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authorEric Laurent <elaurent@google.com>2011-06-18 09:47:37 -0700
committerEric Laurent <elaurent@google.com>2011-07-25 14:37:35 -0700
commit40a09525b7c484764c64daa1ca663a314a1b6d31 (patch)
treebd3c889b82ff41979f512967cdcd106e50489c7e /libaudio/ReSampler.cpp
parentc593b8893fef34a1ff38d617a4e229447a9466cf (diff)
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Audio HAL: added echo reference
Added EchoReference class enabling the input stream to access the audio data written to the output stream and push it as echo reference to the AEC. Also added methods to calculate the echo delay. Moved ReSampler class to a separate source file. Change-Id: I9c3388f39101d567240545eab271eb61c97e7b56
Diffstat (limited to 'libaudio/ReSampler.cpp')
-rw-r--r--libaudio/ReSampler.cpp171
1 files changed, 171 insertions, 0 deletions
diff --git a/libaudio/ReSampler.cpp b/libaudio/ReSampler.cpp
new file mode 100644
index 0000000..aa1d5e0
--- /dev/null
+++ b/libaudio/ReSampler.cpp
@@ -0,0 +1,171 @@
+/*
+** Copyright 2011, The Android Open-Source Project
+**
+** Licensed under the Apache License, Version 2.0 (the "License");
+** you may not use this file except in compliance with the License.
+** You may obtain a copy of the License at
+**
+** http://www.apache.org/licenses/LICENSE-2.0
+**
+** Unless required by applicable law or agreed to in writing, software
+** distributed under the License is distributed on an "AS IS" BASIS,
+** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+** See the License for the specific language governing permissions and
+** limitations under the License.
+*/
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "ReSampler"
+
+#include <utils/Log.h>
+#include "ReSampler.h"
+
+namespace android_audio_legacy {
+
+
+//------------------------------------------------------------------------------
+// speex based resampler
+//------------------------------------------------------------------------------
+
+#define RESAMPLER_QUALITY 2
+
+ReSampler::ReSampler(uint32_t inSampleRate,
+ uint32_t outSampleRate,
+ uint32_t channelCount,
+ BufferProvider* provider)
+ : mStatus(NO_INIT), mSpeexResampler(NULL), mProvider(provider),
+ mInSampleRate(inSampleRate), mOutSampleRate(outSampleRate), mChannelCount(channelCount),
+ mInBuf(NULL), mInBufSize(0)
+{
+ LOGV("ReSampler() cstor %p In SR %d Out SR %d channels %d",
+ this, mInSampleRate, mOutSampleRate, mChannelCount);
+
+ if (mProvider == NULL) {
+ return;
+ }
+
+ int error;
+ mSpeexResampler = speex_resampler_init(channelCount,
+ inSampleRate,
+ outSampleRate,
+ RESAMPLER_QUALITY,
+ &error);
+ if (mSpeexResampler == NULL) {
+ LOGW("ReSampler: Cannot create speex resampler: %s", speex_resampler_strerror(error));
+ return;
+ }
+
+ reset();
+
+ int frames = speex_resampler_get_input_latency(mSpeexResampler);
+ mSpeexDelayNs = (int32_t)((1000000000 * (int64_t)frames) / mInSampleRate);
+ frames = speex_resampler_get_output_latency(mSpeexResampler);
+ mSpeexDelayNs += (int32_t)((1000000000 * (int64_t)frames) / mOutSampleRate);
+
+ mStatus = NO_ERROR;
+}
+
+ReSampler::~ReSampler()
+{
+ free(mInBuf);
+
+ if (mSpeexResampler != NULL) {
+ speex_resampler_destroy(mSpeexResampler);
+ }
+}
+
+void ReSampler::reset()
+{
+ mFramesIn = 0;
+ mFramesRq = 0;
+
+ if (mSpeexResampler != NULL) {
+ speex_resampler_reset_mem(mSpeexResampler);
+ }
+}
+
+int32_t ReSampler::delayNs()
+{
+ int32_t delay = (int32_t)((1000000000 * (int64_t)mFramesIn) / mInSampleRate);
+ delay += mSpeexDelayNs;
+
+ return delay;
+}
+
+// outputs a number of frames less or equal to *outFrameCount and updates *outFrameCount
+// with the actual number of frames produced.
+int ReSampler::resample(int16_t *out, size_t *outFrameCount)
+{
+ if (mStatus != NO_ERROR) {
+ return mStatus;
+ }
+
+ if (out == NULL || outFrameCount == NULL) {
+ return BAD_VALUE;
+ }
+
+ size_t framesRq = *outFrameCount;
+ // update and cache the number of frames needed at the input sampling rate to produce
+ // the number of frames requested at the output sampling rate
+ if (framesRq != mFramesRq) {
+ mFramesNeeded = (framesRq * mOutSampleRate) / mInSampleRate + 1;
+ mFramesRq = framesRq;
+ }
+
+ size_t framesWr = 0;
+ size_t inFrames = 0;
+ while (framesWr < framesRq) {
+ if (mFramesIn < mFramesNeeded) {
+ // make sure that the number of frames present in mInBuf (mFramesIn) is at least
+ // the number of frames needed to produce the number of frames requested at
+ // the output sampling rate
+ if (mInBufSize < mFramesNeeded) {
+ mInBufSize = mFramesNeeded;
+ mInBuf = (int16_t *)realloc(mInBuf, mInBufSize * mChannelCount * sizeof(int16_t));
+ }
+ BufferProvider::Buffer buf;
+ buf.frameCount = mFramesNeeded - mFramesIn;
+ mProvider->getNextBuffer(&buf);
+ if (buf.raw == NULL) {
+ break;
+ }
+ memcpy(mInBuf + mFramesIn * mChannelCount,
+ buf.raw,
+ buf.frameCount * mChannelCount * sizeof(int16_t));
+ mFramesIn += buf.frameCount;
+ mProvider->releaseBuffer(&buf);
+ }
+
+ size_t outFrames = framesRq - framesWr;
+ inFrames = mFramesIn;
+ if (mChannelCount == 1) {
+ speex_resampler_process_int(mSpeexResampler,
+ 0,
+ mInBuf,
+ &inFrames,
+ out + framesWr * mChannelCount,
+ &outFrames);
+ } else {
+ speex_resampler_process_interleaved_int(mSpeexResampler,
+ mInBuf,
+ &inFrames,
+ out + framesWr * mChannelCount,
+ &outFrames);
+ }
+ framesWr += outFrames;
+ mFramesIn -= inFrames;
+ LOGW_IF((framesWr != framesRq) && (mFramesIn != 0),
+ "ReSampler::resample() remaining %d frames in and %d frames out",
+ mFramesIn, (framesRq - framesWr));
+ }
+ if (mFramesIn) {
+ memmove(mInBuf,
+ mInBuf + inFrames * mChannelCount,
+ mFramesIn * mChannelCount * sizeof(int16_t));
+ }
+ *outFrameCount = framesWr;
+
+ return NO_ERROR;
+}
+
+}; // namespace android_audio_legacy