我正在尝试用android中的ffmpeg和opensles播放音频流。问题似乎是将解码和重新采样的帧从ffmpeg传递给opensles，因为我听到的声音听起来像机器人并且有刮痕。
来自ffmpeg的解码帧：
PCM
48000 Hz
S16p
在这种情况下需要打开：
PCM
48000 Hz
S16
开放式设置：
SLDataLocator_AndroidSimpleBufferQueue loc_bufq = {SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, 255};
SLDataFormat_PCM format_pcm = { SL_DATAFORMAT_PCM, 2 , SL_SAMPLINGRATE_48, SL_PCMSAMPLEFORMAT_FIXED_16, SL_PCMSAMPLEFORMAT_FIXED_16,
SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT, SL_BYTEORDER_LITTLEENDIAN};
SLDataSource audioSrc = {&loc_bufq, &format_pcm};
这是重新取样和排队到openles的伪代码：
#define OPENSLES_BUFLEN 10
#define MAX_AUDIO_FRAME_SIZE 192000
DECLARE_ALIGNED(16,uint8_t,audio_buffer)[MAX_AUDIO_FRAME_SIZE * OPENSLES_BUFLEN];
int decode_audio(AVCodecContext * ctx, SwrContext *swr_context, AVPacket *packet, AVFrame * frame){
int got_frame_ptr;
int len = avcodec_decode_audio4(ctx, frame, &got_frame_ptr, packet);
if(!got_frame_ptr)
return -ERROR;
int original_data_size = av_samples_get_buffer_size(NULL, ctx->channels,
frame->nb_samples, ctx->sample_fmt, 1);
uint8_t *audio_buf;
int data_size;
if (swr_context != NULL) {
uint8_t *out[] = { audio_buffer };
int sample_per_buffer_divider = 2* av_get_bytes_per_sample(AV_SAMPLE_FMT_S16);;
int len2 = swr_convert(swr_context, out,
sizeof(audio_buffer) / sample_per_buffer_divider,
frame->extended_data, frame->nb_samples);
if (len2 < 0) {
return -ERROR;
}
if (len2 == sizeof(audio_buffer) / sample_per_buffer_divider) {
swr_init(swr_context);
}
audio_buf = audio_buffer;
data_size = len2 * sample_per_buffer_divider;
}
else {
audio_buf = frame->data[0];
data_size = original_data_size;
}
(*opengSLESData->bqPlayerBufferQueue)->Enqueue(opengSLESData->bqPlayerBufferQueue, audio_buf, data_size)
}
感谢您的帮助，谢谢。
答案 0 :(得分：0)
示例可能有帮助
#include "stdafx.h"
#include <iostream>
extern "C"
{
#include "libavcodec/avcodec.h"
#include "libavformat/avformat.h"
//#include "swscale.h"
#include "libswresample/swresample.h"
};
FILE *fin, *fout;
int ffmpeg_audio_decode( const char * inFile, const char * outFile)
{
// Initialize FFmpeg
av_register_all();
AVFrame* frame = avcodec_alloc_frame();
if (!frame)
{
std::cout << "Error allocating the frame" << std::endl;
return 1;
}
// you can change the file name "01 Push Me to the Floor.wav" to whatever the file is you're reading, like "myFile.ogg" or
// "someFile.webm" and this should still work
AVFormatContext* formatContext = NULL;
//if (avformat_open_input(&formatContext, "01 Push Me to the Floor.wav", NULL, NULL) != 0)
if (avformat_open_input(&formatContext, inFile, NULL, NULL) != 0)
{
av_free(frame);
std::cout << "Error opening the file" << std::endl;
return 1;
}
if (avformat_find_stream_info(formatContext, NULL) < 0)
{
av_free(frame);
av_close_input_file(formatContext);
std::cout << "Error finding the stream info" << std::endl;
return 1;
}
AVStream* audioStream = NULL;
// Find the audio stream (some container files can have multiple streams in them)
for (unsigned int i = 0; i < formatContext->nb_streams; ++i)
{
if (formatContext->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO)
{
audioStream = formatContext->streams[i];
break;
}
}
if (audioStream == NULL)
{
av_free(frame);
av_close_input_file(formatContext);
std::cout << "Could not find any audio stream in the file" << std::endl;
return 1;
}
AVCodecContext* codecContext = audioStream->codec;
codecContext->codec = avcodec_find_decoder(codecContext->codec_id);
if (codecContext->codec == NULL)
{
av_free(frame);
av_close_input_file(formatContext);
std::cout << "Couldn't find a proper decoder" << std::endl;
return 1;
}
else if (avcodec_open2(codecContext, codecContext->codec, NULL) != 0)
{
av_free(frame);
av_close_input_file(formatContext);
std::cout << "Couldn't open the context with the decoder" << std::endl;
return 1;
}
std::cout << "This stream has " << codecContext->channels << " channels and a sample rate of " << codecContext->sample_rate << "Hz" << std::endl;
std::cout << "The data is in the format " << av_get_sample_fmt_name(codecContext->sample_fmt) << std::endl;
//codecContext->sample_fmt = AV_SAMPLE_FMT_S16;
int64_t outChannelLayout = AV_CH_LAYOUT_MONO; //AV_CH_LAYOUT_STEREO;
AVSampleFormat outSampleFormat = AV_SAMPLE_FMT_S16; // Packed audio, non-planar (this is the most common format, and probably what you want; also, WAV needs it)
int outSampleRate = 8000;//44100;
// Note that AVCodecContext::channel_layout may or may not be set by libavcodec. Because of this,
// we won't use it, and will instead try to guess the layout from the number of channels.
SwrContext* swrContext = swr_alloc_set_opts(NULL,
outChannelLayout,
outSampleFormat,
outSampleRate,
av_get_default_channel_layout(codecContext->channels),
codecContext->sample_fmt,
codecContext->sample_rate,
0,
NULL);
if (swrContext == NULL)
{
av_free(frame);
avcodec_close(codecContext);
avformat_close_input(&formatContext);
std::cout << "Couldn't create the SwrContext" << std::endl;
return 1;
}
if (swr_init(swrContext) != 0)
{
av_free(frame);
avcodec_close(codecContext);
avformat_close_input(&formatContext);
swr_free(&swrContext);
std::cout << "Couldn't initialize the SwrContext" << std::endl;
return 1;
}
fout = fopen(outFile, "wb+");
AVPacket packet;
av_init_packet(&packet);
// Read the packets in a loop
while (av_read_frame(formatContext, &packet) == 0)
{
if (packet.stream_index == audioStream->index)
{
AVPacket decodingPacket = packet;
while (decodingPacket.size > 0)
{
// Try to decode the packet into a frame
int frameFinished = 0;
int result = avcodec_decode_audio4(
codecContext,
frame,
&frameFinished,
&decodingPacket);
if (result < 0 || frameFinished == 0)
{
break;
}
unsigned char buffer[100000] = {NULL};
unsigned char* pointers[SWR_CH_MAX] = {NULL};
pointers[0] = &buffer[0];
int numSamplesOut = swr_convert(
swrContext,
pointers,
outSampleRate,
(const unsigned char**)frame->extended_data,
frame->nb_samples);
fwrite(
(short *)buffer,
sizeof(short),
(size_t)numSamplesOut,
fout);
decodingPacket.size -= result;
decodingPacket.data += result;
}
}
// You *must* call av_free_packet() after each call to av_read_frame() or else you'll leak memory
av_free_packet(&packet);
}
// Some codecs will cause frames to be buffered up in the decoding process. If the CODEC_CAP_DELAY flag
// is set, there can be buffered up frames that need to be flushed, so we'll do that
if (codecContext->codec->capabilities & CODEC_CAP_DELAY)
{
av_init_packet(&packet);
// Decode all the remaining frames in the buffer, until the end is reached
int frameFinished = 0;
while (avcodec_decode_audio4(codecContext, frame, &frameFinished, &packet) >= 0 && frameFinished)
{
}
}
// Clean up!
av_free(frame);
avcodec_close(codecContext);
av_close_input_file(formatContext);
fclose(fout);
}