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some refactors and make audio work with opus 

use SWITCH_RTP_MAX_BUF_LEN for video
timestamp has to times 1000 or it too short
neither VLC nor Chrome plays it because they don't support OPUS,
but you can use `ffmpeg -i a.webm b.webm`
This commit is contained in:
Seven Du 2015-03-13 10:58:21 +08:00 committed by Michael Jerris
parent db31cc650e
commit 8d4686aee2
1 changed files with 21 additions and 17 deletions
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36
src/mod/formats/mod_webm/mod_webm.cpp
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@ -49,6 +49,9 @@ SWITCH_MODULE_DEFINITION(mod_webm, mod_webm_load, NULL, NULL);
#define IS_VP8_KEY_FRAME(byte) ((((byte) & 0x01) ^ 0x01) ? true : false)
#define IS_VP9_KEY_FRAME(byte) (((byte) & 0x01) ? true : false)
#define AUDIO_CODEC "OPUS"
// #define AUDIO_CODEC "VORBIS"
struct webm_file_context {
switch_memory_pool_t *pool;
mkvmuxer::AudioTrack* audio;
@ -131,16 +134,16 @@ static switch_status_t webm_file_open(switch_file_handle_t *handle, const char *
mkvmuxer::SegmentInfo* const info = context->segment->GetSegmentInfo();
info->set_timecode_scale(1000000);
info->set_muxing_app("FreeSWITCH");
info->set_writing_app(switch_version_full());
context->audio_track_id = context->segment->AddAudioTrack(handle->samplerate, handle->channels, 1);
context->audio_track_id = context->segment->AddAudioTrack(handle->samplerate, handle->channels, 0);
if (!context->audio_track_id) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error add audio track!\n");
goto end;
}
context->audio = static_cast<mkvmuxer::AudioTrack*>(context->segment->GetTrackByNumber(context->audio_track_id));
context->audio->set_codec_id("A_OPUS"); // or A_VORBIS
// context->audio->set_bit_depth(8);
context->audio->set_codec_id("A_" AUDIO_CODEC);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "sample rate: %d, channels: %d\n", handle->samplerate, handle->channels);
@ -160,7 +163,7 @@ static switch_status_t webm_file_open(switch_file_handle_t *handle, const char *
"useinbandfec=1;minptime=20;ptime=20;samplerate=%d%s", handle->samplerate, handle->channels == 2 ? ",stereo=1" : "");
if (switch_core_codec_init(&context->audio_codec,
"OPUS",
AUDIO_CODEC,
fmtp,
handle->samplerate,
20,//ms
@ -201,7 +204,7 @@ end:
if (context->segment) delete context->segment;
if (context->writer) delete context->writer;
return SWITCH_STATUS_FALSE;
return SWITCH_STATUS_GENERR;
}
static switch_status_t webm_file_truncate(switch_file_handle_t *handle, int64_t offset)
@ -265,12 +268,12 @@ static switch_status_t webm_file_write(switch_file_handle_t *handle, void *data,
data, datalen,
handle->samplerate,
buf, &size, &encoded_rate, NULL);
}
switch_mutex_lock(context->mutex);
if (!context->timer.interval) {
goto end; // block audio before video is there
switch_core_timer_init(&context->timer, "soft", 1, 1000, context->pool);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "init timer\n");
} else if(!context->audio_start) { // try make up some sampels if the video already start
@ -280,16 +283,17 @@ static switch_status_t webm_file_write(switch_file_handle_t *handle, void *data,
if (size > 0) {
// timecode still need to figure out for sync
switch_core_timer_sync(&context->timer);
bool ret = context->segment->AddFrame(buf, size, context->audio_track_id, context->timer.samplecount, true);
// bool ret = context->segment->AddFrame((const uint8_t *)data, used, context->audio_track_id, context->audio_duration * 1000000, is_key);
// switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Writing audio %d bytes, ts: %lld\n", size, context->timer.samplecount * 1000LL);
bool ret = context->segment->AddFrame(buf, size, context->audio_track_id, context->timer.samplecount * 1000LL, true);
// bool ret = context->segment->AddFrame((const uint8_t *)buf, size, context->audio_track_id, context->audio_duration, true);
if (!ret) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error writing audio %d bytes\n", size);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error writing audio %d bytes, pts: %lld or %lld\n", size, context->timer.samplecount * 1000LL, context->audio_duration);
}
context->audio_duration = 0;
}
end:
switch_mutex_unlock(context->mutex);
return status;
@ -379,7 +383,7 @@ static switch_status_t do_write_video(switch_file_handle_t *handle, switch_frame
int duration = 0;
if (!context->timer.interval) {
switch_core_timer_init(&context->timer, "soft", 1, 1, context->pool);
switch_core_timer_init(&context->timer, "soft", 1, 1000, context->pool);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "init timer\n");
} else {
switch_core_timer_sync(&context->timer);
@ -390,10 +394,10 @@ static switch_status_t do_write_video(switch_file_handle_t *handle, switch_frame
// switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "samplecount: %u\n", context->timer.samplecount);
bool ret = false;
ret = context->segment->AddFrame((const uint8_t *)data, used, context->video_track_id, context->timer.samplecount, is_key);
ret = context->segment->AddFrame((const uint8_t *)data, used, context->video_track_id, context->timer.samplecount * 1000LL, is_key);
if (!ret) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error add frame %d bytes, timecode: %llu\n", used, context->timer.samplecount);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error add frame %d bytes, timecode: %llu\n", used, context->timer.samplecount * 1000LL);
switch_goto_status(SWITCH_STATUS_FALSE, end);
}
@ -416,10 +420,10 @@ static switch_status_t webm_file_write_video(switch_file_handle_t *handle, switc
return do_write_video(handle, frame);
} else {
switch_frame_t eframe = { 0 };
uint8_t data[SWITCH_RECOMMENDED_BUFFER_SIZE];
uint8_t data[SWITCH_RTP_MAX_BUF_LEN];
eframe.data = data + 12;
eframe.datalen = SWITCH_RECOMMENDED_BUFFER_SIZE - 12;
eframe.datalen = SWITCH_RTP_MAX_BUF_LEN - 12;
eframe.img = frame->img;
do {
frame->datalen = SWITCH_DEFAULT_VIDEO_SIZE;