From 0fc2cb59d5f77412f5922540d67fea81f4d1744b Mon Sep 17 00:00:00 2001 From: Mathias Agopian Date: Sun, 21 Oct 2012 01:01:38 -0700 Subject: a test app for the resamplers Change-Id: I66852d90d384f1d9e77b51ad1a1ebdbaf61d0607 --- services/audioflinger/test-resample.cpp | 229 ++++++++++++++++++++++++++++++++ 1 file changed, 229 insertions(+) create mode 100644 services/audioflinger/test-resample.cpp (limited to 'services/audioflinger/test-resample.cpp') diff --git a/services/audioflinger/test-resample.cpp b/services/audioflinger/test-resample.cpp new file mode 100644 index 0000000..a55a32b --- /dev/null +++ b/services/audioflinger/test-resample.cpp @@ -0,0 +1,229 @@ +/* + * Copyright (C) 2012 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. + */ + +#include "AudioResampler.h" +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +using namespace android; + +struct HeaderWav { + HeaderWav(size_t size, int nc, int sr, int bits) { + strncpy(RIFF, "RIFF", 4); + chunkSize = size + sizeof(HeaderWav); + strncpy(WAVE, "WAVE", 4); + strncpy(fmt, "fmt ", 4); + fmtSize = 16; + audioFormat = 1; + numChannels = nc; + samplesRate = sr; + byteRate = sr * numChannels * (bits/8); + align = nc*(bits/8); + bitsPerSample = bits; + strncpy(data, "data", 4); + dataSize = size; + } + + char RIFF[4]; // RIFF + uint32_t chunkSize; // File size + char WAVE[4]; // WAVE + char fmt[4]; // fmt\0 + uint32_t fmtSize; // fmt size + uint16_t audioFormat; // 1=PCM + uint16_t numChannels; // num channels + uint32_t samplesRate; // sample rate in hz + uint32_t byteRate; // Bps + uint16_t align; // 2=16-bit mono, 4=16-bit stereo + uint16_t bitsPerSample; // bits per sample + char data[4]; // "data" + uint32_t dataSize; // size +}; + +static int usage(const char* name) { + fprintf(stderr,"Usage: %s [-p] [-h] [-q ] [-i ] [-o ] \n", name); + fprintf(stderr,"-p - enable profiling\n"); + fprintf(stderr,"-h - create wav file\n"); + fprintf(stderr,"-q - resampler quality\n"); + fprintf(stderr," dq : default quality\n"); + fprintf(stderr," lq : low quality\n"); + fprintf(stderr," mq : medium quality\n"); + fprintf(stderr," hq : high quality\n"); + fprintf(stderr," vhq : very high quality\n"); + fprintf(stderr,"-i - input file sample rate\n"); + fprintf(stderr,"-o - output file sample rate\n"); + return -1; +} + +int main(int argc, char* argv[]) { + + bool profiling = false; + bool writeHeader = false; + int input_freq = 0; + int output_freq = 0; + AudioResampler::src_quality quality = AudioResampler::DEFAULT_QUALITY; + + int ch; + while ((ch = getopt(argc, argv, "phq:i:o:")) != -1) { + switch (ch) { + case 'p': + profiling = true; + break; + case 'h': + writeHeader = true; + break; + case 'q': + if (!strcmp(optarg, "dq")) + quality = AudioResampler::DEFAULT_QUALITY; + else if (!strcmp(optarg, "lq")) + quality = AudioResampler::LOW_QUALITY; + else if (!strcmp(optarg, "mq")) + quality = AudioResampler::MED_QUALITY; + else if (!strcmp(optarg, "hq")) + quality = AudioResampler::HIGH_QUALITY; + else if (!strcmp(optarg, "vhq")) + quality = AudioResampler::VERY_HIGH_QUALITY; + else { + usage(argv[0]); + return -1; + } + break; + case 'i': + input_freq = atoi(optarg); + break; + case 'o': + output_freq = atoi(optarg); + break; + case '?': + default: + usage(argv[0]); + return -1; + } + } + argc -= optind; + + if (argc != 2) { + usage(argv[0]); + return -1; + } + + argv += optind; + + // ---------------------------------------------------------- + + struct stat st; + if (stat(argv[0], &st) < 0) { + fprintf(stderr, "stat: %s\n", strerror(errno)); + return -1; + } + + int input_fd = open(argv[0], O_RDONLY); + if (input_fd < 0) { + fprintf(stderr, "open: %s\n", strerror(errno)); + return -1; + } + + size_t input_size = st.st_size; + void* input_vaddr = mmap(0, input_size, PROT_READ, MAP_PRIVATE, input_fd, + 0); + if (input_vaddr == MAP_FAILED ) { + fprintf(stderr, "mmap: %s\n", strerror(errno)); + return -1; + } + +// printf("input sample rate: %d Hz\n", input_freq); +// printf("output sample rate: %d Hz\n", output_freq); +// printf("input mmap: %p, size=%u\n", input_vaddr, input_size); + + // ---------------------------------------------------------- + + class Provider: public AudioBufferProvider { + int16_t* mAddr; + size_t mNumFrames; + public: + Provider(const void* addr, size_t size) { + mAddr = (int16_t*) addr; + mNumFrames = size / sizeof(int16_t); + } + virtual status_t getNextBuffer(Buffer* buffer, + int64_t pts = kInvalidPTS) { + buffer->frameCount = mNumFrames; + buffer->i16 = mAddr; + return NO_ERROR; + } + virtual void releaseBuffer(Buffer* buffer) { + } + } provider(input_vaddr, input_size); + + size_t output_size = 2 * 2 * ((int64_t) input_size * output_freq) + / input_freq; + output_size &= ~7; // always stereo, 32-bits + + void* output_vaddr = malloc(output_size); + memset(output_vaddr, 0, output_size); + + AudioResampler* resampler = AudioResampler::create(16, 1, output_freq, + quality); + + size_t out_frames = output_size/8; + resampler->setSampleRate(input_freq); + resampler->setVolume(0x1000, 0x1000); + resampler->resample((int*) output_vaddr, out_frames, &provider); + + if (profiling) { + memset(output_vaddr, 0, output_size); + timespec start, end; + clock_gettime(CLOCK_MONOTONIC_HR, &start); + resampler->resample((int*) output_vaddr, out_frames, &provider); + clock_gettime(CLOCK_MONOTONIC_HR, &end); + int64_t start_ns = start.tv_sec * 1000000000LL + start.tv_nsec; + int64_t end_ns = end.tv_sec * 1000000000LL + end.tv_nsec; + int64_t time = end_ns - start_ns; + printf("%f Mspl/s\n", out_frames/(time/1e9)/1e6); + } + + // down-mix (we just truncate and keep the left channel) + int32_t* out = (int32_t*) output_vaddr; + int16_t* convert = (int16_t*) malloc(out_frames * sizeof(int16_t)); + for (size_t i = 0; i < out_frames; i++) { + convert[i] = out[i * 2] >> 12; + } + + // write output to disk + int output_fd = open(argv[1], O_WRONLY | O_CREAT | O_TRUNC, + S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH); + if (output_fd < 0) { + fprintf(stderr, "open: %s\n", strerror(errno)); + return -1; + } + + if (writeHeader) { + HeaderWav wav(out_frames*sizeof(int16_t), 1, output_freq, 16); + write(output_fd, &wav, sizeof(wav)); + } + + write(output_fd, convert, out_frames * sizeof(int16_t)); + close(output_fd); + + return 0; +} -- cgit v1.1