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-rw-r--r--webrtc/modules/media_file/BUILD.gn12
-rw-r--r--webrtc/modules/media_file/OWNERS15
-rw-r--r--webrtc/modules/media_file/media_file.gypi12
-rw-r--r--webrtc/modules/media_file/media_file.h (renamed from webrtc/modules/media_file/interface/media_file.h)12
-rw-r--r--webrtc/modules/media_file/media_file_defines.h (renamed from webrtc/modules/media_file/interface/media_file_defines.h)8
-rw-r--r--webrtc/modules/media_file/media_file_impl.cc (renamed from webrtc/modules/media_file/source/media_file_impl.cc)2
-rw-r--r--webrtc/modules/media_file/media_file_impl.h (renamed from webrtc/modules/media_file/source/media_file_impl.h)14
-rw-r--r--webrtc/modules/media_file/media_file_unittest.cc (renamed from webrtc/modules/media_file/source/media_file_unittest.cc)20
-rw-r--r--webrtc/modules/media_file/media_file_utility.cc (renamed from webrtc/modules/media_file/source/media_file_utility.cc)351
-rw-r--r--webrtc/modules/media_file/media_file_utility.h (renamed from webrtc/modules/media_file/source/media_file_utility.h)32
-rw-r--r--webrtc/modules/media_file/source/OWNERS5
11 files changed, 198 insertions, 285 deletions
diff --git a/webrtc/modules/media_file/BUILD.gn b/webrtc/modules/media_file/BUILD.gn
index 05cfb4e555..2a4be728f3 100644
--- a/webrtc/modules/media_file/BUILD.gn
+++ b/webrtc/modules/media_file/BUILD.gn
@@ -14,12 +14,12 @@ config("media_file_config") {
source_set("media_file") {
sources = [
- "interface/media_file.h",
- "interface/media_file_defines.h",
- "source/media_file_impl.cc",
- "source/media_file_impl.h",
- "source/media_file_utility.cc",
- "source/media_file_utility.h",
+ "media_file.h",
+ "media_file_defines.h",
+ "media_file_impl.cc",
+ "media_file_impl.h",
+ "media_file_utility.cc",
+ "media_file_utility.h",
]
if (is_win) {
diff --git a/webrtc/modules/media_file/OWNERS b/webrtc/modules/media_file/OWNERS
index beb9729e04..f6467a4161 100644
--- a/webrtc/modules/media_file/OWNERS
+++ b/webrtc/modules/media_file/OWNERS
@@ -1,5 +1,10 @@
-mflodman@webrtc.org
-perkj@webrtc.org
-niklas.enbom@webrtc.org
-
-per-file BUILD.gn=kjellander@webrtc.org
+mflodman@webrtc.org
+perkj@webrtc.org
+niklas.enbom@webrtc.org
+
+# These are for the common case of adding or renaming files. If you're doing
+# structural changes, please get a review from a reviewer in this file.
+per-file *.gyp=*
+per-file *.gypi=*
+
+per-file BUILD.gn=kjellander@webrtc.org
diff --git a/webrtc/modules/media_file/media_file.gypi b/webrtc/modules/media_file/media_file.gypi
index 4ec80c3c52..94a99a22f1 100644
--- a/webrtc/modules/media_file/media_file.gypi
+++ b/webrtc/modules/media_file/media_file.gypi
@@ -17,12 +17,12 @@
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
],
'sources': [
- 'interface/media_file.h',
- 'interface/media_file_defines.h',
- 'source/media_file_impl.cc',
- 'source/media_file_impl.h',
- 'source/media_file_utility.cc',
- 'source/media_file_utility.h',
+ 'media_file.h',
+ 'media_file_defines.h',
+ 'media_file_impl.cc',
+ 'media_file_impl.h',
+ 'media_file_utility.cc',
+ 'media_file_utility.h',
], # source
# TODO(jschuh): Bug 1348: fix size_t to int truncations.
'msvs_disabled_warnings': [ 4267, ],
diff --git a/webrtc/modules/media_file/interface/media_file.h b/webrtc/modules/media_file/media_file.h
index 5b09ad4383..f6924d6bb0 100644
--- a/webrtc/modules/media_file/interface/media_file.h
+++ b/webrtc/modules/media_file/media_file.h
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
-#define WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
+#ifndef WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_H_
+#define WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_H_
#include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/media_file/interface/media_file_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/media_file/media_file_defines.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -177,4 +177,4 @@ protected:
virtual ~MediaFile() {}
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
+#endif // WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_H_
diff --git a/webrtc/modules/media_file/interface/media_file_defines.h b/webrtc/modules/media_file/media_file_defines.h
index ded71a8ca7..a021a148a5 100644
--- a/webrtc/modules/media_file/interface/media_file_defines.h
+++ b/webrtc/modules/media_file/media_file_defines.h
@@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_
-#define WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_
+#ifndef WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_DEFINES_H_
+#define WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_DEFINES_H_
#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -48,4 +48,4 @@ protected:
FileCallback() {}
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_
+#endif // WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_DEFINES_H_
diff --git a/webrtc/modules/media_file/source/media_file_impl.cc b/webrtc/modules/media_file/media_file_impl.cc
index 50175b86d5..abc7b9d9e0 100644
--- a/webrtc/modules/media_file/source/media_file_impl.cc
+++ b/webrtc/modules/media_file/media_file_impl.cc
@@ -11,7 +11,7 @@
#include <assert.h>
#include "webrtc/base/format_macros.h"
-#include "webrtc/modules/media_file/source/media_file_impl.h"
+#include "webrtc/modules/media_file/media_file_impl.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/file_wrapper.h"
#include "webrtc/system_wrappers/include/tick_util.h"
diff --git a/webrtc/modules/media_file/source/media_file_impl.h b/webrtc/modules/media_file/media_file_impl.h
index cdb54d880d..c23f514c75 100644
--- a/webrtc/modules/media_file/source/media_file_impl.h
+++ b/webrtc/modules/media_file/media_file_impl.h
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_IMPL_H_
-#define WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_IMPL_H_
+#ifndef WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_IMPL_H_
+#define WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_IMPL_H_
#include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/media_file/interface/media_file.h"
-#include "webrtc/modules/media_file/interface/media_file_defines.h"
-#include "webrtc/modules/media_file/source/media_file_utility.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/media_file/media_file.h"
+#include "webrtc/modules/media_file/media_file_defines.h"
+#include "webrtc/modules/media_file/media_file_utility.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
namespace webrtc {
@@ -145,4 +145,4 @@ private:
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_IMPL_H_
+#endif // WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_IMPL_H_
diff --git a/webrtc/modules/media_file/source/media_file_unittest.cc b/webrtc/modules/media_file/media_file_unittest.cc
index 370d13228a..6541a8fb7c 100644
--- a/webrtc/modules/media_file/source/media_file_unittest.cc
+++ b/webrtc/modules/media_file/media_file_unittest.cc
@@ -9,10 +9,9 @@
*/
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/media_file/interface/media_file.h"
+#include "webrtc/modules/media_file/media_file.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
class MediaFileTest : public testing::Test {
protected:
@@ -28,8 +27,14 @@ class MediaFileTest : public testing::Test {
webrtc::MediaFile* media_file_;
};
-TEST_F(MediaFileTest, DISABLED_ON_IOS(
- DISABLED_ON_ANDROID(StartPlayingAudioFileWithoutError))) {
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
+#define MAYBE_StartPlayingAudioFileWithoutError \
+ DISABLED_StartPlayingAudioFileWithoutError
+#else
+#define MAYBE_StartPlayingAudioFileWithoutError \
+ StartPlayingAudioFileWithoutError
+#endif
+TEST_F(MediaFileTest, MAYBE_StartPlayingAudioFileWithoutError) {
// TODO(leozwang): Use hard coded filename here, we want to
// loop through all audio files in future
const std::string audio_file = webrtc::test::ProjectRootPath() +
@@ -47,7 +52,12 @@ TEST_F(MediaFileTest, DISABLED_ON_IOS(
ASSERT_EQ(0, media_file_->StopPlaying());
}
-TEST_F(MediaFileTest, DISABLED_ON_IOS(WriteWavFile)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_WriteWavFile DISABLED_WriteWavFile
+#else
+#define MAYBE_WriteWavFile WriteWavFile
+#endif
+TEST_F(MediaFileTest, MAYBE_WriteWavFile) {
// Write file.
static const size_t kHeaderSize = 44;
static const size_t kPayloadSize = 320;
diff --git a/webrtc/modules/media_file/source/media_file_utility.cc b/webrtc/modules/media_file/media_file_utility.cc
index 61ae442d0e..8a815cc25d 100644
--- a/webrtc/modules/media_file/source/media_file_utility.cc
+++ b/webrtc/modules/media_file/media_file_utility.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/media_file/source/media_file_utility.h"
+#include "webrtc/modules/media_file/media_file_utility.h"
#include <assert.h>
#include <sys/stat.h>
@@ -19,7 +19,7 @@
#include "webrtc/common_audio/wav_header.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/file_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
@@ -38,8 +38,8 @@ struct WAVE_RIFF_header
// the chunk size (16, 18 or 40 byte)
struct WAVE_CHUNK_header
{
- int8_t fmt_ckID[4];
- int32_t fmt_ckSize;
+ int8_t fmt_ckID[4];
+ uint32_t fmt_ckSize;
};
} // unnamed namespace
@@ -79,15 +79,15 @@ int32_t ModuleFileUtility::ReadWavHeader(InStream& wav)
// TODO (hellner): tmpStr and tmpStr2 seems unnecessary here.
char tmpStr[6] = "FOUR";
unsigned char tmpStr2[4];
- int32_t i, len;
+ size_t i;
bool dataFound = false;
bool fmtFound = false;
int8_t dummyRead;
_dataSize = 0;
- len = wav.Read(&RIFFheaderObj, sizeof(WAVE_RIFF_header));
- if(len != sizeof(WAVE_RIFF_header))
+ int len = wav.Read(&RIFFheaderObj, sizeof(WAVE_RIFF_header));
+ if (len != static_cast<int>(sizeof(WAVE_RIFF_header)))
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"Not a wave file (too short)");
@@ -123,14 +123,13 @@ int32_t ModuleFileUtility::ReadWavHeader(InStream& wav)
// in a subroutine.
memcpy(tmpStr2, &CHUNKheaderObj.fmt_ckSize, 4);
CHUNKheaderObj.fmt_ckSize =
- (int32_t) ((uint32_t) tmpStr2[0] +
- (((uint32_t)tmpStr2[1])<<8) +
- (((uint32_t)tmpStr2[2])<<16) +
- (((uint32_t)tmpStr2[3])<<24));
+ (uint32_t)tmpStr2[0] + (((uint32_t)tmpStr2[1]) << 8) +
+ (((uint32_t)tmpStr2[2]) << 16) + (((uint32_t)tmpStr2[3]) << 24);
memcpy(tmpStr, CHUNKheaderObj.fmt_ckID, 4);
- while ((len == sizeof(WAVE_CHUNK_header)) && (!fmtFound || !dataFound))
+ while ((len == static_cast<int>(sizeof(WAVE_CHUNK_header))) &&
+ (!fmtFound || !dataFound))
{
if(strcmp(tmpStr, "fmt ") == 0)
{
@@ -164,9 +163,14 @@ int32_t ModuleFileUtility::ReadWavHeader(InStream& wav)
(int16_t) ((uint32_t)tmpStr2[0] +
(((uint32_t)tmpStr2[1])<<8));
+ if (CHUNKheaderObj.fmt_ckSize < sizeof(WAVE_FMTINFO_header))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Chunk size is too small");
+ return -1;
+ }
for (i = 0;
- i < (CHUNKheaderObj.fmt_ckSize -
- (int32_t)sizeof(WAVE_FMTINFO_header));
+ i < CHUNKheaderObj.fmt_ckSize - sizeof(WAVE_FMTINFO_header);
i++)
{
len = wav.Read(&dummyRead, 1);
@@ -187,7 +191,7 @@ int32_t ModuleFileUtility::ReadWavHeader(InStream& wav)
}
else
{
- for (i = 0; i < (CHUNKheaderObj.fmt_ckSize); i++)
+ for (i = 0; i < CHUNKheaderObj.fmt_ckSize; i++)
{
len = wav.Read(&dummyRead, 1);
if(len != 1)
@@ -203,10 +207,8 @@ int32_t ModuleFileUtility::ReadWavHeader(InStream& wav)
memcpy(tmpStr2, &CHUNKheaderObj.fmt_ckSize, 4);
CHUNKheaderObj.fmt_ckSize =
- (int32_t) ((uint32_t)tmpStr2[0] +
- (((uint32_t)tmpStr2[1])<<8) +
- (((uint32_t)tmpStr2[2])<<16) +
- (((uint32_t)tmpStr2[3])<<24));
+ (uint32_t)tmpStr2[0] + (((uint32_t)tmpStr2[1]) << 8) +
+ (((uint32_t)tmpStr2[2]) << 16) + (((uint32_t)tmpStr2[3]) << 24);
memcpy(tmpStr, CHUNKheaderObj.fmt_ckID, 4);
}
@@ -241,35 +243,17 @@ int32_t ModuleFileUtility::ReadWavHeader(InStream& wav)
}
// Calculate the number of bytes that 10 ms of audio data correspond to.
- if(_wavFormatObj.formatTag == kWavFormatPcm)
- {
- // TODO (hellner): integer division for 22050 and 11025 would yield
- // the same result as the else statement. Remove those
- // special cases?
- if(_wavFormatObj.nSamplesPerSec == 44100)
- {
- _readSizeBytes = 440 * _wavFormatObj.nChannels *
- (_wavFormatObj.nBitsPerSample / 8);
- } else if(_wavFormatObj.nSamplesPerSec == 22050) {
- _readSizeBytes = 220 * _wavFormatObj.nChannels *
- (_wavFormatObj.nBitsPerSample / 8);
- } else if(_wavFormatObj.nSamplesPerSec == 11025) {
- _readSizeBytes = 110 * _wavFormatObj.nChannels *
- (_wavFormatObj.nBitsPerSample / 8);
- } else {
- _readSizeBytes = (_wavFormatObj.nSamplesPerSec/100) *
- _wavFormatObj.nChannels * (_wavFormatObj.nBitsPerSample / 8);
- }
-
- } else {
- _readSizeBytes = (_wavFormatObj.nSamplesPerSec/100) *
- _wavFormatObj.nChannels * (_wavFormatObj.nBitsPerSample / 8);
- }
+ size_t samples_per_10ms =
+ ((_wavFormatObj.formatTag == kWavFormatPcm) &&
+ (_wavFormatObj.nSamplesPerSec == 44100)) ?
+ 440 : static_cast<size_t>(_wavFormatObj.nSamplesPerSec / 100);
+ _readSizeBytes = samples_per_10ms * _wavFormatObj.nChannels *
+ (_wavFormatObj.nBitsPerSample / 8);
return 0;
}
int32_t ModuleFileUtility::InitWavCodec(uint32_t samplesPerSec,
- uint32_t channels,
+ size_t channels,
uint32_t bitsPerSample,
uint32_t formatTag)
{
@@ -376,15 +360,15 @@ int32_t ModuleFileUtility::InitWavReading(InStream& wav,
if(start > 0)
{
uint8_t dummy[WAV_MAX_BUFFER_SIZE];
- int32_t readLength;
+ int readLength;
if(_readSizeBytes <= WAV_MAX_BUFFER_SIZE)
{
while (_playoutPositionMs < start)
{
readLength = wav.Read(dummy, _readSizeBytes);
- if(readLength == _readSizeBytes)
+ if(readLength == static_cast<int>(_readSizeBytes))
{
- _readPos += readLength;
+ _readPos += _readSizeBytes;
_playoutPositionMs += 10;
}
else // Must have reached EOF before start position!
@@ -406,7 +390,7 @@ int32_t ModuleFileUtility::InitWavReading(InStream& wav,
{
return -1;
}
- _bytesPerSample = _wavFormatObj.nBitsPerSample / 8;
+ _bytesPerSample = static_cast<size_t>(_wavFormatObj.nBitsPerSample / 8);
_startPointInMs = start;
@@ -431,9 +415,9 @@ int32_t ModuleFileUtility::ReadWavDataAsMono(
bufferSize);
// The number of bytes that should be read from file.
- const uint32_t totalBytesNeeded = _readSizeBytes;
+ const size_t totalBytesNeeded = _readSizeBytes;
// The number of bytes that will be written to outData.
- const uint32_t bytesRequested = (codec_info_.channels == 2) ?
+ const size_t bytesRequested = (codec_info_.channels == 2) ?
totalBytesNeeded >> 1 : totalBytesNeeded;
if(bufferSize < bytesRequested)
{
@@ -472,7 +456,7 @@ int32_t ModuleFileUtility::ReadWavDataAsMono(
// Output data is should be mono.
if(codec_info_.channels == 2)
{
- for (uint32_t i = 0; i < bytesRequested / _bytesPerSample; i++)
+ for (size_t i = 0; i < bytesRequested / _bytesPerSample; i++)
{
// Sample value is the average of left and right buffer rounded to
// closest integer value. Note samples can be either 1 or 2 byte.
@@ -490,7 +474,7 @@ int32_t ModuleFileUtility::ReadWavDataAsMono(
}
memcpy(outData, _tempData, bytesRequested);
}
- return bytesRequested;
+ return static_cast<int32_t>(bytesRequested);
}
int32_t ModuleFileUtility::ReadWavDataAsStereo(
@@ -534,10 +518,10 @@ int32_t ModuleFileUtility::ReadWavDataAsStereo(
}
// The number of bytes that should be read from file.
- const uint32_t totalBytesNeeded = _readSizeBytes;
+ const size_t totalBytesNeeded = _readSizeBytes;
// The number of bytes that will be written to the left and the right
// buffers.
- const uint32_t bytesRequested = totalBytesNeeded >> 1;
+ const size_t bytesRequested = totalBytesNeeded >> 1;
if(bufferSize < bytesRequested)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
@@ -558,7 +542,7 @@ int32_t ModuleFileUtility::ReadWavDataAsStereo(
// either 1 or 2 bytes
if(_bytesPerSample == 1)
{
- for (uint32_t i = 0; i < bytesRequested; i++)
+ for (size_t i = 0; i < bytesRequested; i++)
{
outDataLeft[i] = _tempData[2 * i];
outDataRight[i] = _tempData[(2 * i) + 1];
@@ -572,35 +556,29 @@ int32_t ModuleFileUtility::ReadWavDataAsStereo(
outDataRight);
// Bytes requested to samples requested.
- uint32_t sampleCount = bytesRequested >> 1;
- for (uint32_t i = 0; i < sampleCount; i++)
+ size_t sampleCount = bytesRequested >> 1;
+ for (size_t i = 0; i < sampleCount; i++)
{
outLeft[i] = sampleData[2 * i];
outRight[i] = sampleData[(2 * i) + 1];
}
} else {
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "ReadWavStereoData: unsupported sample size %d!",
+ "ReadWavStereoData: unsupported sample size %" PRIuS "!",
_bytesPerSample);
assert(false);
return -1;
}
- return bytesRequested;
+ return static_cast<int32_t>(bytesRequested);
}
-int32_t ModuleFileUtility::ReadWavData(
- InStream& wav,
- uint8_t* buffer,
- const uint32_t dataLengthInBytes)
+int32_t ModuleFileUtility::ReadWavData(InStream& wav,
+ uint8_t* buffer,
+ size_t dataLengthInBytes)
{
- WEBRTC_TRACE(
- kTraceStream,
- kTraceFile,
- _id,
- "ModuleFileUtility::ReadWavData(wav= 0x%x, buffer= 0x%x, dataLen= %ld)",
- &wav,
- buffer,
- dataLengthInBytes);
+ WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+ "ModuleFileUtility::ReadWavData(wav= 0x%x, buffer= 0x%x, "
+ "dataLen= %" PRIuS ")", &wav, buffer, dataLengthInBytes);
if(buffer == NULL)
@@ -613,7 +591,7 @@ int32_t ModuleFileUtility::ReadWavData(
// Make sure that a read won't return too few samples.
// TODO (hellner): why not read the remaining bytes needed from the start
// of the file?
- if((_dataSize - _readPos) < (int32_t)dataLengthInBytes)
+ if(_dataSize < (_readPos + dataLengthInBytes))
{
// Rewind() being -1 may be due to the file not supposed to be looped.
if(wav.Rewind() == -1)
@@ -685,8 +663,7 @@ int32_t ModuleFileUtility::InitWavWriting(OutStream& wav,
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)
{
@@ -696,7 +673,8 @@ int32_t ModuleFileUtility::InitWavWriting(OutStream& wav,
{
return -1;
}
- }else if(STR_CASE_CMP(codecInst.plname, "PCMA") == 0)
+ }
+ else if(STR_CASE_CMP(codecInst.plname, "PCMA") == 0)
{
_bytesPerSample = 1;
if(WriteWavHeader(wav, 8000, _bytesPerSample, channels, kWavFormatALaw,
@@ -729,15 +707,9 @@ int32_t ModuleFileUtility::WriteWavData(OutStream& out,
const int8_t* buffer,
const size_t dataLength)
{
- WEBRTC_TRACE(
- kTraceStream,
- kTraceFile,
- _id,
- "ModuleFileUtility::WriteWavData(out= 0x%x, buf= 0x%x, dataLen= %" PRIuS
- ")",
- &out,
- buffer,
- dataLength);
+ WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+ "ModuleFileUtility::WriteWavData(out= 0x%x, buf= 0x%x, "
+ "dataLen= %" PRIuS ")", &out, buffer, dataLength);
if(buffer == NULL)
{
@@ -757,19 +729,19 @@ int32_t ModuleFileUtility::WriteWavData(OutStream& out,
int32_t ModuleFileUtility::WriteWavHeader(
OutStream& wav,
- const uint32_t freq,
- const uint32_t bytesPerSample,
- const uint32_t channels,
- const uint32_t format,
- const uint32_t lengthInBytes)
+ uint32_t freq,
+ size_t bytesPerSample,
+ size_t channels,
+ uint32_t format,
+ size_t lengthInBytes)
{
// Frame size in bytes for 10 ms of audio.
// TODO (hellner): 44.1 kHz has 440 samples frame size. Doesn't seem to
// be taken into consideration here!
- const int32_t frameSize = (freq / 100) * channels;
+ const size_t frameSize = (freq / 100) * channels;
// Calculate the number of full frames that the wave file contain.
- const int32_t dataLengthInBytes = frameSize * (lengthInBytes / frameSize);
+ const size_t dataLengthInBytes = frameSize * (lengthInBytes / frameSize);
uint8_t buf[kWavHeaderSize];
webrtc::WriteWavHeader(buf, channels, freq, static_cast<WavFormat>(format),
@@ -785,8 +757,7 @@ int32_t ModuleFileUtility::UpdateWavHeader(OutStream& wav)
{
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)
{
@@ -839,22 +810,17 @@ int32_t ModuleFileUtility::ReadPreEncodedData(
int8_t* outData,
const size_t bufferSize)
{
- WEBRTC_TRACE(
- kTraceStream,
- kTraceFile,
- _id,
- "ModuleFileUtility::ReadPreEncodedData(in= 0x%x, outData= 0x%x, "
- "bufferSize= %" PRIuS ")",
- &in,
- outData,
- bufferSize);
+ WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+ "ModuleFileUtility::ReadPreEncodedData(in= 0x%x, "
+ "outData= 0x%x, bufferSize= %" PRIuS ")", &in, outData,
+ bufferSize);
if(outData == NULL)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id, "output buffer NULL");
}
- uint32_t frameLen;
+ size_t frameLen;
uint8_t buf[64];
// Each frame has a two byte header containing the frame length.
int32_t res = in.Read(buf, 2);
@@ -874,12 +840,9 @@ int32_t ModuleFileUtility::ReadPreEncodedData(
frameLen = buf[0] + buf[1] * 256;
if(bufferSize < frameLen)
{
- WEBRTC_TRACE(
- kTraceError,
- kTraceFile,
- _id,
- "buffer not large enough to read %d bytes of pre-encoded data!",
- frameLen);
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "buffer not large enough to read %" PRIuS " bytes of "
+ "pre-encoded data!", frameLen);
return -1;
}
return in.Read(outData, frameLen);
@@ -897,24 +860,19 @@ int32_t ModuleFileUtility::InitPreEncodedWriting(
}
_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,
- "ModuleFileUtility::WritePreEncodedData(out= 0x%x, inData= 0x%x, "
- "dataLen= %" PRIuS ")",
- &out,
- buffer,
- dataLength);
+ WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+ "ModuleFileUtility::WritePreEncodedData(out= 0x%x, "
+ "inData= 0x%x, dataLen= %" PRIuS ")", &out, buffer,
+ dataLength);
if(buffer == NULL)
{
@@ -945,15 +903,9 @@ int32_t ModuleFileUtility::InitCompressedReading(
const uint32_t start,
const uint32_t stop)
{
- WEBRTC_TRACE(
- kTraceDebug,
- kTraceFile,
- _id,
- "ModuleFileUtility::InitCompressedReading(in= 0x%x, start= %d,\
- stop= %d)",
- &in,
- start,
- stop);
+ WEBRTC_TRACE(kTraceDebug, kTraceFile, _id,
+ "ModuleFileUtility::InitCompressedReading(in= 0x%x, "
+ "start= %d, stop= %d)", &in, start, stop);
#if defined(WEBRTC_CODEC_ILBC)
int16_t read_len = 0;
@@ -976,9 +928,8 @@ int32_t ModuleFileUtility::InitCompressedReading(
if(cnt==64)
{
return -1;
- } else {
- buf[cnt]=0;
}
+ buf[cnt]=0;
#ifdef WEBRTC_CODEC_ILBC
if(!strcmp("#!iLBC20\n", buf))
@@ -996,14 +947,11 @@ int32_t ModuleFileUtility::InitCompressedReading(
while (_playoutPositionMs <= _startPointInMs)
{
read_len = in.Read(buf, 38);
- if(read_len == 38)
- {
- _playoutPositionMs += 20;
- }
- else
+ if(read_len != 38)
{
return -1;
}
+ _playoutPositionMs += 20;
}
}
}
@@ -1023,14 +971,11 @@ int32_t ModuleFileUtility::InitCompressedReading(
while (_playoutPositionMs <= _startPointInMs)
{
read_len = in.Read(buf, 50);
- if(read_len == 50)
- {
- _playoutPositionMs += 20;
- }
- else
+ if(read_len != 50)
{
return -1;
}
+ _playoutPositionMs += 20;
}
}
}
@@ -1047,17 +992,11 @@ int32_t ModuleFileUtility::ReadCompressedData(InStream& in,
int8_t* outData,
size_t bufferSize)
{
- WEBRTC_TRACE(
- kTraceStream,
- kTraceFile,
- _id,
- "ModuleFileUtility::ReadCompressedData(in=0x%x, outData=0x%x, bytes=%"
- PRIuS ")",
- &in,
- outData,
- bufferSize);
+ WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+ "ModuleFileUtility::ReadCompressedData(in=0x%x, outData=0x%x, "
+ "bytes=%" PRIuS ")", &in, outData, bufferSize);
- uint32_t bytesRead = 0;
+ int bytesRead = 0;
if(! _reading)
{
@@ -1069,8 +1008,8 @@ int32_t ModuleFileUtility::ReadCompressedData(InStream& in,
if((_codecId == kCodecIlbc20Ms) ||
(_codecId == kCodecIlbc30Ms))
{
- uint32_t byteSize = 0;
- if(_codecId == kCodecIlbc30Ms)
+ size_t byteSize = 0;
+ if(_codecId == kCodecIlbc30Ms)
{
byteSize = 50;
}
@@ -1081,20 +1020,20 @@ int32_t ModuleFileUtility::ReadCompressedData(InStream& in,
if(bufferSize < byteSize)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "output buffer is too short to read ILBC compressed\
- data.");
+ "output buffer is too short to read ILBC compressed "
+ "data.");
assert(false);
return -1;
}
bytesRead = in.Read(outData, byteSize);
- if(bytesRead != byteSize)
+ if(bytesRead != static_cast<int>(byteSize))
{
if(!in.Rewind())
{
InitCompressedReading(in, _startPointInMs, _stopPointInMs);
bytesRead = in.Read(outData, byteSize);
- if(bytesRead != byteSize)
+ if(bytesRead != static_cast<int>(byteSize))
{
_reading = false;
return -1;
@@ -1136,9 +1075,8 @@ int32_t ModuleFileUtility::InitCompressedWriting(
const CodecInst& codecInst)
{
WEBRTC_TRACE(kTraceDebug, kTraceFile, _id,
- "ModuleFileUtility::InitCompressedWriting(out= 0x%x,\
- codecName= %s)",
- &out, codecInst.plname);
+ "ModuleFileUtility::InitCompressedWriting(out= 0x%x, "
+ "codecName= %s)", &out, codecInst.plname);
_writing = false;
@@ -1177,15 +1115,9 @@ int32_t ModuleFileUtility::WriteCompressedData(
const int8_t* buffer,
const size_t dataLength)
{
- WEBRTC_TRACE(
- kTraceStream,
- kTraceFile,
- _id,
- "ModuleFileUtility::WriteCompressedData(out= 0x%x, buf= 0x%x, "
- "dataLen= %" PRIuS ")",
- &out,
- buffer,
- dataLength);
+ WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+ "ModuleFileUtility::WriteCompressedData(out= 0x%x, buf= 0x%x, "
+ "dataLen= %" PRIuS ")", &out, buffer, dataLength);
if(buffer == NULL)
{
@@ -1204,19 +1136,12 @@ int32_t ModuleFileUtility::InitPCMReading(InStream& pcm,
const uint32_t stop,
uint32_t freq)
{
- WEBRTC_TRACE(
- kTraceInfo,
- kTraceFile,
- _id,
- "ModuleFileUtility::InitPCMReading(pcm= 0x%x, start=%d, stop=%d,\
- freq=%d)",
- &pcm,
- start,
- stop,
- freq);
+ WEBRTC_TRACE(kTraceInfo, kTraceFile, _id,
+ "ModuleFileUtility::InitPCMReading(pcm= 0x%x, start=%d, "
+ "stop=%d, freq=%d)", &pcm, start, stop, freq);
int8_t dummy[320];
- int32_t read_len;
+ int read_len;
_playoutPositionMs = 0;
_startPointInMs = start;
@@ -1261,14 +1186,11 @@ int32_t ModuleFileUtility::InitPCMReading(InStream& pcm,
while (_playoutPositionMs < _startPointInMs)
{
read_len = pcm.Read(dummy, _readSizeBytes);
- if(read_len == _readSizeBytes)
+ if(read_len != static_cast<int>(_readSizeBytes))
{
- _playoutPositionMs += 10;
- }
- else // Must have reached EOF before start position!
- {
- return -1;
+ return -1; // Must have reached EOF before start position!
}
+ _playoutPositionMs += 10;
}
}
_reading = true;
@@ -1279,23 +1201,17 @@ int32_t ModuleFileUtility::ReadPCMData(InStream& pcm,
int8_t* outData,
size_t bufferSize)
{
- WEBRTC_TRACE(
- kTraceStream,
- kTraceFile,
- _id,
- "ModuleFileUtility::ReadPCMData(pcm= 0x%x, outData= 0x%x, bufSize= %"
- PRIuS ")",
- &pcm,
- outData,
- bufferSize);
+ WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+ "ModuleFileUtility::ReadPCMData(pcm= 0x%x, outData= 0x%x, "
+ "bufSize= %" PRIuS ")", &pcm, outData, bufferSize);
if(outData == NULL)
{
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,"buffer NULL");
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id, "buffer NULL");
}
// Readsize for 10ms of audio data (2 bytes per sample).
- uint32_t bytesRequested = 2 * codec_info_.plfreq / 100;
+ size_t bytesRequested = static_cast<size_t>(2 * codec_info_.plfreq / 100);
if(bufferSize < bytesRequested)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
@@ -1304,8 +1220,8 @@ int32_t ModuleFileUtility::ReadPCMData(InStream& pcm,
return -1;
}
- uint32_t bytesRead = pcm.Read(outData, bytesRequested);
- if(bytesRead < bytesRequested)
+ int bytesRead = pcm.Read(outData, bytesRequested);
+ if(bytesRead < static_cast<int>(bytesRequested))
{
if(pcm.Rewind() == -1)
{
@@ -1320,9 +1236,9 @@ int32_t ModuleFileUtility::ReadPCMData(InStream& pcm,
}
else
{
- int32_t rest = bytesRequested - bytesRead;
- int32_t len = pcm.Read(&(outData[bytesRead]), rest);
- if(len == rest)
+ size_t rest = bytesRequested - bytesRead;
+ int len = pcm.Read(&(outData[bytesRead]), rest);
+ if(len == static_cast<int>(rest))
{
bytesRead += len;
}
@@ -1334,7 +1250,7 @@ int32_t ModuleFileUtility::ReadPCMData(InStream& pcm,
if(bytesRead <= 0)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "ReadPCMData: Failed to rewind audio file.");
+ "ReadPCMData: Failed to rewind audio file.");
return -1;
}
}
@@ -1343,7 +1259,7 @@ int32_t ModuleFileUtility::ReadPCMData(InStream& pcm,
if(bytesRead <= 0)
{
WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
- "ReadPCMData: end of file");
+ "ReadPCMData: end of file");
return -1;
}
_playoutPositionMs += 10;
@@ -1414,15 +1330,9 @@ int32_t ModuleFileUtility::WritePCMData(OutStream& out,
const int8_t* buffer,
const size_t dataLength)
{
- WEBRTC_TRACE(
- kTraceStream,
- kTraceFile,
- _id,
- "ModuleFileUtility::WritePCMData(out= 0x%x, buf= 0x%x, dataLen= %" PRIuS
- ")",
- &out,
- buffer,
- dataLength);
+ WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+ "ModuleFileUtility::WritePCMData(out= 0x%x, buf= 0x%x, "
+ "dataLen= %" PRIuS ")", &out, buffer, dataLength);
if(buffer == NULL)
{
@@ -1585,7 +1495,7 @@ int32_t ModuleFileUtility::FileDurationMs(const char* fileName,
case kFileFormatCompressedFile:
{
int32_t cnt = 0;
- int32_t read_len = 0;
+ int read_len = 0;
char buf[64];
do
{
@@ -1642,15 +1552,8 @@ int32_t ModuleFileUtility::FileDurationMs(const char* fileName,
uint32_t ModuleFileUtility::PlayoutPositionMs()
{
WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
- "ModuleFileUtility::PlayoutPosition()");
+ "ModuleFileUtility::PlayoutPosition()");
- if(_reading)
- {
- return _playoutPositionMs;
- }
- else
- {
- return 0;
- }
+ return _reading ? _playoutPositionMs : 0;
}
} // namespace webrtc
diff --git a/webrtc/modules/media_file/source/media_file_utility.h b/webrtc/modules/media_file/media_file_utility.h
index 2823ceca8a..bc2fa5a2f0 100644
--- a/webrtc/modules/media_file/source/media_file_utility.h
+++ b/webrtc/modules/media_file/media_file_utility.h
@@ -9,13 +9,13 @@
*/
// Note: the class cannot be used for reading and writing at the same time.
-#ifndef WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_
-#define WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_
+#ifndef WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_UTILITY_H_
+#define WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_UTILITY_H_
#include <stdio.h>
#include "webrtc/common_types.h"
-#include "webrtc/modules/media_file/interface/media_file_defines.h"
+#include "webrtc/modules/media_file/media_file_defines.h"
namespace webrtc {
class InStream;
@@ -176,11 +176,11 @@ public:
private:
// Biggest WAV frame supported is 10 ms at 48kHz of 2 channel, 16 bit audio.
- enum{WAV_MAX_BUFFER_SIZE = 480*2*2};
+ static const size_t WAV_MAX_BUFFER_SIZE = 480 * 2 * 2;
int32_t InitWavCodec(uint32_t samplesPerSec,
- uint32_t channels,
+ size_t channels,
uint32_t bitsPerSample,
uint32_t formatTag);
@@ -194,16 +194,16 @@ private:
// stereo. format is the encode format (e.g. PCMU, PCMA, PCM etc).
// lengthInBytes is the number of bytes the audio samples are using up.
int32_t WriteWavHeader(OutStream& stream,
- const uint32_t freqInHz,
- const uint32_t bytesPerSample,
- const uint32_t channels,
- const uint32_t format,
- const uint32_t lengthInBytes);
+ uint32_t freqInHz,
+ size_t bytesPerSample,
+ size_t channels,
+ uint32_t format,
+ size_t lengthInBytes);
// Put dataLengthInBytes of audio data from stream into the audioBuffer.
// The return value is the number of bytes written to audioBuffer.
int32_t ReadWavData(InStream& stream, uint8_t* audioBuffer,
- const uint32_t dataLengthInBytes);
+ size_t dataLengthInBytes);
// Update the current audio codec being used for reading or writing
// according to codecInst.
@@ -254,10 +254,10 @@ private:
// TODO (hellner): why store multiple formats. Just store either codec_info_
// or _wavFormatObj and supply conversion functions.
WAVE_FMTINFO_header _wavFormatObj;
- int32_t _dataSize; // Chunk size if reading a WAV file
+ size_t _dataSize; // Chunk size if reading a WAV file
// Number of bytes to read. I.e. frame size in bytes. May be multiple
// chunks if reading WAV.
- int32_t _readSizeBytes;
+ size_t _readSizeBytes;
int32_t _id;
@@ -270,8 +270,8 @@ private:
MediaFileUtility_CodecType _codecId;
// The amount of bytes, on average, used for one audio sample.
- int32_t _bytesPerSample;
- int32_t _readPos;
+ size_t _bytesPerSample;
+ size_t _readPos;
// Only reading or writing can be enabled, not both.
bool _reading;
@@ -281,4 +281,4 @@ private:
uint8_t _tempData[WAV_MAX_BUFFER_SIZE];
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_
+#endif // WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_UTILITY_H_
diff --git a/webrtc/modules/media_file/source/OWNERS b/webrtc/modules/media_file/source/OWNERS
deleted file mode 100644
index 3ee6b4bf5f..0000000000
--- a/webrtc/modules/media_file/source/OWNERS
+++ /dev/null
@@ -1,5 +0,0 @@
-
-# These are for the common case of adding or renaming files. If you're doing
-# structural changes, please get a review from a reviewer in this file.
-per-file *.gyp=*
-per-file *.gypi=*
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-23 09:23:50 +0000