From 20a1f250c676e54c164745b1f5c4fc8feda8f306 Mon Sep 17 00:00:00 2001 From: wm4 Date: Thu, 19 Apr 2018 20:13:28 +0200 Subject: encode: cosmetics Mostly whitespace changes; some semantic preserving transformations. --- audio/out/ao_lavc.c | 54 ++++++++++++++++++++++++++++------------------------- 1 file changed, 29 insertions(+), 25 deletions(-) (limited to 'audio') diff --git a/audio/out/ao_lavc.c b/audio/out/ao_lavc.c index 830fbf9b72..aaa9e6422c 100644 --- a/audio/out/ao_lavc.c +++ b/audio/out/ao_lavc.c @@ -58,11 +58,13 @@ struct priv { bool shutdown; }; +static void encode(struct ao *ao, double apts, void **data); + static bool supports_format(AVCodec *codec, int format) { for (const enum AVSampleFormat *sampleformat = codec->sample_fmts; sampleformat && *sampleformat != AV_SAMPLE_FMT_NONE; - ++sampleformat) + sampleformat++) { if (af_from_avformat(*sampleformat) == format) return true; @@ -100,9 +102,10 @@ static int init(struct ao *ao) if (encode_lavc_alloc_stream(ao->encode_lavc_ctx, AVMEDIA_TYPE_AUDIO, - &ac->stream, &ac->codec) < 0) { - MP_ERR(ao, "could not get a new audio stream\n"); - goto fail; + &ac->stream, &ac->codec) < 0) + { + MP_ERR(ao, "could not get a new audio stream\n"); + goto fail; } codec = ao->encode_lavc_ctx->ac; @@ -114,7 +117,7 @@ static int init(struct ao *ao) // TODO: Remove this redundancy with encode_lavc_alloc_stream also // setting the time base. - // Using codec->time_bvase is deprecated, but needed for older lavf. + // Using codec->time_base is deprecated, but needed for older lavf. ac->stream->time_base.num = 1; ac->stream->time_base.den = ao->samplerate; ac->codec->time_base.num = 1; @@ -145,10 +148,11 @@ static int init(struct ao *ao) if (ac->codec->frame_size <= 1) ac->pcmhack = av_get_bits_per_sample(ac->codec->codec_id) / 8; - if (ac->pcmhack) + if (ac->pcmhack) { ac->aframesize = 16384; // "enough" - else + } else { ac->aframesize = ac->codec->frame_size; + } // enough frames for at least 0.25 seconds ac->framecount = ceil(ao->samplerate * 0.25 / ac->aframesize); @@ -175,7 +179,6 @@ fail: } // close audio device -static void encode(struct ao *ao, double apts, void **data); static void uninit(struct ao *ao) { struct priv *ac = ao->priv; @@ -245,8 +248,7 @@ static void write_packet(struct ao *ao, AVPacket *packet) ac->savepts = AV_NOPTS_VALUE; - if (encode_lavc_write_frame(ao->encode_lavc_ctx, - ac->stream, packet) < 0) { + if (encode_lavc_write_frame(ao->encode_lavc_ctx, ac->stream, packet) < 0) { MP_ERR(ao, "error writing at %d %d/%d\n", (int) packet->pts, ac->stream->time_base.num, @@ -269,6 +271,7 @@ static void encode_audio_and_write(struct ao *ao, AVFrame *frame) ac->codec->time_base.den); return; } + for (;;) { av_init_packet(&packet); status = avcodec_receive_packet(ac->codec, &packet); @@ -328,29 +331,33 @@ static void encode(struct ao *ao, double apts, void **data) if (ectx->options->rawts || ectx->options->copyts) { // real audio pts - frame->pts = floor(apts * ac->codec->time_base.den / ac->codec->time_base.num + 0.5); + frame->pts = floor(apts * ac->codec->time_base.den / + ac->codec->time_base.num + 0.5); } else { // audio playback time - frame->pts = floor(realapts * ac->codec->time_base.den / ac->codec->time_base.num + 0.5); + frame->pts = floor(realapts * ac->codec->time_base.den / + ac->codec->time_base.num + 0.5); } - int64_t frame_pts = av_rescale_q(frame->pts, ac->codec->time_base, ac->worst_time_base); + int64_t frame_pts = av_rescale_q(frame->pts, ac->codec->time_base, + ac->worst_time_base); if (ac->lastpts != AV_NOPTS_VALUE && frame_pts <= ac->lastpts) { // this indicates broken video // (video pts failing to increase fast enough to match audio) MP_WARN(ao, "audio frame pts went backwards (%d <- %d), autofixed\n", (int)frame->pts, (int)ac->lastpts); frame_pts = ac->lastpts + 1; - frame->pts = av_rescale_q(frame_pts, ac->worst_time_base, ac->codec->time_base); + frame->pts = av_rescale_q(frame_pts, ac->worst_time_base, + ac->codec->time_base); } ac->lastpts = frame_pts; frame->quality = ac->codec->global_quality; encode_audio_and_write(ao, frame); av_frame_free(&frame); - } - else + } else { encode_audio_and_write(ao, NULL); + } } // this should round samples down to frame sizes @@ -400,7 +407,6 @@ static int play(struct ao *ao, void **data, int samples, int flags) } if (ac->worst_time_base.den == 0) { - //if (ac->codec->time_base.num / ac->codec->time_base.den >= ac->stream->time_base.num / ac->stream->time_base.den) if (ac->codec->time_base.num * (double) ac->stream->time_base.den >= ac->stream->time_base.num * (double) ac->codec->time_base.den) { MP_VERBOSE(ao, "NOTE: using codec time base (%d/%d) for pts " @@ -445,8 +451,9 @@ static int play(struct ao *ao, void **data, int samples, int flags) nextpts = pts; if (ectx->discontinuity_pts_offset == MP_NOPTS_VALUE) { ectx->discontinuity_pts_offset = ectx->next_in_pts - nextpts; - } - else if (fabs(nextpts + ectx->discontinuity_pts_offset - ectx->next_in_pts) > 30) { + } else if (fabs(nextpts + ectx->discontinuity_pts_offset - + ectx->next_in_pts) > 30) + { MP_WARN(ao, "detected an unexpected discontinuity (pts jumped by " "%f seconds)\n", nextpts + ectx->discontinuity_pts_offset - ectx->next_in_pts); @@ -454,8 +461,7 @@ static int play(struct ao *ao, void **data, int samples, int flags) } outpts = pts + ectx->discontinuity_pts_offset; - } - else { + } else { outpts = pts; } @@ -488,13 +494,11 @@ static int play(struct ao *ao, void **data, int samples, int flags) pthread_mutex_unlock(&ectx->lock); if (flags & AOPLAY_FINAL_CHUNK) { - if (bufpos < orig_samples) { + if (bufpos < orig_samples) MP_ERR(ao, "did not write enough data at the end\n"); - } } else { - if (bufpos > orig_samples) { + if (bufpos > orig_samples) MP_ERR(ao, "audio buffer overflow (should never happen)\n"); - } } return taken; -- cgit v1.2.3