diff options
Diffstat (limited to 'chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc')
-rw-r--r-- | chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc | 54 |
1 files changed, 26 insertions, 28 deletions
diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc index 3abe4f1ec46..97d87b1b3a4 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc @@ -10,61 +10,59 @@ #include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h" +#include <assert.h> #include <string.h> #include "webrtc/common_audio/resampler/include/resampler.h" -#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" -#include "webrtc/system_wrappers/interface/critical_section_wrapper.h" -#include "webrtc/system_wrappers/interface/trace.h" +#include "webrtc/system_wrappers/interface/logging.h" namespace webrtc { - namespace acm2 { -ACMResampler::ACMResampler() - : resampler_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()) { +ACMResampler::ACMResampler() { } ACMResampler::~ACMResampler() { - delete resampler_crit_sect_; } int ACMResampler::Resample10Msec(const int16_t* in_audio, int in_freq_hz, int out_freq_hz, int num_audio_channels, + int out_capacity_samples, int16_t* out_audio) { - CriticalSectionScoped cs(resampler_crit_sect_); - + int in_length = in_freq_hz * num_audio_channels / 100; + int out_length = out_freq_hz * num_audio_channels / 100; if (in_freq_hz == out_freq_hz) { - size_t length = static_cast<size_t>(in_freq_hz * num_audio_channels / 100); - memcpy(out_audio, in_audio, length * sizeof(int16_t)); - return static_cast<int16_t>(in_freq_hz / 100); + if (out_capacity_samples < in_length) { + assert(false); + return -1; + } + memcpy(out_audio, in_audio, in_length * sizeof(int16_t)); + return in_length / num_audio_channels; } - // |maxLen| is maximum number of samples for 10ms at 48kHz. - int max_len = 480 * num_audio_channels; - int length_in = (in_freq_hz / 100) * num_audio_channels; - int out_len; - - ResamplerType type = (num_audio_channels == 1) ? kResamplerSynchronous : - kResamplerSynchronousStereo; - - if (resampler_.ResetIfNeeded(in_freq_hz, out_freq_hz, type) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0, - "Error in reset of resampler"); + if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz, + num_audio_channels) != 0) { + LOG_FERR3(LS_ERROR, InitializeIfNeeded, in_freq_hz, out_freq_hz, + num_audio_channels); return -1; } - if (resampler_.Push(in_audio, length_in, out_audio, max_len, out_len) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0, - "Error in resampler: resampler.Push"); + out_length = + resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples); + if (out_length == -1) { + LOG_FERR4(LS_ERROR, + Resample, + in_audio, + in_length, + out_audio, + out_capacity_samples); return -1; } - return out_len / num_audio_channels; + return out_length / num_audio_channels; } } // namespace acm2 - } // namespace webrtc |