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libeplayer3: move audio resampling to dedicated ipcm writer

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Origin commit data
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Branch: master
Commit: 21a7d427fd
Author: martii <m4rtii@gmx.de>
Date: 2014-04-05 (Sat, 05 Apr 2014)


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No further description and justification available within origin commit message!

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This commit was generated by Migit
This commit is contained in:
martii
2014-04-05 13:21:58 +02:00
parent 91412a3c1b
commit 1bada7d686
7 changed files with 203 additions and 175 deletions
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165
libeplayer3/container/container_ffmpeg.c
View File

@@ -194,10 +194,10 @@ static char *Codec2Encoding(AVCodecContext * codec, int *version)
return NULL;
}
long long int calcPts(AVStream * stream, int64_t pts)
int64_t calcPts(AVFormatContext *avfc, AVStream * stream, int64_t pts)
{
if (!stream) {
ffmpeg_err("stream / packet null\n");
if (!avfc || !stream) {
ffmpeg_err("context / stream null\n");
return INVALID_PTS_VALUE;
}
@@ -206,7 +206,7 @@ long long int calcPts(AVStream * stream, int64_t pts)
else if (avContext->start_time == AV_NOPTS_VALUE)
pts = 90000.0 * (double) pts * av_q2d(stream->time_base);
else
pts = 90000.0 * (double) pts * av_q2d(stream->time_base) - 90000.0 * avContext->start_time / AV_TIME_BASE;
pts = 90000.0 * (double) pts * av_q2d(stream->time_base) - 90000.0 * avfc->start_time / AV_TIME_BASE;
if (pts & 0x8000000000000000ull)
pts = INVALID_PTS_VALUE;
@@ -234,16 +234,7 @@ static void FFMPEGThread(Context_t * context)
hasPlayThreadStarted = 1;
int64_t currentVideoPts = -1, currentAudioPts = -1, showtime = 0, bofcount = 0;
AudioVideoOut_t avOut;
SwrContext *swr = NULL;
AVFrame *decoded_frame = NULL;
int out_sample_rate = 44100;
int out_channels = 2;
uint64_t out_channel_layout = AV_CH_LAYOUT_STEREO;
int restart_audio_resampling = 0;
int64_t currentVideoPts = 0, currentAudioPts = 0, showtime = 0, bofcount = 0;
ffmpeg_printf(10, "\n");
while (context->playback->isCreationPhase) {
@@ -253,6 +244,9 @@ static void FFMPEGThread(Context_t * context)
ffmpeg_printf(10, "Running!\n");
while (context && context->playback && context->playback->isPlaying && !context->playback->abortRequested) {
AudioVideoOut_t avOut;
avOut.restart_audio_resampling = 0;
//IF MOVIE IS PAUSED, WAIT
if (context->playback->isPaused) {
@@ -321,7 +315,7 @@ static void FFMPEGThread(Context_t * context)
if (res < 0 && context->playback->BackWard)
bofcount = 1;
seek_target = INT64_MIN;
restart_audio_resampling = 1;
avOut.restart_audio_resampling = 1;
// flush streams
unsigned int i;
@@ -370,13 +364,14 @@ static void FFMPEGThread(Context_t * context)
ffmpeg_printf(200, "packet_size %d - index %d\n", packet_size, pid);
if (videoTrack && (videoTrack->Id == pid)) {
currentVideoPts = videoTrack->pts = pts = calcPts(videoTrack->stream, packet.pts);
currentVideoPts = /* CHECK videoTrack->pts = */pts = calcPts(avContext, videoTrack->stream, packet.pts);
ffmpeg_printf(200, "VideoTrack index = %d %lld\n", pid, currentVideoPts);
avOut.data = packet_data;
avOut.len = packet_size;
avOut.pts = pts;
avOut.packet = &packet;
avOut.type = "video";
avOut.stream = videoTrack->stream;
@@ -386,8 +381,9 @@ static void FFMPEGThread(Context_t * context)
ffmpeg_err("writing data to video device failed\n");
}
} else if (audioTrack && (audioTrack->Id == pid)) {
context->currentAudioPtsP = &currentAudioPts; //FIXME, temporary workaround only
if (!context->playback->BackWard) {
currentAudioPts = audioTrack->pts = pts = calcPts(audioTrack->stream, packet.pts);
currentAudioPts = /* CHECK audioTrack->pts = */pts = calcPts(avContext, audioTrack->stream, packet.pts);
ffmpeg_printf(200, "AudioTrack index = %d\n", pid);
if (audioTrack->inject_raw_pcm == 1) {
@@ -401,137 +397,18 @@ static void FFMPEGThread(Context_t * context)
avOut.data = packet_data;
avOut.len = packet_size;
avOut.pts = pts;
avOut.packet = &packet;
avOut.type = "audio";
avOut.stream = audioTrack->stream;
avOut.avfc = avContext;
if (context->output->audio->Write(context, &avOut) < 0)
ffmpeg_err("(raw pcm) writing data to audio device failed\n");
} else if (audioTrack->inject_as_pcm == 1) {
AVCodecContext *c = ((AVStream *) (audioTrack->stream))->codec;
if (restart_audio_resampling) {
restart_audio_resampling = 0;
if (swr) {
swr_free(&swr);
swr = NULL;
}
if (decoded_frame) {
av_frame_free(&decoded_frame);
decoded_frame = NULL;
}
context->output->Command(context, OUTPUT_CLEAR, NULL);
context->output->Command(context, OUTPUT_PLAY, NULL);
AVCodec *codec = avcodec_find_decoder(c->codec_id);
if (!codec || avcodec_open2(c, codec, NULL))
fprintf(stderr, "%s %d: avcodec_open2 failed\n", __func__, __LINE__);
}
while (packet_size > 0) {
int got_frame = 0;
if (!decoded_frame) {
if (!(decoded_frame = av_frame_alloc())) {
fprintf(stderr, "out of memory\n");
exit(1);
}
} else
av_frame_unref(decoded_frame);
int len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &packet);
if (len < 0) {
restart_audio_resampling = 1;
break;
}
packet_data += len;
packet_size -= len;
if (!got_frame)
continue;
int e;
if (!swr) {
int rates[] = { 48000, 96000, 192000, 44100, 88200, 176400, 0 };
int *rate = rates;
int in_rate = c->sample_rate;
while (*rate && ((*rate / in_rate) * in_rate != *rate)
&& (in_rate / *rate) * *rate != in_rate)
rate++;
out_sample_rate = *rate ? *rate : 44100;
swr = swr_alloc();
out_channels = c->channels;
if (c->channel_layout == 0) {
// FIXME -- need to guess, looks pretty much like a bug in the FFMPEG WMA decoder
c->channel_layout = AV_CH_LAYOUT_STEREO;
}
out_channel_layout = c->channel_layout;
// player2 won't play mono
if (out_channel_layout == AV_CH_LAYOUT_MONO) {
out_channel_layout = AV_CH_LAYOUT_STEREO;
out_channels = 2;
}
av_opt_set_int(swr, "in_channel_layout", c->channel_layout, 0);
av_opt_set_int(swr, "out_channel_layout", out_channel_layout, 0);
av_opt_set_int(swr, "in_sample_rate", c->sample_rate, 0);
av_opt_set_int(swr, "out_sample_rate", out_sample_rate, 0);
av_opt_set_int(swr, "in_sample_fmt", c->sample_fmt, 0);
av_opt_set_int(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
e = swr_init(swr);
if (e < 0) {
fprintf(stderr,
"swr_init: %d (icl=%d ocl=%d isr=%d osr=%d isf=%d osf=%d\n",
-e, (int) c->channel_layout,
(int) out_channel_layout, c->sample_rate, out_sample_rate, c->sample_fmt, AV_SAMPLE_FMT_S16);
swr_free(&swr);
swr = NULL;
}
}
uint8_t *output = NULL;
int in_samples = decoded_frame->nb_samples;
int out_samples = av_rescale_rnd(swr_get_delay(swr, c->sample_rate) + in_samples, out_sample_rate, c->sample_rate, AV_ROUND_UP);
e = av_samples_alloc(&output, NULL, out_channels, out_samples, AV_SAMPLE_FMT_S16, 1);
if (e < 0) {
fprintf(stderr, "av_samples_alloc: %d\n", -e);
continue;
}
// FIXME. PTS calculation is probably broken.
int64_t next_in_pts = av_rescale(av_frame_get_best_effort_timestamp(decoded_frame),
((AVStream *) audioTrack->stream)->time_base.num * (int64_t) out_sample_rate * c->sample_rate,
((AVStream *) audioTrack->stream)->time_base.den);
int64_t next_out_pts = av_rescale(swr_next_pts(swr, next_in_pts),
((AVStream *) audioTrack->stream)->time_base.den,
((AVStream *) audioTrack->stream)->time_base.num * (int64_t) out_sample_rate * c->sample_rate);
currentAudioPts = audioTrack->pts = pts = calcPts(audioTrack->stream, next_out_pts);
out_samples = swr_convert(swr, &output, out_samples, (const uint8_t **)
&decoded_frame->data[0], in_samples);
avOut.uSampleRate = out_sample_rate;
avOut.uNoOfChannels = av_get_channel_layout_nb_channels(out_channel_layout);
avOut.uBitsPerSample = 16;
avOut.bLittleEndian = 1;
avOut.data = output;
avOut.len = out_samples * sizeof(short) * out_channels;
avOut.pts = videoTrack ? pts : 0;
avOut.type = "audio";
avOut.stream = audioTrack->stream;
avOut.avfc = avContext;
if (context->output->audio->Write(context, &avOut) < 0)
ffmpeg_err("writing data to audio device failed\n");
av_freep(&output);
}
} else {
avOut.data = packet_data;
avOut.len = packet_size;
avOut.pts = pts;
avOut.packet = &packet;
avOut.type = "audio";
avOut.stream = audioTrack->stream;
avOut.avfc = avContext;
@@ -544,7 +421,7 @@ static void FFMPEGThread(Context_t * context)
float duration = 3.0;
ffmpeg_printf(100, "subtitleTrack->stream %p \n", subtitleTrack->stream);
pts = calcPts(subtitleTrack->stream, packet.pts);
pts = calcPts(avContext, subtitleTrack->stream, packet.pts);
if (duration > 0.0) {
/* is there a decoder ? */
@@ -586,11 +463,6 @@ static void FFMPEGThread(Context_t * context)
dvbsub_ass_clear();
if (swr)
swr_free(&swr);
if (decoded_frame)
av_frame_free(&decoded_frame);
if (context->playback)
context->playback->abortPlayback = 1;
hasPlayThreadStarted = 0;
@@ -903,11 +775,6 @@ int container_ffmpeg_update_tracks(Context_t * context, char *filename)
track.duration = (double) stream->duration * av_q2d(stream->time_base) * 1000.0;
}
if (!strncmp(encoding, "A_IPCM", 6)) {
track.inject_as_pcm = 1;
ffmpeg_printf(10, " Handle inject_as_pcm = %d\n", track.inject_as_pcm);
}
if (context->manager->audio) {
if (context->manager->audio->Command(context, MANAGER_ADD, &track) < 0) {
/* konfetti: fixme: is this a reason to return with error? */

2
libeplayer3/include/common.h
View File

@@ -6,12 +6,14 @@
#include "manager.h"
#include "playback.h"
#include <pthread.h>
#include <stdint.h>
typedef struct Context_s {
PlaybackHandler_t *playback;
ContainerHandler_t *container;
OutputHandler_t *output;
ManagerHandler_t *manager;
int64_t *currentAudioPtsP;
} Context_t;
int container_ffmpeg_update_tracks(Context_t * context, char *filename);

5
libeplayer3/include/manager.h
View File

@@ -38,8 +38,8 @@ typedef struct Track_s {
char *language;
/* length of track */
uint64_t duration;
uint64_t pts;
int64_t duration;
//CHECK int64_t pts;
/* context from ffmpeg */
AVFormatContext *avfc;
@@ -53,7 +53,6 @@ typedef struct Track_s {
/* If player2 or the elf do not support decoding of audio codec set this.
* AVCodec is than used for softdecoding and stream will be injected as PCM */
int inject_as_pcm;
int inject_raw_pcm;
int pending;

6
libeplayer3/include/output.h
View File

@@ -42,15 +42,15 @@ typedef struct {
int uSampleRate;
int uBitsPerSample;
int bLittleEndian;
int restart_audio_resampling;
uint64_t pts;
int64_t pts;
char *type;
/* context from ffmpeg */
AVFormatContext *avfc;
/* stream from ffmpeg */
AVStream *stream;
AVPacket *packet;
} AudioVideoOut_t;
struct Context_s;

10
libeplayer3/include/writer.h
View File

@@ -12,19 +12,23 @@
typedef enum { eNone, eAudio, eVideo, eGfx } eWriterType_t;
struct Context_s;
typedef struct Context_s Context_t;
typedef struct {
int fd;
uint8_t *data;
unsigned int len;
uint64_t Pts;
int64_t Pts;
int uNoOfChannels;
int uSampleRate;
int uBitsPerSample;
int bLittleEndian;
/* context from ffmpeg */
int restart_audio_resampling;
AVFormatContext *avfc;
/* stream from ffmpeg */
AVStream *stream;
AVPacket *packet;
Context_t *context;
} WriterAVCallData_t;
typedef struct WriterCaps_s {

5
libeplayer3/output/linuxdvb.c
View File

@@ -940,6 +940,7 @@ static int Write(Context_t *context, AudioVideoOut_t *out)
call.data = out->data;
call.len = out->len;
call.Pts = out->pts;
call.packet = out->packet;
if (writer->writeData)
res = writer->writeData(&call);
@@ -977,12 +978,14 @@ static int Write(Context_t *context, AudioVideoOut_t *out)
call.data = out->data;
call.len = out->len;
call.Pts = out->pts;
call.packet = out->packet;
call.uNoOfChannels = out->uNoOfChannels;
call.uSampleRate = out->uSampleRate;
call.uBitsPerSample = out->uBitsPerSample;
call.bLittleEndian = out->bLittleEndian;
call.restart_audio_resampling = out->restart_audio_resampling;
call.context = context;
if (writer->writeData)
res = writer->writeData(&call);

185
libeplayer3/output/writer/pcm.c
View File

@@ -289,34 +289,187 @@ static int writeData(WriterAVCallData_t *call)
return size;
}
SwrContext *swr = NULL;
AVFrame *decoded_frame = NULL;
int out_sample_rate = 44100;
int out_channels = 2;
uint64_t out_channel_layout = AV_CH_LAYOUT_STEREO;
int restart_audio_resampling = 0;
static int resetIpcm()
{
if (swr)
swr_free(&swr);
if (decoded_frame)
av_frame_free(&decoded_frame);
return 0;
}
int64_t calcPts(AVFormatContext *, AVStream *, int64_t);
static int writeDataIpcm(WriterAVCallData_t *call)
{
AVCodecContext *c = call->stream->codec;
AVPacket *packet = call->packet;
uint8_t *packet_data = packet->data;
unsigned int packet_size = packet->size;
if (call->restart_audio_resampling)
call->restart_audio_resampling = 1;
if (restart_audio_resampling) {
restart_audio_resampling = 0;
if (swr) {
swr_free(&swr);
swr = NULL;
}
if (decoded_frame) {
av_frame_free(&decoded_frame);
decoded_frame = NULL;
}
call->context->output->Command(call->context, OUTPUT_CLEAR, NULL);
call->context->output->Command(call->context, OUTPUT_PLAY, NULL);
AVCodec *codec = avcodec_find_decoder(c->codec_id);
if (!codec || avcodec_open2(c, codec, NULL))
fprintf(stderr, "%s %d: avcodec_open2 failed\n", __func__, __LINE__);
}
while (packet_size > 0) {
int got_frame = 0;
if (!decoded_frame) {
if (!(decoded_frame = av_frame_alloc())) {
fprintf(stderr, "out of memory\n");
exit(1);
}
} else
av_frame_unref(decoded_frame);
int len = avcodec_decode_audio4(c, decoded_frame, &got_frame, packet);
if (len < 0) {
restart_audio_resampling = 1;
break;
}
packet_data += len;
packet_size -= len;
if (!got_frame)
continue;
int e;
if (!swr) {
int rates[] = { 48000, 96000, 192000, 44100, 88200, 176400, 0 };
int *rate = rates;
int in_rate = c->sample_rate;
while (*rate && ((*rate / in_rate) * in_rate != *rate) && (in_rate / *rate) * *rate != in_rate)
rate++;
out_sample_rate = *rate ? *rate : 44100;
swr = swr_alloc();
out_channels = c->channels;
if (c->channel_layout == 0) {
// FIXME -- need to guess, looks pretty much like a bug in the FFMPEG WMA decoder
c->channel_layout = AV_CH_LAYOUT_STEREO;
}
out_channel_layout = c->channel_layout;
// player2 won't play mono
if (out_channel_layout == AV_CH_LAYOUT_MONO) {
out_channel_layout = AV_CH_LAYOUT_STEREO;
out_channels = 2;
}
av_opt_set_int(swr, "in_channel_layout", c->channel_layout, 0);
av_opt_set_int(swr, "out_channel_layout", out_channel_layout, 0);
av_opt_set_int(swr, "in_sample_rate", c->sample_rate, 0);
av_opt_set_int(swr, "out_sample_rate", out_sample_rate, 0);
av_opt_set_int(swr, "in_sample_fmt", c->sample_fmt, 0);
av_opt_set_int(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
e = swr_init(swr);
if (e < 0) {
fprintf(stderr,
"swr_init: %d (icl=%d ocl=%d isr=%d osr=%d isf=%d osf=%d\n",
-e, (int) c->channel_layout,
(int) out_channel_layout, c->sample_rate, out_sample_rate, c->sample_fmt, AV_SAMPLE_FMT_S16);
swr_free(&swr);
swr = NULL;
}
}
uint8_t *output = NULL;
int in_samples = decoded_frame->nb_samples;
int out_samples = av_rescale_rnd(swr_get_delay(swr, c->sample_rate) + in_samples, out_sample_rate, c->sample_rate, AV_ROUND_UP);
e = av_samples_alloc(&output, NULL, out_channels, out_samples, AV_SAMPLE_FMT_S16, 1);
if (e < 0) {
fprintf(stderr, "av_samples_alloc: %d\n", -e);
continue;
}
// FIXME. PTS calculation is probably broken.
int64_t pts;
int64_t next_in_pts = av_rescale(av_frame_get_best_effort_timestamp(decoded_frame),
call->stream->time_base.num * (int64_t) out_sample_rate * c->sample_rate,
call->stream->time_base.den);
int64_t next_out_pts = av_rescale(swr_next_pts(swr, next_in_pts),
call->stream->time_base.den,
call->stream->time_base.num * (int64_t) out_sample_rate * c->sample_rate);
*(call->context->currentAudioPtsP) = /* audioTrack->pts = */ pts = calcPts(call->avfc, call->stream, next_out_pts);
out_samples = swr_convert(swr, &output, out_samples, (const uint8_t **)
&decoded_frame->data[0], in_samples);
WriterAVCallData_t pcmOut;
pcmOut.fd = call->fd;
pcmOut.uSampleRate = out_sample_rate;
pcmOut.uNoOfChannels = av_get_channel_layout_nb_channels(out_channel_layout);
pcmOut.uBitsPerSample = 16;
pcmOut.bLittleEndian = 1;
pcmOut.data = output;
pcmOut.len = out_samples * sizeof(short) * out_channels;
pcmOut.Pts = pts; // FIXME videoTrack ? pts : 0;
pcmOut.stream = call->stream;
pcmOut.avfc = call->avfc;
pcmOut.packet = NULL;
writeData(&pcmOut);
av_freep(&output);
}
return packet->size;
}
/* ***************************** */
/* Writer Definition */
/* ***************************** */
static WriterCaps_t caps_pcm = {
"pcm",
eAudio,
"A_PCM",
AUDIO_ENCODING_LPCMA
"pcm",
eAudio,
"A_PCM",
AUDIO_ENCODING_LPCMA
};
struct Writer_s WriterAudioPCM = {
&reset,
&writeData,
NULL,
&caps_pcm
&reset,
&writeData,
NULL,
&caps_pcm
};
static WriterCaps_t caps_ipcm = {
"ipcm",
eAudio,
"A_IPCM",
AUDIO_ENCODING_LPCMA
"ipcm",
eAudio,
"A_IPCM",
AUDIO_ENCODING_LPCMA
};
struct Writer_s WriterAudioIPCM = {
&reset,
&writeData,
NULL,
&caps_ipcm
&resetIpcm,
&writeDataIpcm,
NULL,
&caps_ipcm
};