ffmpeg 版本:git clone 于 2014-12-02 ,版本接近2.44,在2.44和2.51之间
SDL版本:SDL 1.2(Centos 6.5软件库的相应版本)
有些旧的ffmpeg播放音频示例中,会存在一些音频可以播放一些不能播放,其中一个我们需要考虑的原因和该注意的地方就是 av_decode_audiole类似函数所获的的AVFrame的格式是否是我们(SDL)所需要的,本例代码用来解决该问题,关键点在于swr_convert函数,代码及注释如下:
</pre><pre name="code" class="cpp">#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libswscale/swscale.h>
#include <libswresample/swresample.h>
#include <libavutil/avstring.h>
#include <libavutil/pixfmt.h>
#include <libavutil/log.h>
#include <SDL/SDL.h>
#include <SDL/SDL_thread.h>
#include <stdio.h>
#include <math.h>
#define SDL_AUDIO_BUFFER_SIZE 1024
#define MAX_AUDIOQ_SIZE (1 * 1024 * 1024)
#define FF_ALLOC_EVENT (SDL_USEREVENT)
#define FF_REFRESH_EVENT (SDL_USEREVENT + 1)
#define FF_QUIT_EVENT (SDL_USEREVENT + 2)
//该字段存在于旧版本的ffmpeg中,此处粘贴过来使用,勿怪!
#define AVCODEC_MAX_AUDIO_FRAME_SIZE 192000 // 1 second of 48khz 32bit audio
typedef struct PacketQueue {
AVPacketList *first_pkt, *last_pkt;
int nb_packets;
int size;
SDL_mutex *mutex;
SDL_cond *cond;
} PacketQueue;
typedef struct VideoState {
char filename[1024];
AVFormatContext *ic;
int videoStream, audioStream;
AVStream *audio_st;
AVFrame *audio_frame;
PacketQueue audioq;
unsigned int audio_buf_size;
unsigned int audio_buf_index;
AVPacket audio_pkt;
uint8_t *audio_pkt_data;
int audio_pkt_size;
uint8_t *audio_buf;
uint8_t *audio_buf1;
DECLARE_ALIGNED(16,uint8_t,audio_buf2) [AVCODEC_MAX_AUDIO_FRAME_SIZE * 4];
enum AVSampleFormat audio_src_fmt;
enum AVSampleFormat audio_tgt_fmt;
int audio_src_channels;
int audio_tgt_channels;
int64_t audio_src_channel_layout;
int64_t audio_tgt_channel_layout;
int audio_src_freq;
int audio_tgt_freq;
struct SwrContext *swr_ctx;
SDL_Thread *parse_tid;
int quit;
} VideoState;
VideoState *global_video_state;
void packet_queue_init(PacketQueue *q) {
memset(q, 0, sizeof(PacketQueue));
q->mutex = SDL_CreateMutex();
q->cond = SDL_CreateCond();
}
int packet_queue_put(PacketQueue *q, AVPacket *pkt) {
AVPacketList *pkt1;
pkt1 = (AVPacketList *) av_malloc(sizeof(AVPacketList));
if (!pkt1) {
return -1;
}
pkt1->pkt = *pkt;
pkt1->next = NULL;
SDL_LockMutex(q->mutex);
if (!q->last_pkt) {
q->first_pkt = pkt1;
} else {
q->last_pkt->next = pkt1;
}
q->last_pkt = pkt1;
q->nb_packets++;
q->size += pkt1->pkt.size;
SDL_CondSignal(q->cond);
SDL_UnlockMutex(q->mutex);
return 0;
}
static int packet_queue_get(PacketQueue *q, AVPacket *pkt, int block) {
AVPacketList *pkt1;
int ret;
SDL_LockMutex(q->mutex);
for (;;) {
if (global_video_state->quit) {
ret = -1;
break;
}
pkt1 = q->first_pkt;
if (pkt1) {
q->first_pkt = pkt1->next;
if (!q->first_pkt) {
q->last_pkt = NULL;
}
q->nb_packets--;
q->size -= pkt1->pkt.size;
*pkt = pkt1->pkt;
av_free(pkt1);
ret = 1;
break;
} else if (!block) {
ret = 0;
break;
} else {
SDL_CondWait(q->cond, q->mutex);
}
}
SDL_UnlockMutex(q->mutex);
return ret;
}
int audio_decode_frame(VideoState *is) {
int len1, len2, decoded_data_size;
AVPacket *pkt = &is->audio_pkt;
int got_frame = 0;
int64_t dec_channel_layout;
int wanted_nb_samples, resampled_data_size;
for (;;) {
while (is->audio_pkt_size > 0) {
if (!is->audio_frame) {
if (!(is->audio_frame = av_frame_alloc())) {
return AVERROR(ENOMEM);
}
} else
av_frame_unref(is->audio_frame);
/**
* 当AVPacket中装得是音频时,有可能一个AVPacket中有多个AVFrame,
* 而某些解码器只会解出第一个AVFrame,这种情况我们必须循环解码出后续AVFrame
*/
len1 = avcodec_decode_audio4(is->audio_st->codec, is->audio_frame,
&got_frame, pkt);
if (len1 < 0) {
// error, skip the frame
is->audio_pkt_size = 0;
break;
}
is->audio_pkt_data += len1;
is->audio_pkt_size -= len1;
if (!got_frame)
continue;
//执行到这里我们得到了一个AVFrame
decoded_data_size = av_samples_get_buffer_size(NULL,
is->audio_frame->channels, is->audio_frame->nb_samples,
is->audio_frame->format, 1);
//得到这个AvFrame的声音布局,比如立体声
dec_channel_layout =
(is->audio_frame->channel_layout
&& is->audio_frame->channels
== av_get_channel_layout_nb_channels(
is->audio_frame->channel_layout)) ?
is->audio_frame->channel_layout :
av_get_default_channel_layout(
is->audio_frame->channels);
//这个AVFrame每个声道的采样数
wanted_nb_samples = is->audio_frame->nb_samples;
/**
* 接下来判断我们之前设置SDL时设置的声音格式(AV_SAMPLE_FMT_S16),声道布局,
* 采样频率,每个AVFrame的每个声道采样数与
* 得到的该AVFrame分别是否相同,如有任意不同,我们就需要swr_convert该AvFrame,
* 然后才能符合之前设置好的SDL的需要,才能播放
*/
if (is->audio_frame->format != is->audio_src_fmt
|| dec_channel_layout != is->audio_src_channel_layout
|| is->audio_frame->sample_rate != is->audio_src_freq
|| (wanted_nb_samples != is->audio_frame->nb_samples
&& !is->swr_ctx)) {
if (is->swr_ctx)
swr_free(&is->swr_ctx);
is->swr_ctx = swr_alloc_set_opts(NULL,
is->audio_tgt_channel_layout, is->audio_tgt_fmt,
is->audio_tgt_freq, dec_channel_layout,
is->audio_frame->format, is->audio_frame->sample_rate,
0, NULL);
if (!is->swr_ctx || swr_init(is->swr_ctx) < 0) {
fprintf(stderr, "swr_init() failed\n");
break;
}
is->audio_src_channel_layout = dec_channel_layout;
is->audio_src_channels = is->audio_st->codec->channels;
is->audio_src_freq = is->audio_st->codec->sample_rate;
is->audio_src_fmt = is->audio_st->codec->sample_fmt;
}
/**
* 如果上面if判断失败,就会初始化好swr_ctx,就会如期进行转换
*/
if (is->swr_ctx) {
// const uint8_t *in[] = { is->audio_frame->data[0] };
const uint8_t **in =
(const uint8_t **) is->audio_frame->extended_data;
uint8_t *out[] = { is->audio_buf2 };
if (wanted_nb_samples != is->audio_frame->nb_samples) {
fprintf(stdout, "swr_set_compensation \n");
if (swr_set_compensation(is->swr_ctx,
(wanted_nb_samples - is->audio_frame->nb_samples)
* is->audio_tgt_freq
/ is->audio_frame->sample_rate,
wanted_nb_samples * is->audio_tgt_freq
/ is->audio_frame->sample_rate) < 0) {
fprintf(stderr, "swr_set_compensation() failed\n");
break;
}
}
/**
* 转换该AVFrame到设置好的SDL需要的样子,有些旧的代码示例最主要就是少了这一部分,
* 往往一些音频能播,一些不能播,这就是原因,比如有些源文件音频恰巧是AV_SAMPLE_FMT_S16的。
* swr_convert 返回的是转换后每个声道(channel)的采样数
*/
len2 = swr_convert(is->swr_ctx, out,
sizeof(is->audio_buf2) / is->audio_tgt_channels
/ av_get_bytes_per_sample(is->audio_tgt_fmt),
in, is->audio_frame->nb_samples);
if (len2 < 0) {
fprintf(stderr, "swr_convert() failed\n");
break;
}
if (len2
== sizeof(is->audio_buf2) / is->audio_tgt_channels
/ av_get_bytes_per_sample(is->audio_tgt_fmt)) {
fprintf(stderr,
"warning: audio buffer is probably too small\n");
swr_init(is->swr_ctx);
}
is->audio_buf = is->audio_buf2;
//每声道采样数 x 声道数 x 每个采样字节数
resampled_data_size = len2 * is->audio_tgt_channels
* av_get_bytes_per_sample(is->audio_tgt_fmt);
} else {
resampled_data_size = decoded_data_size;
is->audio_buf = is->audio_frame->data[0];
}
// We have data, return it and come back for more later
return resampled_data_size;
}
if (pkt->data)
av_free_packet(pkt);
memset(pkt, 0, sizeof(*pkt));
if (is->quit)
return -1;
if (packet_queue_get(&is->audioq, pkt, 1) < 0)
return -1;
is->audio_pkt_data = pkt->data;
is->audio_pkt_size = pkt->size;
}
}
void audio_callback(void *userdata, Uint8 *stream, int len) {
VideoState *is = (VideoState *) userdata;
int len1, audio_data_size;
while (len > 0) {
if (is->audio_buf_index >= is->audio_buf_size) {
audio_data_size = audio_decode_frame(is);
if (audio_data_size < 0) {
/* silence */
is->audio_buf_size = 1024;
memset(is->audio_buf, 0, is->audio_buf_size);
} else {
is->audio_buf_size = audio_data_size;
}
is->audio_buf_index = 0;
}
len1 = is->audio_buf_size - is->audio_buf_index;
if (len1 > len) {
len1 = len;
}
memcpy(stream, (uint8_t *) is->audio_buf + is->audio_buf_index, len1);
len -= len1;
stream += len1;
is->audio_buf_index += len1;
}
}
/**
* 设置SDL播放声音的参数如声音采样格式,声道布局,静音值
*/
int stream_component_open(VideoState *is, int stream_index) {
AVFormatContext *ic = is->ic;
AVCodecContext *codecCtx;
AVCodec *codec;
SDL_AudioSpec wanted_spec, spec;
int64_t wanted_channel_layout = 0;
int wanted_nb_channels;
const int next_nb_channels[] = { 0, 0, 1, 6, 2, 6, 4, 6 };
if (stream_index < 0 || stream_index >= ic->nb_streams) {
return -1;
}
codecCtx = ic->streams[stream_index]->codec;
wanted_nb_channels = codecCtx->channels;
if (!wanted_channel_layout
|| wanted_nb_channels
!= av_get_channel_layout_nb_channels(
wanted_channel_layout)) {
wanted_channel_layout = av_get_default_channel_layout(
wanted_nb_channels);
wanted_channel_layout &= ~AV_CH_LAYOUT_STEREO_DOWNMIX;
}
wanted_spec.channels = av_get_channel_layout_nb_channels(
wanted_channel_layout);
wanted_spec.freq = codecCtx->sample_rate;
if (wanted_spec.freq <= 0 || wanted_spec.channels <= 0) {
fprintf(stderr, "Invalid sample rate or channel count!\n");
return -1;
}
wanted_spec.format = AUDIO_S16SYS;
wanted_spec.silence = 0;
wanted_spec.samples = SDL_AUDIO_BUFFER_SIZE;
wanted_spec.callback = audio_callback;
wanted_spec.userdata = is;
while (SDL_OpenAudio(&wanted_spec, &spec) < 0) {
fprintf(stderr, "SDL_OpenAudio (%d channels): %s\n",
wanted_spec.channels, SDL_GetError());
wanted_spec.channels = next_nb_channels[FFMIN(7, wanted_spec.channels)];
if (!wanted_spec.channels) {
fprintf(stderr,
"No more channel combinations to tyu, audio open failed\n");
return -1;
}
wanted_channel_layout = av_get_default_channel_layout(
wanted_spec.channels);
}
if (spec.format != AUDIO_S16SYS) {
fprintf(stderr, "SDL advised audio format %d is not supported!\n",
spec.format);
return -1;
}
if (spec.channels != wanted_spec.channels) {
wanted_channel_layout = av_get_default_channel_layout(spec.channels);
if (!wanted_channel_layout) {
fprintf(stderr, "SDL advised channel count %d is not supported!\n",
spec.channels);
return -1;
}
}
fprintf(stderr, "%d: wanted_spec.format = %d\n", __LINE__,
wanted_spec.format);
fprintf(stderr, "%d: wanted_spec.samples = %d\n", __LINE__,
wanted_spec.samples);
fprintf(stderr, "%d: wanted_spec.channels = %d\n", __LINE__,
wanted_spec.channels);
fprintf(stderr, "%d: wanted_spec.freq = %d\n", __LINE__, wanted_spec.freq);
fprintf(stderr, "%d: spec.format = %d\n", __LINE__, spec.format);
fprintf(stderr, "%d: spec.samples = %d\n", __LINE__, spec.samples);
fprintf(stderr, "%d: spec.channels = %d\n", __LINE__, spec.channels);
fprintf(stderr, "%d: spec.freq = %d\n", __LINE__, spec.freq);
is->audio_src_fmt = is->audio_tgt_fmt = AV_SAMPLE_FMT_S16;
is->audio_src_freq = is->audio_tgt_freq = spec.freq;
is->audio_src_channel_layout = is->audio_tgt_channel_layout =
wanted_channel_layout;
is->audio_src_channels = is->audio_tgt_channels = spec.channels;
codec = avcodec_find_decoder(codecCtx->codec_id);
if (!codec || (avcodec_open2(codecCtx, codec, NULL) < 0)) {
fprintf(stderr, "Unsupported codec!\n");
return -1;
}
ic->streams[stream_index]->discard = AVDISCARD_DEFAULT;
switch (codecCtx->codec_type) {
case AVMEDIA_TYPE_AUDIO:
is->audioStream = stream_index;
is->audio_st = ic->streams[stream_index];
is->audio_buf_size = 0;
is->audio_buf_index = 0;
memset(&is->audio_pkt, 0, sizeof(is->audio_pkt));
packet_queue_init(&is->audioq);
SDL_PauseAudio(0);
break;
default:
break;
}
}
/**
* demuxing出AVPacket
*/
static int decode_thread(void *arg) {
VideoState *is = (VideoState *) arg;
AVFormatContext *ic = NULL;
AVPacket pkt1, *packet = &pkt1;
int ret, i, audio_index = -1;
is->audioStream = -1;
global_video_state = is;
if (avformat_open_input(&ic, is->filename, NULL, NULL) != 0) {
return -1;
}
is->ic = ic;
if (avformat_find_stream_info(ic, NULL) < 0) {
return -1;
}
av_dump_format(ic, 0, is->filename, 0);
for (i = 0; i < ic->nb_streams; i++) {
if (ic->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO
&& audio_index < 0) {
audio_index = i;
break;
}
}
if (audio_index >= 0) {
stream_component_open(is, audio_index);
}
if (is->audioStream < 0) {
fprintf(stderr, "%s: could not open codecs\n", is->filename);
goto fail;
}
// main decode loop
for (;;) {
if (is->quit)
break;
if (is->audioq.size > MAX_AUDIOQ_SIZE) {
SDL_Delay(10);
continue;
}
ret = av_read_frame(is->ic, packet);
if (ret < 0) {
if (ret == AVERROR_EOF || url_feof(is->ic->pb)) {
break;
}
if (is->ic->pb && is->ic->pb->error) {
break;
}
continue;
}
if (packet->stream_index == is->audioStream) {
packet_queue_put(&is->audioq, packet);
} else {
av_free_packet(packet);
}
}
while (!is->quit) {
SDL_Delay(100);
}
fail: {
SDL_Event event;
event.type = FF_QUIT_EVENT;
event.user.data1 = is;
SDL_PushEvent(&event);
}
return 0;
}
int main(int argc, char *argv[]) {
SDL_Event event;
VideoState *is;
is = (VideoState *) av_mallocz(sizeof(VideoState));
if (argc < 2) {
fprintf(stderr, "Usage: test <file>\n");
exit(1);
}
av_register_all();
if (SDL_Init(SDL_INIT_AUDIO)) {
fprintf(stderr, "Could not initialize SDL - %s\n", SDL_GetError());
exit(1);
}
av_strlcpy(is->filename, argv[1], sizeof(is->filename));
is->parse_tid = SDL_CreateThread(decode_thread, is);
if (!is->parse_tid) {
av_free(is);
return -1;
}
for (;;) {
SDL_WaitEvent(&event);
switch (event.type) {
case FF_QUIT_EVENT:
case SDL_QUIT:
is->quit = 1;
SDL_Quit();
exit(0);
break;
default:
break;
}
}
return 0;
}
FFmpeg版本逐渐更新,代码功能更加丰富和易于使用,掌握音视频基础概念结合ffmpeg就可以方便使用!