zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Revert of Add aecdump support to audioproc_f. (patchset #8 id:200001 of https://codereview.webrtc.org/1409943002/ )

Browse Source 
This is the second revert. The first attempt in https://codereview.webrtc.org/1423693008/
was missing a subtle curly brace caused by a merge conflict.
I'm going to let this one go through the CQ.

Reason for revert:
This breaks iOS GYP generation as described on http://www.webrtc.org/native-code/ios
I'm going to drive getting the build_with_libjingle=1 setting removed from the bots to match the official instructions.

See https://code.google.com/p/webrtc/issues/detail?id=4653 for more context, as this is exactly what that issue tries to solve.

Original issue's description:
> Add aecdump support to audioproc_f.
>
> Add a new interface to abstract away file operations. This CL temporarily
> removes support for dumping the output of reverse streams. It will be easy to
> restore in the new framework, although we may decide to only allow it with
> the aecdump format.
>
> We also now require the user to specify the output format, rather than
> defaulting to the input format.
>
> TEST=Bit-exact output to the previous audioproc_f version using an input wav
> file, and to the legacy audioproc using an aecdump file.
>
> Committed: https://crrev.com/bdafe31b86e9819b0adb9041f87e6194b7422b08
> Cr-Commit-Position: refs/heads/master@{#10460}

TBR=aluebs@webrtc.org,peah@webrtc.org,andrew@webrtc.org
BUG=

Review URL: https://codereview.webrtc.org/1412963007

Cr-Commit-Position: refs/heads/master@{#10532}
This commit is contained in:
kjellander
2015-11-05 12:33:18 -08:00
committed by Commit bot
parent 006d93d3c6
commit b7a5c16d2c
10 changed files with 162 additions and 465 deletions
Download Patch File Download Diff File Expand all files Collapse all files

32
webrtc/modules/audio_processing/test/test_utils.h
View File

@ -43,35 +43,6 @@ class RawFile final {
RTC_DISALLOW_COPY_AND_ASSIGN(RawFile);
};
// Reads ChannelBuffers from a provided WavReader.
class ChannelBufferWavReader final {
public:
explicit ChannelBufferWavReader(rtc::scoped_ptr<WavReader> file);
// Reads data from the file according to the |buffer| format. Returns false if
// a full buffer can't be read from the file.
bool Read(ChannelBuffer<float>* buffer);
private:
rtc::scoped_ptr<WavReader> file_;
std::vector<float> interleaved_;
RTC_DISALLOW_COPY_AND_ASSIGN(ChannelBufferWavReader);
};
// Writes ChannelBuffers to a provided WavWriter.
class ChannelBufferWavWriter final {
public:
explicit ChannelBufferWavWriter(rtc::scoped_ptr<WavWriter> file);
void Write(const ChannelBuffer<float>& buffer);
private:
rtc::scoped_ptr<WavWriter> file_;
std::vector<float> interleaved_;
RTC_DISALLOW_COPY_AND_ASSIGN(ChannelBufferWavWriter);
};
void WriteIntData(const int16_t* data,
size_t length,
WavWriter* wav_file,
@ -147,9 +118,6 @@ std::vector<T> ParseList(const std::string& to_parse) {
std::vector<Point> ParseArrayGeometry(const std::string& mic_positions,
size_t num_mics);
// Same as above, but without the num_mics check for when it isn't available.
std::vector<Point> ParseArrayGeometry(const std::string& mic_positions);
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_