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Libavcodec audio: suport for mp3 + fixes

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This commit is contained in:
Martin Pulec
2014-11-26 13:06:24 +01:00
parent e3d186bdfd
commit 2e833bd261
3 changed files with 25 additions and 5 deletions
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28
src/audio/codec/libavcodec.cpp
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@@ -83,6 +83,7 @@ extern "C" {
#define AV_CODEC_ID_OPUS CODEC_ID_OPUS
#define AV_CODEC_ID_ADPCM_G722 CODEC_ID_ADPCM_G722
#define AV_CODEC_ID_ADPCM_G726 CODEC_ID_ADPCM_G726
#define AV_CODEC_ID_MP3 CODEC_ID_MP3
#endif
using namespace std;
@@ -110,6 +111,7 @@ static std::unordered_map<audio_codec_t, AVCodecID, std::hash<int>> mapping {
#endif
{ AC_G722, AV_CODEC_ID_ADPCM_G722 },
{ AC_G726, AV_CODEC_ID_ADPCM_G726 },
{ AC_MP3, AV_CODEC_ID_MP3 },
};
struct libavcodec_codec_state {
@@ -249,8 +251,24 @@ static bool reinitialize_coder(struct libavcodec_codec_state *s, struct audio_de
break;
}
/// @todo
/// Check also planar formats since we can use both (we encode always one channel only)
if(!check_sample_fmt(s->codec, s->codec_ctx->sample_fmt)) {
s->codec_ctx->sample_fmt = s->codec->sample_fmts[0];
s->codec_ctx->sample_fmt = AV_SAMPLE_FMT_NONE;
int i = 0;
while (s->codec->sample_fmts[i] != AV_SAMPLE_FMT_NONE) {
if (s->codec->sample_fmts[i] == AV_SAMPLE_FMT_FLT ||
s->codec->sample_fmts[i] == AV_SAMPLE_FMT_FLTP ||
s->codec->sample_fmts[i] == AV_SAMPLE_FMT_DBL ||
s->codec->sample_fmts[i] == AV_SAMPLE_FMT_DBLP) {
i++;
continue;
} else {
s->codec_ctx->sample_fmt = s->codec->sample_fmts[i];
break;
}
}
assert(s->codec_ctx->sample_fmt != AV_SAMPLE_FMT_NONE);
s->change_bps_to = av_get_bytes_per_sample(s->codec_ctx->sample_fmt);
}
@@ -300,7 +318,7 @@ static bool reinitialize_coder(struct libavcodec_codec_state *s, struct audio_de
}
s->output_channel.sample_rate = desc.sample_rate;
s->output_channel.bps = desc.bps;
s->output_channel.bps = s->change_bps_to == 0 ? desc.bps : s->change_bps_to;
s->saved_desc = desc;
return true;
@@ -347,8 +365,8 @@ static audio_channel *libavcodec_compress(void *state, audio_channel * channel)
}
if(s->change_bps_to) {
change_bps((char *) s->tmp.data, s->saved_desc.bps, channel->data,
s->change_bps_to, channel->data_len);
change_bps((char *) s->tmp.data + s->tmp.data_len, s->change_bps_to,
channel->data, s->saved_desc.bps, channel->data_len);
s->tmp.data_len += channel->data_len / s->saved_desc.bps * s->change_bps_to;
} else {
memcpy((char *) s->tmp.data + s->tmp.data_len, channel->data, channel->data_len);
@@ -516,7 +534,7 @@ static void libavcodec_done(void *state)
free(s);
}
static audio_codec_t supported_codecs[] = { AC_ALAW, AC_MULAW, AC_ADPCM_IMA_WAV, AC_SPEEX, AC_OPUS, AC_G722, AC_G726, AC_NONE };
static audio_codec_t supported_codecs[] = { AC_ALAW, AC_MULAW, AC_ADPCM_IMA_WAV, AC_SPEEX, AC_OPUS, AC_G722, AC_G726, AC_MP3, AC_NONE };
static int supported_bytes_per_second[] = { 2, 0 };
struct audio_codec libavcodec_audio_codec = {