【Android FFMPEG 开发】FFMPEG 音频重采样 ( 初始化音频重采样上下文 SwrContext | 计算音频延迟 | 计算输出样本个数 | 音频重采样 swr_convert )（一）

SwrContext *swrContext = swr_alloc_set_opts(
        0 ,                     //现在还没有 SwrContext 上下文 , 先传入 0
        //输出的音频参数
        AV_CH_LAYOUT_STEREO ,   //双声道立体声
        AV_SAMPLE_FMT_S16 ,     //采样位数 16 位
        44100 ,                 //输出的采样率
        //从编码器中获取输入音频格式
        avCodecContext->channel_layout, //输入的声道数
        avCodecContext->sample_fmt,     //输入的采样位数
        avCodecContext->sample_rate,    //输入的采样率
        0, 0    //日志参数 设置 0 即可
        );
swr_init(swrContext);

enum AVRounding rnd) av_const
int64_t out_count = av_rescale_rnd(
        avFrame->nb_samples + delay, //本次要处理的数据个数
        44100,
        avFrame->sample_rate ,
        AV_ROUND_UP );

int samples_per_channel_count = swr_convert(
        swrContext ,
        &data,
        out_count ,
        (const uint8_t **)avFrame->data, //普通指针转为 const 指针需要使用 const_cast 转换
        avFrame->nb_samples
        );

samples_per_channel_count 是 每个通道的样本数 ;
pcm_data_bit_size = samples_per_channel_count * 2 * 2;
1

/**
 * Allocate SwrContext if needed and set/reset common parameters.
 *
 * This function does not require s to be allocated with swr_alloc(). On the
 * other hand, swr_alloc() can use swr_alloc_set_opts() to set the parameters
 * on the allocated context.
 *
 * @param s               existing Swr context if available, or NULL if not
 * @param out_ch_layout   output channel layout (AV_CH_LAYOUT_*)
 * @param out_sample_fmt  output sample format (AV_SAMPLE_FMT_*).
 * @param out_sample_rate output sample rate (frequency in Hz)
 * @param in_ch_layout    input channel layout (AV_CH_LAYOUT_*)
 * @param in_sample_fmt   input sample format (AV_SAMPLE_FMT_*).
 * @param in_sample_rate  input sample rate (frequency in Hz)
 * @param log_offset      logging level offset
 * @param log_ctx         parent logging context, can be NULL
 *
 * @see swr_init(), swr_free()
 * @return NULL on error, allocated context otherwise
 */
struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
                                      int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
                                      int64_t  in_ch_layout, enum AVSampleFormat  in_sample_fmt, int  in_sample_rate,
                                      int log_offset, void *log_ctx);

/**
 * Initialize context after user parameters have been set.
 * @note The context must be configured using the AVOption API.
 *
 * @see av_opt_set_int()
 * @see av_opt_set_dict()
 *
 * @param[in,out]   s Swr context to initialize
 * @return AVERROR error code in case of failure.
 */
int swr_init(struct SwrContext *s);

/*
 设置音频重采样的上下文参数
 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
    int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
    int64_t  in_ch_layout, enum AVSampleFormat  in_sample_fmt, int  in_sample_rate,
    int log_offset, void *log_ctx);
 */
swrContext = swr_alloc_set_opts(
        0 ,                     //现在还没有 SwrContext 上下文 , 先传入 0
        //输出的音频参数
        AV_CH_LAYOUT_STEREO ,   //双声道立体声
        AV_SAMPLE_FMT_S16 ,     //采样位数 16 位
        44100 ,                 //输出的采样率
        //从编码器中获取输入音频格式
        avCodecContext->channel_layout, //输入的声道数
        avCodecContext->sample_fmt,     //输入的采样位数
        avCodecContext->sample_rate,    //输入的采样率
        0, 0    //日志参数 设置 0 即可
        );
//注意创建完之后初始化
swr_init(swrContext);

/**
 * Gets the delay the next input sample will experience relative to the next output sample.
 *
 * Swresample can buffer data if more input has been provided than available
 * output space, also converting between sample rates needs a delay.
 * This function returns the sum of all such delays.
 * The exact delay is not necessarily an integer value in either input or
 * output sample rate. Especially when downsampling by a large value, the
 * output sample rate may be a poor choice to represent the delay, similarly
 * for upsampling and the input sample rate.
 *
 * @param s     swr context
 * @param base  timebase in which the returned delay will be:
 *              @li if it's set to 1 the returned delay is in seconds
 *              @li if it's set to 1000 the returned delay is in milliseconds
 *              @li if it's set to the input sample rate then the returned
 *                  delay is in input samples
 *              @li if it's set to the output sample rate then the returned
 *                  delay is in output samples
 *              @li if it's the least common multiple of in_sample_rate and
 *                  out_sample_rate then an exact rounding-free delay will be
 *                  returned
 * @returns     the delay in 1 / @c base units.
 */
int64_t swr_get_delay(struct SwrContext *s, int64_t base);

//OpenSLES 播放器设定播放的音频格式是 立体声 , 44100 Hz 采样 , 16位采样位数
//  解码出来的 AVFrame 中的数据格式不确定 , 需要进行重采样
/*
    int64_t swr_get_delay(
    struct SwrContext *s,
    int64_t base
    );
    转码的过程中 , 输入 10 个数据 , 并不一定都能处理完毕并输出 10 个数据 , 可能处理输出了 8 个数据
    还剩余 2 个数据没有处理
    那么在下一次处理的时候 , 需要将上次没有处理完的两个数据处理了 ;
    如果不处理上次的2个数据 , 那么数据会一直积压 , 如果积压数据过多 , 最终造成很大的延迟 , 甚至崩溃
    因此每次处理的时候 , 都要尝试将上次剩余没有处理的数据加入到本次处理的数据中
    如果计算出的 delay 一直等于 0 , 说明没有积压数据
 */
int64_t delay = swr_get_delay(swrContext , avFrame->sample_rate);