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Fix trivial lint errors in FileRecorder and FilePlayer

Browse Source 
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}
This commit is contained in:
kwiberg
2016-08-22 08:43:54 -07:00
committed by Commit bot
parent 853ecb21f7
commit 4ec01d9c9d
11 changed files with 110 additions and 88 deletions
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2
webrtc/modules/audio_mixer/audio_mixer.cc
View File

@ -298,7 +298,7 @@ int AudioMixer::StartRecordingPlayout(OutStream* stream,
return -1;
}
if (_outputFileRecorderPtr->StartRecordingAudioFile(*stream, *codecInst,
if (_outputFileRecorderPtr->StartRecordingAudioFile(stream, *codecInst,
notificationTime) != 0) {
_engineStatisticsPtr->SetLastError(
VE_BAD_FILE, kTraceError,

30
webrtc/modules/utility/include/file_player.h
View File

@ -44,7 +44,7 @@ class FilePlayer {
// will be set to the number of samples read (not the number of samples per
// channel).
virtual int Get10msAudioFromFile(int16_t* outBuffer,
size_t& lengthInSamples,
size_t* lengthInSamples,
int frequencyInHz) = 0;
// Register callback for receiving file playing notifications.
@ -61,7 +61,7 @@ class FilePlayer {
const CodecInst* codecInst) = 0;
// Note: codecInst is used for pre-encoded files.
virtual int32_t StartPlayingFile(InStream& sourceStream,
virtual int32_t StartPlayingFile(InStream* sourceStream,
uint32_t startPosition,
float volumeScaling,
uint32_t notification,
@ -72,15 +72,37 @@ class FilePlayer {
virtual bool IsPlayingFile() const = 0;
virtual int32_t GetPlayoutPosition(uint32_t& durationMs) = 0;
virtual int32_t GetPlayoutPosition(uint32_t* durationMs) = 0;
// Set audioCodec to the currently used audio codec.
virtual int32_t AudioCodec(CodecInst& audioCodec) const = 0;
virtual int32_t AudioCodec(CodecInst* audioCodec) const = 0;
virtual int32_t Frequency() const = 0;
// Note: scaleFactor is in the range [0.0 - 2.0]
virtual int32_t SetAudioScaling(float scaleFactor) = 0;
// Deprecated functions. Use the functions above with the same name instead.
int Get10msAudioFromFile(int16_t* outBuffer,
size_t& lengthInSamples,
int frequencyInHz) {
return Get10msAudioFromFile(outBuffer, &lengthInSamples, frequencyInHz);
}
int32_t StartPlayingFile(InStream& sourceStream,
uint32_t startPosition,
float volumeScaling,
uint32_t notification,
uint32_t stopPosition,
const CodecInst* codecInst) {
return StartPlayingFile(&sourceStream, startPosition, volumeScaling,
notification, stopPosition, codecInst);
}
int32_t GetPlayoutPosition(uint32_t& durationMs) {
return GetPlayoutPosition(&durationMs);
}
int32_t AudioCodec(CodecInst& audioCodec) const {
return AudioCodec(&audioCodec);
}
};
} // namespace webrtc
#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_

4
webrtc/modules/utility/include/file_recorder.h
View File

@ -38,7 +38,7 @@ class FileRecorder {
const CodecInst& codecInst,
uint32_t notification) = 0;
virtual int32_t StartRecordingAudioFile(OutStream& destStream,
virtual int32_t StartRecordingAudioFile(OutStream* destStream,
const CodecInst& codecInst,
uint32_t notification) = 0;
@ -48,7 +48,7 @@ class FileRecorder {
// Return true if recording.
virtual bool IsRecording() const = 0;
virtual int32_t codec_info(CodecInst& codecInst) const = 0;
virtual int32_t codec_info(CodecInst* codecInst) const = 0;
// Write frame to file. Frame should contain 10ms of un-ecoded audio data.
virtual int32_t RecordAudioToFile(const AudioFrame& frame) = 0;

14
webrtc/modules/utility/source/coder.cc
View File

@ -54,7 +54,7 @@ int32_t AudioCoder::SetDecodeCodec(const CodecInst& codec_inst) {
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 @@ int32_t AudioCoder::Decode(AudioFrame& decoded_audio,
}
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 @@ int32_t AudioCoder::Encode(const AudioFrame& audio,
return -1;
}
encoded_data_ = encoded_data;
encoded_length_in_bytes = encoded_length_in_bytes_;
*encoded_length_in_bytes = encoded_length_in_bytes_;
return 0;
}

8
webrtc/modules/utility/source/coder.h
View File

@ -24,23 +24,23 @@ class AudioFrame;
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,

63
webrtc/modules/utility/source/file_player.cc
View File

@ -27,31 +27,31 @@ namespace {
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,
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);
virtual int32_t StartPlayingFile(InStream& sourceStream,
const CodecInst* codecInst) override;
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);
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::Frequency() const {
}
}
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 @@ int32_t FilePlayerImpl::Get10msAudioFromFile(int16_t* outBuffer,
// L16 is un-encoded data. Just pull 10 ms.
size_t lengthInBytes = sizeof(unresampledAudioFrame.data_);
if (_fileModule.PlayoutAudioData((int8_t*)unresampledAudioFrame.data_,
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 @@ int32_t FilePlayerImpl::Get10msAudioFromFile(int16_t* outBuffer,
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 @@ int32_t FilePlayerImpl::Get10msAudioFromFile(int16_t* outBuffer,
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 @@ int32_t FilePlayerImpl::StartPlayingFile(const char* fileName,
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 @@ int32_t FilePlayerImpl::StartPlayingFile(InStream& sourceStream,
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 @@ int32_t FilePlayerImpl::StartPlayingFile(InStream& sourceStream,
}
} 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 @@ int32_t FilePlayerImpl::StartPlayingFile(InStream& sourceStream,
}
} 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 @@ bool FilePlayerImpl::IsPlayingFile() const {
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() {

4
webrtc/modules/utility/source/file_player_unittests.cc
View File

@ -61,8 +61,8 @@ class FilePlayerTest : public ::testing::Test {
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,

23
webrtc/modules/utility/source/file_recorder.cc
View File

@ -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 @@ class FileRecorderImpl : public FileRecorder {
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 @@ int32_t FileRecorderImpl::StartRecordingAudioFile(const char* fileName,
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 @@ int32_t FileRecorderImpl::RecordAudioToFile(
// 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 @@ int32_t FileRecorderImpl::RecordAudioToFile(
_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 @@ int32_t FileRecorderImpl::SetUpAudioEncoder() {
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;
}

16
webrtc/voice_engine/channel.cc
View File

@ -1835,7 +1835,7 @@ int Channel::StartPlayingFileLocally(InStream* stream,
const uint32_t notificationTime(0);
if (output_file_player_->StartPlayingFile(*stream, startPosition,
if (output_file_player_->StartPlayingFile(stream, startPosition,
volumeScaling, notificationTime,
stopPosition, codecInst) != 0) {
_engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
@ -2025,9 +2025,9 @@ int Channel::StartPlayingFileAsMicrophone(InStream* stream,
const uint32_t notificationTime(0);
if (input_file_player_->StartPlayingFile(*stream, startPosition,
volumeScaling, notificationTime,
stopPosition, codecInst) != 0) {
if (input_file_player_->StartPlayingFile(stream, startPosition, volumeScaling,
notificationTime, stopPosition,
codecInst) != 0) {
_engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
"StartPlayingFile() failed to start "
"file playout");
@ -2183,7 +2183,7 @@ int Channel::StartRecordingPlayout(OutStream* stream,
return -1;
}
if (output_file_recorder_->StartRecordingAudioFile(*stream, *codecInst,
if (output_file_recorder_->StartRecordingAudioFile(stream, *codecInst,
notificationTime) != 0) {
_engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
"StartRecordingPlayout() failed to "
@ -3275,7 +3275,7 @@ int32_t Channel::MixOrReplaceAudioWithFile(int mixingFrequency) {
return -1;
}
if (input_file_player_->Get10msAudioFromFile(fileBuffer.get(), fileSamples,
if (input_file_player_->Get10msAudioFromFile(fileBuffer.get(), &fileSamples,
mixingFrequency) == -1) {
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::MixOrReplaceAudioWithFile() file mixing "
@ -3323,8 +3323,8 @@ int32_t Channel::MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency) {
}
// We should get the frequency we ask for.
if (output_file_player_->Get10msAudioFromFile(fileBuffer.get(), fileSamples,
mixingFrequency) == -1) {
if (output_file_player_->Get10msAudioFromFile(
fileBuffer.get(), &fileSamples, mixingFrequency) == -1) {
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::MixAudioWithFile() file mixing failed");
return -1;

2
webrtc/voice_engine/output_mixer.cc
View File

@ -377,7 +377,7 @@ int OutputMixer::StartRecordingPlayout(OutStream* stream,
return -1;
}
if (output_file_recorder_->StartRecordingAudioFile(*stream, *codecInst,
if (output_file_recorder_->StartRecordingAudioFile(stream, *codecInst,
notificationTime) != 0) {
_engineStatisticsPtr->SetLastError(
VE_BAD_FILE, kTraceError,

12
webrtc/voice_engine/transmit_mixer.cc
View File

@ -568,9 +568,9 @@ int TransmitMixer::StartPlayingFileAsMicrophone(InStream* stream,
const uint32_t notificationTime(0);
if (file_player_->StartPlayingFile(
(InStream&)*stream, startPosition, volumeScaling, notificationTime,
stopPosition, (const CodecInst*)codecInst) != 0) {
if (file_player_->StartPlayingFile(stream, startPosition, volumeScaling,
notificationTime, stopPosition,
(const CodecInst*)codecInst) != 0) {
_engineStatisticsPtr->SetLastError(
VE_BAD_FILE, kTraceError,
"StartPlayingFile() failed to start file playout");
@ -743,7 +743,7 @@ int TransmitMixer::StartRecordingMicrophone(OutStream* stream,
return -1;
}
if (file_recorder_->StartRecordingAudioFile(*stream, *codecInst,
if (file_recorder_->StartRecordingAudioFile(stream, *codecInst,
notificationTime) != 0) {
_engineStatisticsPtr->SetLastError(
VE_BAD_FILE, kTraceError,
@ -912,7 +912,7 @@ int TransmitMixer::StartRecordingCall(OutStream* stream,
return -1;
}
if (file_call_recorder_->StartRecordingAudioFile(*stream, *codecInst,
if (file_call_recorder_->StartRecordingAudioFile(stream, *codecInst,
notificationTime) != 0) {
_engineStatisticsPtr->SetLastError(
VE_BAD_FILE, kTraceError,
@ -1096,7 +1096,7 @@ int32_t TransmitMixer::MixOrReplaceAudioWithFile(
return -1;
}
if (file_player_->Get10msAudioFromFile(fileBuffer.get(), fileSamples,
if (file_player_->Get10msAudioFromFile(fileBuffer.get(), &fileSamples,
mixingFrequency) == -1) {
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
"TransmitMixer::MixOrReplaceAudioWithFile() file"