561 lines
20 KiB
C++
561 lines
20 KiB
C++
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#include <iostream>
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#include <unistd.h>
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#include <cmath>
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#include "timing.h"
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#include "log/logger.h"
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#include "common.h"
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#include <modules/audio_processing/include/audio_processing.h>
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#include <modules/audio_processing/include/config.h>
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#include "alsa_dev.h"
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using namespace std;
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using namespace toolkit;
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#define MIX_INPUT_CHANNELS 2
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#define MIX_INPUT_SAMPLES (10 * MIX_INPUT_SAMPLE_RATE/1000)
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// #define MIX_INPUT_SAMPLES 1024
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#define MIX_INPUT_SAMPLE_RATE 44100
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struct audio_buf_t
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{
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uint8_t* data;
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int index;
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int size;
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};
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struct RtmpConfig {
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char url[1024];
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AVFormatContext *formatCtx;
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AVStream *stream;
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AVCodecContext *codecCtx;
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SwrContext *swrCtx;
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};
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static SampleInfo kPcmSampleInfo;
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//----------------------------------------------
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#define MIN(a, b) ((a) < (b) ? (a) : (b))
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#define MAX(a, b) ((a) > (b) ? (a) : (b))
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// 最大/小音量（db）
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#define MIN_DB (-10)
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#define MAX_DB (60)
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// 最大/小音量： 0: 静音； 100:最大音量
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#define MUTE_VOLUME (0)
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#define MAX_VOLUME (100)
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static int vol_scaler_init(int *scaler, int mindb, int maxdb);
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typedef struct VolumeCtlUnit
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{
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int scaler[MAX_VOLUME + 1]; // 音量表
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int zeroDb; // 0db在scaler中的索引
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// 自定义需要调节的音量
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int micVolume;
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VolumeCtlUnit() {
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// 音量控制器初始化
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zeroDb = vol_scaler_init(scaler, MIN_DB, MAX_DB);
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micVolume = 100;
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}
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} volume_ctl_unit_t;
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static volume_ctl_unit_t kVolCtrlUnit;
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static int vol_scaler_init(int *scaler, int mindb, int maxdb)
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{
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double tabdb[MAX_VOLUME + 1];
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double tabf [MAX_VOLUME + 1];
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int z, i;
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for (i = 0; i < (MAX_VOLUME + 1); i++) {
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// (mindb, maxdb)平均分成(MAX_VOLUME + 1)份
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tabdb[i] = mindb + (maxdb - mindb) * i / (MAX_VOLUME + 1);
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// dB = 20 * log(A1 / A2)，当A1，A2相等时，db为0
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// 这里以(1 << 14)作为原始声音振幅，得到调节后的振幅(A1),将A1存入音量表中
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tabf [i] = pow(10.0, tabdb[i] / 20.0);
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scaler[i] = (int)((1 << 14) * tabf[i]); // Q14 fix point
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}
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z = -mindb * (MAX_VOLUME + 1) / (maxdb - mindb);
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z = MAX(z, 0 );
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z = MIN(z, MAX_VOLUME);
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scaler[0] = 0; // 音量表中，0标识静音
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scaler[z] = (1 << 14);// (mindb, maxdb)的中间值作为0db，即不做增益处理
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return z;
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}
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static void vol_scaler_run(int16_t *buf, int n, int volume)
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{
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/* 简易版
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while (n--) {
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*buf = (*buf) * multiplier / 100.0;
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*buf = std::max((int)*buf, -0x7fff);
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*buf = std::min((int)*buf, 0x7fff);
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buf++;
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}
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*/
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int multiplier = kVolCtrlUnit.scaler[volume];
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if (multiplier > (1 << 14)) {
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int32_t v;
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while (n--) {
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v = ((int32_t)*buf * multiplier) >> 14;
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v = MAX(v,-0x7fff);
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v = MIN(v, 0x7fff);
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*buf++ = (int16_t)v;
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}
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} else if (multiplier < (1 << 14)) {
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while (n--) {
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*buf = ((int32_t)*buf * multiplier) >> 14;
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buf++;
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}
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}
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}
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//----------------------------------------------
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webrtc::AudioProcessing::Config webtcConfigInit()
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{
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webrtc::AudioProcessing::Config apmConfig;
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apmConfig.pipeline.maximum_internal_processing_rate = MIX_INPUT_SAMPLE_RATE;
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apmConfig.pipeline.multi_channel_capture = true;
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apmConfig.pipeline.multi_channel_render = true;
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//PreAmplifier
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apmConfig.pre_amplifier.enabled = false;
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apmConfig.pre_amplifier.fixed_gain_factor = 0.7f;
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//HighPassFilter
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apmConfig.high_pass_filter.enabled = false;
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apmConfig.high_pass_filter.apply_in_full_band = false;
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//EchoCanceller
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apmConfig.echo_canceller.enabled = false;
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apmConfig.echo_canceller.mobile_mode = false;
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apmConfig.echo_canceller.export_linear_aec_output = false;
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apmConfig.echo_canceller.enforce_high_pass_filtering = true;
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//NoiseSuppression
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apmConfig.noise_suppression.enabled = true;
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apmConfig.noise_suppression.level = webrtc::AudioProcessing::Config::NoiseSuppression::kHigh;
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apmConfig.noise_suppression.analyze_linear_aec_output_when_available = false;
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//TransientSuppression
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apmConfig.transient_suppression.enabled = false;
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//VoiceDetection
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apmConfig.voice_detection.enabled = true;
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//GainController1
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apmConfig.gain_controller1.enabled = true;
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// kAdaptiveAnalog 自适应模拟模式
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// kAdaptiveDigital 自适应数字增益模式
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// kFixedDigital 固定数字增益模式
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apmConfig.gain_controller1.mode = webrtc::AudioProcessing::Config::GainController1::kAdaptiveDigital;
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apmConfig.gain_controller1.target_level_dbfs = 3; // 目标音量
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apmConfig.gain_controller1.compression_gain_db = 12; // 增益能力
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apmConfig.gain_controller1.enable_limiter = true; // 压限器开关
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apmConfig.gain_controller1.analog_level_minimum = 0;
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apmConfig.gain_controller1.analog_level_maximum = 496;
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apmConfig.gain_controller1.analog_gain_controller.enabled = true;
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apmConfig.gain_controller1.analog_gain_controller.startup_min_volume = webrtc::kAgcStartupMinVolume;
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apmConfig.gain_controller1.analog_gain_controller.clipped_level_min = webrtc::kClippedLevelMin;
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apmConfig.gain_controller1.analog_gain_controller.enable_agc2_level_estimator = false;
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apmConfig.gain_controller1.analog_gain_controller.enable_digital_adaptive = true;
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//GainController2
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apmConfig.gain_controller2.enabled = false;
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apmConfig.gain_controller2.fixed_digital.gain_db = 0.f;
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apmConfig.gain_controller2.adaptive_digital.enabled = false;
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apmConfig.gain_controller2.adaptive_digital.vad_probability_attack = 1.f;
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apmConfig.gain_controller2.adaptive_digital.level_estimator = webrtc::AudioProcessing::Config::GainController2::kRms;
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apmConfig.gain_controller2.adaptive_digital.level_estimator_adjacent_speech_frames_threshold = 1;
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apmConfig.gain_controller2.adaptive_digital.use_saturation_protector = true;
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apmConfig.gain_controller2.adaptive_digital.initial_saturation_margin_db = 20.f;
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apmConfig.gain_controller2.adaptive_digital.extra_saturation_margin_db = 2.f;
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apmConfig.gain_controller2.adaptive_digital.gain_applier_adjacent_speech_frames_threshold = 1;
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apmConfig.gain_controller2.adaptive_digital.max_gain_change_db_per_second = 3.f;
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apmConfig.gain_controller2.adaptive_digital.max_output_noise_level_dbfs = -50.f;
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//ResidualEchoDetector
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apmConfig.residual_echo_detector.enabled = false;
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//LevelEstimation
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apmConfig.level_estimation.enabled = false;
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return apmConfig;
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}
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void pushDestory(RtmpConfig *config) {
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if (config->formatCtx) {
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if (config->formatCtx->pb)
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avio_close(config->formatCtx->pb);
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avformat_free_context(config->formatCtx);
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}
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if (config->codecCtx) {
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avcodec_close(config->codecCtx);
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avcodec_free_context(&config->codecCtx);
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}
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if (config->swrCtx) {
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swr_close(config->swrCtx);
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swr_free(&config->swrCtx);
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}
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}
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int pushInit(RtmpConfig *config, int channels, AVSampleFormat format, int sample_rate)
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{
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if (nullptr == strstr(config->url, "rtmp://")) {
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PrintE("url error, url: %s\n", config->url);
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return -1;
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}
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AVCodec *codec = nullptr;
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AVCodecContext *codecCtx = nullptr;
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AVFormatContext *afctx = nullptr;
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AVCodecParameters *codecPar = nullptr;
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SwrContext *swrCtx = nullptr;
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AVStream *audio_st = nullptr;
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AVDictionary *opts = nullptr;
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int ret;
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// 打开输出流
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ret = avformat_alloc_output_context2(&afctx, nullptr, "flv", config->url);
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if (ret < 0) {
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PrintE("open output failed.\n");
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goto fail;
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}
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if ( !(afctx->oformat->flags & AVFMT_NOFILE) ) {
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ret = avio_open(&afctx->pb, config->url, AVIO_FLAG_WRITE);
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if (ret < 0) {
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PrintE("avio_open failed.\n");
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goto fail;
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}
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}
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// 创建音频流
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audio_st = avformat_new_stream(afctx, codec);
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if (!audio_st) {
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PrintE("alloc new audio stream failed.\n");
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goto fail;
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}
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// 设置编码参数
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codecPar = afctx->streams[audio_st->index]->codecpar;
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codecPar->codec_id = AV_CODEC_ID_AAC;
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codecPar->codec_type = AVMEDIA_TYPE_AUDIO;
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codecPar->codec_tag = 0;
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codecPar->bit_rate = 128 * 1024;
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codecPar->sample_rate = sample_rate;
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codecPar->channel_layout = av_get_default_channel_layout(channels);
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codecPar->channels = av_get_channel_layout_nb_channels(codecPar->channel_layout);
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codecPar->format = AV_SAMPLE_FMT_FLTP;
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// 编码器初始化
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codec = avcodec_find_encoder(codecPar->codec_id);
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if (!codec) {
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PrintE("find codec aac failed.\n");
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return -1;
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}
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codecCtx = avcodec_alloc_context3(codec);
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if (!codecCtx) {
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PrintE("alloc codec context failed.\n");
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goto fail;
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}
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ret = avcodec_parameters_to_context(codecCtx, codecPar);
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if (ret < 0) {
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PrintE("copt codec params failed.\n");
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goto fail;
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}
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// 禁用缓冲
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av_dict_set(&opts, "fflags", "nobuffer", AV_DICT_MATCH_CASE);
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// av_dict_set(&opts, "rtmp_live", "1", AV_DICT_MATCH_CASE);
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// 打开编码器
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ret = avcodec_open2(codecCtx, codec, &opts);
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if (ret < 0) {
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PrintE("open codec {} failed.\n", codec->id);
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goto fail;
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}
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audio_st->codecpar->codec_tag = 0;
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// 释放字典资源
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av_dict_free(&opts);
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// 打印输出流信息
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av_dump_format(afctx, 0, config->url, 1);
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// 重采样初始化
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swrCtx = swr_alloc_set_opts(nullptr,
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codecCtx->channel_layout,
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codecCtx->sample_fmt,
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codecCtx->sample_rate,
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av_get_default_channel_layout(channels),
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format,
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sample_rate,
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0, nullptr);
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if (!swrCtx) {
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PrintE("swr_alloc_set_opts failed.\n");
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goto fail;
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}
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swr_init(swrCtx);
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config->codecCtx = codecCtx;
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config->formatCtx = afctx;
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config->stream = audio_st;
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config->swrCtx = swrCtx;
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PrintI("rtmp push init ok.\n");
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return 0;
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fail:
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if (afctx) {
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if (afctx->pb)
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avio_close(afctx->pb);
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avformat_free_context(afctx);
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}
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if (codecCtx) {
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avcodec_close(codecCtx);
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avcodec_free_context(&codecCtx);
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}
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if (swrCtx) {
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swr_close(swrCtx);
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swr_free(&swrCtx);
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}
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return -1;
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}
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int main(int argc, char *argv[])
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{
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if (argc < 4) {
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fprintf(stderr, "usage %s card_num left(0) or right(1) url \n", argv[0]);
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return -1;
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}
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//初始化日志系统
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Logger::Instance().add(std::make_shared<ConsoleChannel> ());
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Logger::Instance().add(std::make_shared<FileChannel>());
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Logger::Instance().setWriter(std::make_shared<AsyncLogWriter>());
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// 初始化声卡设备
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|
|
int card = atoi(argv[1]);
|
|||
|
|
int type = atoi(argv[2]);
|
|||
|
|
alsa::Config alsaConfig;
|
|||
|
|
alsaConfig.period_time = MIX_INPUT_SAMPLES * 1000000 / MIX_INPUT_SAMPLE_RATE;
|
|||
|
|
alsaConfig.buffer_time = 5 * alsaConfig.period_time;
|
|||
|
|
alsaConfig.channels = MIX_INPUT_CHANNELS;
|
|||
|
|
alsaConfig.format = SND_PCM_FORMAT_S16_LE;
|
|||
|
|
alsaConfig.rate = MIX_INPUT_SAMPLE_RATE;
|
|||
|
|
if (card < 0)
|
|||
|
|
sprintf(alsaConfig.device, "default");
|
|||
|
|
else
|
|||
|
|
sprintf(alsaConfig.device, "plughw:%d", card);
|
|||
|
|
alsa::AlsaDev usbCaptureDev;
|
|||
|
|
if (usbCaptureDev.applyConfig(alsaConfig) < 0) {
|
|||
|
|
PrintE("alsa config failed.\n");
|
|||
|
|
return -1;
|
|||
|
|
}
|
|||
|
|
PrintI("alsa before init: %s\n", usbCaptureDev.configToString());
|
|||
|
|
if (usbCaptureDev.init(SND_PCM_STREAM_CAPTURE) < 0) {
|
|||
|
|
PrintE("alsa init failed.\n");
|
|||
|
|
return -1;
|
|||
|
|
}
|
|||
|
|
PrintI("alsa init: %s\n", usbCaptureDev.configToString());
|
|||
|
|
|
|||
|
|
// webrtc初始化
|
|||
|
|
webrtc::AudioProcessing *apm = webrtc::AudioProcessingBuilder().Create();
|
|||
|
|
if (!apm) {
|
|||
|
|
PrintI("create apm failed.\n");
|
|||
|
|
return -1;
|
|||
|
|
}
|
|||
|
|
webrtc::AudioProcessing::Config config = webtcConfigInit();
|
|||
|
|
apm->ApplyConfig(config);
|
|||
|
|
apm->Initialize();
|
|||
|
|
apm->set_stream_analog_level(408);
|
|||
|
|
webrtc::StreamConfig capConfig;
|
|||
|
|
capConfig.set_has_keyboard(false);
|
|||
|
|
capConfig.set_num_channels(alsaConfig.channels);
|
|||
|
|
capConfig.set_sample_rate_hz(alsaConfig.rate);
|
|||
|
|
PrintI("webrtc params: {\n%s\n}\n", config.ToString().c_str());
|
|||
|
|
|
|||
|
|
int ret = 0;
|
|||
|
|
uint8_t *capData = nullptr;
|
|||
|
|
int buffer_size = usbCaptureDev.getFrames() * usbCaptureDev.getFrameSize();
|
|||
|
|
capData = (uint8_t *)malloc(buffer_size);
|
|||
|
|
assert(capData);
|
|||
|
|
uint8_t *listenBuffer = nullptr;
|
|||
|
|
size_t listenSize = buffer_size;
|
|||
|
|
listenBuffer = (uint8_t*)malloc(listenSize);
|
|||
|
|
assert(listenBuffer);
|
|||
|
|
|
|||
|
|
|
|||
|
|
// 推流初始化
|
|||
|
|
RtmpConfig rtmp;
|
|||
|
|
AVRational av;
|
|||
|
|
int64_t pts = 0;
|
|||
|
|
AVPacket *pkt = av_packet_alloc();
|
|||
|
|
memset(&rtmp, 0, sizeof(rtmp));
|
|||
|
|
strcpy(rtmp.url, argv[3]);
|
|||
|
|
if (pushInit(&rtmp, MIX_INPUT_CHANNELS, AV_SAMPLE_FMT_S16, MIX_INPUT_SAMPLE_RATE) < 0) {
|
|||
|
|
return -1;
|
|||
|
|
}
|
|||
|
|
av.den = rtmp.codecCtx->sample_rate;
|
|||
|
|
av.num = 1;
|
|||
|
|
AVFrame *inputFrame = av_frame_alloc();
|
|||
|
|
{
|
|||
|
|
inputFrame->sample_rate = rtmp.codecCtx->sample_rate;
|
|||
|
|
inputFrame->format = rtmp.codecCtx->sample_fmt;
|
|||
|
|
inputFrame->channels = rtmp.codecCtx->channels;
|
|||
|
|
inputFrame->nb_samples = 1024;
|
|||
|
|
inputFrame->channel_layout = rtmp.codecCtx->channel_layout;
|
|||
|
|
|
|||
|
|
int size = av_samples_get_buffer_size(nullptr,
|
|||
|
|
rtmp.codecCtx->channels, inputFrame->nb_samples, rtmp.codecCtx->sample_fmt, 1);
|
|||
|
|
uint8_t *buffer = (uint8_t *)av_malloc(size);
|
|||
|
|
avcodec_fill_audio_frame(inputFrame, inputFrame->channels, rtmp.codecCtx->sample_fmt,
|
|||
|
|
(const uint8_t*)buffer, size, 1);
|
|||
|
|
}
|
|||
|
|
AVFrame *outputFrame = av_frame_alloc();
|
|||
|
|
{
|
|||
|
|
outputFrame->format = rtmp.codecCtx->sample_fmt;
|
|||
|
|
outputFrame->channel_layout = rtmp.codecCtx->channel_layout;
|
|||
|
|
outputFrame->sample_rate = rtmp.codecCtx->sample_rate;
|
|||
|
|
outputFrame->nb_samples = rtmp.codecCtx->frame_size;
|
|||
|
|
|
|||
|
|
int output_bz = av_samples_get_buffer_size(NULL, rtmp.codecCtx->channels, rtmp.codecCtx->frame_size, rtmp.codecCtx->sample_fmt, 0);
|
|||
|
|
uint8_t *samples_data = (uint8_t *)av_malloc(output_bz);
|
|||
|
|
avcodec_fill_audio_frame(outputFrame, rtmp.codecCtx->channels, rtmp.codecCtx->sample_fmt, samples_data, output_bz, 0);
|
|||
|
|
}
|
|||
|
|
|
|||
|
|
// 写入帧头
|
|||
|
|
ret = avformat_write_header(rtmp.formatCtx, nullptr);
|
|||
|
|
if (ret < 0) {
|
|||
|
|
PrintE("avformat_write_header failed.\n");
|
|||
|
|
return ret;
|
|||
|
|
}
|
|||
|
|
|
|||
|
|
FILE *ns_fp = fopen("/root/ns_out.pcm", "wb");
|
|||
|
|
FILE *input_fp = fopen("/root/input_out.pcm", "wb");
|
|||
|
|
std::vector<audio_buf_t> ns_out_list;
|
|||
|
|
int frames = 0;
|
|||
|
|
PrintI("input frame samples=%d, output frame samples=%d\n", inputFrame->nb_samples, outputFrame->nb_samples);
|
|||
|
|
PrintI("------------------ start ------------------\n");
|
|||
|
|
while (true)
|
|||
|
|
{
|
|||
|
|
// 采集
|
|||
|
|
size_t read_size = usbCaptureDev.read(capData, buffer_size);
|
|||
|
|
// PrintI("alsa read %d\n", read_size);
|
|||
|
|
if (read_size <= 0) {
|
|||
|
|
msleep(1);
|
|||
|
|
continue;
|
|||
|
|
}
|
|||
|
|
|
|||
|
|
// 分流
|
|||
|
|
if (type >= 0) {
|
|||
|
|
uint8_t *ptr = capData;
|
|||
|
|
for (int i = 0; i < usbCaptureDev.getFrames(); ++i) {
|
|||
|
|
// int size = av_get_bytes_per_sample(AV_SAMPLE_FMT_S16);
|
|||
|
|
int size = usbCaptureDev.getSampleSize();
|
|||
|
|
if (type == 0) {
|
|||
|
|
// 左声道 (2.4G)
|
|||
|
|
memcpy(listenBuffer + i * 2 * size, ptr + i * 2 * size, size);
|
|||
|
|
memcpy(listenBuffer + (i * 2 + 1) * size, ptr + i * 2 * size, size);
|
|||
|
|
} else {
|
|||
|
|
// 右声道 (LINE IN)
|
|||
|
|
memcpy(listenBuffer + i * 2 * size, ptr + (i * 2 + 1) * size, size);
|
|||
|
|
memcpy(listenBuffer + (i * 2 + 1) * size, ptr + (i * 2 + 1) * size, size);
|
|||
|
|
}
|
|||
|
|
}
|
|||
|
|
} else {
|
|||
|
|
memcpy(listenBuffer, capData, read_size * usbCaptureDev.getFrameSize());
|
|||
|
|
}
|
|||
|
|
|
|||
|
|
|
|||
|
|
// vol_scaler_run((int16_t *)listenBuffer, capData->GetSamples() * kPcmSampleInfo.channels, kVolCtrlUnit.micVolume);
|
|||
|
|
|
|||
|
|
// 降噪
|
|||
|
|
apm->ProcessStream((int16_t *)listenBuffer, capConfig, capConfig, (int16_t *)listenBuffer);
|
|||
|
|
{
|
|||
|
|
uint8_t *buffer = (uint8_t *)malloc(listenSize);
|
|||
|
|
memcpy(buffer, listenBuffer, listenSize);
|
|||
|
|
audio_buf_t out;
|
|||
|
|
out.data = buffer;
|
|||
|
|
out.index = 0;
|
|||
|
|
out.size = listenSize;
|
|||
|
|
ns_out_list.emplace_back(out);
|
|||
|
|
}
|
|||
|
|
|
|||
|
|
// 填充音频
|
|||
|
|
if (frames <= 0) frames = inputFrame->nb_samples;
|
|||
|
|
while (frames > 0 && ns_out_list.size() > 0)
|
|||
|
|
{
|
|||
|
|
auto nsData = ns_out_list.begin();
|
|||
|
|
|
|||
|
|
|
|||
|
|
int needSize = frames * sizeof(int16_t) * inputFrame->channels;
|
|||
|
|
int readSize = (nsData->size - nsData->index) >= needSize ? needSize : (nsData->size - nsData->index);
|
|||
|
|
|
|||
|
|
memcpy(inputFrame->data[0] + (inputFrame->nb_samples - frames)*sizeof(int16_t)*inputFrame->channels, nsData->data + nsData->index, readSize);
|
|||
|
|
|
|||
|
|
frames -= readSize/(sizeof(int16_t) * inputFrame->channels);
|
|||
|
|
nsData->index += readSize;
|
|||
|
|
|
|||
|
|
if (nsData->index >= nsData->size) {
|
|||
|
|
free(nsData->data);
|
|||
|
|
ns_out_list.erase(ns_out_list.begin());
|
|||
|
|
}
|
|||
|
|
}
|
|||
|
|
if (frames > 0) continue;
|
|||
|
|
// if (input_fp) fwrite(inputFrame->data[0], 1, inputFrame->nb_samples * inputFrame->channels * sizeof(int16_t), input_fp);
|
|||
|
|
// 重采样
|
|||
|
|
{
|
|||
|
|
const uint8_t** in = (const uint8_t**)inputFrame->data;
|
|||
|
|
uint8_t **out = outputFrame->data;
|
|||
|
|
|
|||
|
|
int len2, out_data_size;
|
|||
|
|
|
|||
|
|
len2 = swr_convert(rtmp.swrCtx, out, outputFrame->nb_samples, in, inputFrame->nb_samples);
|
|||
|
|
if (len2 < 0) {
|
|||
|
|
printf("swr_convert failed. \n");
|
|||
|
|
break;
|
|||
|
|
}
|
|||
|
|
|
|||
|
|
out_data_size = len2 * rtmp.codecCtx->channels * av_get_bytes_per_sample(rtmp.codecCtx->sample_fmt);
|
|||
|
|
// if (ns_fp) fwrite(outputFrame->data[0], 1, out_data_size, ns_fp);
|
|||
|
|
}
|
|||
|
|
|
|||
|
|
|
|||
|
|
// 推流到远端
|
|||
|
|
if (pts > INT64_MAX) pts = 0;
|
|||
|
|
outputFrame->pts = pts;
|
|||
|
|
pts += av_rescale_q(outputFrame->nb_samples, av, rtmp.codecCtx->time_base);
|
|||
|
|
|
|||
|
|
ret = avcodec_send_frame(rtmp.codecCtx, outputFrame);
|
|||
|
|
if (ret < 0) {
|
|||
|
|
PrintE("avcodec_send_frame failed: %d\n", ret);
|
|||
|
|
break;
|
|||
|
|
}
|
|||
|
|
|
|||
|
|
while (ret >= 0) {
|
|||
|
|
ret = avcodec_receive_packet(rtmp.codecCtx, pkt);
|
|||
|
|
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
|
|||
|
|
break;
|
|||
|
|
} else if (ret < 0) {
|
|||
|
|
fprintf(stderr, "Error during encoding\n");
|
|||
|
|
break;
|
|||
|
|
}
|
|||
|
|
|
|||
|
|
// 将数据包时间戳从编码器时间基转换到流时间基
|
|||
|
|
pkt->stream_index = rtmp.stream->index;
|
|||
|
|
av_packet_rescale_ts(pkt, rtmp.codecCtx->time_base, rtmp.stream->time_base);
|
|||
|
|
pkt->duration = av_rescale_q(pkt->duration, rtmp.codecCtx->time_base, rtmp.stream->time_base);
|
|||
|
|
|
|||
|
|
// 写入数据包到输出媒体文件
|
|||
|
|
ret = av_interleaved_write_frame(rtmp.formatCtx, pkt);
|
|||
|
|
if (ret < 0) {
|
|||
|
|
fprintf(stderr, "Error while writing audio frame\n");
|
|||
|
|
break;
|
|||
|
|
}
|
|||
|
|
|
|||
|
|
// 释放数据包
|
|||
|
|
av_packet_unref(pkt);
|
|||
|
|
}
|
|||
|
|
}
|
|||
|
|
// 写入帧尾
|
|||
|
|
av_write_trailer(rtmp.formatCtx);
|
|||
|
|
|
|||
|
|
if (apm) {
|
|||
|
|
delete apm;
|
|||
|
|
apm = nullptr;
|
|||
|
|
}
|
|||
|
|
pushDestory(&rtmp);
|
|||
|
|
av_packet_free(&pkt);
|
|||
|
|
av_frame_free(&inputFrame);
|
|||
|
|
av_frame_free(&outputFrame);
|
|||
|
|
av_free(listenBuffer);
|
|||
|
|
for (auto buffer: ns_out_list) {
|
|||
|
|
free(buffer.data);
|
|||
|
|
}
|
|||
|
|
return 0;
|
|||
|
|
}
|