = 0,
+ size_t num_channels = 0,
bool has_keyboard = false)
: sample_rate_hz_(sample_rate_hz),
num_channels_(num_channels),
@@ -513,14 +513,14 @@
sample_rate_hz_ = value;
num_frames_ = calculate_frames(value);
}
- void set_num_channels(int value) { num_channels_ = value; }
+ void set_num_channels(size_t value) { num_channels_ = value; }
void set_has_keyboard(bool value) { has_keyboard_ = value; }
int sample_rate_hz() const { return sample_rate_hz_; }
// The number of channels in the stream, not including the keyboard channel if
// present.
- int num_channels() const { return num_channels_; }
+ size_t num_channels() const { return num_channels_; }
bool has_keyboard() const { return has_keyboard_; }
size_t num_frames() const { return num_frames_; }
@@ -541,7 +541,7 @@
}
int sample_rate_hz_;
- int num_channels_;
+ size_t num_channels_;
bool has_keyboard_;
size_t num_frames_;
};
diff --git a/webrtc/modules/audio_processing/include/mock_audio_processing.h b/webrtc/modules/audio_processing/include/mock_audio_processing.h
index 4ff52ba..9e1f2d5 100644
--- a/webrtc/modules/audio_processing/include/mock_audio_processing.h
+++ b/webrtc/modules/audio_processing/include/mock_audio_processing.h
@@ -201,11 +201,11 @@
MOCK_CONST_METHOD0(proc_split_sample_rate_hz,
int());
MOCK_CONST_METHOD0(num_input_channels,
- int());
+ size_t());
MOCK_CONST_METHOD0(num_output_channels,
- int());
+ size_t());
MOCK_CONST_METHOD0(num_reverse_channels,
- int());
+ size_t());
MOCK_METHOD1(set_output_will_be_muted,
void(bool muted));
MOCK_CONST_METHOD0(output_will_be_muted,
diff --git a/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc b/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc
index d014ce0..fe964ab 100644
--- a/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc
+++ b/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc
@@ -54,12 +54,12 @@
void IntelligibilityEnhancer::TransformCallback::ProcessAudioBlock(
const complex<float>* const* in_block,
- int in_channels,
+ size_t in_channels,
size_t frames,
- int /* out_channels */,
+ size_t /* out_channels */,
complex<float>* const* out_block) {
RTC_DCHECK_EQ(parent_->freqs_, frames);
- for (int i = 0; i < in_channels; ++i) {
+ for (size_t i = 0; i < in_channels; ++i) {
parent_->DispatchAudio(source_, in_block[i], out_block[i]);
}
}
@@ -129,7 +129,7 @@
void IntelligibilityEnhancer::ProcessRenderAudio(float* const* audio,
int sample_rate_hz,
- int num_channels) {
+ size_t num_channels) {
RTC_CHECK_EQ(sample_rate_hz_, sample_rate_hz);
RTC_CHECK_EQ(num_render_channels_, num_channels);
@@ -138,7 +138,7 @@
}
if (active_) {
- for (int i = 0; i < num_render_channels_; ++i) {
+ for (size_t i = 0; i < num_render_channels_; ++i) {
memcpy(audio[i], temp_render_out_buffer_.channels()[i],
chunk_length_ * sizeof(**audio));
}
@@ -147,7 +147,7 @@
void IntelligibilityEnhancer::AnalyzeCaptureAudio(float* const* audio,
int sample_rate_hz,
- int num_channels) {
+ size_t num_channels) {
RTC_CHECK_EQ(sample_rate_hz_, sample_rate_hz);
RTC_CHECK_EQ(num_capture_channels_, num_channels);
diff --git a/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h b/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
index 1e9e35a..1eb2234 100644
--- a/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
+++ b/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
@@ -47,8 +47,8 @@
gain_change_limit(0.1f),
rho(0.02f) {}
int sample_rate_hz;
- int num_capture_channels;
- int num_render_channels;
+ size_t num_capture_channels;
+ size_t num_render_channels;
intelligibility::VarianceArray::StepType var_type;
float var_decay_rate;
size_t var_window_size;
@@ -63,12 +63,12 @@
// Reads and processes chunk of noise stream in time domain.
void AnalyzeCaptureAudio(float* const* audio,
int sample_rate_hz,
- int num_channels);
+ size_t num_channels);
// Reads chunk of speech in time domain and updates with modified signal.
void ProcessRenderAudio(float* const* audio,
int sample_rate_hz,
- int num_channels);
+ size_t num_channels);
bool active() const;
private:
@@ -85,9 +85,9 @@
// All in frequency domain, receives input |in_block|, applies
// intelligibility enhancement, and writes result to |out_block|.
void ProcessAudioBlock(const std::complex<float>* const* in_block,
- int in_channels,
+ size_t in_channels,
size_t frames,
- int out_channels,
+ size_t out_channels,
std::complex<float>* const* out_block) override;
private:
@@ -144,8 +144,8 @@
const size_t bank_size_; // Num ERB filters.
const int sample_rate_hz_;
const int erb_resolution_;
- const int num_capture_channels_;
- const int num_render_channels_;
+ const size_t num_capture_channels_;
+ const size_t num_render_channels_;
const int analysis_rate_; // Num blocks before gains recalculated.
const bool active_; // Whether render gains are being updated.
diff --git a/webrtc/modules/audio_processing/intelligibility/test/intelligibility_proc.cc b/webrtc/modules/audio_processing/intelligibility/test/intelligibility_proc.cc
index 27d0ab4..4d2f5f4 100644
--- a/webrtc/modules/audio_processing/intelligibility/test/intelligibility_proc.cc
+++ b/webrtc/modules/audio_processing/intelligibility/test/intelligibility_proc.cc
@@ -68,7 +68,7 @@
"Enhanced output. Use '-' to "
"play through aplay immediately.");
-const int kNumChannels = 1;
+const size_t kNumChannels = 1;
// void function for gtest
void void_main(int argc, char* argv[]) {
diff --git a/webrtc/modules/audio_processing/level_estimator_impl.cc b/webrtc/modules/audio_processing/level_estimator_impl.cc
index aa676a8..187873e 100644
--- a/webrtc/modules/audio_processing/level_estimator_impl.cc
+++ b/webrtc/modules/audio_processing/level_estimator_impl.cc
@@ -35,7 +35,7 @@
return;
}
- for (int i = 0; i < audio->num_channels(); i++) {
+ for (size_t i = 0; i < audio->num_channels(); i++) {
rms_->Process(audio->channels_const()[i], audio->num_frames());
}
}
diff --git a/webrtc/modules/audio_processing/noise_suppression_impl.cc b/webrtc/modules/audio_processing/noise_suppression_impl.cc
index a40cd63..de7e856 100644
--- a/webrtc/modules/audio_processing/noise_suppression_impl.cc
+++ b/webrtc/modules/audio_processing/noise_suppression_impl.cc
@@ -52,15 +52,14 @@
NoiseSuppressionImpl::~NoiseSuppressionImpl() {}
-void NoiseSuppressionImpl::Initialize(int channels, int sample_rate_hz) {
- RTC_DCHECK_LE(0, channels);
+void NoiseSuppressionImpl::Initialize(size_t channels, int sample_rate_hz) {
rtc::CritScope cs(crit_);
channels_ = channels;
sample_rate_hz_ = sample_rate_hz;
std::vector<rtc::scoped_ptr<Suppressor>> new_suppressors;
if (enabled_) {
new_suppressors.resize(channels);
- for (int i = 0; i < channels; i++) {
+ for (size_t i = 0; i < channels; i++) {
new_suppressors[i].reset(new Suppressor(sample_rate_hz));
}
}
@@ -77,8 +76,7 @@
}
RTC_DCHECK_GE(160u, audio->num_frames_per_band());
- RTC_DCHECK_EQ(suppressors_.size(),
- static_cast<size_t>(audio->num_channels()));
+ RTC_DCHECK_EQ(suppressors_.size(), audio->num_channels());
for (size_t i = 0; i < suppressors_.size(); i++) {
WebRtcNs_Analyze(suppressors_[i]->state(),
audio->split_bands_const_f(i)[kBand0To8kHz]);
@@ -94,8 +92,7 @@
}
RTC_DCHECK_GE(160u, audio->num_frames_per_band());
- RTC_DCHECK_EQ(suppressors_.size(),
- static_cast<size_t>(audio->num_channels()));
+ RTC_DCHECK_EQ(suppressors_.size(), audio->num_channels());
for (size_t i = 0; i < suppressors_.size(); i++) {
#if defined(WEBRTC_NS_FLOAT)
WebRtcNs_Process(suppressors_[i]->state(),
diff --git a/webrtc/modules/audio_processing/noise_suppression_impl.h b/webrtc/modules/audio_processing/noise_suppression_impl.h
index 491c4d2..debbc61 100644
--- a/webrtc/modules/audio_processing/noise_suppression_impl.h
+++ b/webrtc/modules/audio_processing/noise_suppression_impl.h
@@ -26,7 +26,7 @@
~NoiseSuppressionImpl() override;
// TODO(peah): Fold into ctor, once public API is removed.
- void Initialize(int channels, int sample_rate_hz);
+ void Initialize(size_t channels, int sample_rate_hz);
void AnalyzeCaptureAudio(AudioBuffer* audio);
void ProcessCaptureAudio(AudioBuffer* audio);
@@ -42,7 +42,7 @@
rtc::CriticalSection* const crit_;
bool enabled_ GUARDED_BY(crit_) = false;
Level level_ GUARDED_BY(crit_) = kModerate;
- int channels_ GUARDED_BY(crit_) = 0;
+ size_t channels_ GUARDED_BY(crit_) = 0;
int sample_rate_hz_ GUARDED_BY(crit_) = 0;
std::vector<rtc::scoped_ptr<Suppressor>> suppressors_ GUARDED_BY(crit_);
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(NoiseSuppressionImpl);
diff --git a/webrtc/modules/audio_processing/processing_component.cc b/webrtc/modules/audio_processing/processing_component.cc
index 9e16d7c..7abd8e2 100644
--- a/webrtc/modules/audio_processing/processing_component.cc
+++ b/webrtc/modules/audio_processing/processing_component.cc
@@ -55,12 +55,12 @@
return enabled_;
}
-void* ProcessingComponent::handle(int index) const {
+void* ProcessingComponent::handle(size_t index) const {
assert(index < num_handles_);
return handles_[index];
}
-int ProcessingComponent::num_handles() const {
+size_t ProcessingComponent::num_handles() const {
return num_handles_;
}
@@ -70,12 +70,12 @@
}
num_handles_ = num_handles_required();
- if (num_handles_ > static_cast<int>(handles_.size())) {
+ if (num_handles_ > handles_.size()) {
handles_.resize(num_handles_, NULL);
}
- assert(static_cast<int>(handles_.size()) >= num_handles_);
- for (int i = 0; i < num_handles_; i++) {
+ assert(handles_.size() >= num_handles_);
+ for (size_t i = 0; i < num_handles_; i++) {
if (handles_[i] == NULL) {
handles_[i] = CreateHandle();
if (handles_[i] == NULL) {
@@ -98,8 +98,8 @@
return AudioProcessing::kNoError;
}
- assert(static_cast<int>(handles_.size()) >= num_handles_);
- for (int i = 0; i < num_handles_; i++) {
+ assert(handles_.size() >= num_handles_);
+ for (size_t i = 0; i < num_handles_; i++) {
int err = ConfigureHandle(handles_[i]);
if (err != AudioProcessing::kNoError) {
return GetHandleError(handles_[i]);
diff --git a/webrtc/modules/audio_processing/processing_component.h b/webrtc/modules/audio_processing/processing_component.h
index 291aea3..577f157 100644
--- a/webrtc/modules/audio_processing/processing_component.h
+++ b/webrtc/modules/audio_processing/processing_component.h
@@ -47,21 +47,21 @@
protected:
virtual int Configure();
int EnableComponent(bool enable);
- void* handle(int index) const;
- int num_handles() const;
+ void* handle(size_t index) const;
+ size_t num_handles() const;
private:
virtual void* CreateHandle() const = 0;
virtual int InitializeHandle(void* handle) const = 0;
virtual int ConfigureHandle(void* handle) const = 0;
virtual void DestroyHandle(void* handle) const = 0;
- virtual int num_handles_required() const = 0;
+ virtual size_t num_handles_required() const = 0;
virtual int GetHandleError(void* handle) const = 0;
std::vector<void*> handles_;
bool initialized_;
bool enabled_;
- int num_handles_;
+ size_t num_handles_;
};
} // namespace webrtc
diff --git a/webrtc/modules/audio_processing/splitting_filter.cc b/webrtc/modules/audio_processing/splitting_filter.cc
index 60427e2..46cc935 100644
--- a/webrtc/modules/audio_processing/splitting_filter.cc
+++ b/webrtc/modules/audio_processing/splitting_filter.cc
@@ -16,7 +16,7 @@
namespace webrtc {
-SplittingFilter::SplittingFilter(int num_channels,
+SplittingFilter::SplittingFilter(size_t num_channels,
size_t num_bands,
size_t num_frames)
: num_bands_(num_bands) {
@@ -24,7 +24,7 @@
if (num_bands_ == 2) {
two_bands_states_.resize(num_channels);
} else if (num_bands_ == 3) {
- for (int i = 0; i < num_channels; ++i) {
+ for (size_t i = 0; i < num_channels; ++i) {
three_band_filter_banks_.push_back(new ThreeBandFilterBank(num_frames));
}
}
@@ -58,8 +58,7 @@
void SplittingFilter::TwoBandsAnalysis(const IFChannelBuffer* data,
IFChannelBuffer* bands) {
- RTC_DCHECK_EQ(static_cast<int>(two_bands_states_.size()),
- data->num_channels());
+ RTC_DCHECK_EQ(two_bands_states_.size(), data->num_channels());
for (size_t i = 0; i < two_bands_states_.size(); ++i) {
WebRtcSpl_AnalysisQMF(data->ibuf_const()->channels()[i],
data->num_frames(),
@@ -72,8 +71,7 @@
void SplittingFilter::TwoBandsSynthesis(const IFChannelBuffer* bands,
IFChannelBuffer* data) {
- RTC_DCHECK_EQ(static_cast<int>(two_bands_states_.size()),
- data->num_channels());
+ RTC_DCHECK_EQ(two_bands_states_.size(), data->num_channels());
for (size_t i = 0; i < two_bands_states_.size(); ++i) {
WebRtcSpl_SynthesisQMF(bands->ibuf_const()->channels(0)[i],
bands->ibuf_const()->channels(1)[i],
@@ -86,8 +84,7 @@
void SplittingFilter::ThreeBandsAnalysis(const IFChannelBuffer* data,
IFChannelBuffer* bands) {
- RTC_DCHECK_EQ(static_cast<int>(three_band_filter_banks_.size()),
- data->num_channels());
+ RTC_DCHECK_EQ(three_band_filter_banks_.size(), data->num_channels());
for (size_t i = 0; i < three_band_filter_banks_.size(); ++i) {
three_band_filter_banks_[i]->Analysis(data->fbuf_const()->channels()[i],
data->num_frames(),
@@ -97,8 +94,7 @@
void SplittingFilter::ThreeBandsSynthesis(const IFChannelBuffer* bands,
IFChannelBuffer* data) {
- RTC_DCHECK_EQ(static_cast<int>(three_band_filter_banks_.size()),
- data->num_channels());
+ RTC_DCHECK_EQ(three_band_filter_banks_.size(), data->num_channels());
for (size_t i = 0; i < three_band_filter_banks_.size(); ++i) {
three_band_filter_banks_[i]->Synthesis(bands->fbuf_const()->bands(i),
bands->num_frames_per_band(),
diff --git a/webrtc/modules/audio_processing/splitting_filter.h b/webrtc/modules/audio_processing/splitting_filter.h
index 4698d3f..6b81c2f 100644
--- a/webrtc/modules/audio_processing/splitting_filter.h
+++ b/webrtc/modules/audio_processing/splitting_filter.h
@@ -45,7 +45,7 @@
// used.
class SplittingFilter {
public:
- SplittingFilter(int num_channels, size_t num_bands, size_t num_frames);
+ SplittingFilter(size_t num_channels, size_t num_bands, size_t num_frames);
void Analysis(const IFChannelBuffer* data, IFChannelBuffer* bands);
void Synthesis(const IFChannelBuffer* bands, IFChannelBuffer* data);
diff --git a/webrtc/modules/audio_processing/test/audio_file_processor.cc b/webrtc/modules/audio_processing/test/audio_file_processor.cc
index 4c77356..56e9b4b 100644
--- a/webrtc/modules/audio_processing/test/audio_file_processor.cc
+++ b/webrtc/modules/audio_processing/test/audio_file_processor.cc
@@ -132,7 +132,8 @@
void AecDumpFileProcessor::HandleMessage(const Stream& msg) {
RTC_CHECK(!msg.has_input_data());
- RTC_CHECK_EQ(in_buf_->num_channels(), msg.input_channel_size());
+ RTC_CHECK_EQ(in_buf_->num_channels(),
+ static_cast<size_t>(msg.input_channel_size()));
for (int i = 0; i < msg.input_channel_size(); ++i) {
RTC_CHECK_EQ(in_buf_->num_frames() * sizeof(*in_buf_->channels()[i]),
@@ -157,7 +158,8 @@
void AecDumpFileProcessor::HandleMessage(const ReverseStream& msg) {
RTC_CHECK(!msg.has_data());
- RTC_CHECK_EQ(reverse_buf_->num_channels(), msg.channel_size());
+ RTC_CHECK_EQ(reverse_buf_->num_channels(),
+ static_cast<size_t>(msg.channel_size()));
for (int i = 0; i < msg.channel_size(); ++i) {
RTC_CHECK_EQ(reverse_buf_->num_frames() * sizeof(*in_buf_->channels()[i]),
diff --git a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
index 6eae1e5..94aea17 100644
--- a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
+++ b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
@@ -69,7 +69,7 @@
cb->num_frames(),
cb->num_channels(),
cb_int.channels());
- for (int i = 0; i < cb->num_channels(); ++i) {
+ for (size_t i = 0; i < cb->num_channels(); ++i) {
S16ToFloat(cb_int.channels()[i],
cb->num_frames(),
cb->channels()[i]);
@@ -81,7 +81,7 @@
}
// Number of channels including the keyboard channel.
-int TotalChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
+size_t TotalChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
switch (layout) {
case AudioProcessing::kMono:
return 1;
@@ -131,7 +131,7 @@
}
void SetFrameTo(AudioFrame* frame, int16_t left, int16_t right) {
- ASSERT_EQ(2, frame->num_channels_);
+ ASSERT_EQ(2u, frame->num_channels_);
for (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
frame->data_[i] = left;
frame->data_[i + 1] = right;
@@ -256,10 +256,10 @@
int output_rate,
int reverse_input_rate,
int reverse_output_rate,
- int num_input_channels,
- int num_output_channels,
- int num_reverse_input_channels,
- int num_reverse_output_channels,
+ size_t num_input_channels,
+ size_t num_output_channels,
+ size_t num_reverse_input_channels,
+ size_t num_reverse_output_channels,
StreamDirection file_direction) {
std::ostringstream ss;
ss << name << "_i" << num_input_channels << "_" << input_rate / 1000 << "_ir"
@@ -354,9 +354,9 @@
void Init(int sample_rate_hz,
int output_sample_rate_hz,
int reverse_sample_rate_hz,
- int num_input_channels,
- int num_output_channels,
- int num_reverse_channels,
+ size_t num_input_channels,
+ size_t num_output_channels,
+ size_t num_reverse_channels,
bool open_output_file);
void Init(AudioProcessing* ap);
void EnableAllComponents();
@@ -369,12 +369,12 @@
void ProcessDelayVerificationTest(int delay_ms, int system_delay_ms,
int delay_min, int delay_max);
void TestChangingChannelsInt16Interface(
- int num_channels,
+ size_t num_channels,
AudioProcessing::Error expected_return);
- void TestChangingForwardChannels(int num_in_channels,
- int num_out_channels,
+ void TestChangingForwardChannels(size_t num_in_channels,
+ size_t num_out_channels,
AudioProcessing::Error expected_return);
- void TestChangingReverseChannels(int num_rev_channels,
+ void TestChangingReverseChannels(size_t num_rev_channels,
AudioProcessing::Error expected_return);
void RunQuantizedVolumeDoesNotGetStuckTest(int sample_rate);
void RunManualVolumeChangeIsPossibleTest(int sample_rate);
@@ -395,7 +395,7 @@
rtc::scoped_ptr<ChannelBuffer<float> > float_cb_;
rtc::scoped_ptr<ChannelBuffer<float> > revfloat_cb_;
int output_sample_rate_hz_;
- int num_output_channels_;
+ size_t num_output_channels_;
FILE* far_file_;
FILE* near_file_;
FILE* out_file_;
@@ -479,9 +479,9 @@
void ApmTest::Init(int sample_rate_hz,
int output_sample_rate_hz,
int reverse_sample_rate_hz,
- int num_input_channels,
- int num_output_channels,
- int num_reverse_channels,
+ size_t num_input_channels,
+ size_t num_output_channels,
+ size_t num_reverse_channels,
bool open_output_file) {
SetContainerFormat(sample_rate_hz, num_input_channels, frame_, &float_cb_);
output_sample_rate_hz_ = output_sample_rate_hz;
@@ -813,7 +813,7 @@
}
void ApmTest::TestChangingChannelsInt16Interface(
- int num_channels,
+ size_t num_channels,
AudioProcessing::Error expected_return) {
frame_->num_channels_ = num_channels;
EXPECT_EQ(expected_return, apm_->ProcessStream(frame_));
@@ -821,8 +821,8 @@
}
void ApmTest::TestChangingForwardChannels(
- int num_in_channels,
- int num_out_channels,
+ size_t num_in_channels,
+ size_t num_out_channels,
AudioProcessing::Error expected_return) {
const StreamConfig input_stream = {frame_->sample_rate_hz_, num_in_channels};
const StreamConfig output_stream = {output_sample_rate_hz_, num_out_channels};
@@ -833,7 +833,7 @@
}
void ApmTest::TestChangingReverseChannels(
- int num_rev_channels,
+ size_t num_rev_channels,
AudioProcessing::Error expected_return) {
const ProcessingConfig processing_config = {
{{frame_->sample_rate_hz_, apm_->num_input_channels()},
@@ -854,11 +854,11 @@
TestChangingChannelsInt16Interface(0, apm_->kBadNumberChannelsError);
- for (int i = 1; i < 4; i++) {
+ for (size_t i = 1; i < 4; i++) {
TestChangingChannelsInt16Interface(i, kNoErr);
EXPECT_EQ(i, apm_->num_input_channels());
// We always force the number of reverse channels used for processing to 1.
- EXPECT_EQ(1, apm_->num_reverse_channels());
+ EXPECT_EQ(1u, apm_->num_reverse_channels());
}
}
@@ -869,8 +869,8 @@
TestChangingForwardChannels(0, 1, apm_->kBadNumberChannelsError);
TestChangingReverseChannels(0, apm_->kBadNumberChannelsError);
- for (int i = 1; i < 4; ++i) {
- for (int j = 0; j < 1; ++j) {
+ for (size_t i = 1; i < 4; ++i) {
+ for (size_t j = 0; j < 1; ++j) {
// Output channels much be one or match input channels.
if (j == 1 || i == j) {
TestChangingForwardChannels(i, j, kNoErr);
@@ -879,7 +879,7 @@
EXPECT_EQ(i, apm_->num_input_channels());
EXPECT_EQ(j, apm_->num_output_channels());
// The number of reverse channels used for processing to is always 1.
- EXPECT_EQ(1, apm_->num_reverse_channels());
+ EXPECT_EQ(1u, apm_->num_reverse_channels());
} else {
TestChangingForwardChannels(i, j,
AudioProcessing::kBadNumberChannelsError);
@@ -1293,8 +1293,8 @@
const int kSampleRateHz = 16000;
const size_t kSamplesPerChannel =
static_cast<size_t>(AudioProcessing::kChunkSizeMs * kSampleRateHz / 1000);
- const int kNumInputChannels = 2;
- const int kNumOutputChannels = 1;
+ const size_t kNumInputChannels = 2;
+ const size_t kNumOutputChannels = 1;
const size_t kNumChunks = 700;
const float kScaleFactor = 0.25f;
Config config;
@@ -1327,7 +1327,7 @@
int_data.get(),
float_data.get(),
&src_buf));
- for (int j = 0; j < kNumInputChannels; ++j) {
+ for (size_t j = 0; j < kNumInputChannels; ++j) {
for (size_t k = 0; k < kSamplesPerChannel; ++k) {
src_buf.channels()[j][k] *= kScaleFactor;
}
@@ -1352,7 +1352,7 @@
int_data.get(),
float_data.get(),
&src_buf));
- for (int j = 0; j < kNumInputChannels; ++j) {
+ for (size_t j = 0; j < kNumInputChannels; ++j) {
for (size_t k = 0; k < kSamplesPerChannel; ++k) {
src_buf.channels()[j][k] *= kScaleFactor;
}
@@ -1742,7 +1742,8 @@
const audioproc::ReverseStream msg = event_msg.reverse_stream();
if (msg.channel_size() > 0) {
- ASSERT_EQ(revframe_->num_channels_, msg.channel_size());
+ ASSERT_EQ(revframe_->num_channels_,
+ static_cast<size_t>(msg.channel_size()));
for (int i = 0; i < msg.channel_size(); ++i) {
memcpy(revfloat_cb_->channels()[i],
msg.channel(i).data(),
@@ -1772,7 +1773,8 @@
}
if (msg.input_channel_size() > 0) {
- ASSERT_EQ(frame_->num_channels_, msg.input_channel_size());
+ ASSERT_EQ(frame_->num_channels_,
+ static_cast<size_t>(msg.input_channel_size()));
for (int i = 0; i < msg.input_channel_size(); ++i) {
memcpy(float_cb_->channels()[i],
msg.input_channel(i).data(),
@@ -1930,9 +1932,12 @@
if (test->num_input_channels() != test->num_output_channels())
continue;
- const int num_render_channels = test->num_reverse_channels();
- const int num_input_channels = test->num_input_channels();
- const int num_output_channels = test->num_output_channels();
+ const size_t num_render_channels =
+ static_cast<size_t>(test->num_reverse_channels());
+ const size_t num_input_channels =
+ static_cast<size_t>(test->num_input_channels());
+ const size_t num_output_channels =
+ static_cast<size_t>(test->num_output_channels());
const size_t samples_per_channel = static_cast<size_t>(
test->sample_rate() * AudioProcessing::kChunkSizeMs / 1000);
@@ -1975,7 +1980,7 @@
test->sample_rate(),
LayoutFromChannels(num_output_channels),
float_cb_->channels()));
- for (int j = 0; j < num_output_channels; ++j) {
+ for (size_t j = 0; j < num_output_channels; ++j) {
FloatToS16(float_cb_->channels()[j],
samples_per_channel,
output_cb.channels()[j]);
@@ -2008,7 +2013,7 @@
0.01);
// Reset in case of downmixing.
- frame_->num_channels_ = test->num_input_channels();
+ frame_->num_channels_ = static_cast<size_t>(test->num_input_channels());
}
rewind(far_file_);
rewind(near_file_);
@@ -2069,9 +2074,9 @@
Init(test->sample_rate(),
test->sample_rate(),
test->sample_rate(),
- test->num_input_channels(),
- test->num_output_channels(),
- test->num_reverse_channels(),
+ static_cast<size_t>(test->num_input_channels()),
+ static_cast<size_t>(test->num_output_channels()),
+ static_cast<size_t>(test->num_reverse_channels()),
true);
int frame_count = 0;
@@ -2096,7 +2101,8 @@
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
// Ensure the frame was downmixed properly.
- EXPECT_EQ(test->num_output_channels(), frame_->num_channels_);
+ EXPECT_EQ(static_cast<size_t>(test->num_output_channels()),
+ frame_->num_channels_);
max_output_average += MaxAudioFrame(*frame_);
@@ -2126,7 +2132,7 @@
ASSERT_EQ(frame_size, write_count);
// Reset in case of downmixing.
- frame_->num_channels_ = test->num_input_channels();
+ frame_->num_channels_ = static_cast<size_t>(test->num_input_channels());
frame_count++;
}
max_output_average /= frame_count;
@@ -2350,7 +2356,7 @@
static void SetUpTestCase() {
// Create all needed output reference files.
const int kNativeRates[] = {8000, 16000, 32000, 48000};
- const int kNumChannels[] = {1, 2};
+ const size_t kNumChannels[] = {1, 2};
for (size_t i = 0; i < arraysize(kNativeRates); ++i) {
for (size_t j = 0; j < arraysize(kNumChannels); ++j) {
for (size_t k = 0; k < arraysize(kNumChannels); ++k) {
@@ -2374,10 +2380,10 @@
int output_rate,
int reverse_input_rate,
int reverse_output_rate,
- int num_input_channels,
- int num_output_channels,
- int num_reverse_input_channels,
- int num_reverse_output_channels,
+ size_t num_input_channels,
+ size_t num_output_channels,
+ size_t num_reverse_input_channels,
+ size_t num_reverse_output_channels,
std::string output_file_prefix) {
Config config;
config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
diff --git a/webrtc/modules/audio_processing/test/audioproc_float.cc b/webrtc/modules/audio_processing/test/audioproc_float.cc
index d64b006..a489d25 100644
--- a/webrtc/modules/audio_processing/test/audioproc_float.cc
+++ b/webrtc/modules/audio_processing/test/audioproc_float.cc
@@ -16,6 +16,7 @@
#include "gflags/gflags.h"
#include "webrtc/base/checks.h"
+#include "webrtc/base/format_macros.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/common_audio/wav_file.h"
@@ -26,6 +27,14 @@
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/test/testsupport/trace_to_stderr.h"
+namespace {
+
+bool ValidateOutChannels(const char* flagname, int32_t value) {
+ return value >= 0;
+}
+
+} // namespace
+
DEFINE_string(dump, "", "Name of the aecdump debug file to read from.");
DEFINE_string(i, "", "Name of the capture input stream file to read from.");
DEFINE_string(
@@ -33,6 +42,8 @@
"out.wav",
"Name of the output file to write the processed capture stream to.");
DEFINE_int32(out_channels, 1, "Number of output channels.");
+const bool out_channels_dummy =
+ google::RegisterFlagValidator(&FLAGS_out_channels, &ValidateOutChannels);
DEFINE_int32(out_sample_rate, 48000, "Output sample rate in Hz.");
DEFINE_string(mic_positions, "",
"Space delimited cartesian coordinates of microphones in meters. "
@@ -117,8 +128,8 @@
ap->set_stream_key_pressed(FLAGS_ts);
rtc::scoped_ptr<AudioFileProcessor> processor;
- auto out_file = rtc_make_scoped_ptr(
- new WavWriter(FLAGS_o, FLAGS_out_sample_rate, FLAGS_out_channels));
+ auto out_file = rtc_make_scoped_ptr(new WavWriter(
+ FLAGS_o, FLAGS_out_sample_rate, static_cast<size_t>(FLAGS_out_channels)));
std::cout << FLAGS_o << ": " << out_file->FormatAsString() << std::endl;
if (FLAGS_dump.empty()) {
auto in_file = rtc_make_scoped_ptr(new WavReader(FLAGS_i));
diff --git a/webrtc/modules/audio_processing/test/debug_dump_test.cc b/webrtc/modules/audio_processing/test/debug_dump_test.cc
index d2dd9c8..005faa0 100644
--- a/webrtc/modules/audio_processing/test/debug_dump_test.cc
+++ b/webrtc/modules/audio_processing/test/debug_dump_test.cc
@@ -327,7 +327,8 @@
else
apm_->set_stream_key_pressed(true);
- ASSERT_EQ(input_config_.num_channels(), msg.input_channel_size());
+ ASSERT_EQ(input_config_.num_channels(),
+ static_cast<size_t>(msg.input_channel_size()));
ASSERT_EQ(input_config_.num_frames() * sizeof(float),
msg.input_channel(0).size());
@@ -341,7 +342,8 @@
output_config_, output_->channels()));
// Check that output of APM is bit-exact to the output in the dump.
- ASSERT_EQ(output_config_.num_channels(), msg.output_channel_size());
+ ASSERT_EQ(output_config_.num_channels(),
+ static_cast<size_t>(msg.output_channel_size()));
ASSERT_EQ(output_config_.num_frames() * sizeof(float),
msg.output_channel(0).size());
for (int i = 0; i < msg.output_channel_size(); ++i) {
@@ -355,7 +357,8 @@
ASSERT_TRUE(apm_.get());
ASSERT_GT(msg.channel_size(), 0);
- ASSERT_EQ(reverse_config_.num_channels(), msg.channel_size());
+ ASSERT_EQ(reverse_config_.num_channels(),
+ static_cast<size_t>(msg.channel_size()));
ASSERT_EQ(reverse_config_.num_frames() * sizeof(float),
msg.channel(0).size());
diff --git a/webrtc/modules/audio_processing/test/process_test.cc b/webrtc/modules/audio_processing/test/process_test.cc
index ae6b4dc..6e20a78 100644
--- a/webrtc/modules/audio_processing/test/process_test.cc
+++ b/webrtc/modules/audio_processing/test/process_test.cc
@@ -17,6 +17,7 @@
#include <algorithm>
+#include "webrtc/base/format_macros.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
@@ -159,9 +160,9 @@
int32_t sample_rate_hz = 16000;
- int num_capture_input_channels = 1;
- int num_capture_output_channels = 1;
- int num_render_channels = 1;
+ size_t num_capture_input_channels = 1;
+ size_t num_capture_output_channels = 1;
+ size_t num_render_channels = 1;
int samples_per_channel = sample_rate_hz / 100;
@@ -207,14 +208,14 @@
} else if (strcmp(argv[i], "-ch") == 0) {
i++;
ASSERT_LT(i + 1, argc) << "Specify number of channels after -ch";
- ASSERT_EQ(1, sscanf(argv[i], "%d", &num_capture_input_channels));
+ ASSERT_EQ(1, sscanf(argv[i], "%" PRIuS, &num_capture_input_channels));
i++;
- ASSERT_EQ(1, sscanf(argv[i], "%d", &num_capture_output_channels));
+ ASSERT_EQ(1, sscanf(argv[i], "%" PRIuS, &num_capture_output_channels));
} else if (strcmp(argv[i], "-rch") == 0) {
i++;
ASSERT_LT(i, argc) << "Specify number of channels after -rch";
- ASSERT_EQ(1, sscanf(argv[i], "%d", &num_render_channels));
+ ASSERT_EQ(1, sscanf(argv[i], "%" PRIuS, &num_render_channels));
} else if (strcmp(argv[i], "-aec") == 0) {
ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
@@ -447,10 +448,10 @@
if (verbose) {
printf("Sample rate: %d Hz\n", sample_rate_hz);
- printf("Primary channels: %d (in), %d (out)\n",
+ printf("Primary channels: %" PRIuS " (in), %" PRIuS " (out)\n",
num_capture_input_channels,
num_capture_output_channels);
- printf("Reverse channels: %d \n", num_render_channels);
+ printf("Reverse channels: %" PRIuS "\n", num_render_channels);
}
const std::string out_path = webrtc::test::OutputPath();
@@ -601,14 +602,18 @@
if (msg.has_output_sample_rate()) {
output_sample_rate = msg.output_sample_rate();
}
- output_layout = LayoutFromChannels(msg.num_output_channels());
- ASSERT_EQ(kNoErr, apm->Initialize(
- msg.sample_rate(),
- output_sample_rate,
- reverse_sample_rate,
- LayoutFromChannels(msg.num_input_channels()),
- output_layout,
- LayoutFromChannels(msg.num_reverse_channels())));
+ output_layout =
+ LayoutFromChannels(static_cast<size_t>(msg.num_output_channels()));
+ ASSERT_EQ(kNoErr,
+ apm->Initialize(
+ msg.sample_rate(),
+ output_sample_rate,
+ reverse_sample_rate,
+ LayoutFromChannels(
+ static_cast<size_t>(msg.num_input_channels())),
+ output_layout,
+ LayoutFromChannels(
+ static_cast<size_t>(msg.num_reverse_channels()))));
samples_per_channel = msg.sample_rate() / 100;
far_frame.sample_rate_hz_ = reverse_sample_rate;
@@ -638,9 +643,9 @@
if (!raw_output) {
// The WAV file needs to be reset every time, because it can't change
// its sample rate or number of channels.
- output_wav_file.reset(new WavWriter(out_filename + ".wav",
- output_sample_rate,
- msg.num_output_channels()));
+ output_wav_file.reset(new WavWriter(
+ out_filename + ".wav", output_sample_rate,
+ static_cast<size_t>(msg.num_output_channels())));
}
} else if (event_msg.type() == Event::REVERSE_STREAM) {
diff --git a/webrtc/modules/audio_processing/test/test_utils.cc b/webrtc/modules/audio_processing/test/test_utils.cc
index 74f8b73..0bd7012 100644
--- a/webrtc/modules/audio_processing/test/test_utils.cc
+++ b/webrtc/modules/audio_processing/test/test_utils.cc
@@ -76,7 +76,7 @@
void WriteFloatData(const float* const* data,
size_t samples_per_channel,
- int num_channels,
+ size_t num_channels,
WavWriter* wav_file,
RawFile* raw_file) {
size_t length = num_channels * samples_per_channel;
@@ -116,7 +116,7 @@
sample_rate_hz / 1000;
}
-AudioProcessing::ChannelLayout LayoutFromChannels(int num_channels) {
+AudioProcessing::ChannelLayout LayoutFromChannels(size_t num_channels) {
switch (num_channels) {
case 1:
return AudioProcessing::kMono;
diff --git a/webrtc/modules/audio_processing/test/test_utils.h b/webrtc/modules/audio_processing/test/test_utils.h
index 291e03e..e23beb6 100644
--- a/webrtc/modules/audio_processing/test/test_utils.h
+++ b/webrtc/modules/audio_processing/test/test_utils.h
@@ -79,7 +79,7 @@
void WriteFloatData(const float* const* data,
size_t samples_per_channel,
- int num_channels,
+ size_t num_channels,
WavWriter* wav_file,
RawFile* raw_file);
@@ -93,7 +93,7 @@
template <typename T>
void SetContainerFormat(int sample_rate_hz,
- int num_channels,
+ size_t num_channels,
AudioFrame* frame,
rtc::scoped_ptr<ChannelBuffer<T> >* cb) {
SetFrameSampleRate(frame, sample_rate_hz);
@@ -101,7 +101,7 @@
cb->reset(new ChannelBuffer<T>(frame->samples_per_channel_, num_channels));
}
-AudioProcessing::ChannelLayout LayoutFromChannels(int num_channels);
+AudioProcessing::ChannelLayout LayoutFromChannels(size_t num_channels);
template <typename T>
float ComputeSNR(const T* ref, const T* test, size_t length, float* variance) {
diff --git a/webrtc/modules/audio_processing/test/unpack.cc b/webrtc/modules/audio_processing/test/unpack.cc
index cd9205e..8b2b082 100644
--- a/webrtc/modules/audio_processing/test/unpack.cc
+++ b/webrtc/modules/audio_processing/test/unpack.cc
@@ -17,6 +17,7 @@
#include "gflags/gflags.h"
#include "webrtc/audio_processing/debug.pb.h"
+#include "webrtc/base/format_macros.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_processing/test/protobuf_utils.h"
#include "webrtc/modules/audio_processing/test/test_utils.h"
@@ -79,9 +80,9 @@
size_t reverse_samples_per_channel = 0;
size_t input_samples_per_channel = 0;
size_t output_samples_per_channel = 0;
- int num_reverse_channels = 0;
- int num_input_channels = 0;
- int num_output_channels = 0;
+ size_t num_reverse_channels = 0;
+ size_t num_input_channels = 0;
+ size_t num_output_channels = 0;
rtc::scoped_ptr<WavWriter> reverse_wav_file;
rtc::scoped_ptr<WavWriter> input_wav_file;
rtc::scoped_ptr<WavWriter> output_wav_file;
@@ -117,7 +118,7 @@
}
rtc::scoped_ptr<const float* []> data(
new const float* [num_reverse_channels]);
- for (int i = 0; i < num_reverse_channels; ++i) {
+ for (size_t i = 0; i < num_reverse_channels; ++i) {
data[i] = reinterpret_cast<const float*>(msg.channel(i).data());
}
WriteFloatData(data.get(),
@@ -148,7 +149,7 @@
}
rtc::scoped_ptr<const float* []> data(
new const float* [num_input_channels]);
- for (int i = 0; i < num_input_channels; ++i) {
+ for (size_t i = 0; i < num_input_channels; ++i) {
data[i] = reinterpret_cast<const float*>(msg.input_channel(i).data());
}
WriteFloatData(data.get(),
@@ -172,7 +173,7 @@
}
rtc::scoped_ptr<const float* []> data(
new const float* [num_output_channels]);
- for (int i = 0; i < num_output_channels; ++i) {
+ for (size_t i = 0; i < num_output_channels; ++i) {
data[i] =
reinterpret_cast<const float*>(msg.output_channel(i).data());
}
@@ -268,11 +269,14 @@
" Reverse sample rate: %d\n",
reverse_sample_rate);
num_input_channels = msg.num_input_channels();
- fprintf(settings_file, " Input channels: %d\n", num_input_channels);
+ fprintf(settings_file, " Input channels: %" PRIuS "\n",
+ num_input_channels);
num_output_channels = msg.num_output_channels();
- fprintf(settings_file, " Output channels: %d\n", num_output_channels);
+ fprintf(settings_file, " Output channels: %" PRIuS "\n",
+ num_output_channels);
num_reverse_channels = msg.num_reverse_channels();
- fprintf(settings_file, " Reverse channels: %d\n", num_reverse_channels);
+ fprintf(settings_file, " Reverse channels: %" PRIuS "\n",
+ num_reverse_channels);
fprintf(settings_file, "\n");
diff --git a/webrtc/modules/audio_processing/vad/voice_activity_detector.cc b/webrtc/modules/audio_processing/vad/voice_activity_detector.cc
index ef56a35..fc9d103 100644
--- a/webrtc/modules/audio_processing/vad/voice_activity_detector.cc
+++ b/webrtc/modules/audio_processing/vad/voice_activity_detector.cc
@@ -18,7 +18,7 @@
namespace {
const size_t kMaxLength = 320;
-const int kNumChannels = 1;
+const size_t kNumChannels = 1;
const double kDefaultVoiceValue = 1.0;
const double kNeutralProbability = 0.5;
diff --git a/webrtc/modules/include/module_common_types.h b/webrtc/modules/include/module_common_types.h
index 3a63af0..89c5f1b 100644
--- a/webrtc/modules/include/module_common_types.h
+++ b/webrtc/modules/include/module_common_types.h
@@ -28,7 +28,7 @@
uint8_t numEnergy; // number of valid entries in arrOfEnergy
uint8_t arrOfEnergy[kRtpCsrcSize]; // one energy byte (0-9) per channel
bool isCNG; // is this CNG
- uint8_t channel; // number of channels 2 = stereo
+ size_t channel; // number of channels 2 = stereo
};
const int16_t kNoPictureId = -1;
@@ -508,7 +508,7 @@
void UpdateFrame(int id, uint32_t timestamp, const int16_t* data,
size_t samples_per_channel, int sample_rate_hz,
SpeechType speech_type, VADActivity vad_activity,
- int num_channels = 1, uint32_t energy = -1);
+ size_t num_channels = 1, uint32_t energy = -1);
AudioFrame& Append(const AudioFrame& rhs);
@@ -532,7 +532,7 @@
int16_t data_[kMaxDataSizeSamples];
size_t samples_per_channel_;
int sample_rate_hz_;
- int num_channels_;
+ size_t num_channels_;
SpeechType speech_type_;
VADActivity vad_activity_;
// Note that there is no guarantee that |energy_| is correct. Any user of this
@@ -574,7 +574,7 @@
int sample_rate_hz,
SpeechType speech_type,
VADActivity vad_activity,
- int num_channels,
+ size_t num_channels,
uint32_t energy) {
id_ = id;
timestamp_ = timestamp;
@@ -585,7 +585,6 @@
num_channels_ = num_channels;
energy_ = energy;
- assert(num_channels >= 0);
const size_t length = samples_per_channel * num_channels;
assert(length <= kMaxDataSizeSamples);
if (data != NULL) {
@@ -610,7 +609,6 @@
energy_ = src.energy_;
interleaved_ = src.interleaved_;
- assert(num_channels_ >= 0);
const size_t length = samples_per_channel_ * num_channels_;
assert(length <= kMaxDataSizeSamples);
memcpy(data_, src.data_, sizeof(int16_t) * length);
diff --git a/webrtc/modules/media_file/media_file_utility.cc b/webrtc/modules/media_file/media_file_utility.cc
index 1c2f7fd..8a815cc 100644
--- a/webrtc/modules/media_file/media_file_utility.cc
+++ b/webrtc/modules/media_file/media_file_utility.cc
@@ -253,7 +253,7 @@
}
int32_t ModuleFileUtility::InitWavCodec(uint32_t samplesPerSec,
- uint32_t channels,
+ size_t channels,
uint32_t bitsPerSample,
uint32_t formatTag)
{
@@ -663,8 +663,7 @@
return -1;
}
_writing = false;
- uint32_t channels = (codecInst.channels == 0) ?
- 1 : codecInst.channels;
+ size_t channels = (codecInst.channels == 0) ? 1 : codecInst.channels;
if(STR_CASE_CMP(codecInst.plname, "PCMU") == 0)
{
@@ -732,7 +731,7 @@
OutStream& wav,
uint32_t freq,
size_t bytesPerSample,
- uint32_t channels,
+ size_t channels,
uint32_t format,
size_t lengthInBytes)
{
@@ -758,7 +757,7 @@
{
return -1;
}
- uint32_t channels = (codec_info_.channels == 0) ? 1 : codec_info_.channels;
+ size_t channels = (codec_info_.channels == 0) ? 1 : codec_info_.channels;
if(STR_CASE_CMP(codec_info_.plname, "L16") == 0)
{
@@ -861,13 +860,13 @@
}
_writing = true;
_bytesWritten = 1;
- out.Write(&_codecId, 1);
- return 0;
+ out.Write(&_codecId, 1);
+ return 0;
}
int32_t ModuleFileUtility::WritePreEncodedData(
OutStream& out,
- const int8_t* buffer,
+ const int8_t* buffer,
const size_t dataLength)
{
WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
@@ -1010,7 +1009,7 @@
(_codecId == kCodecIlbc30Ms))
{
size_t byteSize = 0;
- if(_codecId == kCodecIlbc30Ms)
+ if(_codecId == kCodecIlbc30Ms)
{
byteSize = 50;
}
diff --git a/webrtc/modules/media_file/media_file_utility.h b/webrtc/modules/media_file/media_file_utility.h
index 46ec340..bc2fa5a 100644
--- a/webrtc/modules/media_file/media_file_utility.h
+++ b/webrtc/modules/media_file/media_file_utility.h
@@ -180,7 +180,7 @@
int32_t InitWavCodec(uint32_t samplesPerSec,
- uint32_t channels,
+ size_t channels,
uint32_t bitsPerSample,
uint32_t formatTag);
@@ -196,7 +196,7 @@
int32_t WriteWavHeader(OutStream& stream,
uint32_t freqInHz,
size_t bytesPerSample,
- uint32_t channels,
+ size_t channels,
uint32_t format,
size_t lengthInBytes);
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h b/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
index 8f58e02..fae8641 100644
--- a/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
@@ -29,7 +29,7 @@
virtual bool PayloadIsCompatible(const RtpUtility::Payload& payload,
const uint32_t frequency,
- const uint8_t channels,
+ const size_t channels,
const uint32_t rate) const = 0;
virtual void UpdatePayloadRate(RtpUtility::Payload* payload,
@@ -39,7 +39,7 @@
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const int8_t payloadType,
const uint32_t frequency,
- const uint8_t channels,
+ const size_t channels,
const uint32_t rate) const = 0;
virtual int GetPayloadTypeFrequency(
@@ -61,7 +61,7 @@
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const int8_t payload_type,
const uint32_t frequency,
- const uint8_t channels,
+ const size_t channels,
const uint32_t rate,
bool* created_new_payload_type);
@@ -71,7 +71,7 @@
int32_t ReceivePayloadType(
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const uint32_t frequency,
- const uint8_t channels,
+ const size_t channels,
const uint32_t rate,
int8_t* payload_type) const;
@@ -173,7 +173,7 @@
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const size_t payload_name_length,
const uint32_t frequency,
- const uint8_t channels,
+ const size_t channels,
const uint32_t rate);
bool IsRtxInternal(const RTPHeader& header) const;
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_receiver.h b/webrtc/modules/rtp_rtcp/include/rtp_receiver.h
index d257a30..0640d5c 100644
--- a/webrtc/modules/rtp_rtcp/include/rtp_receiver.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_receiver.h
@@ -61,7 +61,7 @@
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const int8_t payload_type,
const uint32_t frequency,
- const uint8_t channels,
+ const size_t channels,
const uint32_t rate) = 0;
// De-registers |payload_type| from the payload registry.
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
index d2ac62e..fad97f1 100644
--- a/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -35,7 +35,7 @@
struct AudioPayload {
uint32_t frequency;
- uint8_t channels;
+ size_t channels;
uint32_t rate;
};
@@ -210,7 +210,7 @@
const int8_t payloadType,
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const int frequency,
- const uint8_t channels,
+ const size_t channels,
const uint32_t rate) = 0;
virtual void OnIncomingSSRCChanged(const uint32_t ssrc) = 0;
@@ -333,7 +333,7 @@
int32_t OnInitializeDecoder(const int8_t payloadType,
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const int frequency,
- const uint8_t channels,
+ const size_t channels,
const uint32_t rate) override {
return 0;
}
diff --git a/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h b/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h
index e6a76d0..011829c 100644
--- a/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h
+++ b/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h
@@ -23,7 +23,7 @@
MOCK_CONST_METHOD4(PayloadIsCompatible,
bool(const RtpUtility::Payload& payload,
const uint32_t frequency,
- const uint8_t channels,
+ const size_t channels,
const uint32_t rate));
MOCK_CONST_METHOD2(UpdatePayloadRate,
void(RtpUtility::Payload* payload, const uint32_t rate));
@@ -34,7 +34,7 @@
RtpUtility::Payload*(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const int8_t payloadType,
const uint32_t frequency,
- const uint8_t channels,
+ const size_t channels,
const uint32_t rate));
};
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
index f7cdc4c..ce0bcd7 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
@@ -40,7 +40,7 @@
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const int8_t payload_type,
const uint32_t frequency,
- const uint8_t channels,
+ const size_t channels,
const uint32_t rate,
bool* created_new_payload) {
assert(payload_type >= 0);
@@ -139,7 +139,7 @@
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const size_t payload_name_length,
const uint32_t frequency,
- const uint8_t channels,
+ const size_t channels,
const uint32_t rate) {
RtpUtility::PayloadTypeMap::iterator iterator = payload_type_map_.begin();
for (; iterator != payload_type_map_.end(); ++iterator) {
@@ -171,7 +171,7 @@
int32_t RTPPayloadRegistry::ReceivePayloadType(
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const uint32_t frequency,
- const uint8_t channels,
+ const size_t channels,
const uint32_t rate,
int8_t* payload_type) const {
assert(payload_type);
@@ -388,7 +388,7 @@
bool PayloadIsCompatible(const RtpUtility::Payload& payload,
const uint32_t frequency,
- const uint8_t channels,
+ const size_t channels,
const uint32_t rate) const override {
return
payload.audio &&
@@ -407,7 +407,7 @@
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const int8_t payloadType,
const uint32_t frequency,
- const uint8_t channels,
+ const size_t channels,
const uint32_t rate) const override {
RtpUtility::Payload* payload = new RtpUtility::Payload;
payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
@@ -431,7 +431,7 @@
bool PayloadIsCompatible(const RtpUtility::Payload& payload,
const uint32_t frequency,
- const uint8_t channels,
+ const size_t channels,
const uint32_t rate) const override {
return !payload.audio;
}
@@ -445,7 +445,7 @@
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const int8_t payloadType,
const uint32_t frequency,
- const uint8_t channels,
+ const size_t channels,
const uint32_t rate) const override {
RtpVideoCodecTypes videoType = kRtpVideoGeneric;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
index f19c3da..b73666d 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
@@ -25,7 +25,7 @@
using ::testing::_;
static const char* kTypicalPayloadName = "name";
-static const uint8_t kTypicalChannels = 1;
+static const size_t kTypicalChannels = 1;
static const int kTypicalFrequency = 44000;
static const int kTypicalRate = 32 * 1024;
@@ -90,7 +90,7 @@
TEST_F(RtpPayloadRegistryTest, AudioRedWorkProperly) {
const uint8_t kRedPayloadType = 127;
const int kRedSampleRate = 8000;
- const int kRedChannels = 1;
+ const size_t kRedChannels = 1;
const int kRedBitRate = 0;
// This creates an audio RTP payload strategy.
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
index 93b115a..2e21f23 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
@@ -96,7 +96,7 @@
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const int8_t payload_type,
const uint32_t frequency,
- const uint8_t channels,
+ const size_t channels,
const uint32_t rate) {
CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
@@ -252,7 +252,7 @@
bool new_ssrc = false;
bool re_initialize_decoder = false;
char payload_name[RTP_PAYLOAD_NAME_SIZE];
- uint8_t channels = 1;
+ size_t channels = 1;
uint32_t rate = 0;
{
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
index eaa07d9..5cf94c2 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
@@ -36,7 +36,7 @@
int32_t RegisterReceivePayload(const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const int8_t payload_type,
const uint32_t frequency,
- const uint8_t channels,
+ const size_t channels,
const uint32_t rate) override;
int32_t DeRegisterReceivePayload(const int8_t payload_type) override;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 6ad666b..f4933af 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -296,7 +296,7 @@
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
int8_t payload_number,
uint32_t frequency,
- uint8_t channels,
+ size_t channels,
uint32_t rate) {
assert(payload_name);
CriticalSectionScoped cs(send_critsect_.get());
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index 2aa7f4c..3c62336 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -116,7 +116,7 @@
int32_t RegisterPayload(
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const int8_t payload_type, const uint32_t frequency,
- const uint8_t channels, const uint32_t rate);
+ const size_t channels, const uint32_t rate);
int32_t DeRegisterSendPayload(const int8_t payload_type);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
index d361443..2aa4961 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -66,7 +66,7 @@
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const int8_t payloadType,
const uint32_t frequency,
- const uint8_t channels,
+ const size_t channels,
const uint32_t rate,
RtpUtility::Payload** payload) {
if (RtpUtility::StringCompare(payloadName, "cn", 2)) {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
index a3cee5e..1e96d17 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -29,7 +29,7 @@
int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
int8_t payloadType,
uint32_t frequency,
- uint8_t channels,
+ size_t channels,
uint32_t rate,
RtpUtility::Payload** payload);
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
index 69ed843..634969b 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
@@ -64,7 +64,7 @@
int32_t OnInitializeDecoder(const int8_t payloadType,
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const int frequency,
- const uint8_t channels,
+ const size_t channels,
const uint32_t rate) override {
if (payloadType == 96) {
EXPECT_EQ(test_rate, rate) <<
diff --git a/webrtc/system_wrappers/include/aligned_array.h b/webrtc/system_wrappers/include/aligned_array.h
index e985e88..a2ffe99 100644
--- a/webrtc/system_wrappers/include/aligned_array.h
+++ b/webrtc/system_wrappers/include/aligned_array.h
@@ -20,20 +20,20 @@
// aligned to the given byte alignment.
template<typename T> class AlignedArray {
public:
- AlignedArray(int rows, size_t cols, size_t alignment)
+ AlignedArray(size_t rows, size_t cols, size_t alignment)
: rows_(rows),
cols_(cols) {
RTC_CHECK_GT(alignment, 0u);
head_row_ = static_cast<T**>(AlignedMalloc(rows_ * sizeof(*head_row_),
alignment));
- for (int i = 0; i < rows_; ++i) {
+ for (size_t i = 0; i < rows_; ++i) {
head_row_[i] = static_cast<T*>(AlignedMalloc(cols_ * sizeof(**head_row_),
alignment));
}
}
~AlignedArray() {
- for (int i = 0; i < rows_; ++i) {
+ for (size_t i = 0; i < rows_; ++i) {
AlignedFree(head_row_[i]);
}
AlignedFree(head_row_);
@@ -47,27 +47,27 @@
return head_row_;
}
- T* Row(int row) {
+ T* Row(size_t row) {
RTC_CHECK_LE(row, rows_);
return head_row_[row];
}
- const T* Row(int row) const {
+ const T* Row(size_t row) const {
RTC_CHECK_LE(row, rows_);
return head_row_[row];
}
- T& At(int row, size_t col) {
+ T& At(size_t row, size_t col) {
RTC_CHECK_LE(col, cols_);
return Row(row)[col];
}
- const T& At(int row, size_t col) const {
+ const T& At(size_t row, size_t col) const {
RTC_CHECK_LE(col, cols_);
return Row(row)[col];
}
- int rows() const {
+ size_t rows() const {
return rows_;
}
@@ -76,7 +76,7 @@
}
private:
- int rows_;
+ size_t rows_;
size_t cols_;
T** head_row_;
};
diff --git a/webrtc/system_wrappers/source/aligned_array_unittest.cc b/webrtc/system_wrappers/source/aligned_array_unittest.cc
index eb3ad88..01238f8 100644
--- a/webrtc/system_wrappers/source/aligned_array_unittest.cc
+++ b/webrtc/system_wrappers/source/aligned_array_unittest.cc
@@ -27,7 +27,7 @@
TEST(AlignedArrayTest, CheckAlignment) {
AlignedArray<bool> arr(10, 7, 128);
ASSERT_TRUE(IsAligned(arr.Array(), 128));
- for (int i = 0; i < 10; ++i) {
+ for (size_t i = 0; i < 10; ++i) {
ASSERT_TRUE(IsAligned(arr.Row(i), 128));
ASSERT_EQ(arr.Row(i), arr.Array()[i]);
}
@@ -36,13 +36,13 @@
TEST(AlignedArrayTest, CheckOverlap) {
AlignedArray<size_t> arr(10, 7, 128);
- for (int i = 0; i < 10; ++i) {
+ for (size_t i = 0; i < 10; ++i) {
for (size_t j = 0; j < 7; ++j) {
arr.At(i, j) = 20 * i + j;
}
}
- for (int i = 0; i < 10; ++i) {
+ for (size_t i = 0; i < 10; ++i) {
for (size_t j = 0; j < 7; ++j) {
ASSERT_EQ(arr.At(i, j), 20 * i + j);
ASSERT_EQ(arr.Row(i)[j], 20 * i + j);
@@ -53,7 +53,7 @@
TEST(AlignedArrayTest, CheckRowsCols) {
AlignedArray<bool> arr(10, 7, 128);
- ASSERT_EQ(arr.rows(), 10);
+ ASSERT_EQ(arr.rows(), 10u);
ASSERT_EQ(arr.cols(), 7u);
}
diff --git a/webrtc/tools/agc/activity_metric.cc b/webrtc/tools/agc/activity_metric.cc
index 55280fb..2cb0a1b 100644
--- a/webrtc/tools/agc/activity_metric.cc
+++ b/webrtc/tools/agc/activity_metric.cc
@@ -56,7 +56,7 @@
// silence frame. Otherwise true VAD would drift with respect to the audio.
// We only consider mono inputs.
static void DitherSilence(AudioFrame* frame) {
- ASSERT_EQ(1, frame->num_channels_);
+ ASSERT_EQ(1u, frame->num_channels_);
const double kRmsSilence = 5;
const double sum_squared_silence = kRmsSilence * kRmsSilence *
frame->samples_per_channel_;
diff --git a/webrtc/tools/agc/agc_harness.cc b/webrtc/tools/agc/agc_harness.cc
index 3dcc954..0d35d4b 100644
--- a/webrtc/tools/agc/agc_harness.cc
+++ b/webrtc/tools/agc/agc_harness.cc
@@ -12,6 +12,7 @@
#include "gflags/gflags.h"
#include "webrtc/base/checks.h"
+#include "webrtc/base/format_macros.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/system_wrappers/include/trace.h"
@@ -176,8 +177,8 @@
printf("Codecs:\n");
for (int i = 0; i < codec_->NumOfCodecs(); i++) {
RTC_CHECK_EQ(0, codec_->GetCodec(i, params));
- printf("%d %s/%d/%d\n", params.pltype, params.plname, params.plfreq,
- params.channels);
+ printf("%d %s/%d/%" PRIuS "\n", params.pltype, params.plname,
+ params.plfreq, params.channels);
}
}
diff --git a/webrtc/video/vie_channel.cc b/webrtc/video/vie_channel.cc
index 401cba8..bc23c9d 100644
--- a/webrtc/video/vie_channel.cc
+++ b/webrtc/video/vie_channel.cc
@@ -1184,7 +1184,7 @@
const int8_t payload_type,
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const int frequency,
- const uint8_t channels,
+ const size_t channels,
const uint32_t rate) {
LOG(LS_INFO) << "OnInitializeDecoder " << static_cast<int>(payload_type)
<< " " << payload_name;
diff --git a/webrtc/video/vie_channel.h b/webrtc/video/vie_channel.h
index 25d06b9..4ba3948 100644
--- a/webrtc/video/vie_channel.h
+++ b/webrtc/video/vie_channel.h
@@ -190,7 +190,7 @@
int32_t OnInitializeDecoder(const int8_t payload_type,
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const int frequency,
- const uint8_t channels,
+ const size_t channels,
const uint32_t rate) override;
void OnIncomingSSRCChanged(const uint32_t ssrc) override;
void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) override;
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 64b40a8..b1b55e8 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -419,11 +419,11 @@
int8_t payloadType,
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
int frequency,
- uint8_t channels,
+ size_t channels,
uint32_t rate) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::OnInitializeDecoder(payloadType=%d, "
- "payloadName=%s, frequency=%u, channels=%u, rate=%u)",
+ "payloadName=%s, frequency=%u, channels=%" PRIuS ", rate=%u)",
payloadType, payloadName, frequency, channels, rate);
CodecInst receiveCodec = {0};
@@ -459,7 +459,7 @@
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::OnReceivedPayloadData(payloadSize=%" PRIuS ","
- " payloadType=%u, audioChannel=%u)",
+ " payloadType=%u, audioChannel=%" PRIuS ")",
payloadSize,
rtpHeader->header.payloadType,
rtpHeader->type.Audio.channel);
@@ -1079,8 +1079,8 @@
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId,_channelId),
- "Channel::Init() unable to register %s (%d/%d/%d/%d) "
- "to RTP/RTCP receiver",
+ "Channel::Init() unable to register %s "
+ "(%d/%d/%" PRIuS "/%d) to RTP/RTCP receiver",
codec.plname, codec.pltype, codec.plfreq,
codec.channels, codec.rate);
}
@@ -1088,8 +1088,8 @@
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice,
VoEId(_instanceId,_channelId),
- "Channel::Init() %s (%d/%d/%d/%d) has been added to "
- "the RTP/RTCP receiver",
+ "Channel::Init() %s (%d/%d/%" PRIuS "/%d) has been "
+ "added to the RTP/RTCP receiver",
codec.plname, codec.pltype, codec.plfreq,
codec.channels, codec.rate);
}
@@ -1591,7 +1591,7 @@
CodecInst codec;
int32_t samplingFreqHz(-1);
- const int kMono = 1;
+ const size_t kMono = 1;
if (frequency == kFreq32000Hz)
samplingFreqHz = 32000;
else if (frequency == kFreq16000Hz)
@@ -3355,7 +3355,7 @@
void Channel::Demultiplex(const int16_t* audio_data,
int sample_rate,
size_t number_of_frames,
- int number_of_channels) {
+ size_t number_of_channels) {
CodecInst codec;
GetSendCodec(codec);
@@ -3842,7 +3842,7 @@
sample < _audioFrame.samples_per_channel_;
sample++)
{
- for (int channel = 0;
+ for (size_t channel = 0;
channel < _audioFrame.num_channels_;
channel++)
{
@@ -3976,7 +3976,8 @@
kTraceVoice,
VoEId(_instanceId, _channelId),
"Channel::RegisterReceiveCodecsToRTPModule() unable"
- " to register %s (%d/%d/%d/%d) to RTP/RTCP receiver",
+ " to register %s (%d/%d/%" PRIuS "/%d) to RTP/RTCP "
+ "receiver",
codec.plname, codec.pltype, codec.plfreq,
codec.channels, codec.rate);
}
@@ -3986,7 +3987,7 @@
kTraceVoice,
VoEId(_instanceId, _channelId),
"Channel::RegisterReceiveCodecsToRTPModule() %s "
- "(%d/%d/%d/%d) has been added to the RTP/RTCP "
+ "(%d/%d/%" PRIuS "/%d) has been added to the RTP/RTCP "
"receiver",
codec.plname, codec.pltype, codec.plfreq,
codec.channels, codec.rate);
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 9184b93..d15f9db 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -381,7 +381,7 @@
int32_t OnInitializeDecoder(int8_t payloadType,
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
int frequency,
- uint8_t channels,
+ size_t channels,
uint32_t rate) override;
void OnIncomingSSRCChanged(uint32_t ssrc) override;
void OnIncomingCSRCChanged(uint32_t CSRC, bool added) override;
@@ -451,7 +451,7 @@
void Demultiplex(const int16_t* audio_data,
int sample_rate,
size_t number_of_frames,
- int number_of_channels);
+ size_t number_of_channels);
uint32_t PrepareEncodeAndSend(int mixingFrequency);
uint32_t EncodeAndSend();
diff --git a/webrtc/voice_engine/output_mixer.cc b/webrtc/voice_engine/output_mixer.cc
index d3a4f7f..0dacf35 100644
--- a/webrtc/voice_engine/output_mixer.cc
+++ b/webrtc/voice_engine/output_mixer.cc
@@ -10,6 +10,7 @@
#include "webrtc/voice_engine/output_mixer.h"
+#include "webrtc/base/format_macros.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/utility/include/audio_frame_operations.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
@@ -462,11 +463,12 @@
}
int OutputMixer::GetMixedAudio(int sample_rate_hz,
- int num_channels,
+ size_t num_channels,
AudioFrame* frame) {
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::GetMixedAudio(sample_rate_hz=%d, num_channels=%d)",
- sample_rate_hz, num_channels);
+ WEBRTC_TRACE(
+ kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
+ "OutputMixer::GetMixedAudio(sample_rate_hz=%d, num_channels=%" PRIuS ")",
+ sample_rate_hz, num_channels);
// --- Record playout if enabled
{
diff --git a/webrtc/voice_engine/output_mixer.h b/webrtc/voice_engine/output_mixer.h
index c042a03..91387e6 100644
--- a/webrtc/voice_engine/output_mixer.h
+++ b/webrtc/voice_engine/output_mixer.h
@@ -63,7 +63,7 @@
int32_t SetAnonymousMixabilityStatus(MixerParticipant& participant,
bool mixable);
- int GetMixedAudio(int sample_rate_hz, int num_channels,
+ int GetMixedAudio(int sample_rate_hz, size_t num_channels,
AudioFrame* audioFrame);
// VoEVolumeControl
diff --git a/webrtc/voice_engine/test/auto_test/standard/codec_test.cc b/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
index 5ab6d58..3a3d830 100644
--- a/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
+++ b/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
@@ -50,7 +50,7 @@
TEST_F(CodecTest, PcmuIsDefaultCodecAndHasTheRightValues) {
EXPECT_EQ(0, voe_codec_->GetSendCodec(channel_, codec_instance_));
- EXPECT_EQ(1, codec_instance_.channels);
+ EXPECT_EQ(1u, codec_instance_.channels);
EXPECT_EQ(160, codec_instance_.pacsize);
EXPECT_EQ(8000, codec_instance_.plfreq);
EXPECT_EQ(0, codec_instance_.pltype);
diff --git a/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc b/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc
index 0021a4b..ccfe3c2 100644
--- a/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc
+++ b/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc
@@ -19,6 +19,7 @@
#include "gflags/gflags.h"
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/format_macros.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/call/rtc_event_log.h"
#include "webrtc/engine_configurations.h"
@@ -113,8 +114,8 @@
int res = codec->GetCodec(i, codec_params);
VALIDATE;
SetStereoIfOpus(opus_stereo, &codec_params);
- printf("%2d. %3d %s/%d/%d \n", i, codec_params.pltype, codec_params.plname,
- codec_params.plfreq, codec_params.channels);
+ printf("%2d. %3d %s/%d/%" PRIuS " \n", i, codec_params.pltype,
+ codec_params.plname, codec_params.plfreq, codec_params.channels);
}
}
diff --git a/webrtc/voice_engine/transmit_mixer.cc b/webrtc/voice_engine/transmit_mixer.cc
index 2241f46..1204b04 100644
--- a/webrtc/voice_engine/transmit_mixer.cc
+++ b/webrtc/voice_engine/transmit_mixer.cc
@@ -300,7 +300,8 @@
return 0;
}
-void TransmitMixer::GetSendCodecInfo(int* max_sample_rate, int* max_channels) {
+void TransmitMixer::GetSendCodecInfo(int* max_sample_rate,
+ size_t* max_channels) {
*max_sample_rate = 8000;
*max_channels = 1;
for (ChannelManager::Iterator it(_channelManagerPtr); it.IsValid();
@@ -318,7 +319,7 @@
int32_t
TransmitMixer::PrepareDemux(const void* audioSamples,
size_t nSamples,
- uint8_t nChannels,
+ size_t nChannels,
uint32_t samplesPerSec,
uint16_t totalDelayMS,
int32_t clockDrift,
@@ -327,7 +328,7 @@
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
"TransmitMixer::PrepareDemux(nSamples=%" PRIuS ", "
- "nChannels=%u, samplesPerSec=%u, totalDelayMS=%u, "
+ "nChannels=%" PRIuS ", samplesPerSec=%u, totalDelayMS=%u, "
"clockDrift=%d, currentMicLevel=%u)",
nSamples, nChannels, samplesPerSec, totalDelayMS, clockDrift,
currentMicLevel);
@@ -432,8 +433,8 @@
}
void TransmitMixer::DemuxAndMix(const int voe_channels[],
- int number_of_voe_channels) {
- for (int i = 0; i < number_of_voe_channels; ++i) {
+ size_t number_of_voe_channels) {
+ for (size_t i = 0; i < number_of_voe_channels; ++i) {
voe::ChannelOwner ch = _channelManagerPtr->GetChannel(voe_channels[i]);
voe::Channel* channel_ptr = ch.channel();
if (channel_ptr) {
@@ -465,8 +466,8 @@
}
void TransmitMixer::EncodeAndSend(const int voe_channels[],
- int number_of_voe_channels) {
- for (int i = 0; i < number_of_voe_channels; ++i) {
+ size_t number_of_voe_channels) {
+ for (size_t i = 0; i < number_of_voe_channels; ++i) {
voe::ChannelOwner ch = _channelManagerPtr->GetChannel(voe_channels[i]);
voe::Channel* channel_ptr = ch.channel();
if (channel_ptr && channel_ptr->Sending())
@@ -698,8 +699,7 @@
const uint32_t notificationTime(0); // Not supported in VoE
CodecInst dummyCodec = { 100, "L16", 16000, 320, 1, 320000 };
- if (codecInst != NULL &&
- (codecInst->channels < 0 || codecInst->channels > 2))
+ if (codecInst != NULL && codecInst->channels > 2)
{
_engineStatisticsPtr->SetLastError(
VE_BAD_ARGUMENT, kTraceError,
@@ -1133,10 +1133,10 @@
void TransmitMixer::GenerateAudioFrame(const int16_t* audio,
size_t samples_per_channel,
- int num_channels,
+ size_t num_channels,
int sample_rate_hz) {
int codec_rate;
- int num_codec_channels;
+ size_t num_codec_channels;
GetSendCodecInfo(&codec_rate, &num_codec_channels);
stereo_codec_ = num_codec_channels == 2;
diff --git a/webrtc/voice_engine/transmit_mixer.h b/webrtc/voice_engine/transmit_mixer.h
index 071d91d..0aee106 100644
--- a/webrtc/voice_engine/transmit_mixer.h
+++ b/webrtc/voice_engine/transmit_mixer.h
@@ -52,7 +52,7 @@
int32_t PrepareDemux(const void* audioSamples,
size_t nSamples,
- uint8_t nChannels,
+ size_t nChannels,
uint32_t samplesPerSec,
uint16_t totalDelayMS,
int32_t clockDrift,
@@ -63,12 +63,12 @@
int32_t DemuxAndMix();
// Used by the Chrome to pass the recording data to the specific VoE
// channels for demux.
- void DemuxAndMix(const int voe_channels[], int number_of_voe_channels);
+ void DemuxAndMix(const int voe_channels[], size_t number_of_voe_channels);
int32_t EncodeAndSend();
// Used by the Chrome to pass the recording data to the specific VoE
// channels for encoding and sending to the network.
- void EncodeAndSend(const int voe_channels[], int number_of_voe_channels);
+ void EncodeAndSend(const int voe_channels[], size_t number_of_voe_channels);
// Must be called on the same thread as PrepareDemux().
uint32_t CaptureLevel() const;
@@ -170,11 +170,11 @@
// Gets the maximum sample rate and number of channels over all currently
// sending codecs.
- void GetSendCodecInfo(int* max_sample_rate, int* max_channels);
+ void GetSendCodecInfo(int* max_sample_rate, size_t* max_channels);
void GenerateAudioFrame(const int16_t audioSamples[],
size_t nSamples,
- int nChannels,
+ size_t nChannels,
int samplesPerSec);
int32_t RecordAudioToFile(uint32_t mixingFrequency);
diff --git a/webrtc/voice_engine/utility.cc b/webrtc/voice_engine/utility.cc
index eb442ec..605e553 100644
--- a/webrtc/voice_engine/utility.cc
+++ b/webrtc/voice_engine/utility.cc
@@ -34,12 +34,12 @@
void RemixAndResample(const int16_t* src_data,
size_t samples_per_channel,
- int num_channels,
+ size_t num_channels,
int sample_rate_hz,
PushResampler<int16_t>* resampler,
AudioFrame* dst_frame) {
const int16_t* audio_ptr = src_data;
- int audio_ptr_num_channels = num_channels;
+ size_t audio_ptr_num_channels = num_channels;
int16_t mono_audio[AudioFrame::kMaxDataSizeSamples];
// Downmix before resampling.
@@ -68,8 +68,7 @@
<< ", dst_frame->data_ = " << dst_frame->data_;
assert(false);
}
- dst_frame->samples_per_channel_ =
- static_cast<size_t>(out_length / audio_ptr_num_channels);
+ dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;
// Upmix after resampling.
if (num_channels == 1 && dst_frame->num_channels_ == 2) {
@@ -81,9 +80,9 @@
}
void MixWithSat(int16_t target[],
- int target_channel,
+ size_t target_channel,
const int16_t source[],
- int source_channel,
+ size_t source_channel,
size_t source_len) {
assert(target_channel == 1 || target_channel == 2);
assert(source_channel == 1 || source_channel == 2);
diff --git a/webrtc/voice_engine/utility.h b/webrtc/voice_engine/utility.h
index cc44533..4139f05 100644
--- a/webrtc/voice_engine/utility.h
+++ b/webrtc/voice_engine/utility.h
@@ -40,15 +40,15 @@
// parameters.
void RemixAndResample(const int16_t* src_data,
size_t samples_per_channel,
- int num_channels,
+ size_t num_channels,
int sample_rate_hz,
PushResampler<int16_t>* resampler,
AudioFrame* dst_frame);
void MixWithSat(int16_t target[],
- int target_channel,
+ size_t target_channel,
const int16_t source[],
- int source_channel,
+ size_t source_channel,
size_t source_len);
} // namespace voe
diff --git a/webrtc/voice_engine/voe_base_impl.cc b/webrtc/voice_engine/voe_base_impl.cc
index 0b6c13c..3e5cfbb 100644
--- a/webrtc/voice_engine/voe_base_impl.cc
+++ b/webrtc/voice_engine/voe_base_impl.cc
@@ -82,7 +82,7 @@
int32_t VoEBaseImpl::RecordedDataIsAvailable(const void* audioSamples,
const size_t nSamples,
const size_t nBytesPerSample,
- const uint8_t nChannels,
+ const size_t nChannels,
const uint32_t samplesPerSec,
const uint32_t totalDelayMS,
const int32_t clockDrift,
@@ -97,23 +97,22 @@
int32_t VoEBaseImpl::NeedMorePlayData(const size_t nSamples,
const size_t nBytesPerSample,
- const uint8_t nChannels,
+ const size_t nChannels,
const uint32_t samplesPerSec,
void* audioSamples,
size_t& nSamplesOut,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) {
- GetPlayoutData(static_cast<int>(samplesPerSec), static_cast<int>(nChannels),
- nSamples, true, audioSamples,
- elapsed_time_ms, ntp_time_ms);
+ GetPlayoutData(static_cast<int>(samplesPerSec), nChannels, nSamples, true,
+ audioSamples, elapsed_time_ms, ntp_time_ms);
nSamplesOut = audioFrame_.samples_per_channel_;
return 0;
}
int VoEBaseImpl::OnDataAvailable(const int voe_channels[],
- int number_of_voe_channels,
+ size_t number_of_voe_channels,
const int16_t* audio_data, int sample_rate,
- int number_of_channels,
+ size_t number_of_channels,
size_t number_of_frames,
int audio_delay_milliseconds, int volume,
bool key_pressed, bool need_audio_processing) {
@@ -128,7 +127,7 @@
// No need to go through the APM, demultiplex the data to each VoE channel,
// encode and send to the network.
- for (int i = 0; i < number_of_voe_channels; ++i) {
+ for (size_t i = 0; i < number_of_voe_channels; ++i) {
// TODO(ajm): In the case where multiple channels are using the same codec
// rate, this path needlessly does extra conversions. We should convert once
// and share between channels.
@@ -142,14 +141,14 @@
void VoEBaseImpl::OnData(int voe_channel, const void* audio_data,
int bits_per_sample, int sample_rate,
- int number_of_channels, size_t number_of_frames) {
+ size_t number_of_channels, size_t number_of_frames) {
PushCaptureData(voe_channel, audio_data, bits_per_sample, sample_rate,
number_of_channels, number_of_frames);
}
void VoEBaseImpl::PushCaptureData(int voe_channel, const void* audio_data,
int bits_per_sample, int sample_rate,
- int number_of_channels,
+ size_t number_of_channels,
size_t number_of_frames) {
voe::ChannelOwner ch = shared_->channel_manager().GetChannel(voe_channel);
voe::Channel* channel_ptr = ch.channel();
@@ -165,7 +164,7 @@
void VoEBaseImpl::PullRenderData(int bits_per_sample,
int sample_rate,
- int number_of_channels,
+ size_t number_of_channels,
size_t number_of_frames,
void* audio_data, int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) {
@@ -699,8 +698,8 @@
}
int VoEBaseImpl::ProcessRecordedDataWithAPM(
- const int voe_channels[], int number_of_voe_channels,
- const void* audio_data, uint32_t sample_rate, uint8_t number_of_channels,
+ const int voe_channels[], size_t number_of_voe_channels,
+ const void* audio_data, uint32_t sample_rate, size_t number_of_channels,
size_t number_of_frames, uint32_t audio_delay_milliseconds,
int32_t clock_drift, uint32_t volume, bool key_pressed) {
assert(shared_->transmit_mixer() != nullptr);
@@ -765,7 +764,7 @@
return 0;
}
-void VoEBaseImpl::GetPlayoutData(int sample_rate, int number_of_channels,
+void VoEBaseImpl::GetPlayoutData(int sample_rate, size_t number_of_channels,
size_t number_of_frames, bool feed_data_to_apm,
void* audio_data, int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) {
diff --git a/webrtc/voice_engine/voe_base_impl.h b/webrtc/voice_engine/voe_base_impl.h
index 167aacd..58e0387 100644
--- a/webrtc/voice_engine/voe_base_impl.h
+++ b/webrtc/voice_engine/voe_base_impl.h
@@ -57,7 +57,7 @@
int32_t RecordedDataIsAvailable(const void* audioSamples,
const size_t nSamples,
const size_t nBytesPerSample,
- const uint8_t nChannels,
+ const size_t nChannels,
const uint32_t samplesPerSec,
const uint32_t totalDelayMS,
const int32_t clockDrift,
@@ -66,17 +66,17 @@
uint32_t& newMicLevel) override;
int32_t NeedMorePlayData(const size_t nSamples,
const size_t nBytesPerSample,
- const uint8_t nChannels,
+ const size_t nChannels,
const uint32_t samplesPerSec,
void* audioSamples,
size_t& nSamplesOut,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) override;
int OnDataAvailable(const int voe_channels[],
- int number_of_voe_channels,
+ size_t number_of_voe_channels,
const int16_t* audio_data,
int sample_rate,
- int number_of_channels,
+ size_t number_of_channels,
size_t number_of_frames,
int audio_delay_milliseconds,
int current_volume,
@@ -86,17 +86,17 @@
const void* audio_data,
int bits_per_sample,
int sample_rate,
- int number_of_channels,
+ size_t number_of_channels,
size_t number_of_frames) override;
void PushCaptureData(int voe_channel,
const void* audio_data,
int bits_per_sample,
int sample_rate,
- int number_of_channels,
+ size_t number_of_channels,
size_t number_of_frames) override;
void PullRenderData(int bits_per_sample,
int sample_rate,
- int number_of_channels,
+ size_t number_of_channels,
size_t number_of_frames,
void* audio_data,
int64_t* elapsed_time_ms,
@@ -124,12 +124,12 @@
// It returns new AGC microphone volume or 0 if no volume changes
// should be done.
int ProcessRecordedDataWithAPM(
- const int voe_channels[], int number_of_voe_channels,
- const void* audio_data, uint32_t sample_rate, uint8_t number_of_channels,
+ const int voe_channels[], size_t number_of_voe_channels,
+ const void* audio_data, uint32_t sample_rate, size_t number_of_channels,
size_t number_of_frames, uint32_t audio_delay_milliseconds,
int32_t clock_drift, uint32_t volume, bool key_pressed);
- void GetPlayoutData(int sample_rate, int number_of_channels,
+ void GetPlayoutData(int sample_rate, size_t number_of_channels,
size_t number_of_frames, bool feed_data_to_apm,
void* audio_data, int64_t* elapsed_time_ms,
int64_t* ntp_time_ms);
diff --git a/webrtc/voice_engine/voe_codec_impl.cc b/webrtc/voice_engine/voe_codec_impl.cc
index 162f1c2..6eb11b7 100644
--- a/webrtc/voice_engine/voe_codec_impl.cc
+++ b/webrtc/voice_engine/voe_codec_impl.cc
@@ -10,6 +10,7 @@
#include "webrtc/voice_engine/voe_codec_impl.h"
+#include "webrtc/base/format_macros.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
@@ -64,7 +65,7 @@
"SetSendCodec(channel=%d, codec)", channel);
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
"codec: plname=%s, pacsize=%d, plfreq=%d, pltype=%d, "
- "channels=%d, rate=%d",
+ "channels=%" PRIuS ", rate=%d",
codec.plname, codec.pacsize, codec.plfreq, codec.pltype,
codec.channels, codec.rate);
if (!_shared->statistics().Initialized()) {
@@ -161,7 +162,7 @@
WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
"SetRecPayloadType(channel=%d, codec)", channel);
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "codec: plname=%s, plfreq=%d, pltype=%d, channels=%u, "
+ "codec: plname=%s, plfreq=%d, pltype=%d, channels=%" PRIuS ", "
"pacsize=%d, rate=%d",
codec.plname, codec.plfreq, codec.pltype, codec.channels,
codec.pacsize, codec.rate);
