Fix trivial lint errors in FileRecorder and FilePlayer
Mostly, it's about replacing mutable reference arguments with pointer
arguments, and replacing C style casts with C++ style casts.
Review-Url: https://codereview.webrtc.org/2056653002
Cr-Commit-Position: refs/heads/master@{#13849}
diff --git a/webrtc/modules/utility/source/coder.cc b/webrtc/modules/utility/source/coder.cc
index f2ae43e..f72d03b 100644
--- a/webrtc/modules/utility/source/coder.cc
+++ b/webrtc/modules/utility/source/coder.cc
@@ -54,7 +54,7 @@
return 0;
}
-int32_t AudioCoder::Decode(AudioFrame& decoded_audio,
+int32_t AudioCoder::Decode(AudioFrame* decoded_audio,
uint32_t samp_freq_hz,
const int8_t* incoming_payload,
size_t payload_length) {
@@ -68,22 +68,22 @@
}
bool muted;
int32_t ret =
- acm_->PlayoutData10Ms((uint16_t)samp_freq_hz, &decoded_audio, &muted);
+ acm_->PlayoutData10Ms((uint16_t)samp_freq_hz, decoded_audio, &muted);
RTC_DCHECK(!muted);
return ret;
}
-int32_t AudioCoder::PlayoutData(AudioFrame& decoded_audio,
- uint16_t& samp_freq_hz) {
+int32_t AudioCoder::PlayoutData(AudioFrame* decoded_audio,
+ uint16_t samp_freq_hz) {
bool muted;
- int32_t ret = acm_->PlayoutData10Ms(samp_freq_hz, &decoded_audio, &muted);
+ int32_t ret = acm_->PlayoutData10Ms(samp_freq_hz, decoded_audio, &muted);
RTC_DCHECK(!muted);
return ret;
}
int32_t AudioCoder::Encode(const AudioFrame& audio,
int8_t* encoded_data,
- size_t& encoded_length_in_bytes) {
+ size_t* encoded_length_in_bytes) {
// Fake a timestamp in case audio doesn't contain a correct timestamp.
// Make a local copy of the audio frame since audio is const
AudioFrame audio_frame;
@@ -98,7 +98,7 @@
return -1;
}
encoded_data_ = encoded_data;
- encoded_length_in_bytes = encoded_length_in_bytes_;
+ *encoded_length_in_bytes = encoded_length_in_bytes_;
return 0;
}
diff --git a/webrtc/modules/utility/source/coder.h b/webrtc/modules/utility/source/coder.h
index 5f44190..4855a00 100644
--- a/webrtc/modules/utility/source/coder.h
+++ b/webrtc/modules/utility/source/coder.h
@@ -24,23 +24,23 @@
class AudioCoder : public AudioPacketizationCallback {
public:
- AudioCoder(uint32_t instance_id);
+ explicit AudioCoder(uint32_t instance_id);
~AudioCoder();
int32_t SetEncodeCodec(const CodecInst& codec_inst);
int32_t SetDecodeCodec(const CodecInst& codec_inst);
- int32_t Decode(AudioFrame& decoded_audio,
+ int32_t Decode(AudioFrame* decoded_audio,
uint32_t samp_freq_hz,
const int8_t* incoming_payload,
size_t payload_length);
- int32_t PlayoutData(AudioFrame& decoded_audio, uint16_t& samp_freq_hz);
+ int32_t PlayoutData(AudioFrame* decoded_audio, uint16_t samp_freq_hz);
int32_t Encode(const AudioFrame& audio,
int8_t* encoded_data,
- size_t& encoded_length_in_bytes);
+ size_t* encoded_length_in_bytes);
protected:
int32_t SendData(FrameType frame_type,
diff --git a/webrtc/modules/utility/source/file_player.cc b/webrtc/modules/utility/source/file_player.cc
index 4b33929..75b7214 100644
--- a/webrtc/modules/utility/source/file_player.cc
+++ b/webrtc/modules/utility/source/file_player.cc
@@ -27,31 +27,31 @@
class FilePlayerImpl : public FilePlayer {
public:
FilePlayerImpl(uint32_t instanceID, FileFormats fileFormat);
- ~FilePlayerImpl();
+ ~FilePlayerImpl() override;
- virtual int Get10msAudioFromFile(int16_t* outBuffer,
- size_t& lengthInSamples,
- int frequencyInHz);
- virtual int32_t RegisterModuleFileCallback(FileCallback* callback);
- virtual int32_t StartPlayingFile(const char* fileName,
- bool loop,
- uint32_t startPosition,
- float volumeScaling,
- uint32_t notification,
- uint32_t stopPosition,
- const CodecInst* codecInst);
- virtual int32_t StartPlayingFile(InStream& sourceStream,
- uint32_t startPosition,
- float volumeScaling,
- uint32_t notification,
- uint32_t stopPosition,
- const CodecInst* codecInst);
- virtual int32_t StopPlayingFile();
- virtual bool IsPlayingFile() const;
- virtual int32_t GetPlayoutPosition(uint32_t& durationMs);
- virtual int32_t AudioCodec(CodecInst& audioCodec) const;
- virtual int32_t Frequency() const;
- virtual int32_t SetAudioScaling(float scaleFactor);
+ int Get10msAudioFromFile(int16_t* outBuffer,
+ size_t* lengthInSamples,
+ int frequencyInHz) override;
+ int32_t RegisterModuleFileCallback(FileCallback* callback) override;
+ int32_t StartPlayingFile(const char* fileName,
+ bool loop,
+ uint32_t startPosition,
+ float volumeScaling,
+ uint32_t notification,
+ uint32_t stopPosition,
+ const CodecInst* codecInst) override;
+ int32_t StartPlayingFile(InStream* sourceStream,
+ uint32_t startPosition,
+ float volumeScaling,
+ uint32_t notification,
+ uint32_t stopPosition,
+ const CodecInst* codecInst) override;
+ int32_t StopPlayingFile() override;
+ bool IsPlayingFile() const override;
+ int32_t GetPlayoutPosition(uint32_t* durationMs) override;
+ int32_t AudioCodec(CodecInst* audioCodec) const override;
+ int32_t Frequency() const override;
+ int32_t SetAudioScaling(float scaleFactor) override;
private:
int32_t SetUpAudioDecoder();
@@ -108,13 +108,13 @@
}
}
-int32_t FilePlayerImpl::AudioCodec(CodecInst& audioCodec) const {
- audioCodec = _codec;
+int32_t FilePlayerImpl::AudioCodec(CodecInst* audioCodec) const {
+ *audioCodec = _codec;
return 0;
}
int32_t FilePlayerImpl::Get10msAudioFromFile(int16_t* outBuffer,
- size_t& lengthInSamples,
+ size_t* lengthInSamples,
int frequencyInHz) {
if (_codec.plfreq == 0) {
LOG(LS_WARNING) << "Get10msAudioFromFile() playing not started!"
@@ -129,13 +129,14 @@
// L16 is un-encoded data. Just pull 10 ms.
size_t lengthInBytes = sizeof(unresampledAudioFrame.data_);
- if (_fileModule.PlayoutAudioData((int8_t*)unresampledAudioFrame.data_,
- lengthInBytes) == -1) {
+ if (_fileModule.PlayoutAudioData(
+ reinterpret_cast<int8_t*>(unresampledAudioFrame.data_),
+ lengthInBytes) == -1) {
// End of file reached.
return -1;
}
if (lengthInBytes == 0) {
- lengthInSamples = 0;
+ *lengthInSamples = 0;
return 0;
}
// One sample is two bytes.
@@ -150,15 +151,15 @@
if (++_numberOf10MsInDecoder >= _numberOf10MsPerFrame) {
_numberOf10MsInDecoder = 0;
size_t bytesFromFile = sizeof(encodedBuffer);
- if (_fileModule.PlayoutAudioData((int8_t*)encodedBuffer, bytesFromFile) ==
- -1) {
+ if (_fileModule.PlayoutAudioData(reinterpret_cast<int8_t*>(encodedBuffer),
+ bytesFromFile) == -1) {
// End of file reached.
return -1;
}
encodedLengthInBytes = bytesFromFile;
}
- if (_audioDecoder.Decode(unresampledAudioFrame, frequencyInHz,
- (int8_t*)encodedBuffer,
+ if (_audioDecoder.Decode(&unresampledAudioFrame, frequencyInHz,
+ reinterpret_cast<int8_t*>(encodedBuffer),
encodedLengthInBytes) == -1) {
return -1;
}
@@ -178,7 +179,7 @@
unresampledAudioFrame.samples_per_channel_, outBuffer,
MAX_AUDIO_BUFFER_IN_SAMPLES, outLen);
- lengthInSamples = outLen;
+ *lengthInSamples = outLen;
if (_scaling != 1.0) {
for (size_t i = 0; i < outLen; i++) {
@@ -270,7 +271,7 @@
return 0;
}
-int32_t FilePlayerImpl::StartPlayingFile(InStream& sourceStream,
+int32_t FilePlayerImpl::StartPlayingFile(InStream* sourceStream,
uint32_t startPosition,
float volumeScaling,
uint32_t notification,
@@ -304,7 +305,7 @@
return -1;
}
if (_fileModule.StartPlayingAudioStream(
- sourceStream, notification, _fileFormat, &codecInstL16,
+ *sourceStream, notification, _fileFormat, &codecInstL16,
startPosition, stopPosition) == -1) {
LOG(LS_ERROR) << "StartPlayingFile() failed to initialize stream "
<< "playout.";
@@ -312,7 +313,7 @@
}
} else if (_fileFormat == kFileFormatPreencodedFile) {
- if (_fileModule.StartPlayingAudioStream(sourceStream, notification,
+ if (_fileModule.StartPlayingAudioStream(*sourceStream, notification,
_fileFormat, codecInst) == -1) {
LOG(LS_ERROR) << "StartPlayingFile() failed to initialize stream "
<< "playout.";
@@ -320,7 +321,7 @@
}
} else {
CodecInst* no_inst = NULL;
- if (_fileModule.StartPlayingAudioStream(sourceStream, notification,
+ if (_fileModule.StartPlayingAudioStream(*sourceStream, notification,
_fileFormat, no_inst, startPosition,
stopPosition) == -1) {
LOG(LS_ERROR) << "StartPlayingFile() failed to initialize stream "
@@ -348,8 +349,8 @@
return _fileModule.IsPlaying();
}
-int32_t FilePlayerImpl::GetPlayoutPosition(uint32_t& durationMs) {
- return _fileModule.PlayoutPositionMs(durationMs);
+int32_t FilePlayerImpl::GetPlayoutPosition(uint32_t* durationMs) {
+ return _fileModule.PlayoutPositionMs(*durationMs);
}
int32_t FilePlayerImpl::SetUpAudioDecoder() {
diff --git a/webrtc/modules/utility/source/file_player_unittests.cc b/webrtc/modules/utility/source/file_player_unittests.cc
index 8c70952..022bec0 100644
--- a/webrtc/modules/utility/source/file_player_unittests.cc
+++ b/webrtc/modules/utility/source/file_player_unittests.cc
@@ -61,8 +61,8 @@
for (int i = 0; i < output_length_ms / 10; ++i) {
int16_t out[10 * kSampleRateHz / 1000] = {0};
size_t num_samples;
- EXPECT_EQ(0,
- player_->Get10msAudioFromFile(out, num_samples, kSampleRateHz));
+ EXPECT_EQ(
+ 0, player_->Get10msAudioFromFile(out, &num_samples, kSampleRateHz));
checksum.Update(out, num_samples * sizeof(out[0]));
if (FLAGS_file_player_output) {
ASSERT_EQ(num_samples,
diff --git a/webrtc/modules/utility/source/file_recorder.cc b/webrtc/modules/utility/source/file_recorder.cc
index 3ba7967..c28c2d1 100644
--- a/webrtc/modules/utility/source/file_recorder.cc
+++ b/webrtc/modules/utility/source/file_recorder.cc
@@ -16,10 +16,8 @@
#include "webrtc/common_audio/resampler/include/resampler.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
-#include "webrtc/engine_configurations.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/media_file/media_file.h"
-#include "webrtc/modules/media_file/media_file.h"
#include "webrtc/modules/media_file/media_file_defines.h"
#include "webrtc/modules/utility/source/coder.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
@@ -48,12 +46,12 @@
int32_t StartRecordingAudioFile(const char* fileName,
const CodecInst& codecInst,
uint32_t notificationTimeMs) override;
- int32_t StartRecordingAudioFile(OutStream& destStream,
+ int32_t StartRecordingAudioFile(OutStream* destStream,
const CodecInst& codecInst,
uint32_t notificationTimeMs) override;
int32_t StopRecording() override;
bool IsRecording() const override;
- int32_t codec_info(CodecInst& codecInst) const override;
+ int32_t codec_info(CodecInst* codecInst) const override;
int32_t RecordAudioToFile(const AudioFrame& frame) override;
private:
@@ -120,12 +118,12 @@
return retVal;
}
-int32_t FileRecorderImpl::StartRecordingAudioFile(OutStream& destStream,
+int32_t FileRecorderImpl::StartRecordingAudioFile(OutStream* destStream,
const CodecInst& codecInst,
uint32_t notificationTimeMs) {
codec_info_ = codecInst;
int32_t retVal = _moduleFile->StartRecordingAudioStream(
- destStream, _fileFormat, codecInst, notificationTimeMs);
+ *destStream, _fileFormat, codecInst, notificationTimeMs);
if (retVal == 0) {
retVal = SetUpAudioEncoder();
@@ -193,13 +191,13 @@
// Encode the audio data before writing to file. Don't encode if the codec
// is PCM.
// NOTE: stereo recording is only supported for WAV files.
- // TODO (hellner): WAV expect PCM in little endian byte order. Not
+ // TODO(hellner): WAV expect PCM in little endian byte order. Not
// "encoding" with PCM coder should be a problem for big endian systems.
size_t encodedLenInBytes = 0;
if (_fileFormat == kFileFormatPreencodedFile ||
STR_CASE_CMP(codec_info_.plname, "L16") != 0) {
- if (_audioEncoder.Encode(*ptrAudioFrame, _audioBuffer, encodedLenInBytes) ==
- -1) {
+ if (_audioEncoder.Encode(*ptrAudioFrame, _audioBuffer,
+ &encodedLenInBytes) == -1) {
LOG(LS_WARNING) << "RecordAudioToFile() codec " << codec_info_.plname
<< " not supported or failed to encode stream.";
return -1;
@@ -212,7 +210,8 @@
_audioResampler.Push(
ptrAudioFrame->data_,
ptrAudioFrame->samples_per_channel_ * ptrAudioFrame->num_channels_,
- (int16_t*)_audioBuffer, MAX_AUDIO_BUFFER_IN_BYTES, outLen);
+ reinterpret_cast<int16_t*>(_audioBuffer), MAX_AUDIO_BUFFER_IN_BYTES,
+ outLen);
encodedLenInBytes = outLen * sizeof(int16_t);
}
@@ -239,11 +238,11 @@
return 0;
}
-int32_t FileRecorderImpl::codec_info(CodecInst& codecInst) const {
+int32_t FileRecorderImpl::codec_info(CodecInst* codecInst) const {
if (codec_info_.plfreq == 0) {
return -1;
}
- codecInst = codec_info_;
+ *codecInst = codec_info_;
return 0;
}