From 3eceac2eab0b42ee082a0b615ebf40a21f0fb915 Mon Sep 17 00:00:00 2001 From: wm4 Date: Wed, 7 Dec 2016 19:44:29 +0100 Subject: Remove compatibility things Possible with bumped FFmpeg/Libav. These are just the simple cases. --- audio/decode/ad_lavc.c | 22 +--------------------- audio/decode/ad_spdif.c | 10 ---------- audio/filter/af_lavcac3enc.c | 16 ---------------- audio/filter/af_lavrresample.c | 5 ----- audio/out/ao_lavc.c | 23 ----------------------- 5 files changed, 1 insertion(+), 75 deletions(-) (limited to 'audio') diff --git a/audio/decode/ad_lavc.c b/audio/decode/ad_lavc.c index e28558414d..276ce69739 100644 --- a/audio/decode/ad_lavc.c +++ b/audio/decode/ad_lavc.c @@ -207,7 +207,6 @@ static int decode_packet(struct dec_audio *da, struct demux_packet *mpkt, if (priv->needs_reset) control(da, ADCTRL_RESET, NULL); -#if HAVE_AVCODEC_NEW_CODEC_API int ret = avcodec_send_packet(avctx, &pkt); if (ret >= 0 || ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) { if (ret >= 0 && mpkt) @@ -220,24 +219,6 @@ static int decode_packet(struct dec_audio *da, struct demux_packet *mpkt, if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) ret = 0; } -#else - int ret = avcodec_decode_audio4(avctx, priv->avframe, &got_frame, &pkt); - if (mpkt) { - // At least "shorten" decodes sub-frames, instead of the whole packet. - // At least "mpc8" can return 0 and wants the packet again next time. - if (ret >= 0) { - ret = FFMIN(ret, mpkt->len); // sanity check against decoder overreads - mpkt->buffer += ret; - mpkt->len -= ret; - mpkt->pts = MP_NOPTS_VALUE; // don't reset PTS next time - } - // LATM may need many packets to find mux info - if (ret == AVERROR(EAGAIN)) { - mpkt->len = 0; - return 0; - } - } -#endif if (ret < 0) { MP_ERR(da, "Error decoding audio.\n"); return -1; @@ -245,8 +226,7 @@ static int decode_packet(struct dec_audio *da, struct demux_packet *mpkt, if (!got_frame) return 0; - double out_pts = mp_pts_from_av(MP_AVFRAME_DEC_PTS(priv->avframe), - &priv->codec_timebase); + double out_pts = mp_pts_from_av(priv->avframe->pts, &priv->codec_timebase); struct mp_audio *mpframe = mp_audio_from_avframe(priv->avframe); if (!mpframe) diff --git a/audio/decode/ad_spdif.c b/audio/decode/ad_spdif.c index 56e4a8102d..e15aca5c53 100644 --- a/audio/decode/ad_spdif.c +++ b/audio/decode/ad_spdif.c @@ -116,16 +116,10 @@ static int determine_codec_profile(struct dec_audio *da, AVPacket *pkt) goto done; } -#if HAVE_AVCODEC_NEW_CODEC_API if (avcodec_send_packet(ctx, pkt) < 0) goto done; if (avcodec_receive_frame(ctx, frame) < 0) goto done; -#else - int got_frame = 0; - if (avcodec_decode_audio4(ctx, frame, &got_frame, pkt) < 1 || !got_frame) - goto done; -#endif profile = ctx->profile; @@ -178,11 +172,7 @@ static int init_filter(struct dec_audio *da, AVPacket *pkt) if (!stream) goto fail; -#if HAVE_AVCODEC_HAS_CODECPAR stream->codecpar->codec_id = spdif_ctx->codec_id; -#else - stream->codec->codec_id = spdif_ctx->codec_id; -#endif AVDictionary *format_opts = NULL; diff --git a/audio/filter/af_lavcac3enc.c b/audio/filter/af_lavcac3enc.c index 0a7c5d4440..9df5adb96f 100644 --- a/audio/filter/af_lavcac3enc.c +++ b/audio/filter/af_lavcac3enc.c @@ -280,7 +280,6 @@ static int filter_out(struct af_instance *af) AVPacket pkt = {0}; av_init_packet(&pkt); -#if HAVE_AVCODEC_NEW_CODEC_API // Send input as long as it wants. while (1) { err = read_input_frame(af, frame); @@ -310,21 +309,6 @@ static int filter_out(struct af_instance *af) MP_FATAL(af, "Encode failed.\n"); goto done; } -#else - err = read_input_frame(af, frame); - if (err < 0) - goto done; - if (err == 0) - goto done; - err = -1; - int ok; - int lavc_ret = avcodec_encode_audio2(s->lavc_actx, &pkt, frame, &ok); - s->input->samples = 0; - if (lavc_ret < 0 || !ok) { - MP_FATAL(af, "Encode failed.\n"); - goto done; - } -#endif MP_DBG(af, "avcodec_encode_audio got %d, pending %d.\n", pkt.size, s->pending->samples + s->input->samples); diff --git a/audio/filter/af_lavrresample.c b/audio/filter/af_lavrresample.c index dc5d1a0d23..828be66247 100644 --- a/audio/filter/af_lavrresample.c +++ b/audio/filter/af_lavrresample.c @@ -111,12 +111,7 @@ static double get_delay(struct af_resample *s) } static int get_out_samples(struct af_resample *s, int in_samples) { -#if LIBSWRESAMPLE_VERSION_MAJOR > 1 || LIBSWRESAMPLE_VERSION_MINOR >= 2 return swr_get_out_samples(s->avrctx, in_samples); -#else - return av_rescale_rnd(in_samples, s->out_rate, s->in_rate, AV_ROUND_UP) - + swr_get_delay(s->avrctx, s->out_rate); -#endif } #endif diff --git a/audio/out/ao_lavc.c b/audio/out/ao_lavc.c index 8ae1317407..4dbc55a369 100644 --- a/audio/out/ao_lavc.c +++ b/audio/out/ao_lavc.c @@ -258,7 +258,6 @@ static void encode_audio_and_write(struct ao *ao, AVFrame *frame) struct priv *ac = ao->priv; AVPacket packet = {0}; -#if HAVE_AVCODEC_NEW_CODEC_API int status = avcodec_send_frame(ac->codec, frame); if (status < 0) { MP_ERR(ao, "error encoding at %d %d/%d\n", @@ -297,28 +296,6 @@ static void encode_audio_and_write(struct ao *ao, AVFrame *frame) write_packet(ao, &packet); av_packet_unref(&packet); } -#else - av_init_packet(&packet); - int got_packet = 0; - int status = avcodec_encode_audio2(ac->codec, &packet, frame, &got_packet); - if (status < 0) { - MP_ERR(ao, "error encoding at %d %d/%d\n", - frame ? (int) frame->pts : -1, - ac->codec->time_base.num, - ac->codec->time_base.den); - return; - } - if (!got_packet) { - return; - } - if (frame) { - if (ac->savepts == AV_NOPTS_VALUE) - ac->savepts = frame->pts; - } - encode_lavc_write_stats(ao->encode_lavc_ctx, ac->codec); - write_packet(ao, &packet); - av_packet_unref(&packet); -#endif } // must get exactly ac->aframesize amount of data -- cgit v1.2.3