File_Stream_deal.c
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#include "AudioDeal/FF_Statement.h"
#include "../Resample/pcm_Resample.h"
#include "AudioDeal/AudioDeal.h"
#include "AudioDeal/Filter/FF_Filter.h"
#include "JZsdkLib.h"
#include "AudioDeal/Alsa/pcm_AlsaPlay.h"
/******************************
*
* 用于处理mp3流的.c
* 输入mp3数据 → 解码成pcm → pcm流处理模块
*
* ******************************/
static AVCodecParserContext *parser = NULL;
static AVCodecContext *cdc_ctx = NULL;
static AVPacket *pkt;
static AVFrame *decoded_frame = NULL;
static const AVCodec *codec;
T_JZsdkReturnCode Stream_Player_decode(struct AudioDealInfo *AD_Info, AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame);
T_JZsdkReturnCode File_Stream_deal_Init(enum AVCodecID id)
{
pkt = av_packet_alloc();
if(!pkt)
{
JZSDK_LOG_ERROR("av_packet_alloc failed.");
}
codec = avcodec_find_decoder(id);
if (!codec) {
JZSDK_LOG_ERROR("Codec not found\n");
}
parser = av_parser_init(codec->id);
if (!parser) {
JZSDK_LOG_ERROR("Parser not found\n");
}
cdc_ctx = avcodec_alloc_context3(codec);
if (!cdc_ctx) {
JZSDK_LOG_ERROR("Could not allocate audio codec context\n");
}
/* open it */
if (avcodec_open2(cdc_ctx, codec, NULL) < 0)
{
JZSDK_LOG_ERROR("Could not open codec\n");
}
JZSDK_LOG_INFO("File_Stream_deal_Init success");
}
//输入mp3的实时数据，以及本次数据的长度
T_JZsdkReturnCode mp3_Stream_Interface_Mp3Data(struct AudioDealInfo *AD_Info, unsigned int in_sampleRate, unsigned char *data, int dataSize)
{
//JZSDK_LOG_DEBUG("mp3_Stream_Interface_Mp3Data");
T_JZsdkReturnCode ret = JZ_ERROR_SYSTEM_MODULE_CODE_SUCCESS;
//将数据输入到
while(dataSize > 0 && (AD_Info->AudioDeal_ResampleAndFilter_Execute_Flag == JZ_FLAGCODE_ON))
{
//printf("开始解码\n");
//如果解码器不存在，初始化解码器
if (!decoded_frame)
{
if (!(decoded_frame = av_frame_alloc()))
{
JZSDK_LOG_ERROR("Could not allocate audio frame\n");
return JZ_ERROR_SYSTEM_MODULE_CODE_FAILURE;
}
}
//printf("alloc后\n");
//检查参数，并将正确的数据输入到pkt中
//parser 解析器
//cdc_ctx 上下文
//pkt输出的数据指针
//data datasize 输入的数据指针
//pts、dts、pos：时间戳和位置信息，一般可以设置为AV_NOPTS_VALUE和0。
int ret = av_parser_parse2(parser, cdc_ctx, &pkt->data, &pkt->size, data, dataSize, AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);
if (ret < 0) {
JZSDK_LOG_ERROR("Error while parsing\n");
return JZ_ERROR_SYSTEM_MODULE_CODE_FAILURE;
}
//printf("av_parser_parse2\n");
//重置重采样器
FF_Resample_Reset(AD_Info, in_sampleRate, (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO, AV_SAMPLE_FMT_S16);
//printf("FF_Resample_Reset\n");
//检查滤波器
FF_Filter_Init(AD_Info, 0x01);
//printf("FF_Filter_Init %d \n", pkt->size );
//数据指针 往后一个解析长度
//长度指针 减少一个被解析数据的长度
data += ret;
dataSize -= ret;
//如果输出有长度 解码输出的数据
if (pkt->size > 0)
{
ret = Stream_Player_decode(AD_Info, cdc_ctx, pkt, decoded_frame);
if(ret == JZ_ERROR_SYSTEM_MODULE_CODE_HARDLY_FAILURE)
{
return JZ_ERROR_SYSTEM_MODULE_CODE_HARDLY_FAILURE;
}
}
}
}
//
T_JZsdkReturnCode Stream_Player_decode(struct AudioDealInfo *AD_Info, AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame)
{
int ret;
//JZSDK_LOG_DEBUG("Stream_Player_decode");
//发送数据包给解码器解码，已将数据解码为pcm原始数据
ret = avcodec_send_packet(dec_ctx, pkt);
if (ret < 0)
{
JZSDK_LOG_ERROR("Error submitting the packet to the decoder, ret=%d\n",ret);
//return JZ_ERROR_SYSTEM_MODULE_CODE_HARDLY_FAILURE;
return JZ_ERROR_SYSTEM_MODULE_CODE_FAILURE;
}
//申请内存
AVFrame *eq_frame = av_frame_alloc();
AVFrame *temp_frame = av_frame_alloc();
//JZSDK_LOG_DEBUG("alloc");
/* read all the output frames (in general there may be any number of them */
//读取输出的帧
while ( (ret >= 0) && (AD_Info->AudioDeal_ResampleAndFilter_Execute_Flag == JZ_FLAGCODE_ON) )
{
//从解码器中读取解码后的音频帧数据
ret = avcodec_receive_frame(dec_ctx, frame);
//输出帧不可用 、 输出帧 已用完
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
{
continue;;
}
else if (ret < 0)
{
JZSDK_LOG_ERROR("Error during decoding\n");
//释放掉输出的变量
av_frame_free(&eq_frame);
av_frame_free(&temp_frame);
//JZSDK_LOG_DEBUG("提前释放后\n");
return JZ_ERROR_SYSTEM_MODULE_CODE_FAILURE;
}
int out_nb_samples = 0;
//JZSDK_LOG_DEBUG("播放前");
//重采样解码后的数据
void* resampledData = FF_Resample_Send_And_Get_ResampleData(AD_Info, frame->data, frame->nb_samples, &out_nb_samples);
//滤波
if(AD_Info->FilterMode != JZ_FLAGCODE_OFF)
{
temp_frame->data[0] = (unsigned char*)resampledData;
temp_frame->nb_samples = out_nb_samples;
//将临时帧 放入 均衡滤波器
FF_Filter_push_frame_to_fliter(AD_Info, temp_frame);
while(AD_Info->AudioDeal_ResampleAndFilter_Execute_Flag == JZ_FLAGCODE_ON)
{
//得到滤波器输出的音频帧 eq_frame
int fret = FF_Filter_get_frame_from_filter(AD_Info, eq_frame);
if (fret != JZ_ERROR_SYSTEM_MODULE_CODE_SUCCESS)
{
break;
}
//播放改滤波后的帧
Pcm_AlsaPlay(AD_Info, (unsigned char*)eq_frame->data[0], eq_frame->nb_samples);
av_frame_unref(eq_frame);
}
av_frame_unref(temp_frame);
}
else //不滤波
{
//直接播放
//JZSDK_LOG_INFO("播放 %d 数据",out_nb_samples);
Pcm_AlsaPlay(AD_Info ,resampledData, out_nb_samples);
}
FF_Resample_freeReasmpleData(resampledData);
}
//JZSDK_LOG_DEBUG("怀疑释放前\n");
//释放掉输出的变量
av_frame_unref(temp_frame);
av_frame_unref(eq_frame);
//JZSDK_LOG_DEBUG("怀疑释放后\n");
}
T_JZsdkReturnCode mp3_Stream_Interface_Mp3Data_Stop(struct AudioDealInfo *AD_Info)
{
// pkt->data = NULL;
// pkt->size = 0;
// Stream_Player_decode(AD_Info, cdc_ctx, pkt, decoded_frame);
}