diff options
author | Anton Khirnov <anton@khirnov.net> | 2011-11-21 17:39:22 +0400 |
---|---|---|
committer | Anton Khirnov <anton@khirnov.net> | 2011-11-21 21:12:38 +0400 |
commit | ded28ba35b14b55302e54d35a29d6f920850770b (patch) | |
tree | c87bd6e6f05987708ed6b8c7c031ea2e71ffea59 /avconv.c | |
parent | 78162b4ea28abebe1c7b87b88cd7731fd3b41073 (diff) |
avconv: split audio transcoding out of output_packet().
Diffstat (limited to 'avconv.c')
-rw-r--r-- | avconv.c | 205 |
1 files changed, 112 insertions, 93 deletions
@@ -1615,6 +1615,109 @@ static void rate_emu_sleep(InputStream *ist) } } +static int transcode_audio(InputStream *ist, AVPacket *pkt, int *got_output) +{ + static unsigned int samples_size = 0; + int bps = av_get_bytes_per_sample(ist->st->codec->sample_fmt); + uint8_t *decoded_data_buf = NULL; + int decoded_data_size = 0; + int i, ret; + + if (pkt && samples_size < FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE)) { + av_free(samples); + samples_size = FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE); + samples = av_malloc(samples_size); + } + decoded_data_size = samples_size; + + ret = avcodec_decode_audio3(ist->st->codec, samples, &decoded_data_size, + pkt); + if (ret < 0) + return ret; + pkt->data += ret; + pkt->size -= ret; + *got_output = decoded_data_size > 0; + + /* Some bug in mpeg audio decoder gives */ + /* decoded_data_size < 0, it seems they are overflows */ + if (!*got_output) { + /* no audio frame */ + return 0; + } + + decoded_data_buf = (uint8_t *)samples; + ist->next_pts += ((int64_t)AV_TIME_BASE/bps * decoded_data_size) / + (ist->st->codec->sample_rate * ist->st->codec->channels); + + // preprocess audio (volume) + if (audio_volume != 256) { + switch (ist->st->codec->sample_fmt) { + case AV_SAMPLE_FMT_U8: + { + uint8_t *volp = samples; + for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) { + int v = (((*volp - 128) * audio_volume + 128) >> 8) + 128; + *volp++ = av_clip_uint8(v); + } + break; + } + case AV_SAMPLE_FMT_S16: + { + int16_t *volp = samples; + for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) { + int v = ((*volp) * audio_volume + 128) >> 8; + *volp++ = av_clip_int16(v); + } + break; + } + case AV_SAMPLE_FMT_S32: + { + int32_t *volp = samples; + for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) { + int64_t v = (((int64_t)*volp * audio_volume + 128) >> 8); + *volp++ = av_clipl_int32(v); + } + break; + } + case AV_SAMPLE_FMT_FLT: + { + float *volp = samples; + float scale = audio_volume / 256.f; + for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) { + *volp++ *= scale; + } + break; + } + case AV_SAMPLE_FMT_DBL: + { + double *volp = samples; + double scale = audio_volume / 256.; + for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) { + *volp++ *= scale; + } + break; + } + default: + av_log(NULL, AV_LOG_FATAL, + "Audio volume adjustment on sample format %s is not supported.\n", + av_get_sample_fmt_name(ist->st->codec->sample_fmt)); + exit_program(1); + } + } + + rate_emu_sleep(ist); + + for (i = 0; i < nb_output_streams; i++) { + OutputStream *ost = &output_streams[i]; + + if (!check_output_constraints(ist, ost) || !ost->encoding_needed) + continue; + do_audio_out(output_files[ost->file_index].ctx, ost, ist, + decoded_data_buf, decoded_data_size); + } + return 0; +} + /* pkt = NULL means EOF (needed to flush decoder buffers) */ static int output_packet(InputStream *ist, int ist_index, OutputStream *ost_table, int nb_ostreams, @@ -1625,7 +1728,6 @@ static int output_packet(InputStream *ist, int ist_index, int ret = 0, i; int got_output; void *buffer_to_free = NULL; - static unsigned int samples_size= 0; AVSubtitle subtitle, *subtitle_to_free; int64_t pkt_pts = AV_NOPTS_VALUE; #if CONFIG_AVFILTER @@ -1634,7 +1736,6 @@ static int output_packet(InputStream *ist, int ist_index, float quality; AVPacket avpkt; - int bps = av_get_bytes_per_sample(ist->st->codec->sample_fmt); if(ist->next_pts == AV_NOPTS_VALUE) ist->next_pts= ist->pts; @@ -1656,8 +1757,6 @@ static int output_packet(InputStream *ist, int ist_index, //while we have more to decode or while the decoder did output something on EOF while (ist->decoding_needed && (avpkt.size > 0 || (!pkt && got_output))) { - uint8_t *decoded_data_buf; - int decoded_data_size; AVFrame *decoded_frame, *filtered_frame; handle_eof: ist->pts= ist->next_pts; @@ -1667,38 +1766,19 @@ static int output_packet(InputStream *ist, int ist_index, "Multiple frames in a packet from stream %d\n", pkt->stream_index); ist->showed_multi_packet_warning=1; + // XXX temporary hack, will be turned to a switch() once all codec + // types are split out + if (ist->st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { + ret = transcode_audio(ist, &avpkt, &got_output); + if (ret < 0) + return ret; + continue; + } + /* decode the packet if needed */ decoded_frame = filtered_frame = NULL; - decoded_data_buf = NULL; /* fail safe */ - decoded_data_size= 0; subtitle_to_free = NULL; switch(ist->st->codec->codec_type) { - case AVMEDIA_TYPE_AUDIO:{ - if(pkt && samples_size < FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE)) { - samples_size = FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE); - av_free(samples); - samples= av_malloc(samples_size); - } - decoded_data_size= samples_size; - /* XXX: could avoid copy if PCM 16 bits with same - endianness as CPU */ - ret = avcodec_decode_audio3(ist->st->codec, samples, &decoded_data_size, - &avpkt); - if (ret < 0) - return ret; - avpkt.data += ret; - avpkt.size -= ret; - got_output = decoded_data_size > 0; - /* Some bug in mpeg audio decoder gives */ - /* decoded_data_size < 0, it seems they are overflows */ - if (!got_output) { - /* no audio frame */ - continue; - } - decoded_data_buf = (uint8_t *)samples; - ist->next_pts += ((int64_t)AV_TIME_BASE/bps * decoded_data_size) / - (ist->st->codec->sample_rate * ist->st->codec->channels); - break;} case AVMEDIA_TYPE_VIDEO: if (!(decoded_frame = avcodec_alloc_frame())) return AVERROR(ENOMEM); @@ -1743,64 +1823,6 @@ static int output_packet(InputStream *ist, int ist_index, return -1; } - // preprocess audio (volume) - if (ist->st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { - if (audio_volume != 256) { - switch (ist->st->codec->sample_fmt) { - case AV_SAMPLE_FMT_U8: - { - uint8_t *volp = samples; - for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) { - int v = (((*volp - 128) * audio_volume + 128) >> 8) + 128; - *volp++ = av_clip_uint8(v); - } - break; - } - case AV_SAMPLE_FMT_S16: - { - int16_t *volp = samples; - for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) { - int v = ((*volp) * audio_volume + 128) >> 8; - *volp++ = av_clip_int16(v); - } - break; - } - case AV_SAMPLE_FMT_S32: - { - int32_t *volp = samples; - for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) { - int64_t v = (((int64_t)*volp * audio_volume + 128) >> 8); - *volp++ = av_clipl_int32(v); - } - break; - } - case AV_SAMPLE_FMT_FLT: - { - float *volp = samples; - float scale = audio_volume / 256.f; - for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) { - *volp++ *= scale; - } - break; - } - case AV_SAMPLE_FMT_DBL: - { - double *volp = samples; - double scale = audio_volume / 256.; - for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) { - *volp++ *= scale; - } - break; - } - default: - av_log(NULL, AV_LOG_FATAL, - "Audio volume adjustment on sample format %s is not supported.\n", - av_get_sample_fmt_name(ist->st->codec->sample_fmt)); - exit_program(1); - } - } - } - /* frame rate emulation */ rate_emu_sleep(ist); @@ -1846,9 +1868,6 @@ static int output_packet(InputStream *ist, int ist_index, av_assert0(ist->decoding_needed); switch(ost->st->codec->codec_type) { - case AVMEDIA_TYPE_AUDIO: - do_audio_out(os, ost, ist, decoded_data_buf, decoded_data_size); - break; case AVMEDIA_TYPE_VIDEO: #if CONFIG_AVFILTER if (ost->picref->video && !ost->frame_aspect_ratio) |