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| author | andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | 2013-04-29 17:27:29 +0000 |
|---|---|---|
| committer | andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | 2013-04-29 17:27:29 +0000 |
| commit | b6fadb16521d2d174c19a6fa881523fced10c6c9 (patch) | |
| tree | 2f6c76c43890f6e90e50e0a1b8b351432051bded /voice_engine/output_mixer.cc | |
| parent | c12e655e176f5a6f7892625d783661634cb7a891 (diff) | |
| download | webrtc-b6fadb16521d2d174c19a6fa881523fced10c6c9.tar.gz | |
Add a wrapper around PushSincResampler and the old Resampler.
The old resampler is used whenever it supports the requested rates. Otherwise
the sinc resampler is enabled.
Integrated with output_mixer in order to test the change through
output_mixer_unittest. The sinc resampler will not yet be used, since we don't
feed VoE with any rates that trigger it.
BUG=webrtc:1395
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1355004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3915 4adac7df-926f-26a2-2b94-8c16560cd09d
Diffstat (limited to 'voice_engine/output_mixer.cc')
| -rw-r--r-- | voice_engine/output_mixer.cc | 22 |
1 files changed, 11 insertions, 11 deletions
diff --git a/voice_engine/output_mixer.cc b/voice_engine/output_mixer.cc index a1245649..a8e41779 100644 --- a/voice_engine/output_mixer.cc +++ b/voice_engine/output_mixer.cc @@ -8,16 +8,16 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "output_mixer.h" +#include "webrtc/voice_engine/output_mixer.h" -#include "audio_processing.h" -#include "audio_frame_operations.h" -#include "critical_section_wrapper.h" -#include "file_wrapper.h" -#include "output_mixer_internal.h" -#include "statistics.h" -#include "trace.h" -#include "voe_external_media.h" +#include "webrtc/modules/audio_processing/include/audio_processing.h" +#include "webrtc/modules/utility/interface/audio_frame_operations.h" +#include "webrtc/system_wrappers/interface/critical_section_wrapper.h" +#include "webrtc/system_wrappers/interface/file_wrapper.h" +#include "webrtc/system_wrappers/interface/trace.h" +#include "webrtc/voice_engine/include/voe_external_media.h" +#include "webrtc/voice_engine/output_mixer_internal.h" +#include "webrtc/voice_engine/statistics.h" namespace webrtc { @@ -528,7 +528,7 @@ int OutputMixer::GetMixedAudio(int sample_rate_hz, frame->sample_rate_hz_ = sample_rate_hz; // TODO(andrew): Ideally the downmixing would occur much earlier, in // AudioCodingModule. - return RemixAndResample(_audioFrame, &_resampler, frame); + return RemixAndResample(_audioFrame, &resampler_, frame); } int32_t @@ -602,7 +602,7 @@ void OutputMixer::APMAnalyzeReverseStream() { AudioFrame frame; frame.num_channels_ = 1; frame.sample_rate_hz_ = _audioProcessingModulePtr->sample_rate_hz(); - if (RemixAndResample(_audioFrame, &_apmResampler, &frame) == -1) + if (RemixAndResample(_audioFrame, &audioproc_resampler_, &frame) == -1) return; if (_audioProcessingModulePtr->AnalyzeReverseStream(&frame) == -1) { |