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Misc. small cleanups.

Browse Source 
* Better param names
* Avoid using negative values for (bogus) placeholder channel counts (mostly in tests).  Since channels will be changing to size_t, negative values will be illegal; it's sufficient to use 0 in these cases.
* Use arraysize()
* Use size_t for counting frames, samples, blocks, buffers, and bytes -- most of these are already size_t in most places, this just fixes some stragglers
* reinterpret_cast<int64_t>(void*) is not necessarily safe; use uintptr_t instead
* Remove unnecessary code, e.g. dead code, needlessly long/repetitive code, or function overrides that exactly match the base definition
* Fix indenting
* Use uint32_t for timestamps (matching how it's already a uint32_t in most places)
* Spelling
* RTC_CHECK_EQ(expected, actual)
* Rewrap
* Use .empty()
* Be more pedantic about matching int/int32_t/
* Remove pointless consts on input parameters to functions
* Add missing sanity checks

All this was found in the course of constructing https://codereview.webrtc.org/1316523002/ , and is being landed separately first.

BUG=none
TEST=none

Review URL: https://codereview.webrtc.org/1534193008

Cr-Commit-Position: refs/heads/master@{#11191}
This commit is contained in:
pkasting
2016-01-08 13:50:27 -08:00
committed by Commit bot
parent 5de688ed34
commit 25702cb162
51 changed files with 445 additions and 608 deletions
Download Patch File Download Diff File Expand all files Collapse all files

7
webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
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@ -83,7 +83,8 @@ size_t AudioEncoderPcm::Max10MsFramesInAPacket() const {
}
int AudioEncoderPcm::GetTargetBitrate() const {
return 8 * BytesPerSample() * SampleRateHz() * NumChannels();
return static_cast<int>(
8 * BytesPerSample() * SampleRateHz() * NumChannels());
}
AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeInternal(
@ -122,7 +123,7 @@ size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
return WebRtcG711_EncodeA(audio, input_len, encoded);
}
int AudioEncoderPcmA::BytesPerSample() const {
size_t AudioEncoderPcmA::BytesPerSample() const {
return 1;
}
@ -135,7 +136,7 @@ size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
return WebRtcG711_EncodeU(audio, input_len, encoded);
}
int AudioEncoderPcmU::BytesPerSample() const {
size_t AudioEncoderPcmU::BytesPerSample() const {
return 1;
}

6
webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
View File

@ -54,7 +54,7 @@ class AudioEncoderPcm : public AudioEncoder {
size_t input_len,
uint8_t* encoded) = 0;
virtual int BytesPerSample() const = 0;
virtual size_t BytesPerSample() const = 0;
private:
const int sample_rate_hz_;
@ -83,7 +83,7 @@ class AudioEncoderPcmA final : public AudioEncoderPcm {
size_t input_len,
uint8_t* encoded) override;
int BytesPerSample() const override;
size_t BytesPerSample() const override;
private:
static const int kSampleRateHz = 8000;
@ -105,7 +105,7 @@ class AudioEncoderPcmU final : public AudioEncoderPcm {
size_t input_len,
uint8_t* encoded) override;
int BytesPerSample() const override;
size_t BytesPerSample() const override;
private:
static const int kSampleRateHz = 8000;

2
webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
View File

@ -92,7 +92,7 @@ float IsacSpeedTest::DecodeABlock(const uint8_t* bit_stream,
value = WebRtcIsacfix_Decode(ISACFIX_main_inst_, bit_stream, encoded_bytes,
out_data, &audio_type);
clocks = clock() - clocks;
EXPECT_EQ(output_length_sample_, value);
EXPECT_EQ(output_length_sample_, static_cast<size_t>(value));
return 1000.0 * clocks / CLOCKS_PER_SEC;
}

15
webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
View File

@ -137,15 +137,14 @@ AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeInternal(
uint8_t* encoded) {
if (input_buffer_.empty())
first_timestamp_in_buffer_ = rtp_timestamp;
RTC_DCHECK_EQ(static_cast<size_t>(SamplesPer10msFrame()), audio.size());
RTC_DCHECK_EQ(SamplesPer10msFrame(), audio.size());
input_buffer_.insert(input_buffer_.end(), audio.cbegin(), audio.cend());
if (input_buffer_.size() <
(static_cast<size_t>(Num10msFramesPerPacket()) * SamplesPer10msFrame())) {
(Num10msFramesPerPacket() * SamplesPer10msFrame())) {
return EncodedInfo();
}
RTC_CHECK_EQ(
input_buffer_.size(),
static_cast<size_t>(Num10msFramesPerPacket()) * SamplesPer10msFrame());
RTC_CHECK_EQ(input_buffer_.size(),
Num10msFramesPerPacket() * SamplesPer10msFrame());
int status = WebRtcOpus_Encode(
inst_, &input_buffer_[0],
rtc::CheckedDivExact(input_buffer_.size(),
@ -214,11 +213,11 @@ void AudioEncoderOpus::SetTargetBitrate(int bits_per_second) {
RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, config_.bitrate_bps));
}
int AudioEncoderOpus::Num10msFramesPerPacket() const {
return rtc::CheckedDivExact(config_.frame_size_ms, 10);
size_t AudioEncoderOpus::Num10msFramesPerPacket() const {
return static_cast<size_t>(rtc::CheckedDivExact(config_.frame_size_ms, 10));
}
int AudioEncoderOpus::SamplesPer10msFrame() const {
size_t AudioEncoderOpus::SamplesPer10msFrame() const {
return rtc::CheckedDivExact(kSampleRateHz, 100) * config_.num_channels;
}

4
webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
View File

@ -85,8 +85,8 @@ class AudioEncoderOpus final : public AudioEncoder {
bool dtx_enabled() const { return config_.dtx_enabled; }
private:
int Num10msFramesPerPacket() const;
int SamplesPer10msFrame() const;
size_t Num10msFramesPerPacket() const;
size_t SamplesPer10msFrame() const;
bool RecreateEncoderInstance(const Config& config);
Config config_;

2
webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
View File

@ -77,7 +77,7 @@ float OpusSpeedTest::DecodeABlock(const uint8_t* bit_stream,
value = WebRtcOpus_Decode(opus_decoder_, bit_stream, encoded_bytes, out_data,
&audio_type);
clocks = clock() - clocks;
EXPECT_EQ(output_length_sample_, value);
EXPECT_EQ(output_length_sample_, static_cast<size_t>(value));
return 1000.0 * clocks / CLOCKS_PER_SEC;
}

2
webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
View File

@ -22,7 +22,7 @@ size_t AudioEncoderPcm16B::EncodeCall(const int16_t* audio,
return WebRtcPcm16b_Encode(audio, input_len, encoded);
}
int AudioEncoderPcm16B::BytesPerSample() const {
size_t AudioEncoderPcm16B::BytesPerSample() const {
return 2;
}

2
webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
View File

@ -37,7 +37,7 @@ class AudioEncoderPcm16B final : public AudioEncoderPcm {
size_t input_len,
uint8_t* encoded) override;
int BytesPerSample() const override;
size_t BytesPerSample() const override;
private:
RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcm16B);

9
webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
View File

@ -23,8 +23,10 @@ AudioCodecSpeedTest::AudioCodecSpeedTest(int block_duration_ms,
: block_duration_ms_(block_duration_ms),
input_sampling_khz_(input_sampling_khz),
output_sampling_khz_(output_sampling_khz),
input_length_sample_(block_duration_ms_ * input_sampling_khz_),
output_length_sample_(block_duration_ms_ * output_sampling_khz_),
input_length_sample_(
static_cast<size_t>(block_duration_ms_ * input_sampling_khz_)),
output_length_sample_(
static_cast<size_t>(block_duration_ms_ * output_sampling_khz_)),
data_pointer_(0),
loop_length_samples_(0),
max_bytes_(0),
@ -65,8 +67,7 @@ void AudioCodecSpeedTest::SetUp() {
memcpy(&in_data_[loop_length_samples_], &in_data_[0],
input_length_sample_ * channels_ * sizeof(int16_t));
max_bytes_ =
static_cast<size_t>(input_length_sample_ * channels_ * sizeof(int16_t));
max_bytes_ = input_length_sample_ * channels_ * sizeof(int16_t);
out_data_.reset(new int16_t[output_length_sample_ * channels_]);
bit_stream_.reset(new uint8_t[max_bytes_]);

4
webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
View File

@ -55,10 +55,10 @@ class AudioCodecSpeedTest : public testing::TestWithParam<coding_param> {
int output_sampling_khz_;
// Number of samples-per-channel in a frame.
int input_length_sample_;
size_t input_length_sample_;
// Expected output number of samples-per-channel in a frame.
int output_length_sample_;
size_t output_length_sample_;
rtc::scoped_ptr<int16_t[]> in_data_;
rtc::scoped_ptr<int16_t[]> out_data_;