源码分析6"/>
ffplay源码分析6
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ffplay是FFmpeg工程自带的简单播放器，使用FFmpeg提供的解码器和SDL库进行视频播放。本文基于FFmpeg工程4.1版本进行分析，其中ffplay源码清单如下：
.1/fftools/ffplay.c
在尝试分析源码前，可先阅读如下参考文章作为铺垫：
[1]. 雷霄骅，视音频编解码技术零基础学习方法
[2]. 视频编解码基础概念
[3]. 色彩空间与像素格式
[4]. 音频参数解析
[5]. FFmpeg基础概念
“ffplay源码分析”系列文章如下：
[1]. ffplay源码分析1-概述
[2]. ffplay源码分析2-数据结构
[3]. ffplay源码分析3-代码框架
[4]. ffplay源码分析4-音视频同步
[5]. ffplay源码分析5-图像格式转换
[6]. ffplay源码分析6-音频重采样
[7]. ffplay源码分析7-播放控制
6. 音频重采样
FFmpeg解码得到的音频帧的格式未必能被SDL支持，在这种情况下，需要进行音频重采样，即将音频帧格式转换为SDL支持的音频格式，否则是无法正常播放的。
音频重采样涉及两个步骤：
1) 打开音频设备时进行的准备工作：确定SDL支持的音频格式，作为后期音频重采样的目标格式
2) 音频播放线程中，取出音频帧后，若有需要(音频帧格式与SDL支持音频格式不匹配)则进行重采样，否则直接输出
6.1 打开音频设备
音频设备的打开实际是在解复用线程中实现的。解复用线程中先打开音频设备(设定音频回调函数供SDL音频播放线程回调)，然后再创建音频解码线程。调用链如下：
main() -->
stream_open() -->
read_thread() -->
stream_component_open() -->audio_open(is, channel_layout, nb_channels, sample_rate, &is->audio_tgt);decoder_start(&is->auddec, audio_thread, is);audio_open()函数填入期望的音频参数，打开音频设备后，将实际的音频参数存入输出参数is->audio_tgt中，后面音频播放线程用会用到此参数，使用此参数将原始音频数据重采样，转换为音频设备支持的格式。
static int audio_open(void *opaque, int64_t wanted_channel_layout, int wanted_nb_channels, int wanted_sample_rate, struct AudioParams *audio_hw_params)
{SDL_AudioSpec wanted_spec, spec;const char *env;static const int next_nb_channels[] = {0, 0, 1, 6, 2, 6, 4, 6};static const int next_sample_rates[] = {0, 44100, 48000, 96000, 192000};int next_sample_rate_idx = FF_ARRAY_ELEMS(next_sample_rates) - 1;env = SDL_getenv("SDL_AUDIO_CHANNELS");if (env) { // 若环境变量有设置，优先从环境变量取得声道数和声道布局wanted_nb_channels = atoi(env);wanted_channel_layout = av_get_default_channel_layout(wanted_nb_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;}// 根据channel_layout获取nb_channels，当传入参数wanted_nb_channels不匹配时，此处会作修正wanted_nb_channels = av_get_channel_layout_nb_channels(wanted_channel_layout);wanted_spec.channels = wanted_nb_channels; // 声道数wanted_spec.freq = wanted_sample_rate; // 采样率if (wanted_spec.freq <= 0 || wanted_spec.channels <= 0) {av_log(NULL, AV_LOG_ERROR, "Invalid sample rate or channel count!\n");return -1;}while (next_sample_rate_idx && next_sample_rates[next_sample_rate_idx] >= wanted_spec.freq)next_sample_rate_idx--; // 从采样率数组中找到第一个不大于传入参数wanted_sample_rate的值// 音频采样格式有两大类型：planar和packed，假设一个双声道音频文件，一个左声道采样点记作L，一个右声道采样点记作R，则：// planar存储格式：(plane1)LLLLLLLL...LLLL (plane2)RRRRRRRR...RRRR// packed存储格式：(plane1)LRLRLRLR...........................LRLR// 在这两种采样类型下，又细分多种采样格式，如AV_SAMPLE_FMT_S16、AV_SAMPLE_FMT_S16P等，注意SDL2.0目前不支持planar格式// channel_layout是int64_t类型，表示音频声道布局，每bit代表一个特定的声道，参考channel_layout.h中的定义，一目了然// 数据量(bits/秒) = 采样率(Hz) * 采样深度(bit) * 声道数wanted_spec.format = AUDIO_S16SYS; // 采样格式：S表带符号，16是采样深度(位深)，SYS表采用系统字节序，这个宏在SDL中定义wanted_spec.silence = 0; // 静音值wanted_spec.samples = FFMAX(SDL_AUDIO_MIN_BUFFER_SIZE, 2 << av_log2(wanted_spec.freq / SDL_AUDIO_MAX_CALLBACKS_PER_SEC)); // SDL声音缓冲区尺寸，单位是单声道采样点尺寸x声道数wanted_spec.callback = sdl_audio_callback; // 回调函数，若为NULL，则应使用SDL_QueueAudio()机制wanted_spec.userdata = opaque; // 提供给回调函数的参数// 打开音频设备并创建音频处理线程。期望的参数是wanted_spec，实际得到的硬件参数是spec// 1) SDL提供两种使音频设备取得音频数据方法：// a. push，SDL以特定的频率调用回调函数，在回调函数中取得音频数据// b. pull，用户程序以特定的频率调用SDL_QueueAudio()，向音频设备提供数据。此种情况wanted_spec.callback=NULL// 2) 音频设备打开后播放静音，不启动回调，调用SDL_PauseAudio(0)后启动回调，开始正常播放音频// SDL_OpenAudioDevice()第一个参数为NULL时，等价于SDL_OpenAudio()while (!(audio_dev = SDL_OpenAudioDevice(NULL, 0, &wanted_spec, &spec, SDL_AUDIO_ALLOW_FREQUENCY_CHANGE | SDL_AUDIO_ALLOW_CHANNELS_CHANGE))) {av_log(NULL, AV_LOG_WARNING, "SDL_OpenAudio (%d channels, %d Hz): %s\n",wanted_spec.channels, wanted_spec.freq, SDL_GetError());// 如果打开音频设备失败，则尝试用不同的声道数或采样率再试打开音频设备，这里有些奇怪，暂不深究wanted_spec.channels = next_nb_channels[FFMIN(7, wanted_spec.channels)];if (!wanted_spec.channels) {wanted_spec.freq = next_sample_rates[next_sample_rate_idx--];wanted_spec.channels = wanted_nb_channels;if (!wanted_spec.freq) {av_log(NULL, AV_LOG_ERROR,"No more combinations to try, audio open failed\n");return -1;}}wanted_channel_layout = av_get_default_channel_layout(wanted_spec.channels);}// 检查打开音频设备的实际参数：采样格式if (spec.format != AUDIO_S16SYS) {av_log(NULL, AV_LOG_ERROR,"SDL advised audio format %d is not supported!\n", spec.format);return -1;}// 检查打开音频设备的实际参数：声道数if (spec.channels != wanted_spec.channels) {wanted_channel_layout = av_get_default_channel_layout(spec.channels);if (!wanted_channel_layout) {av_log(NULL, AV_LOG_ERROR,"SDL advised channel count %d is not supported!\n", spec.channels);return -1;}}// wanted_spec是期望的参数，spec是实际的参数，wanted_spec和spec都是SDL中的结构。// 此处audio_hw_params是FFmpeg中的参数，输出参数供上级函数使用audio_hw_params->fmt = AV_SAMPLE_FMT_S16;audio_hw_params->freq = spec.freq;audio_hw_params->channel_layout = wanted_channel_layout;audio_hw_params->channels = spec.channels;audio_hw_params->frame_size = av_samples_get_buffer_size(NULL, audio_hw_params->channels, 1, audio_hw_params->fmt, 1);audio_hw_params->bytes_per_sec = av_samples_get_buffer_size(NULL, audio_hw_params->channels, audio_hw_params->freq, audio_hw_params->fmt, 1);if (audio_hw_params->bytes_per_sec <= 0 || audio_hw_params->frame_size <= 0) {av_log(NULL, AV_LOG_ERROR, "av_samples_get_buffer_size failed\n");return -1;}return spec.size;
}打开音频设备，涉及到FFmpeg中音频存储的基础概念，为稍显清晰，将相关注释摘抄如下：
6.1.1 音频格式相关
**planar&packed** 音频采样格式有两大类型：planar和packed，假设一个双声道音频文件，一个左声道采样点记作L，一个右声道采样点记作R，则： planar存储格式：(plane1)LLLLLLLL...LLLL (plane2)RRRRRRRR...RRRR packed存储格式：(plane1)LRLRLRLR...........................LRLR 在这两种采样类型下，又细分多种采样格式，如AV_SAMPLE_FMT_S16、AV_SAMPLE_FMT_S16P等，注意SDL2.0目前不支持planar格式 SDL中定义音频参数数据结构定义如下： /*** The calculated values in this structure are calculated by SDL_OpenAudio().** For multi-channel audio, the default SDL channel mapping is:* 2: FL FR (stereo)* 3: FL FR LFE (2.1 surround)* 4: FL FR BL BR (quad)* 5: FL FR FC BL BR (quad + center)* 6: FL FR FC LFE SL SR (5.1 surround - last two can also be BL BR)* 7: FL FR FC LFE BC SL SR (6.1 surround)* 8: FL FR FC LFE BL BR SL SR (7.1 surround)*/
typedef struct SDL_AudioSpec
{int freq; /**< DSP frequency -- samples per second */SDL_AudioFormat format; /**< Audio data format */Uint8 channels; /**< Number of channels: 1 mono, 2 stereo */Uint8 silence; /**< Audio buffer silence value (calculated) */Uint16 samples; /**< Audio buffer size in sample FRAMES (total samples divided by channel count) */Uint16 padding; /**< Necessary for some compile environments */Uint32 size; /**< Audio buffer size in bytes (calculated) */SDL_AudioCallback callback; /**< Callback that feeds the audio device (NULL to use SDL_QueueAudio()). */void *userdata; /**< Userdata passed to callback (ignored for NULL callbacks). */
} SDL_AudioSpec;SDL音频格式定义如下： /*** \brief Audio format flags.** These are what the 16 bits in SDL_AudioFormat currently mean...* (Unspecified bits are always zero).** \verbatim++-----------------------sample is signed if set|||| ++-----------sample is bigendian if set|| |||| || ++---sample is float if set|| || |||| || || +---sample bit size---+|| || || | |15 14 13 12 11 10 09 08 07 06 05 04 03 02 01 00\endverbatim** There are macros in SDL 2.0 and later to query these bits.*/
typedef Uint16 SDL_AudioFormat;/*** \name Audio format flags** Defaults to LSB byte order.*/
/* @{ */
#define AUDIO_U8 0x0008 /**< Unsigned 8-bit samples */
#define AUDIO_S8 0x8008 /**< Signed 8-bit samples */
#define AUDIO_U16LSB 0x0010 /**< Unsigned 16-bit samples */
#define AUDIO_S16LSB 0x8010 /**< Signed 16-bit samples */
#define AUDIO_U16MSB 0x1010 /**< As above, but big-endian byte order */
#define AUDIO_S16MSB 0x9010 /**< As above, but big-endian byte order */
#define AUDIO_U16 AUDIO_U16LSB
#define AUDIO_S16 AUDIO_S16LSB
/* @} */FFmpeg中定义音频参数的相关数据结构为： // 这个结构是在ffplay.c中定义的：
typedef struct AudioParams {int freq;int channels;int64_t channel_layout;enum AVSampleFormat fmt;int frame_size;int bytes_per_sec;
} AudioParams;/*** Audio sample formats** - 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.** - 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.** - The data layout as used in av_samples_fill_arrays() and elsewhere in FFmpeg* (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 bitsAV_SAMPLE_FMT_S16, ///< signed 16 bitsAV_SAMPLE_FMT_S32, ///< signed 32 bitsAV_SAMPLE_FMT_FLT, ///< floatAV_SAMPLE_FMT_DBL, ///< doubleAV_SAMPLE_FMT_U8P, ///< unsigned 8 bits, planarAV_SAMPLE_FMT_S16P, ///< signed 16 bits, planarAV_SAMPLE_FMT_S32P, ///< signed 32 bits, planarAV_SAMPLE_FMT_FLTP, ///< float, planarAV_SAMPLE_FMT_DBLP, ///< double, planarAV_SAMPLE_FMT_S64, ///< signed 64 bitsAV_SAMPLE_FMT_S64P, ///< signed 64 bits, planarAV_SAMPLE_FMT_NB ///< Number of sample formats. DO NOT USE if linking dynamically
}; **channel_layout** channel_layout是int64_t类型，表示音频声道布局，每bit代表一个特定的声道，参考channel_layout.h中的定义：/*** @defgroup channel_masks Audio channel masks** A channel layout is a 64-bits integer with a bit set for every channel.* The number of bits set must be equal to the number of channels.* The value 0 means that the channel layout is not known.* @note this data structure is not powerful enough to handle channels* combinations that have the same channel multiple times, such as* dual-mono.** @{*/
#define AV_CH_FRONT_LEFT 0x00000001
#define AV_CH_FRONT_RIGHT 0x00000002
#define AV_CH_FRONT_CENTER 0x00000004
#define AV_CH_LOW_FREQUENCY 0x00000008
#define AV_CH_BACK_LEFT 0x00000010
#define AV_CH_BACK_RIGHT 0x00000020
#define AV_CH_FRONT_LEFT_OF_CENTER 0x00000040
#define AV_CH_FRONT_RIGHT_OF_CENTER 0x00000080
#define AV_CH_BACK_CENTER 0x00000100
#define AV_CH_SIDE_LEFT 0x00000200
#define AV_CH_SIDE_RIGHT 0x00000400
#define AV_CH_TOP_CENTER 0x00000800
#define AV_CH_TOP_FRONT_LEFT 0x00001000
#define AV_CH_TOP_FRONT_CENTER 0x00002000
#define AV_CH_TOP_FRONT_RIGHT 0x00004000
#define AV_CH_TOP_BACK_LEFT 0x00008000
#define AV_CH_TOP_BACK_CENTER 0x00010000
#define AV_CH_TOP_BACK_RIGHT 0x00020000
#define AV_CH_STEREO_LEFT 0x20000000 ///< Stereo downmix.
#define AV_CH_STEREO_RIGHT 0x40000000 ///< See AV_CH_STEREO_LEFT.
#define AV_CH_WIDE_LEFT 0x0000000080000000ULL
#define AV_CH_WIDE_RIGHT 0x0000000100000000ULL
#define AV_CH_SURROUND_DIRECT_LEFT 0x0000000200000000ULL
#define AV_CH_SURROUND_DIRECT_RIGHT 0x0000000400000000ULL
#define AV_CH_LOW_FREQUENCY_2 0x0000000800000000ULL/** Channel mask value used for AVCodecContext.request_channel_layoutto indicate that the user requests the channel order of the decoder outputto be the native codec channel order. */
#define AV_CH_LAYOUT_NATIVE 0x8000000000000000ULL/*** @}* @defgroup channel_mask_c Audio channel layouts* @{* */
#define AV_CH_LAYOUT_MONO (AV_CH_FRONT_CENTER)
#define AV_CH_LAYOUT_STEREO (AV_CH_FRONT_LEFT|AV_CH_FRONT_RIGHT)
#define AV_CH_LAYOUT_2POINT1 (AV_CH_LAYOUT_STEREO|AV_CH_LOW_FREQUENCY)
#define AV_CH_LAYOUT_2_1 (AV_CH_LAYOUT_STEREO|AV_CH_BACK_CENTER)
#define AV_CH_LAYOUT_SURROUND (AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER)
#define AV_CH_LAYOUT_3POINT1 (AV_CH_LAYOUT_SURROUND|AV_CH_LOW_FREQUENCY)
#define AV_CH_LAYOUT_4POINT0 (AV_CH_LAYOUT_SURROUND|AV_CH_BACK_CENTER)
#define AV_CH_LAYOUT_4POINT1 (AV_CH_LAYOUT_4POINT0|AV_CH_LOW_FREQUENCY)
#define AV_CH_LAYOUT_2_2 (AV_CH_LAYOUT_STEREO|AV_CH_SIDE_LEFT|AV_CH_SIDE_RIGHT)
#define AV_CH_LAYOUT_QUAD (AV_CH_LAYOUT_STEREO|AV_CH_BACK_LEFT|AV_CH_BACK_RIGHT)
#define AV_CH_LAYOUT_5POINT0 (AV_CH_LAYOUT_SURROUND|AV_CH_SIDE_LEFT|AV_CH_SIDE_RIGHT)
#define AV_CH_LAYOUT_5POINT1 (AV_CH_LAYOUT_5POINT0|AV_CH_LOW_FREQUENCY)
#define AV_CH_LAYOUT_5POINT0_BACK (AV_CH_LAYOUT_SURROUND|AV_CH_BACK_LEFT|AV_CH_BACK_RIGHT)
#define AV_CH_LAYOUT_5POINT1_BACK (AV_CH_LAYOUT_5POINT0_BACK|AV_CH_LOW_FREQUENCY)
#define AV_CH_LAYOUT_6POINT0 (AV_CH_LAYOUT_5POINT0|AV_CH_BACK_CENTER)
#define AV_CH_LAYOUT_6POINT0_FRONT (AV_CH_LAYOUT_2_2|AV_CH_FRONT_LEFT_OF_CENTER|AV_CH_FRONT_RIGHT_OF_CENTER)
#define AV_CH_LAYOUT_HEXAGONAL (AV_CH_LAYOUT_5POINT0_BACK|AV_CH_BACK_CENTER)
#define AV_CH_LAYOUT_6POINT1 (AV_CH_LAYOUT_5POINT1|AV_CH_BACK_CENTER)
#define AV_CH_LAYOUT_6POINT1_BACK (AV_CH_LAYOUT_5POINT1_BACK|AV_CH_BACK_CENTER)
#define AV_CH_LAYOUT_6POINT1_FRONT (AV_CH_LAYOUT_6POINT0_FRONT|AV_CH_LOW_FREQUENCY)
#define AV_CH_LAYOUT_7POINT0 (AV_CH_LAYOUT_5POINT0|AV_CH_BACK_LEFT|AV_CH_BACK_RIGHT)
#define AV_CH_LAYOUT_7POINT0_FRONT (AV_CH_LAYOUT_5POINT0|AV_CH_FRONT_LEFT_OF_CENTER|AV_CH_FRONT_RIGHT_OF_CENTER)
#define AV_CH_LAYOUT_7POINT1 (AV_CH_LAYOUT_5POINT1|AV_CH_BACK_LEFT|AV_CH_BACK_RIGHT)
#define AV_CH_LAYOUT_7POINT1_WIDE (AV_CH_LAYOUT_5POINT1|AV_CH_FRONT_LEFT_OF_CENTER|AV_CH_FRONT_RIGHT_OF_CENTER)
#define AV_CH_LAYOUT_7POINT1_WIDE_BACK (AV_CH_LAYOUT_5POINT1_BACK|AV_CH_FRONT_LEFT_OF_CENTER|AV_CH_FRONT_RIGHT_OF_CENTER)
#define AV_CH_LAYOUT_OCTAGONAL (AV_CH_LAYOUT_5POINT0|AV_CH_BACK_LEFT|AV_CH_BACK_CENTER|AV_CH_BACK_RIGHT)
#define AV_CH_LAYOUT_HEXADECAGONAL (AV_CH_LAYOUT_OCTAGONAL|AV_CH_WIDE_LEFT|AV_CH_WIDE_RIGHT|AV_CH_TOP_BACK_LEFT|AV_CH_TOP_BACK_RIGHT|AV_CH_TOP_BACK_CENTER|AV_CH_TOP_FRONT_CENTER|AV_CH_TOP_FRONT_LEFT|AV_CH_TOP_FRONT_RIGHT)
#define AV_CH_LAYOUT_STEREO_DOWNMIX (AV_CH_STEREO_LEFT|AV_CH_STEREO_RIGHT)6.1.2 打开音频设备
打开音频设备并创建音频处理线程，通过调用SDL_OpenAudio()或SDL_OpenAudioDevice()实现。输入参数是预期的参数，输出参数是实际参数 1) SDL提供两种使音频设备取得音频数据方法： a. push，SDL以特定的频率调用回调函数，在回调函数中取得音频数据 b. pull，用户程序以特定的频率调用SDL_QueueAudio()，向音频设备提供数据。此种情况wanted_spec.callback=NULL 2) 音频设备打开后播放静音，不启动回调，调用SDL_PauseAudio(0)后启动回调，开始正常播放音频 SDL_OpenAudioDevice()第一个参数为NULL时，等价于SDL_OpenAudio() 6.2 音频重采样
音频重采样在audio_decode_frame()中实现，audio_decode_frame()就是从音频frame队列中取出一个frame，按指定格式经过重采样后输出。audio_decode_frame()函数名起得不太好，它只是进行重采样，并不进行解码，叫audio_resample_frame()可能更贴切。
重采样的细节很琐碎，直接看注释：
/*** Decode one audio frame and return its uncompressed size.** The processed audio frame is decoded, converted if required, and* stored in is->audio_buf, with size in bytes given by the return* value.*/
static int audio_decode_frame(VideoState *is)
{int data_size, resampled_data_size;int64_t dec_channel_layout;av_unused double audio_clock0;int wanted_nb_samples;Frame *af;if (is->paused)return -1;do {
#if defined(_WIN32)while (frame_queue_nb_remaining(&is->sampq) == 0) {if ((av_gettime_relative() - audio_callback_time) > 1000000LL * is->audio_hw_buf_size / is->audio_tgt.bytes_per_sec / 2)return -1;av_usleep (1000);}
#endif// 若队列头部可读，则由af指向可读帧if (!(af = frame_queue_peek_readable(&is->sampq)))return -1;frame_queue_next(&is->sampq);} while (af->serial != is->audioq.serial);// 根据frame中指定的音频参数获取缓冲区的大小data_size = av_samples_get_buffer_size(NULL, af->frame->channels, // 本行两参数：linesize，声道数af->frame->nb_samples, // 本行一参数：本帧中包含的单个声道中的样本数af->frame->format, 1); // 本行两参数：采样格式，不对齐// 获取声道布局dec_channel_layout =(af->frame->channel_layout && af->frame->channels == av_get_channel_layout_nb_channels(af->frame->channel_layout)) ?af->frame->channel_layout : av_get_default_channel_layout(af->frame->channels);// 获取样本数校正值：若同步时钟是音频，则不调整样本数；否则根据同步需要调整样本数wanted_nb_samples = synchronize_audio(is, af->frame->nb_samples);// is->audio_tgt是SDL可接受的音频帧数，是audio_open()中取得的参数// 在audio_open()函数中又有“is->audio_src = is->audio_tgt”// 此处表示：如果frame中的音频参数 == is->audio_src == is->audio_tgt，那音频重采样的过程就免了(因此时is->swr_ctr是NULL)// 否则使用frame(源)和is->audio_tgt(目标)中的音频参数来设置is->swr_ctx，并使用frame中的音频参数来赋值is->audio_srcif (af->frame->format != is->audio_src.fmt ||dec_channel_layout != is->audio_src.channel_layout ||af->frame->sample_rate != is->audio_src.freq ||(wanted_nb_samples != af->frame->nb_samples && !is->swr_ctx)) {swr_free(&is->swr_ctx);// 使用frame(源)和is->audio_tgt(目标)中的音频参数来设置is->swr_ctxis->swr_ctx = swr_alloc_set_opts(NULL,is->audio_tgt.channel_layout, is->audio_tgt.fmt, is->audio_tgt.freq,dec_channel_layout, af->frame->format, af->frame->sample_rate,0, NULL);if (!is->swr_ctx || swr_init(is->swr_ctx) < 0) {av_log(NULL, AV_LOG_ERROR,"Cannot create sample rate converter for conversion of %d Hz %s %d channels to %d Hz %s %d channels!\n",af->frame->sample_rate, av_get_sample_fmt_name(af->frame->format), af->frame->channels,is->audio_tgt.freq, av_get_sample_fmt_name(is->audio_tgt.fmt), is->audio_tgt.channels);swr_free(&is->swr_ctx);return -1;}// 使用frame中的参数更新is->audio_src，第一次更新后后面基本不用执行此if分支了，因为一个音频流中各frame通用参数一样is->audio_src.channel_layout = dec_channel_layout;is->audio_src.channels = af->frame->channels;is->audio_src.freq = af->frame->sample_rate;is->audio_src.fmt = af->frame->format;}if (is->swr_ctx) {// 重采样输入参数1：输入音频样本数是af->frame->nb_samples// 重采样输入参数2：输入音频缓冲区const uint8_t **in = (const uint8_t **)af->frame->extended_data;// 重采样输出参数1：输出音频缓冲区尺寸// 重采样输出参数2：输出音频缓冲区uint8_t **out = &is->audio_buf1;// 重采样输出参数：输出音频样本数(多加了256个样本)int out_count = (int64_t)wanted_nb_samples * is->audio_tgt.freq / af->frame->sample_rate + 256;// 重采样输出参数：输出音频缓冲区尺寸(以字节为单位)int out_size = av_samples_get_buffer_size(NULL, is->audio_tgt.channels, out_count, is->audio_tgt.fmt, 0);int len2;if (out_size < 0) {av_log(NULL, AV_LOG_ERROR, "av_samples_get_buffer_size() failed\n");return -1;}// 如果frame中的样本数经过校正，则条件成立if (wanted_nb_samples != af->frame->nb_samples) {// 重采样补偿：不清楚参数怎么算的if (swr_set_compensation(is->swr_ctx, (wanted_nb_samples - af->frame->nb_samples) * is->audio_tgt.freq / af->frame->sample_rate, wanted_nb_samples * is->audio_tgt.freq / af->frame->sample_rate) < 0) {av_log(NULL, AV_LOG_ERROR, "swr_set_compensation() failed\n");return -1;}}av_fast_malloc(&is->audio_buf1, &is->audio_buf1_size, out_size);if (!is->audio_buf1)return AVERROR(ENOMEM);// 音频重采样：返回值是重采样后得到的音频数据中单个声道的样本数len2 = swr_convert(is->swr_ctx, out, out_count, in, af->frame->nb_samples);if (len2 < 0) {av_log(NULL, AV_LOG_ERROR, "swr_convert() failed\n");return -1;}if (len2 == out_count) {av_log(NULL, AV_LOG_WARNING, "audio buffer is probably too small\n");if (swr_init(is->swr_ctx) < 0)swr_free(&is->swr_ctx);}is->audio_buf = is->audio_buf1;// 重采样返回的一帧音频数据大小(以字节为单位)resampled_data_size = len2 * is->audio_tgt.channels * av_get_bytes_per_sample(is->audio_tgt.fmt);} else {// 未经重采样，则将指针指向frame中的音频数据is->audio_buf = af->frame->data[0];resampled_data_size = data_size;}audio_clock0 = is->audio_clock;/* update the audio clock with the pts */if (!isnan(af->pts))is->audio_clock = af->pts + (double) af->frame->nb_samples / af->frame->sample_rate;elseis->audio_clock = NAN;is->audio_clock_serial = af->serial;
#ifdef DEBUG{static double last_clock;printf("audio: delay=%0.3f clock=%0.3f clock0=%0.3f\n",is->audio_clock - last_clock,is->audio_clock, audio_clock0);last_clock = is->audio_clock;}
#endifreturn resampled_data_size;
}转载于:.html
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