Question

这是libavutil/samplefmt.h的代码段：

/**
 * Audio Sample Formats
 *
 * @par
 * The data described by the sample format is always in native-endian order.
 * Sample values can be expressed by native C types, hence the lack of a signed
 * 24-bit sample format even though it is a common raw audio data format.
 *
 * @par
 * The floating-point formats are based on full volume being in the range
 * [-1.0, 1.0]. Any values outside this range are beyond full volume level.
 *
 * @par
 * The data layout as used in av_samples_fill_arrays() and elsewhere in Libav
 * (such as AVFrame in libavcodec) is as follows:
 *
 * @par
 * For planar sample formats, each audio channel is in a separate data plane,
 * and linesize is the buffer size, in bytes, for a single plane. All data
 * planes must be the same size. For packed sample formats, only the first data
 * plane is used, and samples for each channel are interleaved. In this case,
 * linesize is the buffer size, in bytes, for the 1 plane.
 */
enum AVSampleFormat {
    AV_SAMPLE_FMT_NONE = -1,
    AV_SAMPLE_FMT_U8,          ///< unsigned 8 bits
    AV_SAMPLE_FMT_S16,         ///< signed 16 bits
    AV_SAMPLE_FMT_S32,         ///< signed 32 bits
    AV_SAMPLE_FMT_FLT,         ///< float
    AV_SAMPLE_FMT_DBL,         ///< double

    AV_SAMPLE_FMT_U8P,         ///< unsigned 8 bits, planar
    AV_SAMPLE_FMT_S16P,        ///< signed 16 bits, planar
    AV_SAMPLE_FMT_S32P,        ///< signed 32 bits, planar
    AV_SAMPLE_FMT_FLTP,        ///< float, planar
    AV_SAMPLE_FMT_DBLP,        ///< double, planar

    AV_SAMPLE_FMT_NB           ///< Number of sample formats. DO NOT USE if linking dynamically
};

它特别提到即使它是一种常见的原始音频数据格式，也会丢失24位。因此，如果我使用libav / ffmpeg导出到音频文件，我将如何使用24位音频？

导出音频文件如下所示：

AVCodec *codec = get_codec();
AVOutputFormat *oformat = get_output_format();
AVFormatContext *fmt_ctx = avformat_alloc_context();
assert(fmt_ctx);
int err = avio_open(&fmt_ctx->pb, get_output_filename(), AVIO_FLAG_WRITE);
assert(err >= 0);
fmt_ctx->oformat = oformat;
AVStream *stream = avformat_new_stream(fmt_ctx, codec);
assert(stream);
AVCodecContext *codec_ctx = stream->codec;
codec_ctx->bit_rate = get_export_bit_rate();

// How to set this to 24 bit instead of 32?
codec_ctx->sample_fmt = AV_SAMPLE_FMT_S32;

codec_ctx->sample_rate = get_sample_rate();
codec_ctx->channel_layout = get_channel_layout()
codec_ctx->channels = get_channel_count();
codec_ctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;

Answer 1

使用AV_SAMPLE_FMT_S32并将ctx-＆gt; bits_per_raw_sample设置为24.音频需要在32位整数的MSB中，即。在最后用0填充。

如何用libav / ffmpeg编码24位音频？

1 个答案: