From 60f60cc76e43197a0825ffac9aff0b7007a94175 Mon Sep 17 00:00:00 2001 From: Bent Bisballe Nyeng Date: Wed, 20 Nov 2013 12:57:17 +0100 Subject: Framework API now complete in its first iteration. So far with support for raw, opus and amrwb. --- test/test_opus.cc | 272 ++++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 272 insertions(+) create mode 100644 test/test_opus.cc (limited to 'test/test_opus.cc') diff --git a/test/test_opus.cc b/test/test_opus.cc new file mode 100644 index 0000000..68ea0f4 --- /dev/null +++ b/test/test_opus.cc @@ -0,0 +1,272 @@ +/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */ +/* vim: set et sw=2 ts=2: */ +/*************************************************************************** + * test_init.cc + * + * Mon Sep 2 14:02:16 CEST 2013 + * Copyright 2013 Bent Bisballe Nyeng + * deva@aasimon.org + ****************************************************************************/ + +/* + * This file is part of lrtp. + * + * lrtp is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * lrtp is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with lrtp; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ +#include + +#include + +#include +#include + +#include +#include +#include +#include + +#define KEY "123456789012345678901234567890123456789012345678901234567890" +#define SSRC 1234567890 + +#define FS 8000 + +#define F1 440 +#define AF1 0.3 + +#define F2 500 +#define AF2 0.7 + +void dump(const char *title, const char *buf, size_t size) +{ + printf("%12s: ", title); + for(int i = 0; i < size; i++) { + if(i % 8 == 0) printf(" "); + printf("%02x ", (unsigned char)*buf++); + } + printf("\n"); +} + +class Audio { +public: + Audio() { + ao_initialize(); + + device = NULL; + ao_sample_format format; + + int default_driver = ao_default_driver_id(); + if(default_driver == -1) { + printf("Error could not default driver.\n"); + return; + } + printf("Default driver: %d\n", default_driver); + + format.bits = 16; + format.channels = 2; + format.rate = FS; + format.byte_format = AO_FMT_LITTLE; + + device = ao_open_live(default_driver, &format, NULL); + if(device == NULL) { + printf("Error opening device.\n"); + return; + } + } + + ~Audio() { + if(device) ao_close(device); + ao_shutdown(); + } + + void play(char *pcm, size_t size) { + ao_play(device, pcm, size); + } + +private: + ao_device *device; +}; + +int main() +{ + size_t channels = 2; + size_t ms[] = { 120, 240, 480, 960, 1920, 2880 }; + + std::vector packets; + unsigned int csrc = 42; + + double sin_x = 0; + size_t ts = 0; + + printf("========== Encode ==========\n"); + + { // Encode + struct lrtp_t *lrtp = lrtp_init(KEY, SSRC); + + struct lrtp_profile_t *profile = + lrtp_create_profile(lrtp, PROFILE_OPUS, csrc, + //OPTION_RAW_PKG_SIZE, pkg_size, + OPTION_END); + + char packet[16*1024]; + size_t packetsize = sizeof(packet); + + int err; + OpusEncoder *opus = opus_encoder_create(FS, channels, + OPUS_APPLICATION_AUDIO, &err); + printf("Opus create err: %d\n", err); + + opus_encoder_ctl(opus, OPUS_SET_BITRATE(64000));// [500;512000] + opus_encoder_ctl(opus, OPUS_SET_COMPLEXITY(10)); // [0;10] + opus_encoder_ctl(opus, OPUS_SET_SIGNAL(OPUS_SIGNAL_MUSIC)); + + int cnt = 0; + size_t timestamp = 0; + for(unsigned int ts = 0; ts < 120; ts++) { + printf("packet #%d\n", ts); + + size_t idx = rand() % (sizeof(ms)/sizeof(size_t)); + printf("idx: %d\n", idx); + size_t pcmsize = ms[idx] / (48000.0 / FS); // Number of samples pr channel + short *pcm = new short[100000/*pcmsize * channels*/]; + for(int i = 0 ; i < pcmsize; i++) { + sin_x++; + + double amp1 = sin((2*M_PI/(double)FS)*(double)sin_x * AF1) * SHRT_MAX; + double amp2 = sin((2*M_PI/(double)FS)*(double)sin_x * AF2) * SHRT_MAX; + + pcm[i*2] = (short)(sin(2*M_PI/FS*(double)sin_x * F1) * amp1); + pcm[i*2+1] = (short)(sin(2*M_PI/FS*(double)sin_x * F2) * amp2); + } + + // Master timestamp is sample number in 48kHz (Opus RFC states this) + timestamp += pcmsize * 48000 / FS; + + // size_t pcmsize = pcmsize * channels * sizeof(short); + + char frame[pcmsize]; + int framesize = sizeof(frame); + framesize = opus_encode(opus, pcm, pcmsize, + (unsigned char*)frame, framesize); + + if(framesize < 0) { + printf("Opus error: %s\n", opus_strerror(framesize)); + } + + printf("Opus Packet: %d bytes from %d bytes\n", pcmsize, framesize); + + int ret = lrtp_enqueue_frame(profile, frame, framesize); + while( (ret = lrtp_pack(lrtp, packet, sizeof(packet))) != 0) { + std::string p; + p.append(packet, ret); + packets.push_back(p); + } + + delete[] pcm; + } + + opus_encoder_destroy(opus); + lrtp_destroy_profile(lrtp, csrc); + lrtp_close(lrtp); + } + + printf("========== Decode ==========\n"); + + { // Decode + struct lrtp_t *lrtp = lrtp_init(KEY, SSRC); + + struct lrtp_profile_t *profile = + lrtp_create_profile(lrtp, PROFILE_OPUS, csrc, + //OPTION_RAW_PKG_SIZE, pkg_size, + OPTION_END); + + int err; + OpusDecoder *opus = opus_decoder_create(FS, channels, &err); + printf("Opus create err: %d\n", err); + + int idx = (sizeof(ms)/sizeof(size_t)) - 1; + printf("idx: %d\n", idx); + + Audio audio; + + char frame[16*1024]; + + int cnt = 0; + std::vector::iterator i = packets.begin(); + while(i != packets.end()) { + size_t packetsize = i->size(); + const char *packet = i->data(); + unsigned int ts; + + printf("unpack sz: %d - %p\n", packetsize, packet); + + lrtp_unpack(lrtp, packet, packetsize); + int ret; + while((ret = lrtp_dequeue_frame(lrtp, frame, sizeof(frame), &csrc, &ts)) + != 0) { + size_t pcmsize = 16*1024;//ms[idx] / (48000 / FS); + short *pcm = new short[pcmsize * channels]; + printf("pcmsize %d\n", pcmsize); fflush(stdout); + int res = opus_decode(opus, (const unsigned char*)frame, ret, + pcm, pcmsize, 0); + + printf("Decompressed %d bytes\n", res); + // pcmsize = res * channels * sizeof(short); + + audio.play((char *)pcm, res * channels * sizeof(short)); + + delete[] pcm; + } + + i++; + } + + + /* + + std::vector::iterator i = packets.begin(); + while(i != packets.end()) { + size_t packetsize = i->size(); + printf("unpack sz: %d\n", packetsize); + const char *packet = i->data(); + unsigned int ts; + + framesize = sizeof(frame); + + lrtp_unpack(lrtp, packet, packetsize, frame, &framesize, &csrc, &ts); + printf("Got %d bytes, csrc %d, ts: %d\n", framesize, csrc, ts); + + size_t pcmsize = 16*1024;//ms[idx] / (48000 / FS); + short *pcm = new short[pcmsize * channels]; + printf("pcmsize %d\n", pcmsize); fflush(stdout); + int res = opus_decode(opus, (const unsigned char*)frame, framesize, + pcm, pcmsize, 0); + framesize = sizeof(frame); + + printf("Decompressed %d bytes\n", res); + // pcmsize = res * channels * sizeof(short); + + audio.play((char *)pcm, res * channels * sizeof(short)); + + delete[] pcm; + i++; + } + */ + opus_decoder_destroy(opus); + lrtp_destroy_profile(lrtp, csrc); + lrtp_close(lrtp); + } + + return 0; +} -- cgit v1.2.3