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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
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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;
}