From df383a5a54582811e5e038efc557172b8ec69dd1 Mon Sep 17 00:00:00 2001 From: Andy Hung Date: Wed, 9 Apr 2014 19:10:15 -0700 Subject: Update test-resample to handle multichannel Option -c # specifies number of channels (mono default). Option -s to specify stereo is removed (-c 2 replaces). Option -h to specify WAV header is removed (WAV is now default). Change-Id: Iba4b83806028a8a9c1ddba6f555182d214ef73ff Signed-off-by: Andy Hung --- services/audioflinger/test-resample.cpp | 105 +++++++++++++++----------------- 1 file changed, 48 insertions(+), 57 deletions(-) (limited to 'services/audioflinger') diff --git a/services/audioflinger/test-resample.cpp b/services/audioflinger/test-resample.cpp index d1de95d..e14b4ae 100644 --- a/services/audioflinger/test-resample.cpp +++ b/services/audioflinger/test-resample.cpp @@ -36,16 +36,16 @@ using namespace android; static bool gVerbose = false; static int usage(const char* name) { - fprintf(stderr,"Usage: %s [-p] [-f] [-F] [-h] [-v] [-s] [-q {dq|lq|mq|hq|vhq|dlq|dmq|dhq}]" + fprintf(stderr,"Usage: %s [-p] [-f] [-F] [-v] [-c channels]" + " [-q {dq|lq|mq|hq|vhq|dlq|dmq|dhq}]" " [-i input-sample-rate] [-o output-sample-rate]" " [-O csv] [-P csv] []" " \n", name); fprintf(stderr," -p enable profiling\n"); fprintf(stderr," -f enable filter profiling\n"); fprintf(stderr," -F enable floating point -q {dlq|dmq|dhq} only"); - fprintf(stderr," -h create wav file\n"); fprintf(stderr," -v verbose : log buffer provider calls\n"); - fprintf(stderr," -s stereo (ignored if input file is specified)\n"); + fprintf(stderr," -c # channels (1-2 for lq|mq|hq; 1-8 for dlq|dmq|dhq)\n"); fprintf(stderr," -q resampler quality\n"); fprintf(stderr," dq : default quality\n"); fprintf(stderr," lq : low quality\n"); @@ -102,11 +102,9 @@ int parseCSV(const char *string, Vector& values) } int main(int argc, char* argv[]) { - const char* const progname = argv[0]; bool profileResample = false; bool profileFilter = false; - bool writeHeader = false; bool useFloat = false; int channels = 1; int input_freq = 0; @@ -116,7 +114,7 @@ int main(int argc, char* argv[]) { Vector Pvalues; int ch; - while ((ch = getopt(argc, argv, "pfFhvsq:i:o:O:P:")) != -1) { + while ((ch = getopt(argc, argv, "pfFvc:q:i:o:O:P:")) != -1) { switch (ch) { case 'p': profileResample = true; @@ -127,14 +125,11 @@ int main(int argc, char* argv[]) { case 'F': useFloat = true; break; - case 'h': - writeHeader = true; - break; case 'v': gVerbose = true; break; - case 's': - channels = 2; + case 'c': + channels = atoi(optarg); break; case 'q': if (!strcmp(optarg, "dq")) @@ -183,6 +178,11 @@ int main(int argc, char* argv[]) { } } + if (channels < 1 + || channels > (quality < AudioResampler::DYN_LOW_QUALITY ? 2 : 8)) { + fprintf(stderr, "invalid number of audio channels %d\n", channels); + return -1; + } if (useFloat && quality < AudioResampler::DYN_LOW_QUALITY) { fprintf(stderr, "float processing is only possible for dynamic resamplers\n"); return -1; @@ -234,20 +234,20 @@ int main(int argc, char* argv[]) { double t = double(i) / input_freq; double y = sin(M_PI * k * t * t); int16_t yi = floor(y * 32767.0 + 0.5); - for (size_t j=0 ; j<(size_t)channels ; j++) { - in[i*channels + j] = yi / (1+j); // right ch. 1/2 left ch. + for (int j = 0; j < channels; j++) { + in[i*channels + j] = yi / (1 + j); } } } - size_t frame_size = channels * sizeof(int16_t); - size_t input_frames = input_size / frame_size; + size_t input_framesize = channels * sizeof(int16_t); + size_t input_frames = input_size / input_framesize; // For float processing, convert input int16_t to float array if (useFloat) { void *new_vaddr; - frame_size = channels * sizeof(float); - input_size = input_frames * frame_size; + input_framesize = channels * sizeof(float); + input_size = input_frames * input_framesize; new_vaddr = malloc(input_size); memcpy_to_float_from_i16(reinterpret_cast(new_vaddr), reinterpret_cast(input_vaddr), input_frames * channels); @@ -323,15 +323,17 @@ int main(int argc, char* argv[]) { void reset() { mNextFrame = 0; } - } provider(input_vaddr, input_frames, frame_size, Pvalues); + } provider(input_vaddr, input_frames, input_framesize, Pvalues); if (gVerbose) { printf("%zu input frames\n", input_frames); } int bit_depth = useFloat ? 32 : 16; - size_t output_size = 2 * 4 * ((int64_t) input_frames * output_freq) / input_freq; - output_size &= ~7; // always stereo, 32-bits + int output_channels = channels > 2 ? channels : 2; // output is at least stereo samples + size_t output_framesize = output_channels * (useFloat ? sizeof(float) : sizeof(int32_t)); + size_t output_frames = ((int64_t) input_frames * output_freq) / input_freq; + size_t output_size = output_frames * output_framesize; if (profileFilter) { // Check how fast sample rate changes are that require filter changes. @@ -380,7 +382,7 @@ int main(int argc, char* argv[]) { void* output_vaddr = malloc(output_size); AudioResampler* resampler = AudioResampler::create(bit_depth, channels, output_freq, quality); - size_t out_frames = output_size/8; + /* set volume precision to 12 bits, so the volume scale is 1<<12. * The output int32_t is represented as Q4.27, with 4 bits of guard @@ -422,7 +424,7 @@ int main(int argc, char* argv[]) { for (int n = 0; n < trials; ++n) { clock_gettime(CLOCK_MONOTONIC, &start); for (int i = 0; i < looplimit; ++i) { - resampler->resample((int*) output_vaddr, out_frames, &provider); + resampler->resample((int*) output_vaddr, output_frames, &provider); provider.reset(); // during benchmarking reset only the provider } clock_gettime(CLOCK_MONOTONIC, &end); @@ -435,26 +437,26 @@ int main(int argc, char* argv[]) { } // Mfrms/s is "Millions of output frames per second". printf("quality: %d channels: %d msec: %" PRId64 " Mfrms/s: %.2lf\n", - quality, channels, time/1000000, out_frames * looplimit / (time / 1e9) / 1e6); + quality, channels, time/1000000, output_frames * looplimit / (time / 1e9) / 1e6); resampler->reset(); } memset(output_vaddr, 0, output_size); if (gVerbose) { - printf("resample() %zu output frames\n", out_frames); + printf("resample() %zu output frames\n", output_frames); } if (Ovalues.isEmpty()) { - Ovalues.push(out_frames); + Ovalues.push(output_frames); } - for (size_t i = 0, j = 0; i < out_frames; ) { + for (size_t i = 0, j = 0; i < output_frames; ) { size_t thisFrames = Ovalues[j++]; if (j >= Ovalues.size()) { j = 0; } - if (thisFrames == 0 || thisFrames > out_frames - i) { - thisFrames = out_frames - i; + if (thisFrames == 0 || thisFrames > output_frames - i) { + thisFrames = output_frames - i; } - resampler->resample((int*) output_vaddr + 2*i, thisFrames, &provider); + resampler->resample((int*) output_vaddr + output_channels*i, thisFrames, &provider); i += thisFrames; } if (gVerbose) { @@ -471,20 +473,20 @@ int main(int argc, char* argv[]) { // which is then converted to int16_t for final storage. if (useFloat) { memcpy_to_q4_27_from_float(reinterpret_cast(output_vaddr), - reinterpret_cast(output_vaddr), out_frames * 2); // stereo samples + reinterpret_cast(output_vaddr), output_frames * output_channels); } - // mono takes left channel only - // stereo right channel is half amplitude of stereo left channel (due to input creation) + // mono takes left channel only (out of stereo output pair) + // stereo and multichannel preserve all channels. int32_t* out = (int32_t*) output_vaddr; - int16_t* convert = (int16_t*) malloc(out_frames * channels * sizeof(int16_t)); + int16_t* convert = (int16_t*) malloc(output_frames * channels * sizeof(int16_t)); // round to half towards zero and saturate at int16 (non-dithered) const int roundVal = (1<<(volumePrecision-1)) - 1; // volumePrecision > 0 - for (size_t i = 0; i < out_frames; i++) { + for (size_t i = 0; i < output_frames; i++) { for (int j = 0; j < channels; j++) { - int32_t s = out[i * 2 + j] + roundVal; // add offset here + int32_t s = out[i * output_channels + j] + roundVal; // add offset here if (s < 0) { s = (s + 1) >> volumePrecision; // round to 0 if (s < -32768) { @@ -501,29 +503,18 @@ int main(int argc, char* argv[]) { } // write output to disk - if (writeHeader) { - SF_INFO info; - info.frames = 0; - info.samplerate = output_freq; - info.channels = channels; - info.format = SF_FORMAT_WAV | SF_FORMAT_PCM_16; - SNDFILE *sf = sf_open(file_out, SFM_WRITE, &info); - if (sf == NULL) { - perror(file_out); - return EXIT_FAILURE; - } - (void) sf_writef_short(sf, convert, out_frames); - sf_close(sf); - } else { - int output_fd = open(file_out, O_WRONLY | O_CREAT | O_TRUNC, - S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH); - if (output_fd < 0) { - perror(file_out); - return EXIT_FAILURE; - } - write(output_fd, convert, out_frames * channels * sizeof(int16_t)); - close(output_fd); + SF_INFO info; + info.frames = 0; + info.samplerate = output_freq; + info.channels = channels; + info.format = SF_FORMAT_WAV | SF_FORMAT_PCM_16; + SNDFILE *sf = sf_open(file_out, SFM_WRITE, &info); + if (sf == NULL) { + perror(file_out); + return EXIT_FAILURE; } + (void) sf_writef_short(sf, convert, output_frames); + sf_close(sf); return EXIT_SUCCESS; } -- cgit v1.1