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@ -48,18 +48,21 @@ static int get_format_from_sample_fmt(const char **fmt, enum AVSampleFormat samp |
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} |
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int init_audio_codec(AVFormatContext *fmt_ctx, AVCodecContext **avcodec_ctx, bool is_encoder) { |
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AVCodec *codec = is_encoder ? avcodec_find_encoder(fmt_ctx->audio_codec_id) |
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: avcodec_find_decoder(fmt_ctx->audio_codec_id); |
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AVCodecContext *codec_ctx = nullptr; |
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for (int i = 0; i < fmt_ctx->nb_streams; ++i) { |
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if (fmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) { |
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codec_ctx = fmt_ctx->streams[i]->codec; |
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} |
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} |
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AVCodec *codec = is_encoder ? avcodec_find_encoder(codec_ctx->codec_id) |
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: avcodec_find_decoder(codec_ctx->codec_id); |
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if (!codec) { |
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ALOGE("can't found codec id=%d\n", fmt_ctx->audio_codec_id); |
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ALOGE("can't found codec id=%d\n", codec_ctx->codec_id); |
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return -1; |
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} |
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AVCodecContext *codec_ctx = avcodec_alloc_context3(codec); |
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if (!codec_ctx) { |
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ALOGE("avcodec_alloc_context3 fail!\n"); |
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return -2; |
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} |
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int ret = avcodec_open2(codec_ctx, codec, nullptr); |
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if (ret < 0) |
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ALOGE("avcodec_open2 fail:%d", ret); |
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*avcodec_ctx = codec_ctx; |
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return ret; |
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} |
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@ -95,10 +98,9 @@ int init_audio_muxer(AVFormatContext **ofmt_ctx, const char* filename) { |
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int resampling(const char *src_filename, const char *dst_filename, int dst_rate) |
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{ |
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int src_rate = 0; |
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int src_nb_samples = 0; |
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int src_rate; |
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int64_t src_ch_layout = AV_CH_LAYOUT_STEREO; |
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enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_S16; |
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enum AVSampleFormat src_sample_fmt; |
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int dst_bufsize; |
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int dst_linesize; |
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@ -129,6 +131,10 @@ int resampling(const char *src_filename, const char *dst_filename, int dst_rate) |
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avformat_find_stream_info(iformat_ctx, nullptr); |
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frame = av_frame_alloc(); |
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opacket = av_packet_alloc(); |
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init_audio_decoder(iformat_ctx, &icodec_ctx); |
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src_rate = icodec_ctx->sample_rate; |
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src_ch_layout = (int64_t) icodec_ctx->channel_layout; |
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src_sample_fmt = icodec_ctx->sample_fmt; |
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/* create resample context */ |
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swr_ctx = swr_alloc(); |
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@ -153,12 +159,6 @@ int resampling(const char *src_filename, const char *dst_filename, int dst_rate) |
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goto end; |
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} |
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/* compute the number of converted samples: buffering is avoided
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* ensuring that the output buffer will contain at least all the |
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* converted input samples */ |
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max_dst_nb_samples = dst_nb_samples = |
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(int) av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP); |
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/* buffer is going to be directly written to a raw-audio file, no alignment */ |
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dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout); |
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ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels, |
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@ -172,13 +172,20 @@ int resampling(const char *src_filename, const char *dst_filename, int dst_rate) |
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if (ret < 0) { |
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goto end; |
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} |
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init_audio_decoder(iformat_ctx, &icodec_ctx); |
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init_audio_encoder(oformat_ctx, &ocodec_ctx); |
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while (av_read_frame(iformat_ctx, &packet) >= 0) { |
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ret = avcodec_decode_audio4(icodec_ctx, frame, &got_frame_ptr, &packet); |
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if (ret < 0) { |
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ALOGE("decode audio error:%d\n", ret); |
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continue; |
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} |
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ALOGE("decode succ, pts=%ld\n", frame->pts); |
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/* compute destination number of samples */ |
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dst_nb_samples = (int) av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) + |
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src_nb_samples, dst_rate, src_rate, AV_ROUND_UP); |
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frame->nb_samples, dst_rate, src_rate, AV_ROUND_UP); |
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if (dst_nb_samples > max_dst_nb_samples) { |
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av_freep(&dst_data[0]); |
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ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels, |
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@ -188,12 +195,6 @@ int resampling(const char *src_filename, const char *dst_filename, int dst_rate) |
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max_dst_nb_samples = dst_nb_samples; |
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} |
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ret = avcodec_decode_audio4(icodec_ctx, frame, &got_frame_ptr, &packet); |
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if (ret < 0) { |
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ALOGE("decode audio error:%d\n", ret); |
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continue; |
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} |
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/* convert to destination format */ |
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ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)frame->data, frame->nb_samples); |
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if (ret < 0) { |
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