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-rw-r--r--audio/out/ao_lavc.c240
1 files changed, 55 insertions, 185 deletions
diff --git a/audio/out/ao_lavc.c b/audio/out/ao_lavc.c
index aaa9e6422c..e18db667a3 100644
--- a/audio/out/ao_lavc.c
+++ b/audio/out/ao_lavc.c
@@ -40,8 +40,8 @@
#include "common/encode_lavc.h"
struct priv {
- AVStream *stream;
- AVCodecContext *codec;
+ struct encoder_context *enc;
+
int pcmhack;
int aframesize;
int aframecount;
@@ -53,14 +53,13 @@ struct priv {
double expected_next_pts;
AVRational worst_time_base;
- int worst_time_base_is_stream;
bool shutdown;
};
static void encode(struct ao *ao, double apts, void **data);
-static bool supports_format(AVCodec *codec, int format)
+static bool supports_format(const AVCodec *codec, int format)
{
for (const enum AVSampleFormat *sampleformat = codec->sample_fmts;
sampleformat && *sampleformat != AV_SAMPLE_FMT_NONE;
@@ -72,7 +71,7 @@ static bool supports_format(AVCodec *codec, int format)
return false;
}
-static void select_format(struct ao *ao, AVCodec *codec)
+static void select_format(struct ao *ao, const AVCodec *codec)
{
int formats[AF_FORMAT_COUNT + 1];
af_get_best_sample_formats(ao->format, formats);
@@ -88,70 +87,53 @@ static void select_format(struct ao *ao, AVCodec *codec)
// open & setup audio device
static int init(struct ao *ao)
{
- struct priv *ac = talloc_zero(ao, struct priv);
- AVCodec *codec;
-
- ao->priv = ac;
+ struct priv *ac = ao->priv;
- if (!encode_lavc_available(ao->encode_lavc_ctx)) {
- MP_ERR(ao, "the option --o (output file) must be specified\n");
+ ac->enc = encoder_context_alloc(ao->encode_lavc_ctx, STREAM_AUDIO, ao->log);
+ if (!ac->enc)
return -1;
- }
+ talloc_steal(ac, ac->enc);
- pthread_mutex_lock(&ao->encode_lavc_ctx->lock);
-
- 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;
- }
-
- codec = ao->encode_lavc_ctx->ac;
+ AVCodecContext *encoder = ac->enc->encoder;
+ const AVCodec *codec = encoder->codec;
int samplerate = af_select_best_samplerate(ao->samplerate,
codec->supported_samplerates);
if (samplerate > 0)
ao->samplerate = samplerate;
- // TODO: Remove this redundancy with encode_lavc_alloc_stream also
- // setting the time base.
- // 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;
- ac->codec->time_base.den = ao->samplerate;
+ encoder->time_base.num = 1;
+ encoder->time_base.den = ao->samplerate;
- ac->codec->sample_rate = ao->samplerate;
+ encoder->sample_rate = ao->samplerate;
struct mp_chmap_sel sel = {0};
mp_chmap_sel_add_any(&sel);
if (!ao_chmap_sel_adjust2(ao, &sel, &ao->channels, false))
goto fail;
mp_chmap_reorder_to_lavc(&ao->channels);
- ac->codec->channels = ao->channels.num;
- ac->codec->channel_layout = mp_chmap_to_lavc(&ao->channels);
+ encoder->channels = ao->channels.num;
+ encoder->channel_layout = mp_chmap_to_lavc(&ao->channels);
- ac->codec->sample_fmt = AV_SAMPLE_FMT_NONE;
+ encoder->sample_fmt = AV_SAMPLE_FMT_NONE;
select_format(ao, codec);
ac->sample_size = af_fmt_to_bytes(ao->format);
- ac->codec->sample_fmt = af_to_avformat(ao->format);
- ac->codec->bits_per_raw_sample = ac->sample_size * 8;
+ encoder->sample_fmt = af_to_avformat(ao->format);
+ encoder->bits_per_raw_sample = ac->sample_size * 8;
- if (encode_lavc_open_codec(ao->encode_lavc_ctx, ac->codec) < 0)
+ if (!encoder_init_codec_and_muxer(ac->enc))
goto fail;
ac->pcmhack = 0;
- if (ac->codec->frame_size <= 1)
- ac->pcmhack = av_get_bits_per_sample(ac->codec->codec_id) / 8;
+ if (encoder->frame_size <= 1)
+ ac->pcmhack = av_get_bits_per_sample(encoder->codec_id) / 8;
if (ac->pcmhack) {
ac->aframesize = 16384; // "enough"
} else {
- ac->aframesize = ac->codec->frame_size;
+ ac->aframesize = encoder->frame_size;
}
// enough frames for at least 0.25 seconds
@@ -169,7 +151,6 @@ static int init(struct ao *ao)
if (ao->channels.num > AV_NUM_DATA_POINTERS)
goto fail;
- pthread_mutex_unlock(&ao->encode_lavc_ctx->lock);
return 0;
fail:
@@ -184,28 +165,17 @@ static void uninit(struct ao *ao)
struct priv *ac = ao->priv;
struct encode_lavc_context *ectx = ao->encode_lavc_ctx;
- if (!ac || ac->shutdown)
- return;
-
- pthread_mutex_lock(&ectx->lock);
+ if (!ac->shutdown) {
+ double outpts = ac->expected_next_pts;
- if (!encode_lavc_start(ectx)) {
- MP_WARN(ao, "not even ready to encode audio at end -> dropped\n");
+ pthread_mutex_lock(&ectx->lock);
+ if (!ac->enc->options->rawts && ac->enc->options->copyts)
+ outpts += ectx->discontinuity_pts_offset;
pthread_mutex_unlock(&ectx->lock);
- return;
- }
- if (ac->stream) {
- double outpts = ac->expected_next_pts;
- if (!ectx->options->rawts && ectx->options->copyts)
- outpts += ectx->discontinuity_pts_offset;
- outpts += encode_lavc_getoffset(ectx, ac->codec);
+ outpts += encoder_get_offset(ac->enc);
encode(ao, outpts, NULL);
}
-
- pthread_mutex_unlock(&ectx->lock);
-
- ac->shutdown = true;
}
// return: how many samples can be played without blocking
@@ -216,106 +186,21 @@ static int get_space(struct ao *ao)
return ac->aframesize * ac->framecount;
}
-static void write_packet(struct ao *ao, AVPacket *packet)
-{
- // TODO: Can we unify this with the equivalent video code path?
- struct priv *ac = ao->priv;
-
- packet->stream_index = ac->stream->index;
- if (packet->pts != AV_NOPTS_VALUE) {
- packet->pts = av_rescale_q(packet->pts,
- ac->codec->time_base,
- ac->stream->time_base);
- } else {
- // Do we need this at all? Better be safe than sorry...
- MP_WARN(ao, "encoder lost pts, why?\n");
- if (ac->savepts != MP_NOPTS_VALUE) {
- packet->pts = av_rescale_q(ac->savepts,
- ac->codec->time_base,
- ac->stream->time_base);
- }
- }
- if (packet->dts != AV_NOPTS_VALUE) {
- packet->dts = av_rescale_q(packet->dts,
- ac->codec->time_base,
- ac->stream->time_base);
- }
- if (packet->duration > 0) {
- packet->duration = av_rescale_q(packet->duration,
- ac->codec->time_base,
- ac->stream->time_base);
- }
-
- ac->savepts = AV_NOPTS_VALUE;
-
- 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,
- ac->stream->time_base.den);
- return;
- }
-}
-
-static void encode_audio_and_write(struct ao *ao, AVFrame *frame)
-{
- // TODO: Can we unify this with the equivalent video code path?
- struct priv *ac = ao->priv;
- AVPacket packet = {0};
-
- int status = avcodec_send_frame(ac->codec, frame);
- 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;
- }
-
- for (;;) {
- av_init_packet(&packet);
- status = avcodec_receive_packet(ac->codec, &packet);
- if (status == AVERROR(EAGAIN)) { // No more packets for now.
- if (frame == NULL) {
- MP_ERR(ao, "sent flush frame, got EAGAIN");
- }
- break;
- }
- if (status == AVERROR_EOF) { // No more packets, ever.
- if (frame != NULL) {
- MP_ERR(ao, "sent audio frame, got EOF");
- }
- break;
- }
- 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);
- break;
- }
- 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);
- }
-}
-
// must get exactly ac->aframesize amount of data
static void encode(struct ao *ao, double apts, void **data)
{
struct priv *ac = ao->priv;
struct encode_lavc_context *ectx = ao->encode_lavc_ctx;
+ AVCodecContext *encoder = ac->enc->encoder;
double realapts = ac->aframecount * (double) ac->aframesize /
ao->samplerate;
ac->aframecount++;
+ pthread_mutex_lock(&ectx->lock);
if (data)
ectx->audio_pts_offset = realapts - apts;
+ pthread_mutex_unlock(&ectx->lock);
if(data) {
AVFrame *frame = av_frame_alloc();
@@ -329,17 +214,17 @@ static void encode(struct ao *ao, double apts, void **data)
frame->linesize[0] = frame->nb_samples * ao->sstride;
- if (ectx->options->rawts || ectx->options->copyts) {
+ if (ac->enc->options->rawts || ac->enc->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 * encoder->time_base.den /
+ encoder->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 * encoder->time_base.den /
+ encoder->time_base.num + 0.5);
}
- int64_t frame_pts = av_rescale_q(frame->pts, ac->codec->time_base,
+ int64_t frame_pts = av_rescale_q(frame->pts, encoder->time_base,
ac->worst_time_base);
if (ac->lastpts != AV_NOPTS_VALUE && frame_pts <= ac->lastpts) {
// this indicates broken video
@@ -348,15 +233,15 @@ static void encode(struct ao *ao, double apts, void **data)
(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);
+ encoder->time_base);
}
ac->lastpts = frame_pts;
- frame->quality = ac->codec->global_quality;
- encode_audio_and_write(ao, frame);
+ frame->quality = encoder->global_quality;
+ encoder_encode(ac->enc, frame);
av_frame_free(&frame);
} else {
- encode_audio_and_write(ao, NULL);
+ encoder_encode(ac->enc, NULL);
}
}
@@ -365,20 +250,16 @@ static void encode(struct ao *ao, double apts, void **data)
static int play(struct ao *ao, void **data, int samples, int flags)
{
struct priv *ac = ao->priv;
+ struct encoder_context *enc = ac->enc;
struct encode_lavc_context *ectx = ao->encode_lavc_ctx;
int bufpos = 0;
double nextpts;
double outpts;
int orig_samples = samples;
+ // for ectx PTS fields
pthread_mutex_lock(&ectx->lock);
- if (!encode_lavc_start(ectx)) {
- MP_WARN(ao, "not ready yet for encoding audio\n");
- pthread_mutex_unlock(&ectx->lock);
- return 0;
- }
-
double pts = ectx->last_audio_in_pts;
pts += ectx->samples_since_last_pts / (double)ao->samplerate;
@@ -407,26 +288,10 @@ 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 * (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 "
- "adjustment; the stream base (%d/%d) is not worse.\n",
- (int)ac->codec->time_base.num,
- (int)ac->codec->time_base.den,
- (int)ac->stream->time_base.num,
- (int)ac->stream->time_base.den);
- ac->worst_time_base = ac->codec->time_base;
- ac->worst_time_base_is_stream = 0;
- } else {
- MP_WARN(ao, "NOTE: not using codec time base (%d/%d) for pts "
- "adjustment; the stream base (%d/%d) is worse.\n",
- (int)ac->codec->time_base.num,
- (int)ac->codec->time_base.den,
- (int)ac->stream->time_base.num,
- (int)ac->stream->time_base.den);
- ac->worst_time_base = ac->stream->time_base;
- ac->worst_time_base_is_stream = 1;
- }
+ // We don't know the muxer time_base anymore, and can't, because we
+ // might start encoding before the muxer is opened. (The muxer decides
+ // the final AVStream.time_base when opening the muxer.)
+ ac->worst_time_base = enc->encoder->time_base;
// NOTE: we use the following "axiom" of av_rescale_q:
// if time base A is worse than time base B, then
@@ -446,7 +311,7 @@ static int play(struct ao *ao, void **data, int samples, int flags)
}
// Fix and apply the discontinuity pts offset.
- if (!ectx->options->rawts && ectx->options->copyts) {
+ if (!enc->options->rawts && enc->options->copyts) {
// fix the discontinuity pts offset
nextpts = pts;
if (ectx->discontinuity_pts_offset == MP_NOPTS_VALUE) {
@@ -465,8 +330,10 @@ static int play(struct ao *ao, void **data, int samples, int flags)
outpts = pts;
}
+ pthread_mutex_unlock(&ectx->lock);
+
// Shift pts by the pts offset first.
- outpts += encode_lavc_getoffset(ectx, ac->codec);
+ outpts += encoder_get_offset(enc);
while (samples - bufpos >= ac->aframesize) {
void *start[MP_NUM_CHANNELS] = {0};
@@ -479,8 +346,10 @@ static int play(struct ao *ao, void **data, int samples, int flags)
// Calculate expected pts of next audio frame (input side).
ac->expected_next_pts = pts + bufpos / (double) ao->samplerate;
+ pthread_mutex_lock(&ectx->lock);
+
// Set next allowed input pts value (input side).
- if (!ectx->options->rawts && ectx->options->copyts) {
+ if (!enc->options->rawts && enc->options->copyts) {
nextpts = ac->expected_next_pts + ectx->discontinuity_pts_offset;
if (nextpts > ectx->next_in_pts)
ectx->next_in_pts = nextpts;
@@ -513,6 +382,7 @@ const struct ao_driver audio_out_lavc = {
.encode = true,
.description = "audio encoding using libavcodec",
.name = "lavc",
+ .priv_size = sizeof(struct priv),
.init = init,
.uninit = uninit,
.get_space = get_space,
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-24 06:30:11 +0000