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Audio codec: use vector to hold tmp data

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This commit is contained in:
Martin Pulec
2020-08-25 14:47:29 +02:00
parent 5a8531916d
commit 7a1c2bcf7e
1 changed files with 21 additions and 26 deletions
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47
src/audio/codec/libavcodec.cpp
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@@ -57,8 +57,10 @@ extern "C" {
}
#include <array>
#include <vector>
#include <memory>
#include <type_traits>
#include <unordered_map>
#include <vector>
#include "audio/audio.h"
#include "audio/codec.h"
@@ -111,7 +113,7 @@ static std::unordered_map<audio_codec_t, codec_param, std::hash<int>> mapping {
};
struct libavcodec_codec_state {
uint32_t magic;
uint32_t magic = MAGIC;
pthread_mutex_t *libav_global_lock;
AVCodecContext *codec_ctx;
AVCodec *codec;
@@ -121,9 +123,9 @@ struct libavcodec_codec_state {
struct audio_desc saved_desc;
audio_channel tmp;
char *tmp_data; ///< tmp.data, but non-const qualified
vector<char> tmp_data = vector<char>(TMP_DATA_LEN); ///< tmp.data, but non-const qualified
audio_channel output_channel;
char *output_channel_data; ///< output_channel.data, but non-const qualified
vector<char> output_channel_data = vector<char>(TMP_DATA_LEN); ///< output_channel.data, but non-const qualified
void *samples;
@@ -132,6 +134,7 @@ struct libavcodec_codec_state {
bool context_initialized;
audio_codec_direction_t direction;
};
static_assert(is_aggregate_v<libavcodec_codec_state>, "ensure aggregate to allow aggregate initialization");
/**
* Initializates selected audio codec
@@ -165,8 +168,7 @@ static void *libavcodec_init(audio_codec_t audio_codec, audio_codec_direction_t
avcodec_register_all();
#endif
struct libavcodec_codec_state *s = (struct libavcodec_codec_state *)
calloc(1, sizeof(struct libavcodec_codec_state));
auto *s = new libavcodec_codec_state{};
s->direction = direction;
if(direction == AUDIO_CODER) {
if (preferred_encoder) {
@@ -183,7 +185,7 @@ static void *libavcodec_init(audio_codec_t audio_codec, audio_codec_direction_t
fprintf(stderr, "Your Libavcodec build doesn't contain codec \"%s\".\n",
get_name_to_audio_codec(audio_codec));
}
free(s);
delete s;
return NULL;
} else {
if (!silent) {
@@ -192,14 +194,13 @@ static void *libavcodec_init(audio_codec_t audio_codec, audio_codec_direction_t
}
}
s->magic = MAGIC;
s->libav_global_lock = rm_acquire_shared_lock(LAVCD_LOCK_NAME);
s->codec_ctx = avcodec_alloc_context3(s->codec);
if(!s->codec_ctx) { // not likely :)
if (!silent) {
fprintf(stderr, "Could not allocate audio codec context\n");
}
free(s);
delete s;
return NULL;
}
@@ -211,12 +212,8 @@ static void *libavcodec_init(audio_codec_t audio_codec, audio_codec_direction_t
s->av_frame = av_frame_alloc();
memset(&s->tmp, 0, sizeof(audio_channel));
memset(&s->output_channel, 0, sizeof(audio_channel));
s->tmp_data = static_cast<char *>(malloc(TMP_DATA_LEN));
s->tmp.data = s->tmp_data;
s->output_channel_data = static_cast<char *>(malloc(TMP_DATA_LEN));
s->output_channel.data = s->output_channel_data;
s->tmp.data = s->tmp_data.data();
s->output_channel.data = s->output_channel_data.data();
if(direction == AUDIO_CODER) {
s->output_channel.codec = audio_codec;
@@ -407,7 +404,7 @@ static audio_channel *libavcodec_compress(void *state, audio_channel * channel)
LOG(LOG_LEVEL_ERROR) << MOD_NAME << "Auxiliary buffer overflow!\n";
return {};
}
int2float(s->tmp_data + s->tmp.data_len, channel->data, channel->data_len);
int2float(s->tmp_data.data() + s->tmp.data_len, channel->data, channel->data_len);
s->tmp.data_len += channel->data_len;
} else {
size_t data_len = channel->data_len / channel->bps * 4;
@@ -417,16 +414,16 @@ static audio_channel *libavcodec_compress(void *state, audio_channel * channel)
LOG(LOG_LEVEL_ERROR) << MOD_NAME << "Auxiliary buffer overflow!\n";
return {};
}
int2float(s->tmp_data + s->tmp.data_len, tmp.get(), data_len);
int2float(s->tmp_data.data() + s->tmp.data_len, tmp.get(), data_len);
s->tmp.data_len += data_len;
}
} else {
change_bps(s->tmp_data + s->tmp.data_len, s->output_channel.bps,
change_bps(s->tmp_data.data() + s->tmp.data_len, s->output_channel.bps,
channel->data, s->saved_desc.bps, channel->data_len);
s->tmp.data_len += channel->data_len / s->saved_desc.bps * s->output_channel.bps;
}
} else {
memcpy(s->tmp_data + s->tmp.data_len, channel->data, channel->data_len);
memcpy(s->tmp_data.data() + s->tmp.data_len, channel->data, channel->data_len);
s->tmp.data_len += channel->data_len;
}
}
@@ -451,7 +448,7 @@ static audio_channel *libavcodec_compress(void *state, audio_channel * channel)
LOG(LOG_LEVEL_ERROR) << MOD_NAME << "Output buffer overflow!\n";
return {};
}
memcpy(s->output_channel_data + s->output_channel.data_len,
memcpy(s->output_channel_data.data() + s->output_channel.data_len,
pkt.data, pkt.size);
s->output_channel.data_len += pkt.size;
av_packet_unref(&pkt);
@@ -493,7 +490,7 @@ static audio_channel *libavcodec_compress(void *state, audio_channel * channel)
}
s->tmp.data_len -= offset;
memmove(s->tmp_data, s->tmp.data + offset, s->tmp.data_len);
memmove(s->tmp_data.data(), s->tmp.data + offset, s->tmp.data_len);
///fprintf(stderr, "%d %d\n", i++% 2, s->output_channel.data_len);
if(s->output_channel.data_len) {
@@ -559,7 +556,7 @@ static audio_channel *libavcodec_decompress(void *state, audio_channel * channel
int data_size = av_samples_get_buffer_size(NULL, channels,
s->av_frame->nb_samples,
s->codec_ctx->sample_fmt, 1);
memcpy(s->output_channel_data + offset, s->av_frame->data[0],
memcpy(s->output_channel_data.data() + offset, s->av_frame->data[0],
data_size);
offset += len;
s->output_channel.data_len += data_size;
@@ -594,7 +591,7 @@ static audio_channel *libavcodec_decompress(void *state, audio_channel * channel
s->codec_ctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
unique_ptr<char []> int32_data(unique_ptr<char []>(new char [s->output_channel.data_len]));
float2int(int32_data.get(), s->output_channel.data, s->output_channel.data_len);
memcpy(s->output_channel_data, int32_data.get(), s->output_channel.data_len);
memcpy(s->output_channel_data.data(), int32_data.get(), s->output_channel.data_len);
s->output_channel.bps = 4;
} else {
s->output_channel.bps =
@@ -671,12 +668,10 @@ static void libavcodec_done(void *state)
pthread_mutex_unlock(s->libav_global_lock);
rm_release_shared_lock(LAVCD_LOCK_NAME);
free(s->output_channel_data);
free(s->tmp_data);
av_freep(&s->samples);
av_frame_free(&s->av_frame);
free(s);
delete s;
}
static const audio_codec_t supported_codecs[] = { AC_ALAW, AC_MULAW, AC_SPEEX, AC_OPUS, AC_G722, AC_FLAC, AC_MP3, AC_AAC, AC_NONE };